| 01:16.17 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 01:16.17 | *** topic/#asterisk is #asterisk Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 10.0.0-beta1 (2011/07/22), 1.8.6.0 (2011/08/31), dahdi-linux 2.5.0.1 (2011/09/08), dahdi-tools 2.5.0.1 (2011/09/08), libpri 1.4.12 (2011/07/06) -=- Visit the official Asterisk wiki: wiki.asterisk.org -=- http://xkcd.com/806/ |
| 01:28.52 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 01:39.34 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 01:56.09 | *** join/#asterisk jayson_r (~jayson@cpe-071-076-046-081.sc.res.rr.com) |
| 02:02.20 | ChannelZ | It's The Law |
| 02:02.55 | *** join/#asterisk jayson_r (~jayson@cpe-071-076-046-081.sc.res.rr.com) |
| 02:09.45 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 02:15.46 | luke-jr | I don't see how to limit calls to/from extensions using GROUPs⦠all the examples break as soon as someone transfers |
| 02:16.47 | carrar | SIP Devices? |
| 02:16.54 | luke-jr | ye |
| 02:16.56 | luke-jr | s |
| 02:17.29 | p3nguin | Set the group, then check the count. |
| 02:17.32 | luke-jr | Polycom SIP devices, I think |
| 02:17.33 | carrar | Could use ChanIsAvail |
| 02:17.40 | luke-jr | p3nguin: that works, until it gets transferred |
| 02:17.58 | p3nguin | What happens when you transfer? |
| 02:18.15 | luke-jr | p3nguin: it leaves the original caller in the group for the original destination |
| 02:18.35 | luke-jr | so the transferred destination is left with none in the group, and the original destination is callless and "busy' |
| 02:19.00 | carrar | You just want to know if they are on the phone right? |
| 02:19.11 | luke-jr | yeah |
| 02:19.17 | carrar | Use ChanIsAvail |
| 02:19.24 | p3nguin | Why would you use groupcount for that? |
| 02:22.00 | luke-jr | p3nguin: that was the suggestion on the note deprecating call-limit |
| 02:22.13 | carrar | Get Coding! |
| 02:22.16 | p3nguin | But call limit isn't for checking if someone is on the phone, either. |
| 02:22.19 | carrar | time is wasting |
| 02:22.22 | p3nguin | So there's some confusion. |
| 02:23.07 | p3nguin | Using GROUP_COUNT is the new way to limit calls, but that doesn't have anything to do with checking if someone is on the phone. |
| 02:25.23 | p3nguin | Either you've gone about something the wrong way, or you gave the wrong answer when you were asked if you just wanted to know if someone is on the phone. |
| 02:28.29 | luke-jr | ChanIsAvail always reports it as available ;) |
| 02:28.52 | carrar | So find out what you are ding wrong |
| 02:28.59 | p3nguin | probably because it's available. |
| 02:29.21 | luke-jr | p3nguin: I'm just trying to send to busy-voicemail if they're on the line |
| 02:29.34 | carrar | read up on that application |
| 02:29.35 | p3nguin | turn off call waiting on the phones. |
| 02:29.54 | carrar | core show application ChanIsAvail |
| 02:30.13 | p3nguin | Use DIALSTATUS to send to the appropriate place after they are found to be unavailable. |
| 02:30.17 | luke-jr | DEVICE_STATE seems to be usable |
| 02:30.29 | luke-jr | p3nguin: DIALSTATUS is never BUSY on these phones |
| 02:30.35 | p3nguin | TURN OFF CALL WAITING |
| 02:31.01 | luke-jr | p3nguin: I don't know how. |
| 02:31.03 | p3nguin | If you have call waiting, the phone is obviously not going to be busy. |
| 02:31.09 | p3nguin | That's the point of call waiting. |
| 02:31.19 | luke-jr | p3nguin: the phones themselves don't make any indication to the user that there's an attempted call |
| 02:31.31 | p3nguin | No call waiting tone? |
| 02:31.34 | luke-jr | nope |
| 02:31.50 | p3nguin | What kind of phone doesn't have call waiting tones when there is a second call? |
| 02:31.53 | luke-jr | no idea |
| 02:32.10 | p3nguin | You're actually working on this system, aren't you? |
| 02:32.31 | carrar | luke-jr, perhaps you should just hire someone to do this for you |
| 02:32.41 | p3nguin | If so, surely you can figure out what kind of phones they are. |
| 02:33.18 | p3nguin | Even the shittiest of phones send an intelligible user agent info. |
| 02:33.33 | carrar | I'm able to do exactly what you want with ChanIsAvail just fine |
| 02:33.45 | carrar | I'll do it for $5,000,000 |
| 02:34.13 | p3nguin | You shouldn't have to, though, and if the channel is available (which it apparently is), ChanIsAvail will always return that it is available. |
| 02:34.22 | carrar | *sigh* |
| 02:34.34 | carrar | ok both of you |
| 02:34.50 | p3nguin | If I have call waiting enabled, my channel will always be available unless I enable DND-busy. |
| 02:35.21 | carrar | cause thats how you wrote your dialplan |
| 02:35.21 | luke-jr | [22:30:17] <luke-jr> DEVICE_STATE seems to be usable |
| 02:35.26 | carrar | didn't need too |
| 02:35.36 | p3nguin | Call waiting has very little to do with dial plan. |
| 02:36.04 | carrar | Though most people like the call waiting tone |
| 02:36.05 | p3nguin | The same dialplan that is used to send one call is used to send every subsequent call. |
| 02:36.09 | carrar | I do |
| 02:36.22 | carrar | but some don't want to be bothered by a second call |
| 02:36.32 | carrar | so ChanIsAvail solves that |
| 02:36.49 | p3nguin | If my phone is able to accept a second call, I'd like to know I am getting a call to be able to answer it, divert it, or ignore it. |
| 02:37.16 | p3nguin | But if I don't want a second call, I have to turn off call waiting. |
| 02:37.32 | p3nguin | Or I can rewrite the dialplan to use GROUP_COUNT for each device. |
| 02:37.42 | p3nguin | and limit it to 1 |
| 02:37.48 | carrar | groupcount is not the way to do that |
| 02:38.05 | p3nguin | ChanIsAvail isn't either, since the channel is always available if call waiting is enabled. |
| 02:38.10 | carrar | ChanIsAvail IS |
| 02:38.14 | carrar | it works great |
| 02:38.17 | carrar | I use it |
| 02:38.32 | carrar | I've done this before |
| 02:38.41 | carrar | I have customers use it |
| 02:38.44 | carrar | So yes |
| 02:38.46 | carrar | it works |
| 02:38.49 | p3nguin | Explain to me how ChanIsAvail is going to report back that the phone is busy when there is another "line" on it able to accept a call. |
| 02:38.52 | carrar | it does exactly what he is asking for |
| 02:39.12 | carrar | ChanIsAvail can tell you if the device is in use |
| 02:39.24 | carrar | if it is, then don't send a call |
| 02:39.48 | carrar | I invite you to read: |
| 02:39.48 | carrar | core show application ChanIsAvail |
| 02:40.33 | p3nguin | You're apparently missing the key point I'm trying to make. If the phone has call waiting, the channel is always available. |
| 02:40.36 | carrar | and I think your both capable of figuring out how to use it correctly |
| 02:40.52 | carrar | He does not want to use callwaiting on the phone |
| 02:41.07 | p3nguin | But he said he doesn't know how to turn it off. So it is on until then. |
| 02:41.21 | p3nguin | So it's on, and the channel will always be available. |
| 02:41.22 | carrar | Ok whatever, I'm done |
| 02:42.10 | carrar | He can read his phone manual if he wants |
| 02:42.41 | p3nguin | He said he can't even figure out what kind of phones they are, so don't be too sure. |
| 02:42.41 | *** join/#asterisk jwiggins (~James@gateway/tor-sasl/jwiggins) |
| 02:42.48 | carrar | heh |
| 02:43.16 | p3nguin | I've never before seen anyone that can't figure out what kind of SIP phones are in use. |
| 02:43.19 | carrar | Very capable phones he has |
| 02:43.19 | carrar | <luke-jr> Polycom SIP devices, I think |
| 02:43.23 | p3nguin | Or at least have some idea, even if it's wrong. |
| 02:43.41 | p3nguin | Heck, if he has Polycoms, there's a setting for almost everything. |
| 02:44.23 | carrar | sip show peer <peer name>, look at Useragent |
| 02:44.33 | carrar | what does it say luke-jr |
| 02:44.53 | p3nguin | But why would there be no callwaiting tones on additional calls? There's a setting to suppress that? |
| 02:45.12 | p3nguin | No beep, no flashy-flashy? |
| 02:47.35 | carrar | call.callWaiting.ring |
| 02:47.53 | carrar | (beep, ring, silent) |
| 02:48.10 | carrar | beep is not set |
| 02:48.12 | carrar | if |
| 02:48.48 | carrar | (for polycoms) |
| 02:49.56 | p3nguin | I wish my phones could ring instead of beep. |
| 02:50.03 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 02:50.09 | carrar | How annouying would that be |
| 02:50.19 | p3nguin | quite |
| 02:51.57 | p3nguin | My main reason is for when I am listening to a conference (muted, of course), and I get another call. |
| 02:52.04 | p3nguin | I sometimes don't notice the tone. |
| 02:52.20 | carrar | and then what? |
| 02:52.24 | p3nguin | Especially if I walk away from my desk for a minute and there's a call. |
| 02:52.27 | carrar | put the conf on hold? |
| 02:52.31 | p3nguin | Yes. |
| 02:52.35 | carrar | playing on hold music? |
| 02:52.40 | p3nguin | I'm muted. |
| 02:52.46 | carrar | no |
| 02:52.50 | p3nguin | yeah, I am. |
| 02:52.54 | carrar | muted via the conf muting? |
| 02:52.56 | p3nguin | yes |
| 02:53.02 | carrar | not the phone mute, ok |
| 02:53.14 | p3nguin | If the admin unmutes me while they are on hold, that's their problem. |
| 02:53.57 | p3nguin | Nah, I don't rely on phone mute on the conf because it always goes unmute when I switch from speaker to headset or something like that. |
| 02:54.06 | p3nguin | I just mute/unmute with dtmf as needed. |
| 02:54.24 | carrar | Use a cisco phone so you have a shinny red button lite up |
| 02:55.01 | p3nguin | I have a Cisco phone, but that doesn't help when I go away from my desk with the conf on. |
| 02:55.11 | p3nguin | I can't hear the beep or see the light. |
| 02:55.22 | p3nguin | If it would give a normal ring, that would be perfect. |
| 02:56.09 | p3nguin | But I don't really want to get a Polycom phone just for that. |
| 02:56.38 | luke-jr | PolycomSoundPointIP-SPIP_330-UA/2.2.0.0047 |
| 02:56.42 | carrar | Well |
| 02:56.44 | carrar | There you go |
| 02:56.52 | carrar | You have lots of call waiting options |
| 02:56.58 | luke-jr | DEVICE_STATE is easier |
| 02:56.59 | carrar | none, beep or a ring |
| 02:57.31 | carrar | or if you really just don't want to give the end user a option for call waiting use ChanIsAvail |
| 02:57.59 | p3nguin | Just disable call waiting first. |
| 02:58.15 | p3nguin | Then you have the choice of ChanIsAvail or using DIALSTATUS. |
| 03:00.42 | p3nguin | Hmm. I guess now we have to carry guns when we travel by bus. |
| 03:01.05 | carrar | Should always asert your right to carry |
| 03:01.27 | p3nguin | Some goofball in Springfield, MO, shot and killed someone on a Greyhound. |
| 03:01.40 | carrar | You need a permit to carryconcealed there? |
| 03:01.52 | p3nguin | Yeah |
| 03:01.57 | carrar | same here |
| 03:02.09 | carrar | Washington state |
| 03:03.05 | carrar | we have a Reciprocity agreements with MO |
| 03:03.16 | carrar | so my CPL is good there also |
| 03:03.42 | p3nguin | In MO it's not hard to get the permit, but on the IL side of the river, the lawmakers have been fighting over right to carry for some time now. |
| 03:04.14 | carrar | yeah none with IL |
| 03:04.42 | p3nguin | We have very strict rules concerning carrying weapons in IL. I think they call it like Six Seconds to Safety or some bullshit like that. |
| 03:05.17 | p3nguin | It has to be in a special type of carrying case on your person (fanny pack-like), and it can't be loaded. |
| 03:05.25 | p3nguin | Lot of good that'll do when needed. |
| 03:06.00 | carrar | haha |
| 03:06.37 | p3nguin | "Oh hang on for a second while I pull out my disassembled firearm, put it together, and load it up... then you can try to mug me." |
| 03:07.37 | ChannelZ | But it's for the children! |
| 03:08.24 | carrar | how did children survive up to this point in time anyhow! |
| 03:08.33 | p3nguin | One of the rules for CCW in MO that I found very amusing is that you are required to carry your permit with you when carrying your weapon, but if you don't have your permit, it's not a crime. |
| 03:08.41 | ChannelZ | More of them drown in buckets of water |
| 03:09.09 | carrar | shouldn't be acrime |
| 03:09.39 | carrar | They should outlaw buckets of water! |
| 03:09.44 | carrar | permit required |
| 03:09.57 | carrar | WBP |
| 03:10.10 | ChannelZ | Yes. Home Depot is a death factory |
| 03:10.16 | p3nguin | I think there is a 3 or 6$ fine for not carrying your permit when you are carrying a weapon. |
| 03:10.39 | carrar | can always open carry in our state without a permit |
| 03:10.49 | carrar | thats fun to do |
| 03:10.56 | carrar | cops love it |
| 03:11.06 | ChannelZ | And soccer moms |
| 03:11.20 | p3nguin | It's silly. They put a CCW endorsement on drivers license, so I think that should be good enough. |
| 03:12.08 | ChannelZ | What's silly is thugs that don't give a shit about gun-free zones or permits or anything. |
| 03:12.20 | ChannelZ | Meanwhile I have to pay $150 for a permit |
| 03:12.22 | carrar | http://www.king5.com/home/related/Raw-Conversation-recorded-by-man-carrying-gun-into-coffee-shop-104215743.html |
| 03:12.28 | carrar | http://www.youtube.com/watch?v=DL7gB-M61MI |
| 03:12.28 | carrar | http://www.youtube.com/watch?v=3fbKZ2kJp8s |
| 03:12.31 | carrar | good stuff |
| 03:13.15 | p3nguin | In order to carry openly in Washington, do I have to be a resident or the state or be a resident of a state which has CCW permits? |
| 03:13.26 | carrar | Just a resident |
| 03:13.41 | p3nguin | So I couldn't get away with it if I go visit you. |
| 03:13.47 | carrar | having a CCW is just if it's concealed |
| 03:13.56 | carrar | but it's just nice to have it anyways |
| 03:14.13 | p3nguin | I don't know which is more desired, concealed or open. |
| 03:14.23 | carrar | well I would have to re-read the law on open carry if you are not a resident |
| 03:14.53 | ChannelZ | "IANAL" |
| 03:15.00 | carrar | open is better if you want to let people know that they probably are not going to robb you with a knife |
| 03:15.10 | p3nguin | I'd think concealed would potentially get you into a place where you needed to use the weapon, where open carry would be more of a deterent. |
| 03:15.24 | ChannelZ | "Guns make people nervous and when people have them, we're going to follow up an ask questions." |
| 03:15.35 | carrar | yeah, screw their rights! |
| 03:15.37 | carrar | heh |
| 03:15.49 | ChannelZ | Yeah lady, well your Suburban XLT that you can barely see over the steering wheel of makes me nervous, lady |
| 03:15.57 | p3nguin | But then again, the possibilty of someone having a concealed firearm is also a deterent for some of the inexperienced criminals. |
| 03:16.13 | p3nguin | And why do I keep misspelling deterrent? |
| 03:16.19 | ChannelZ | Detergent |
| 03:16.23 | p3nguin | :/ |
| 03:16.35 | carrar | Deoderant |
| 03:16.44 | carrar | thats misspelled too |
| 03:19.20 | ChannelZ | Spelling is for chumps! |
| 03:22.56 | ChannelZ | if anyone wants Googly Plus invites, yell. Not that they're hard to get |
| 03:23.25 | p3nguin | I thought we already had everyone on it that wanted to be on it. |
| 03:23.29 | carrar | I don't have a faceplace login, not sure I need google either |
| 03:23.50 | ChannelZ | yeah it's kind of pointless why it's "closed" |
| 03:23.52 | ChannelZ | still |
| 03:23.55 | carrar | Besides, I have my own social web site |
| 03:24.25 | p3nguin | I don't do any other social networking stuff, but I decided to do G+ and be a rebel. |
| 03:24.25 | ChannelZ | ratemyrack.com |
| 03:24.53 | ChannelZ | oh wow that site still exists |
| 03:25.19 | carrar | Can't really see that going away anytime soon |
| 03:26.41 | ChannelZ | there is always a fresh supply of racks |
| 03:28.29 | ChannelZ | oh, here. http://www.nraila.org/Legislation/Federal/Read.aspx?id=7072 |
| 03:34.09 | ChannelZ | ouch |
| 03:39.17 | *** join/#asterisk ferdna (~yup@cpe-67-10-220-35.elp.res.rr.com) |
| 03:44.55 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 04:05.37 | *** join/#asterisk coppice (~chatzilla@116.92.16.50) |
| 04:26.16 | *** join/#asterisk dijib (~nobodysho@bas10-kitchener06-1176138825.dsl.bell.ca) |
| 04:26.33 | dijib | anybody running * on centos in here? |
| 04:27.37 | p3nguin | Does the OS have that great of an impact on asterisk that you won't accept help from those who are not running CentOS? |
| 04:30.17 | ChannelZ | I run .50 CentOS |
| 04:31.12 | p3nguin | Get two hundred of them and you can have a DollarOS. |
| 04:32.04 | ChannelZ | and some shitty auto-tune rap |
| 04:33.00 | coppice | most people with a strong opinion run 2CentOS |
| 04:33.54 | dijib | no im having an issue running * as a daemon still |
| 04:34.14 | dijib | it runs on boot. but i cant call in out internal or anything. |
| 04:34.20 | dijib | until i run a -c |
| 04:34.31 | dijib | then it spawns a new instance |
| 04:34.55 | dijib | safe_asterisk starts as root, asterisk starts as user asterisk |
| 04:35.06 | dijib | ive done all the chkconfig stuff |
| 04:36.03 | p3nguin | Start over. The problem is that it *does* work? |
| 04:38.22 | dijib | when i start from cold boot. safe_asterisk & asterisk both run. but phones have no action. i call my line and its dead air... |
| 04:38.36 | dijib | not sure what else to say. no worky but the process is running |
| 04:40.32 | p3nguin | If you kill off any and all asterisk-related processes, can you start it with asterisk -G asterisk -U asterisk -vvvvddddddddg ? |
| 04:41.18 | dijib | whats g u d? |
| 04:41.44 | p3nguin | "man asterisk" doesn't know? |
| 04:42.38 | dijib | apparently not |
| 04:43.08 | dijib | [root@trunk ~]# man asterisk |
| 04:43.08 | dijib | -bash: man: command not found |
| 04:43.09 | dijib | [root@trunk ~]# man /usr/sbin/asterisk |
| 04:43.09 | dijib | -bash: man: command not found |
| 04:44.19 | dijib | -d debug, -g dump core if crash and.... u>...? |
| 04:44.41 | dijib | there isnt a u in the online man page but i assume is unnatend? |
| 04:44.45 | dijib | its |
| 04:45.00 | dijib | am i flooding? |
| 04:46.22 | p3nguin | I didn't say to use an option u. |
| 04:46.22 | dijib | user |
| 04:46.27 | p3nguin | because there isn't one. |
| 04:47.01 | dijib | saying g requires an argument. .. dump path? |
| 04:48.03 | p3nguin | Must be a new feature; mine does not require any path. |
| 04:49.29 | dijib | in lower case it doesnt your right |
| 04:50.31 | dijib | whats L in in ps -L |
| 04:51.50 | dijib | Unable to access the running directory (Permission denied). Changing to '/' for compatibility. |
| 04:51.57 | dijib | what dir's do i have to chmod? i |
| 04:52.04 | dijib | i figured it was a permission issue |
| 04:52.15 | dijib | anything * |
| 04:52.35 | dijib | /var/spool/asterisk /var/lib/asterisk |
| 04:52.43 | dijib | what else am i missing? |
| 04:55.11 | dijib | nevermidn let me figure this out |
| 04:55.14 | p3nguin | chown -R asterisk:asterisk /etc/asterisk /var/lib/asterisk /var/log/asterisk /var/spool/asterisk /var/run/asterisk |
| 04:55.18 | p3nguin | oops |
| 04:55.25 | p3nguin | I retract that statement. |
| 04:55.35 | p3nguin | You'll figure it out. |
| 04:58.45 | dijib | k rebooting to see |
| 05:08.13 | dijib | im still getting that permisson error |
| 05:08.20 | dijib | after those chown |
| 05:09.53 | *** join/#asterisk raden (~J@66-168-15-100.dhcp.stpt.wi.charter.com) |
| 05:11.50 | p3nguin | I'm guessing that means asterisk is trying to access your current working directory, and it can't. |
| 05:12.06 | p3nguin | I'm not sure why 1.8 does that -- 1.4 has never done that to me. |
| 05:12.21 | p3nguin | 1.8 always seems to, though. |
| 05:12.50 | p3nguin | But that message has never prevented asterisk from running for me. |
| 05:19.28 | *** join/#asterisk nix8n82-phone (~hmg@75-174-132-115.chyn.qwest.net) |
| 05:20.30 | dijib | still not working |
| 05:20.59 | dijib | it looks like my dialplan has been loaded cuz i try my extensions and it tries to call and if i use a non defined extension i get an error |
| 05:21.07 | dijib | my voicemail immediatly hangs up though |
| 05:21.08 | p3nguin | What happens when you start it with asterisk -G asterisk -U asterisk -vvvvddddddddg ? |
| 05:21.13 | dijib | it runs |
| 05:21.18 | p3nguin | Perfect. |
| 05:21.37 | dijib | its running right now though, it just doesnt work |
| 05:21.56 | p3nguin | I don't know what "doesnt work" means. You have to be very specific. |
| 05:21.59 | *** join/#asterisk godmachine-x6 (~godmachin@unaffiliated/godmachine-x6) |
| 05:22.09 | dijib | i can register sip clients |
| 05:22.18 | p3nguin | That's good. |
| 05:22.28 | dijib | but all i get is dead air |
| 05:23.07 | p3nguin | If you pick up a phone and call an extension which dials another phone, you hear nothing as the other phone rings? |
| 05:23.25 | dijib | phone doesnt even ring |
| 05:23.28 | dijib | i just hear nothing |
| 05:23.54 | p3nguin | So nothing happens at all. Show me the extension you're calling. |
| 05:23.56 | *** join/#asterisk Syrex (~syrex@dsl-146-17-198.telkomadsl.co.za) |
| 05:24.16 | *** join/#asterisk joako (~joako@opensuse/member/joak0) |
| 05:24.36 | dijib | ok yes it rings |
| 05:24.58 | p3nguin | The target phone rings, but you hear nothing while you wait. Is that right? |
| 05:25.04 | dijib | yes |
| 05:25.05 | joako | I just want to say Asterisk works well for me. 2231343 calls processed System uptime: 4 weeks, 1 day, 10 hours, 59 minutes, 48 seconds |
| 05:25.15 | dijib | not while i wait, call established... i hear nothing |
| 05:25.18 | p3nguin | What kind of phone are you using to call from? |
| 05:25.34 | dijib | software to software |
| 05:25.40 | p3nguin | While you were waiting on someone to answer, was there a ringing sound on your side? |
| 05:25.57 | dijib | how do you have 2million calls? |
| 05:26.18 | p3nguin | lots of calls per hour |
| 05:26.28 | dijib | what are you a callcenter for? |
| 05:27.02 | p3nguin | While you were waiting on someone to answer, was there a ringing sound on your side? |
| 05:27.25 | dijib | 3156 calls per hour |
| 05:29.54 | joako | I use alarm panels and the AlarmReceiver app as a sort of telematics system |
| 05:37.16 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 05:45.29 | *** join/#asterisk godmachine-x6 (~godmachin@unaffiliated/godmachine-x6) |
| 05:50.52 | dijib | 4 S asterisk 1308 1304 19 80 0 - 7668 - 14752 0 01:41 ? 00:00:55 /usr/sbin/asterisk -f -U asterisk -vvvvg -c |
| 05:51.01 | dijib | and i still cant asterisk -r |
| 05:51.05 | dijib | howcome? |
| 05:51.11 | p3nguin | Did you run it like I said? |
| 05:51.46 | dijib | im running it how safe_asterisk runs it on boot |
| 05:52.33 | p3nguin | What is your astrundir set to in asterisk.conf? |
| 05:53.14 | dijib | /var/run/asterisk |
| 05:53.38 | p3nguin | What files are in that directory? |
| 05:54.04 | dijib | asterisk.pid |
| 05:56.16 | p3nguin | If you run asterisk the way I said, do you still only have that one file in /var/run/asterisk? |
| 05:58.46 | dijib | no i get an additional asterisk.ctl |
| 05:59.59 | p3nguin | Good. So asterisk is not broken. |
| 06:00.12 | p3nguin | If the ctl exists, you should be able to connect with asterisk -r |
| 06:00.48 | dijib | i can but not running through safe_asterisk |
| 06:02.00 | p3nguin | I'd like to see the output of ls -dl /var/run/asterisk |
| 06:02.25 | dijib | running how? |
| 06:02.46 | dijib | 770 |
| 06:02.49 | dijib | is the permission |
| 06:02.52 | p3nguin | Just the output of that command. |
| 06:03.17 | dijib | owen and asterisk:asterisk |
| 06:03.20 | dijib | is owener group |
| 06:03.36 | p3nguin | I just want to see the output of the command. |
| 06:03.47 | dijib | drwxrwx---. 2 asterisk asterisk 4096 Sep 10 01:53 /var/run/asterisk/ |
| 06:03.56 | dijib | 770 500:500 |
| 06:04.04 | p3nguin | I realize it's hard, but I knew you could do it! |
| 06:04.27 | dijib | im telling you what it said with 770 500:500 |
| 06:04.38 | p3nguin | I just asked for the output. |
| 06:04.46 | dijib | its the same things bro |
| 06:05.01 | p3nguin | If you don't want to cooperate, that's fine. Good luck! |
| 06:08.22 | dijib | are you normally this short with people? |
| 06:08.46 | carrar | Step 1) FIX |
| 06:08.48 | p3nguin | Short? I asked you THREE TIMES before you gave me the output. |
| 06:08.49 | carrar | Step 2) IT |
| 06:08.53 | carrar | Step 3) FIX IT |
| 06:09.03 | p3nguin | Step 4) Profit? |
| 06:09.16 | carrar | well he can't get to setp 1 |
| 06:09.17 | carrar | step 1 |
| 06:09.25 | dijib | 770 500:500 was the output.. it just wasnt in a format you were willing to accept |
| 06:09.28 | p3nguin | oh |
| 06:09.30 | p3nguin | Good point. |
| 06:09.37 | carrar | Who the hell says "770 500:500" |
| 06:09.41 | carrar | hahah |
| 06:09.46 | p3nguin | There is absolutely no possible way that command outputted that data. |
| 06:09.49 | p3nguin | None. |
| 06:09.51 | p3nguin | At all. |
| 06:10.11 | dijib | then whats the drwxrwx--- mean then? |
| 06:10.18 | dijib | =770 |
| 06:10.23 | carrar | SMRT |
| 06:10.28 | p3nguin | But that's fine, you aren't required to accept my help. |
| 06:10.36 | p3nguin | I'm also not required to provide it. |
| 06:10.46 | dijib | alright then man. |
| 06:10.47 | carrar | 0770 |
| 06:10.58 | dijib | why 0? |
| 06:11.32 | p3nguin | man ls? |
| 06:12.32 | p3nguin | Nope, I was mistaken. It isn't in there. |
| 06:12.37 | carrar | Maybe you should run Asterisk from rpm |
| 06:12.45 | carrar | man, I can't believe I just said that |
| 06:13.09 | p3nguin | His asterisk seems to run fine the way I told him to run it, but he wants to use safe_asterisk instead. |
| 06:13.36 | carrar | safe_asterisk should work great on CentOS |
| 06:13.46 | p3nguin | His must be a special case. |
| 06:14.28 | dijib | safe_asterisk monitors if asterisk is running. for some reason i cant -r into it and the system doesnt work ok. |
| 06:14.36 | dijib | dont worry about it, im too much trouble |
| 06:15.54 | p3nguin | If I used safe_asterisk, I would be more likely to understand why it doesn't create the socket... but I don't, so I don't. |
| 06:16.29 | p3nguin | I should look into it, though. It is apparently a liked way to run asterisk. |
| 06:18.17 | dijib | i think i found my error |
| 06:18.46 | dijib | nope nevermind |
| 06:26.50 | ChannelZ | PEBKAC? |
| 06:27.23 | p3nguin | It looks like safe_asterisk should be run as "safe_asterisk -U asterisk -G asterisk" at least. |
| 06:28.35 | p3nguin | Or change ASTARGS in the script itself. |
| 06:31.16 | p3nguin | It's not a bad looking script, but I don't know that it will give me any real benefit over how I currently run asterisk. |
| 06:34.15 | *** join/#asterisk timahvo1 (~rogue@41.223.57.75) |
| 06:40.20 | *** join/#asterisk godmachine-x6 (~godmachin@unaffiliated/godmachine-x6) |
| 06:41.10 | dijib | i changed ASTARGS with -G asterisk -U asterisk still with same result. no .ctl |
| 06:42.44 | *** join/#asterisk kvad12 (~sysop@174.143.237.76) |
| 06:43.27 | kvad12 | Hey guys anyone have any exp running asterisk on XEN? If so how does it run any issues. Looking to migrate cluster. |
| 06:51.15 | *** join/#asterisk godmachine-x6 (~godmachin@unaffiliated/godmachine-x6) |
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| 08:20.45 | justdave | anyone know if the kernel modules in the asterisk yum repo are ever going to get recompiled against a recent kernel? |
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| 08:42.42 | devil_evoxxx | hi all |
| 08:44.12 | devil_evoxxx | a question, in other country ( no italy ) operator that provide E1/T1 access is able to permit incoming calls from different district (for example a E1 for local prefix 06 and another E1 for local prefix 05) |
| 08:52.13 | tzafrir | devil_evoxxx, and the question is? |
| 08:56.54 | devil_evoxxx | we are a wisp in italy and we have a TDM stream into our farm. We splice a 1gbit ethernet and a E1 stream. But for incoming calls on E1 the only prefix is the local prefix where we are |
| 08:57.05 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 08:57.11 | devil_evoxxx | and wee need more local prefix for neighbor city for offer voip services |
| 09:14.04 | *** join/#asterisk Fritz09 (~Adium@pop1-2174.catv.wtnet.de) |
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| 10:49.34 | TomTom | hi there, i have a beginners problem. i want that asterisk does react on capi/isdn calls, but it says "no extension found". i've made a rather simple setup. any idea what may be still wrong? config -> http://nopaste.info/d23861ce00.html |
| 11:29.12 | *** join/#asterisk cyford (~cyford@c-24-99-17-185.hsd1.ga.comcast.net) |
| 11:30.03 | pa | hi, question: i updated asterisk some time ago, now it is 1.8.6 (on ubuntu natty) |
| 11:30.17 | pa | i also installed the pkg asterisk-mp3 to be able to play mp3s |
| 11:30.43 | pa | but i am not sure where to place my files: if i try to reach my old working extension now, i get: |
| 11:31.07 | pa | <PROTECTED> |
| 11:47.54 | *** join/#asterisk lanning (~lanning@173-8-187-197-SFBA.hfc.comcastbusiness.net) |
| 11:49.03 | *** join/#asterisk m29poff (~m29poff@125.213.206.77.rev.sfr.net) |
| 11:53.53 | wdoekes2 | pa: /var/lib/asterisk/sounds[/en] ? |
| 11:55.44 | pa | ah yes |
| 11:55.46 | pa | right |
| 11:55.54 | pa | but then the problem was that it doesnt read the mp3 |
| 11:55.55 | pa | i think |
| 11:56.01 | pa | i converted to gsm and it worked |
| 12:17.07 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
| 12:20.35 | *** join/#asterisk godmachine-x6 (~godmachin@unaffiliated/godmachine-x6) |
| 12:21.48 | mtbf | I'm trying to troubleshoot some issue with app_rxfax on some old box with 1.2.0, while analysing logs I found out, since some day the only message in the log related to the app_rxfax is : app_rxfax.c: Got hangup , no configuration files were modified since that time, anyone saw simmilar problem? |
| 12:44.38 | *** part/#asterisk StaRetji (~BigAll@80.93.240.171) |
| 12:47.34 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
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| 13:10.32 | *** join/#asterisk eduzimrs (~eduzimrs@201.22.86.124.static.gvt.net.br) |
| 13:10.59 | eduzimrs | hi, im having problem with my sip realtime cache... anyone can help? |
| 13:11.07 | eduzimrs | using mysql |
| 13:12.57 | eduzimrs | when my mysql database is down, all my sip clients get disconnect from * |
| 13:16.39 | *** join/#asterisk irroot (~irroot@41.55.58.47) |
| 13:16.49 | *** join/#asterisk blitzrage (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 13:16.49 | *** mode/#asterisk [+o blitzrage] by ChanServ |
| 13:20.00 | dwayne | eduzimrs, just a guess, but it is probably not a good idea to bring your database down while using realtime |
| 13:22.41 | *** join/#asterisk adeel (~adeel@184.175.36.92) |
| 13:22.43 | eduzimrs | ok, but im doing tests, it should be in cache no? when rtcachefriends=yes is set. |
| 13:24.18 | eduzimrs | dwayne: my sip clients shouldnt get disconnect from * |
| 13:50.31 | *** join/#asterisk krish|wired-in (~krish|wir@183.83.188.83) |
| 13:50.46 | krish|wired-in | guys, does this work - https://issues.asterisk.org/jira/browse/ASTERISK-3435 |
| 13:51.52 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 13:51.58 | krish|wired-in | how can I get the uptime in unix format |
| 14:01.52 | krish|wired-in | guys, how to query dialplan variables |
| 14:01.56 | krish|wired-in | like ${EPOCH} |
| 14:02.51 | *** join/#asterisk dlynes_laptop (~dlynes@bas8-hamilton14-3096452550.dsl.bell.ca) |
| 14:05.06 | *** join/#asterisk _omer (~omer@182.185.220.157) |
| 14:05.19 | _omer | is there any specific room for a2billing ? |
| 14:08.24 | _omer | http://212.108.128.50/ <---- how to add such kind of POPUP window security on A2Billing Admin Folder .... |
| 14:20.02 | *** join/#asterisk brdude (~brdude@c-24-5-194-184.hsd1.ca.comcast.net) |
| 14:23.25 | _omer | anyone ??? |
| 14:42.29 | *** join/#asterisk cyford (~cyford@c-24-99-17-185.hsd1.ga.comcast.net) |
| 14:47.10 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
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| 14:50.58 | *** part/#asterisk dlisenby (4bb63d33@gateway/web/freenode/ip.75.182.61.51) |
| 14:52.16 | simplydrew | does anyone have experience with working with chan_sccp? I have it installed, but am having some issues |
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| 15:34.53 | *** join/#asterisk k3asd` (~k3asd@static-94-32-127-180.clienti.tiscali.it) |
| 15:35.20 | k3asd` | hi |
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| 15:55.24 | _omer | http://212.108.128.50/ <---- how to add such kind of POPUP window security on A2Billing Admin Folder .... |
| 15:55.43 | *** join/#asterisk freeedrich| (~eeePC@hansaserver.de) |
| 15:56.03 | WIMPy | _omer: Connection refused |
| 15:56.45 | WIMPy | For popups I use System(). |
| 16:02.25 | mtbf | I'm trying to troubleshoot some issue with app_rxfax on some old box with 1.2.0, while analysing logs I found out, since some day the only message in the log related to the app_rxfax is : app_rxfax.c: Got hangup , no configuration files were modified since that time, anyone saw simmilar problem? |
| 16:04.58 | *** join/#asterisk MarKsaitis (~MarKsaiti@cpc4-rdng22-2-0-cust646.15-3.cable.virginmedia.com) |
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| 16:16.54 | *** part/#asterisk Cain (~Geek@unaffiliated/cain) |
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| 16:47.02 | p3nguin | simplydrew: What's wrong with it today? |
| 16:47.21 | HenriqueAtila | Hello :) |
| 16:47.53 | simplydrew | p3nguin: I've checked and rechecked everything, but when I actually use the command you recommended of "module load chan_sccp.so" I get an error of "WARNING[771]: loader.c:446 load_dynamic_module: Error loading module 'chan_sccp.so': /usr/lib/asterisk/modules/chan_sccp.so: undefined symbol: manager_event" |
| 16:48.26 | pabelanger | what version of Asterisk? |
| 16:48.34 | p3nguin | Okay, so that's still the same problem as before. Did you try using either of the other two branches instead of the stable branch? |
| 16:48.59 | simplydrew | p3nguin: I wasn't sure how exactly to go about that, but I remembered you saying something about it |
| 16:48.59 | p3nguin | I suggested you use stable branch 3.0, but there is dev branch 3.1 and dev branch 4.0. |
| 16:49.11 | simplydrew | p3nguin: how would I go to the dev branch of 4.0? |
| 16:49.49 | simplydrew | pabelanger: 1.6.0.26 |
| 16:50.17 | p3nguin | svn co https://chan-sccp-b.svn.sourceforge.net/svnroot/chan-sccp-b/branches/V3.1/ chan-sccp-b_V3.1 |
| 16:50.36 | p3nguin | and |
| 16:50.42 | p3nguin | svn co https://chan-sccp-b.svn.sourceforge.net/svnroot/chan-sccp-b/trunk chan-sccp-b_V4.0 |
| 16:50.51 | p3nguin | for those two branches |
| 16:50.59 | pabelanger | my bad, I just noticed it was chan_sccp |
| 16:51.06 | HenriqueAtila | What distro is better for run asterisk? |
| 16:51.20 | p3nguin | henriqueatila: Spin the wheel. |
| 16:51.26 | simplydrew | okay, I'll give 4.0 a try. should I go back and remove the previous version? |
| 16:52.01 | p3nguin | simplydrew: You only need to remove it if you want to. When you install the new one, it will overwrite the existing files. |
| 16:52.13 | simplydrew | p3nguin: gotcha, okay. just wanted to confirm |
| 16:52.42 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 16:55.18 | simplydrew | p3nguin: after doing ./configure, make clean, I did a make install and get these errors: http://pastebin.com/Z7aWLtkY |
| 16:57.09 | p3nguin | I guess I need to install some asterisk testing instances so I can see what works where for me. |
| 16:57.32 | p3nguin | I use Asterisk 1.4 branch and the stable 3.0 branch doesn't give me any problems. |
| 16:57.53 | simplydrew | hmm |
| 17:06.16 | pabelanger | simplydrew: you are missing the Asterisk header files |
| 17:06.53 | simplydrew | pabelanger: how would I go about resolving that? |
| 17:07.14 | pabelanger | download asterisk |
| 17:09.55 | simplydrew | pabelanger: okay, I assume i need to go up to 1.8 since I'm currently on 1.6? |
| 17:10.11 | simplydrew | wasn't sure of the best safe way to do that without disturbing anything |
| 17:10.39 | pabelanger | how did you install asterisk? |
| 17:11.11 | simplydrew | well, this is a trixbox install, so it technically did it |
| 17:11.50 | pabelanger | so, you need to figure out how trixbox installs the header files, or get the source code for that version of Asterisk installed |
| 17:13.38 | simplydrew | those would be located in /usr/include/asterisk, correct? it appears that a lot are in there. not sure what I'm looking at in there though. lots of .h files |
| 17:13.52 | pabelanger | yes, .h are header files |
| 17:19.56 | *** join/#asterisk Cain (~Geek@unaffiliated/cain) |
| 17:40.27 | *** join/#asterisk greenwolf (penguin@cpe-74-77-221-5.buffalo.res.rr.com) |
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| 17:52.44 | carrar | Y*A*W*N |
| 17:54.37 | *** join/#asterisk Ashutto (~Ashetic@2001:b00:7c00:600:6ef0:49ff:fe7d:3d35) |
| 17:54.41 | Ashutto | Hello |
| 17:57.52 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 17:58.31 | Ashutto | i'm behind several nat levels i cannot control, my only "public" interface free from nat&firewalling is IPv6 only. Is it possible to have a working asterisk in this conditions? |
| 17:59.40 | WIMPy | Sure. But using SIP to the outside will at least be tricky. |
| 18:00.01 | *** join/#asterisk Assuero (~Assuero@187.127.163.55) |
| 18:11.28 | p3nguin | I'd still like to know why when I get voice mail, I get msg0000 in the voicemail directory, but msg0001 is the file that is emailed. There is always an offset of 1 between the message files in asterisk and the message file that is emailed. |
| 18:16.02 | ChannelZ | it's wrong? |
| 18:16.17 | p3nguin | Uh yeah it's wrong. |
| 18:16.38 | samandiriel | but it feels so right...! |
| 18:16.40 | ChannelZ | I mean it's actually mailing you the wrong file, it's not just the name in the email? |
| 18:17.03 | p3nguin | The content of the sound file is right, but the name of the file is wrong. |
| 18:18.02 | p3nguin | If I get a voicemail, it will be recorded as msg0000.wav, for example. When completed, it will be emailed to me as msg0001.wav rather than the correct file name. |
| 18:18.53 | p3nguin | It has been this way through several versions. I've asked about it multiple times, but no one has ever been able to tell me how to fix it. |
| 18:19.13 | p3nguin | I'll patch it locally if someone can tell me where the problem lies. |
| 18:19.33 | ChannelZ | I've never had that. |
| 18:19.48 | p3nguin | I'm looking at app_voicemail.c right now hoping something pops out at me. |
| 18:19.56 | ChannelZ | I'd have to look at the source, I wonder if you have an old .txt file lying around causing it to get confused |
| 18:20.01 | samandiriel | why does it matter, so long as it's consistent? |
| 18:20.10 | p3nguin | because it's wrong. |
| 18:20.14 | ChannelZ | samandiriel: don't even ask |
| 18:20.24 | *** join/#asterisk corretico (~luis@201.201.44.82) |
| 18:20.26 | ChannelZ | Are you using database storage? |
| 18:20.39 | p3nguin | no |
| 18:23.31 | p3nguin | I guess I need to check others' mail boxes before I make the assumption that it is broken globally. |
| 18:27.06 | *** join/#asterisk corretico (~luis@201.201.44.82) |
| 18:28.41 | p3nguin | Okay, I found a user with no mails in the INBOX. I called and left a message. I see msg0000.WAV and msg0000.txt in the INBOX. |
| 18:29.07 | p3nguin | Now to determine what file name is emailed. |
| 18:30.27 | ChannelZ | I can't even trick mine on purpose so far. |
| 18:33.04 | anonymouz666 | using Local/ channels as a realtime queue members is weird |
| 18:33.16 | anonymouz666 | it does not enter in the context you write in Local/ |
| 18:34.19 | p3nguin | It works fine from the conf; I've never messed with realtime. |
| 18:35.15 | anonymouz666 | indeed, it works fine from the conf. |
| 18:37.10 | ChannelZ | Well now I've screwed up my whole mailbox |
| 18:37.17 | p3nguin | :( |
| 18:37.41 | ChannelZ | well in so much as I deleted all my messages but there are still files there |
| 18:37.58 | ChannelZ | I confused it by making empty .txt files |
| 18:38.16 | ChannelZ | there now it's right |
| 18:39.24 | ChannelZ | You mentioned on the test box it recorded msg0000.WAV but before you said the attachments were named .wav |
| 18:39.51 | p3nguin | I used msg0000.wav as the example to describe the problem. I'm using WAV only. |
| 18:40.56 | p3nguin | I seem to remember a time where I actually used wav and WAV, and both files would be emailed with a name offset of 1. |
| 18:44.43 | ChannelZ | dunno. If your INBOX is completely empty of files and you do a test and it's offset, I have no idea what is happening |
| 18:46.54 | *** join/#asterisk SuPrSluG (~SuPrSluG@8.22.96.106) |
| 18:47.29 | p3nguin | I'm still waiting to find out what file was emailed. msg0000.txt and msg0000.WAV are in the INBOX, but I bet msg0001.WAV is what shows up in email, since that is what happens to mine every time, regardless of the msg index number. |
| 18:48.34 | p3nguin | I don't even remember if there has ever been a version I used that didn't do this. |
| 18:49.01 | p3nguin | When I first noticed it, I rolled back a few asterisk versions and it was still doing it, so I assumed it was doing it all along and I just didn't pay attention. |
| 18:52.11 | *** join/#asterisk m29poff (~m29poff@125.213.206.77.rev.sfr.net) |
| 18:58.55 | *** join/#asterisk greenwolf (~chatzilla@cpe-74-77-221-5.buffalo.res.rr.com) |
| 19:05.14 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
| 19:08.29 | *** join/#asterisk rotten777 (~matthew@fl-67-233-23-154.dhcp.embarqhsd.net) |
| 19:11.54 | rotten777 | anyone awake? |
| 19:13.50 | WIMPy | *zzZzZZ* |
| 19:14.01 | rotten777 | darn :-X |
| 19:15.45 | rotten777 | i'm just a beginner with some stupid questions... what harm would it do to be awake? ;) |
| 19:16.37 | rotten777 | i have a DID at a voip provider, a single sip phone (polycom ip 330), and a single debian based server. trying to get dialing working and evidently i'm not very smart |
| 19:17.47 | *** join/#asterisk Hanumaan (~Hanumaan@dslb-094-216-161-171.pools.arcor-ip.net) |
| 19:20.00 | WIMPy | That was not a question. |
| 19:20.31 | WIMPy | But I've got one: Why do you want to add a server between your ITSP and your phone? |
| 19:20.46 | rotten777 | I don't but I can't get broadvoice to work with the polycom ip 330 |
| 19:21.02 | rotten777 | i'm brand spanking new to voip but not to linux or networking by any means |
| 19:21.18 | rotten777 | so i'm probably just reading the conf instructions wrong in the polycom gui |
| 19:23.03 | p3nguin | New message 1 in mailbox. Attached is msg0001.WAV. So it's not just my box. |
| 19:34.55 | ChannelZ | is your email command doing something insane? |
| 19:35.42 | p3nguin | mailcmd = /usr/bin/msmtp -t |
| 19:35.52 | ChannelZ | hm. |
| 19:35.53 | ChannelZ | shrugs |
| 19:36.02 | p3nguin | It's beyond me. |
| 19:44.07 | *** join/#asterisk timahvo1 (~rogue@41.223.57.78) |
| 19:44.57 | *** join/#asterisk garymc (~chatzilla@host109-155-155-5.range109-155.btcentralplus.com) |
| 19:46.41 | *** join/#asterisk SuPrSluG (~SuPrSluG@8.22.96.106) |
| 19:50.20 | rotten777 | is there a way if my polycom or softphone are even registering on my asterisk box? |
| 19:50.33 | p3nguin | Is there a way for it to do what? |
| 19:50.54 | rotten777 | is there a way to check the active sip devices on asterisk |
| 19:51.05 | p3nguin | sip show peers |
| 19:52.20 | rotten777 | hmm ok they both show up |
| 19:52.37 | rotten777 | i can't dial either extension though... i hate being the noob |
| 19:52.56 | p3nguin | What context did you assign for the phones? |
| 19:53.36 | rotten777 | [testing] |
| 19:53.36 | rotten777 | exten=>1001,1,Dial(SIP/byrdits) |
| 19:53.37 | rotten777 | exten=>1002,1,Dial(SIP/softphone) |
| 19:53.46 | rotten777 | is that what you're asking? |
| 19:54.07 | p3nguin | In sip.conf, you assigned a context for each phone. What context did you assign? |
| 19:54.20 | rotten777 | testing |
| 19:54.24 | p3nguin | context= something |
| 19:54.55 | p3nguin | In sip show peers, does byrdits show an IP address? |
| 19:55.02 | rotten777 | no |
| 19:55.08 | p3nguin | Then it isn't registered. |
| 19:55.10 | p3nguin | And you cannot call it. |
| 19:55.37 | rotten777 | Name/username Host Dyn Nat ACL Port Status |
| 19:55.37 | rotten777 | byrdits/byrdits (Unspecified) D 5060 Unmonitored |
| 19:55.37 | rotten777 | softphone/softphone 192.168.77.2 D 5060 Unmonitored |
| 19:55.37 | rotten777 | 2 sip peers [Monitored: 0 online, 0 offline Unmonitored: 2 online, 0 offline] |
| 19:55.46 | rotten777 | so it isn't seeing my hardware phone... |
| 19:56.22 | p3nguin | byrdits isn't registered. Can you use byrdits to call softphone by extension 1002? |
| 19:56.30 | rotten777 | i can't call either from either |
| 19:56.40 | rotten777 | do i need to set the outbound proxy on the polycom to the ip of the asterisk box? this is all on a private subnet btw.. no nat or anything |
| 19:57.15 | p3nguin | You have to set the server address, but I don't know if you have to specify it again in the outbound proxy field. I don't think you do. |
| 19:57.39 | rotten777 | ok under lines i have line 1 with the address user id and password completed |
| 19:58.19 | rotten777 | ahh i didn't have it in server 1 or server 2... let me try server 1 with the ip |
| 19:58.25 | ChannelZ | can't figure out why he keeps getting "Correct auth, but based on stale nonce" from Zoiper lately |
| 19:59.45 | rotten777 | [Sep 10 15:59:11] NOTICE[25588]: chan_sip.c:21763 handle_request_register: Registration from '<sip:192.168.77.21@192.168.77.21>' failed for '192.168.77.22' - No matching peer found |
| 19:59.51 | *** part/#asterisk didnot (~didnot@unaffiliated/didnot) |
| 19:59.53 | ChannelZ | ta-da! |
| 20:00.05 | rotten777 | what is that? |
| 20:00.19 | p3nguin | Did you specify your sip user name in two places? |
| 20:00.31 | p3nguin | address and auth name |
| 20:00.54 | rotten777 | address was the ip i thought |
| 20:01.00 | rotten777 | but i am wrong |
| 20:01.01 | rotten777 | haha |
| 20:01.09 | p3nguin | I'm pretty sure address is your sip name. |
| 20:01.20 | rotten777 | ok putting in the username in address and auth name as the username as well |
| 20:01.31 | rotten777 | yeah i'm a complete noob to this. thanks for the help btw |
| 20:02.28 | rotten777 | Name/username Host Dyn Nat ACL Port Status |
| 20:02.28 | rotten777 | byrdits/byrdits 192.168.77.22 D 5060 Unmonitored |
| 20:02.28 | rotten777 | softphone/softphone 192.168.77.2 D 5060 Unmonitored |
| 20:02.28 | rotten777 | 2 sip peers [Monitored: 0 online, 0 offline Unmonitored: 2 online, 0 offline] |
| 20:02.36 | p3nguin | That's good! |
| 20:02.56 | rotten777 | ok when dialing from the hardware phone to the softphone i get circuit busy tone |
| 20:03.17 | p3nguin | and the other way? |
| 20:03.35 | rotten777 | no ring or circuit busy or anything dead air |
| 20:03.59 | p3nguin | Are those the correct addresses? |
| 20:04.04 | ChannelZ | probably dialplan hasn't even sent the dial |
| 20:04.05 | p3nguin | for the phone |
| 20:04.07 | rotten777 | oh snap wait |
| 20:04.08 | p3nguin | s |
| 20:04.09 | rotten777 | found it |
| 20:04.38 | rotten777 | sweet! it works |
| 20:04.45 | rotten777 | both ways |
| 20:04.52 | p3nguin | What was the problem? |
| 20:04.58 | ChannelZ | congrats - now you're bi! |
| 20:05.19 | rotten777 | softphone wasn't placing call it was just synth tone & not actually talking to asterisk |
| 20:05.47 | rotten777 | well crap |
| 20:05.52 | rotten777 | now get circuit busy again |
| 20:06.34 | rotten777 | [Sep 10 16:05:36] WARNING[25633]: app_dial.c:1747 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown) |
| 20:06.50 | p3nguin | The device seems to be gone. |
| 20:08.42 | rotten777 | ok now that's reliable |
| 20:08.47 | rotten777 | not sure why it failed the once |
| 20:09.02 | rotten777 | now if i can figure out how to use my broadvoice account to dial to the ptsn |
| 20:09.06 | p3nguin | network issues? |
| 20:09.30 | rotten777 | i guess... the asterisk box, my softphone and my hardphone are all in the same gigabit switch |
| 20:09.36 | rotten777 | nothing layer 3 in between them |
| 20:09.48 | p3nguin | That's easy. Start by rethinking context hierarchy. |
| 20:10.07 | rotten777 | any pointers? |
| 20:10.10 | p3nguin | Create a new sip peer for the ITSP. Create a new context for outgoing calls via that peer. |
| 20:10.10 | rotten777 | or places to read up |
| 20:10.12 | *** join/#asterisk Fritz09 (~Adium@pop1-2174.catv.wtnet.de) |
| 20:10.17 | p3nguin | ~book |
| 20:10.17 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://ofps.oreilly.com/titles/9780596517342/ or ~buybook |
| 20:10.27 | p3nguin | There's a huge book with all the details. |
| 20:10.42 | p3nguin | Or I can share some examples with you. |
| 20:10.53 | rotten777 | yeah examples sound good |
| 20:11.48 | rotten777 | i basically have the 1 broadvoice did via sip and 1 server and 1 hardware phone |
| 20:11.49 | p3nguin | example sip.conf: http://pastebin.com/tER2jGnY |
| 20:11.59 | rotten777 | k reading now |
| 20:14.23 | p3nguin | example extensions.conf: http://pastebin.com/Piqv4Egj |
| 20:15.00 | rotten777 | ok to get the asterisk server to talk to itsp over nat do i need to specify the external ip? |
| 20:15.08 | p3nguin | In these examples, voipms is the peer I'm using for outbound calls. |
| 20:15.12 | p3nguin | ~sipnat |
| 20:15.12 | infobot | [~sipnat] Quick guide on configuring Asterisk + SIP behind NAT: http://www.aocomputing.net/?p=3 otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions . Typically, you MUST configure these settings with appropriate values in the [general] section of sip.conf: nat, canreinvite, externhost or externip, and localnet. |
| 20:16.00 | p3nguin | Which asterisk version are you using? |
| 20:16.32 | rotten777 | latest with ubuntu server |
| 20:16.36 | rotten777 | let me check |
| 20:16.39 | p3nguin | core show version |
| 20:16.50 | rotten777 | 1.6.2.9 |
| 20:17.12 | WIMPy | That is not very new. |
| 20:18.27 | p3nguin | 15 months old |
| 20:18.42 | p3nguin | Lots of bug fixes in 15 months. |
| 20:19.10 | rotten777 | its ubuntu natty |
| 20:19.13 | rotten777 | they're that far off eh? |
| 20:19.48 | p3nguin | I don't know when Ubuntu Nasty was released. |
| 20:23.39 | rotten777 | alright i'm kind of confused... |
| 20:23.45 | rotten777 | i'm reading the dialplan basics |
| 20:23.54 | rotten777 | i have the broadvoice peer loaded |
| 20:24.03 | *** join/#asterisk dlisenby (4bb63d33@gateway/web/freenode/ip.75.182.61.51) |
| 20:24.23 | p3nguin | Did you use a register statement for broadvoice? |
| 20:25.01 | rotten777 | [broadvoice] |
| 20:25.02 | rotten777 | type=peer |
| 20:25.02 | rotten777 | disallow=all |
| 20:25.02 | rotten777 | allow=ulaw |
| 20:25.02 | rotten777 | contet=from-broadvoice |
| 20:25.02 | rotten777 | dtmfmode=rfc2833 |
| 20:25.04 | rotten777 | host=sip.broadvoice.com |
| 20:25.07 | dlisenby | p3nguin, thanks for your help the other night with Callcentric. I went out and acquired a sip trunk with Nextiva. |
| 20:25.08 | rotten777 | nat=yes |
| 20:25.09 | p3nguin | Don't flood us. |
| 20:25.13 | p3nguin | ~pb |
| 20:25.13 | infobot | [~pb] 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.us , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ ; or install pastebinit with yum or aptitude. |
| 20:25.48 | p3nguin | Broadvoice is not behind nat, so nat=yes is wrong. |
| 20:26.27 | rotten777 | ok the phone is do i need to append that? |
| 20:27.09 | p3nguin | If they are on the same network segment as asterisk, you can probably omit the nat line in the phone entries. |
| 20:27.20 | rotten777 | gotcha so they should be fine |
| 20:27.33 | p3nguin | But you need to finish configuring your broadvoice peer entry. |
| 20:27.53 | p3nguin | You didn't specify your username or password for them, so you won't be able to authenticate when sending calls. |
| 20:28.02 | p3nguin | See my example. |
| 20:28.49 | p3nguin | And you need to send a registration unless you are doing IP auth, which I doubt you are. |
| 20:30.30 | rotten777 | looking at the register statement in that example.. what is the data prior to the dns entry of the sip server |
| 20:30.39 | rotten777 | register => 105245:2FsuonGrOuq4r@chicago.voip.ms |
| 20:31.01 | p3nguin | register => username:password@host |
| 20:31.53 | rotten777 | ok so i have to fille the username variable, the secret variable and the register variable? |
| 20:32.11 | p3nguin | Where did variables come into play? |
| 20:32.43 | rotten777 | gah yeah i'm reading it as a script or something i mean the statements for username secret and register |
| 20:32.46 | p3nguin | If you are not doing IP auth, you have to register (by using a register statement) to be able to receive calls. |
| 20:33.10 | p3nguin | And several providers require that you are registered before you can send calls. |
| 20:33.20 | rotten777 | they hold redundant data and i wanted to make sure that i wasn't adding username and secret statements in there if it was only needed in register |
| 20:33.25 | rotten777 | gotcha |
| 20:33.54 | p3nguin | The peer entry below is used to match the calls coming in as well as "route" calls going out. |
| 20:34.15 | p3nguin | username/password will be required in it for sending calls out, because they will want authentication. |
| 20:35.36 | rotten777 | ok so i need the 2 peers in sip, 1 for the itsp, 1 for the phone... i think i'm good. let me see if the peers show up in asterisk |
| 20:35.56 | p3nguin | And the register statement in the general section |
| 20:36.10 | p3nguin | You can see if you are registered to the ITSP with sip show registry. |
| 20:36.27 | rotten777 | 0 sip regs |
| 20:37.34 | p3nguin | The register statement has to be in the [general] section of sip.conf, before any peer entries, and before [authentication] if it exists. |
| 20:38.20 | rotten777 | ok it is showing in the regs now |
| 20:38.33 | p3nguin | You forgot to run sip reload? |
| 20:38.40 | rotten777 | i restarted the asterisk service |
| 20:38.49 | p3nguin | That's a lot of work just to reload sip. |
| 20:38.52 | p3nguin | sip reload |
| 20:38.55 | rotten777 | haha ok |
| 20:39.10 | p3nguin | And when you changed extensions.conf, use dialplan reload. |
| 20:39.21 | rotten777 | ok |
| 20:39.24 | p3nguin | s/changed/change/ |
| 20:40.04 | p3nguin | So you've registered to the ITSP, and you're ready to look at the dial plan? |
| 20:40.09 | rotten777 | yup! |
| 20:40.32 | p3nguin | Have you created your [from-broadvoice] context in extensions.conf? |
| 20:40.54 | rotten777 | doing that now |
| 20:41.37 | *** join/#asterisk SuPrSluG (~SuPrSluG@8.22.96.106) |
| 20:42.00 | rotten777 | is that the DID# on the exten statement on the from-ipkall ? |
| 20:42.31 | p3nguin | Yes. My ITSPs send calls to my phone number, so the extension is the DID number. |
| 20:43.24 | rotten777 | gotcha. the "30" behind the sip address. what does this signify |
| 20:43.39 | p3nguin | SIP/jack,30? |
| 20:43.48 | rotten777 | yes |
| 20:43.51 | p3nguin | 30 second timeout for dialing a phone named "jack" |
| 20:44.11 | p3nguin | After 30 seconds, it'll progress to the next line if there is one. |
| 20:44.42 | rotten777 | ok |
| 20:45.32 | *** join/#asterisk DrDigi (~mmurphy@50-73-49-97-static.hfc.comcastbusiness.net) |
| 20:45.32 | rotten777 | so 4 exten statements |
| 20:45.52 | p3nguin | where? |
| 20:45.59 | rotten777 | from-broadvoice |
| 20:46.18 | p3nguin | You probably just need one extension, but that one extension will have several priorities. |
| 20:46.28 | rotten777 | i see some statements with verbose playback congestion playtones... are these needed? |
| 20:47.17 | p3nguin | I have no idea what I was doing there. |
| 20:47.27 | rotten777 | lol k |
| 20:47.30 | p3nguin | But now I have to go look at my actual dial plan to see if I am doing that in there too. |
| 20:48.25 | p3nguin | Oh, I see what I was doing. |
| 20:48.51 | p3nguin | I was trying out both ways of providing congestion tones and forgot to decide on one when I pasted that. |
| 20:49.02 | rotten777 | gotcha |
| 20:49.16 | p3nguin | Sometimes I do silly things like that. |
| 20:49.19 | rotten777 | ok i have the from-broadvoice finished |
| 20:49.33 | rotten777 | hey at least you can understand what you're looking at |
| 20:50.35 | rotten777 | isn-outbound is the context for the outgoing to the itsp |
| 20:50.36 | rotten777 | ? |
| 20:50.54 | p3nguin | If you didn't figure it out already, I am defining any used phone number explicitly and then any number I am not defining will end up on the pattern _X. to play a message saying the number is not in service. |
| 20:50.59 | p3nguin | ~isn |
| 20:50.59 | infobot | methinks isn is ITAD Subscriber Number (see 'itad'). An ISN is a method of dialing SIP URI's via a standard keypad on a telephone. Because of the alphanumeric nature of SIP URIs, it is difficult to dial them via the keypad on your phone. The use of ISN numbers simplifies this by utilizing DNS lookups to map the ISN number to a domain. See http://www.freenum.org for more information. |
| 20:51.24 | rotten777 | ah |
| 20:51.30 | p3nguin | You don't need ISN for broadvoice. |
| 20:52.10 | rotten777 | ahh voipms-outbound? |
| 20:52.20 | p3nguin | That's where I dial out to the PSTN. |
| 20:52.31 | rotten777 | i think i'm able to see the logic in the statements a lot better now |
| 20:52.56 | rotten777 | so you have an optional 1 at the beginning of the number... it adds the local area code if needed |
| 20:53.03 | p3nguin | I'm supporting 11-digit, 10-digit, 7-digit in the local area code, plus international dialing. |
| 20:54.15 | rotten777 | wow asterisk is a lot more powerful than i thought |
| 20:54.20 | p3nguin | So if you dial 1+area code+ 7 digits, it matches. If you don't dial the 1 but just the area code and 7 digits, it matches. If you just dial 7 digits, it dials those 7 digits in your local area code that you write into that line. |
| 20:54.39 | rotten777 | my plan with broadvoice is unlimited in the state of FL |
| 20:54.56 | rotten777 | i'll have to find a list of FL area codes and put it in here... anything outside of florida i'll use my cell |
| 20:55.44 | rotten777 | hmm ok dialplan reloaded |
| 20:55.50 | rotten777 | i have the broadvoice-outbound in there |
| 20:55.59 | rotten777 | i don't get anything when i dial |
| 20:56.02 | p3nguin | http://www.50states.com/areacodes/florida.htm |
| 20:56.12 | p3nguin | 16 of them |
| 20:57.18 | rotten777 | cool |
| 20:57.23 | p3nguin | You probably didn't configure the rest of the context hierarchy well enough to match the extensions in your broadvoice-outbound context. |
| 20:57.38 | p3nguin | I prefer to assign a context 'phones' to any phone. |
| 20:58.02 | p3nguin | Then in the phones context in extensions.conf, I use includes to specify what those phones can do. |
| 20:58.30 | p3nguin | See line 157-161 in my extensions.conf example. |
| 20:59.02 | rotten777 | ahh my internal is a different name and i have the outbound now in there |
| 20:59.41 | p3nguin | You have to build a good hierarchy so that no incoming calls ever have the ability to dial back outbound. |
| 21:00.08 | p3nguin | Otherwise, you run the risk of someone running out your prepaid balance or running up a huge bill if you pay for calls you already made. |
| 21:00.40 | rotten777 | oh nice haha |
| 21:00.52 | rotten777 | ok i've got that stuff done and reloaded the dialplan |
| 21:00.57 | p3nguin | In most cases, phones will never have a reason to call your own DIDs, so they don't have a reason to be able to call your inbound context. |
| 21:01.09 | rotten777 | yeah |
| 21:01.29 | p3nguin | In cases where you do have to call your own numbers, we re-think the way to accomplish it. |
| 21:02.07 | p3nguin | Okay, so includes are in place and calling a phone number does what? |
| 21:02.13 | rotten777 | dead air |
| 21:02.19 | rotten777 | i did show channels |
| 21:02.23 | p3nguin | core set verbose 4 |
| 21:02.28 | rotten777 | k |
| 21:02.33 | p3nguin | call again. |
| 21:02.57 | rotten777 | > doing dnsmgr_lookup for 'sip.broadvoice.com' |
| 21:03.03 | rotten777 | sticks there |
| 21:03.17 | rotten777 | broadvoice does say to use a proxy... i'm assuming i should have added that |
| 21:03.47 | rotten777 | outbound proxy server |
| 21:03.51 | p3nguin | They didn't offer you a configuration sample so you'd know how to use their service? |
| 21:04.18 | rotten777 | checking right now |
| 21:04.40 | p3nguin | If you didn't see any call progression appear in the console when verbose was turned up, the next thing to do is check the sip debug to see why there is no call starting. sip set debug on |
| 21:11.18 | rotten777 | they have an asterisk conf example of sorts on the site |
| 21:12.03 | p3nguin | Is it where I can see it? |
| 21:12.33 | rotten777 | http://pastebin.com/SinFTbmE |
| 21:14.52 | rotten777 | http://www.broadvoice.com/support_install_asterisk.html |
| 21:15.22 | p3nguin | So they want you to register using a proxy that is the same as the host. How weird. |
| 21:15.42 | p3nguin | proxy sip.broadvoice.com and host sip.broadvoice.com |
| 21:15.49 | p3nguin | Does not make sense to me. |
| 21:15.53 | rotten777 | yeah i have the feeling the service might suck but it was a quick google... nobody i know has any idea about voip |
| 21:16.03 | p3nguin | ~itsp-us |
| 21:16.15 | p3nguin | ~itsplist-us |
| 21:16.15 | infobot | Here are some popular ITSPs (USA) starting with the more respected ones: http://www.teliax.com , http://www.voicepulse.com/connect/ , http://www.broadvoice.com , http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net , http://voip.ms and http://flowroute.com |
| 21:16.27 | rotten777 | yeah now the sip registration is failing too with their instructions |
| 21:16.32 | rotten777 | ugh |
| 21:16.59 | p3nguin | I'd put it back like it was and then check the sip debug to see what is going wrong. |
| 21:17.44 | p3nguin | They may require you to use fromdomain and fromuser in the peer entry. |
| 21:17.55 | rotten777 | yeah i have those there |
| 21:18.11 | rotten777 | fromdomain is sip.broadvoice.com fromuser is my # |
| 21:18.24 | rotten777 | but username is also my # |
| 21:19.39 | p3nguin | sip set debug on |
| 21:19.41 | p3nguin | make a call. |
| 21:20.02 | rotten777 | [Sep 10 17:19:47] NOTICE[26257]: chan_sip.c:11696 sip_reg_timeout: -- Registration for '<8636584102>@sip.broadvoice.com@sip.broadvoice.com' timed out, trying again (Attempt #19) |
| 21:20.02 | rotten777 | <PROTECTED> |
| 21:20.03 | p3nguin | When it fails, hang up. Copy everything and paste in the pastebin. |
| 21:20.07 | rotten777 | ahh ok |
| 21:20.36 | p3nguin | Fix the register statement first. |
| 21:20.51 | p3nguin | They'll most likely want you to be registered in order to send calls. |
| 21:22.52 | rotten777 | [Sep 10 17:22:32] NOTICE[26257]: chan_sip.c:18392 handle_response_register: Outbound Registration: Expiry for sip.broadvoice.com is 30 sec (Scheduling reregistration in 24 s) |
| 21:23.02 | rotten777 | it seems like the hosts file addition they asked for killed the sip registration success |
| 21:23.22 | p3nguin | I have no idea why they would have asked you to do that. |
| 21:23.37 | p3nguin | But that message does not say it failed. |
| 21:23.47 | p3nguin | It says the expiration is in 30 seconds. |
| 21:24.02 | rotten777 | ok i tried to make a call and get nothing |
| 21:24.18 | p3nguin | If you have sip debug enabled, you should see something. |
| 21:24.23 | rotten777 | request sent is the state it is in |
| 21:24.26 | rotten777 | yeah i see a ton of stuff |
| 21:24.32 | rotten777 | i still have the verbose to 4 |
| 21:25.44 | rotten777 | http://pastebin.com/ua1gUu3t |
| 21:27.02 | p3nguin | It says it is transmitting nat to a public IP address. Did you remember to put nat=no in your broadvoice peer? |
| 21:27.35 | rotten777 | i just added nat=no |
| 21:27.37 | rotten777 | again |
| 21:27.44 | rotten777 | reload and looks like it quieted down |
| 21:28.05 | rotten777 | now the same thing |
| 21:28.05 | p3nguin | It's doing a lot of REGISTER stuff. I'd undo whatever I did to make it not register. |
| 21:28.19 | p3nguin | You had it registering before, so revert back. |
| 21:29.21 | p3nguin | I've never seen an ITSP that knows how to configure asterisk for the end user. It's kind of ridiculous. |
| 21:29.30 | ChannelZ | If it ain't broke, try harderf |
| 21:29.46 | rotten777 | i dont know what else i can change back |
| 21:29.52 | rotten777 | i thought it was back how we had it |
| 21:30.18 | p3nguin | You said something about /etc/hosts. Did you undo the changes there? You changed the register statment. Did you undo that, too? |
| 21:30.27 | rotten777 | yes i removed the hosts entry |
| 21:30.38 | *** part/#asterisk rotten777 (~matthew@fl-67-233-23-154.dhcp.embarqhsd.net) |
| 21:30.42 | *** join/#asterisk rotten777 (~matthew@fl-67-233-23-154.dhcp.embarqhsd.net) |
| 21:30.48 | rotten777 | whoops |
| 21:30.49 | rotten777 | lol |
| 21:31.49 | rotten777 | does the register statement whitespace matter? |
| 21:32.17 | p3nguin | The only whitespace should be around the => |
| 21:32.21 | *** join/#asterisk X-Rob (~Rob@eth2083.qld.adsl.internode.on.net) |
| 21:32.27 | rotten777 | yeah thats where it is |
| 21:32.31 | p3nguin | register => user:pass@host |
| 21:32.44 | p3nguin | or sometimes... |
| 21:32.47 | p3nguin | register => user:pass@host/extension |
| 21:33.15 | rotten777 | it all looks fine to me |
| 21:33.28 | *** join/#asterisk rdegges (~rdegges@69.164.197.143) |
| 21:33.50 | p3nguin | After sip reload, sip show registry still says request sent? |
| 21:34.13 | rotten777 | ahh registered now |
| 21:34.17 | p3nguin | It usually only takes a few seconds for the registration to succeed and that status changes to registered. |
| 21:34.35 | rotten777 | ok so it shows registered now |
| 21:34.52 | p3nguin | That should quiet some of the REGISTER packets in the sip debug. Enable sip debug and make a call. |
| 21:36.42 | rotten777 | http://pastebin.com/tVaKNtDX |
| 21:37.40 | p3nguin | Still just a crapload of register stuff. |
| 21:37.43 | p3nguin | No call at all. |
| 21:38.48 | rotten777 | http://pastebin.com/uSc96J7t |
| 21:38.53 | rotten777 | did i screw something up here |
| 21:40.06 | p3nguin | That actually looks quite nice. |
| 21:40.18 | p3nguin | But you're missing some parts of extensions.conf. |
| 21:40.27 | rotten777 | that was just the bottom |
| 21:40.31 | p3nguin | Did you omit them for the paste, or did you leave them out? |
| 21:40.37 | rotten777 | yeah just for the paste |
| 21:40.41 | p3nguin | okay |
| 21:41.07 | p3nguin | Now your phones' entries in sip.conf... they have context=phones? |
| 21:41.26 | rotten777 | context=bits |
| 21:41.31 | rotten777 | no wonder no calls are being placed... |
| 21:41.31 | p3nguin | That's a problem. |
| 21:42.39 | rotten777 | do they need multiple context statements or just the 1 |
| 21:43.04 | p3nguin | You can only have one. |
| 21:43.14 | rotten777 | yeah they're set to context=phones |
| 21:43.21 | rotten777 | dialplan and sip reload |
| 21:43.53 | rotten777 | seems the same |
| 21:44.36 | *** join/#asterisk SuPrSluG (~SuPrSluG@8.22.96.106) |
| 21:45.08 | rotten777 | http://pastebin.com/FG1H4PkU |
| 21:45.19 | rotten777 | it is showing registered |
| 21:45.48 | p3nguin | Still just a bunch of register crap. |
| 21:45.55 | p3nguin | No calls in that paste either. |
| 21:46.47 | rotten777 | user=phone on the peer for broadvoice is fine? |
| 21:47.19 | p3nguin | You don't need to set that explicitly, but if you are seeing that info in the debug it is fine. |
| 21:48.44 | rotten777 | ok weird |
| 21:48.50 | rotten777 | softphone works fine |
| 21:49.00 | rotten777 | polycom doesn't |
| 21:51.44 | rotten777 | I don't get that. they're on the same subnet calling into the same asterisk server calling to the same number |
| 21:52.11 | p3nguin | Are the peer entries for the two phones pretty much the same? |
| 21:52.33 | rotten777 | they're exactly the same |
| 21:52.36 | rotten777 | other than the username |
| 21:56.34 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 22:01.18 | *** join/#asterisk felimwhiteley (~quassel@46.7.101.58) |
| 22:01.18 | rotten777 | bam fixed |
| 22:01.18 | rotten777 | p3nguin you're the man |
| 22:01.43 | p3nguin | What was causing the Polycom to fail? |
| 22:02.11 | rotten777 | there's 2 pages of network/sip stuff in it's gui. i removed it all and re-added everything |
| 22:02.15 | rotten777 | something was goofed up and not sure what |
| 22:02.18 | rotten777 | but it was in the polycom |
| 22:02.22 | rotten777 | not in asterisk |
| 22:04.21 | rotten777 | now i have another problem though... outbound works great but i get the broadvoice voicemail when i dial in from the outside |
| 22:04.42 | p3nguin | sip set debug on |
| 22:04.45 | p3nguin | make a call. |
| 22:05.39 | rotten777 | http://pastebin.com/GdenCYHK |
| 22:05.47 | rotten777 | i see the invite |
| 22:05.48 | rotten777 | but |
| 22:06.54 | p3nguin | They are sending to extension s rather than to your phone number. Change your register statment to include /extension on the end of it. The extension in the register needs to be your phone number. |
| 22:07.05 | rotten777 | k |
| 22:07.10 | p3nguin | I don't know what they expect you to do in the case of more than one phone number. |
| 22:07.15 | p3nguin | I hate ITSPs that do that. |
| 22:08.20 | rotten777 | hmm ok i have the /npanxx1234 on the extension |
| 22:08.32 | rotten777 | on the register statement rather |
| 22:08.50 | p3nguin | As long as it's your phone number and the extension is also your phone number, that should solve it. sip reload, make a call. |
| 22:09.21 | rdegges | Hey all. |
| 22:09.56 | rotten777 | http://pastebin.com/bp4694Dy |
| 22:10.06 | rdegges | Have any of you got experience using MeetMe under high loads with DAHDI dummy? |
| 22:10.27 | rdegges | I'm trying to find out if using dahdi dummy is a sufficient timing backend for high-volume meetme usage |
| 22:10.56 | rdegges | I'm trying to support 100+ callers per meetme room, but having a really hard time doing that currently using dahdi dummy. |
| 22:11.07 | rdegges | I'm not sure if it's dahdi dummy that's the problem, or my version of asterisk, or my hardware. |
| 22:11.15 | rdegges | I'm using 1.6.2 atm. |
| 22:11.24 | rdegges | Any advice or pointers I could use to help narrow it down a bit? |
| 22:11.30 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 22:15.03 | *** join/#asterisk Azrael808 (~peter@cpc17-walt12-2-0-cust657.13-2.cable.virginmedia.com) |
| 22:19.10 | rotten777 | p3nguin: the inbound context from broadvoice has just the phone number.. the sip invite shows "phone#@ip:port".. does that need to be reflected in the context? |
| 22:19.53 | p3nguin | No. The extension can only be the phone number. |
| 22:20.12 | p3nguin | Well, it could be anything you make it, but it does not include the IP address or the port number. |
| 22:20.30 | p3nguin | I'm still waiting to see debug of a failed call. |
| 22:22.28 | rotten777 | http://pastebin.com/s5xRMEK4 |
| 22:22.36 | rotten777 | thats straight to voicemail at the broadvoice switch |
| 22:23.00 | p3nguin | That's not a call. |
| 22:23.48 | p3nguin | I don't understand what's going on. |
| 22:24.29 | rotten777 | when i call from my cell phone to the did # i go straigh to voicemail |
| 22:24.37 | rotten777 | it doesn't come to asterisk |
| 22:24.41 | rotten777 | or to my polycom |
| 22:26.15 | p3nguin | I'm sure it's something going on with broadvoice, but I don't know what needs to be done to overcome it. I'm glad I have an ITSP that cooperates. |
| 22:30.44 | rotten777 | i disabled their voicemail and now get a busy tone when calling in |
| 22:32.55 | rotten777 | now outgoing doesn't work |
| 22:32.55 | rotten777 | http://pastebin.com/jpLKf7dL |
| 22:50.08 | *** join/#asterisk jayson_r (~jayson@cpe-071-076-046-081.sc.res.rr.com) |
| 22:57.45 | p3nguin | It looks like it is finally at least trying to start a call. |
| 22:58.04 | rotten777 | what was the tilde command you did earlier that showed itsp's |
| 22:58.13 | p3nguin | Did you forward the RTP port range in addition to the SIP port? |
| 22:58.13 | rotten777 | i'm about done with broadvoice |
| 22:58.17 | rotten777 | whats the rtp port range? |
| 22:58.18 | p3nguin | ~itsplist-us |
| 22:58.18 | infobot | Here are some popular ITSPs (USA) starting with the more respected ones: http://www.teliax.com , http://www.voicepulse.com/connect/ , http://www.broadvoice.com , http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net , http://voip.ms and http://flowroute.com |
| 22:58.46 | p3nguin | It is normally UDP 10000-20000. Check rtp.conf to verify. |
| 22:59.54 | p3nguin | Of the ones in that list, I use voip.ms and flowroute. |
| 23:01.21 | rotten777 | yeah now i'm getting errors with the sip registry |
| 23:02.07 | p3nguin | My primary is VoIP.ms, because it just works. |
| 23:04.29 | p3nguin | In the several years I've been using them, I have had very few problems. The biggest problem is typical of every service provider -- they don't know shit about how to do anything. |
| 23:06.55 | rotten777 | yeah i'm pinging their servers and somehow i get less latency to new york than i do to miami... miami is about a 3 hour drive from here... amazing |
| 23:07.07 | rotten777 | i wonder if i can get a refund for the crap services |
| 23:07.13 | rotten777 | you would recommend voip.ms |
| 23:07.14 | rotten777 | ? |
| 23:07.23 | p3nguin | Absolutely. |
| 23:07.37 | p3nguin | I buy all my DIDs from them right now. |
| 23:08.53 | p3nguin | I'm thinking about getting a DID from Flowroute pretty soon, though. I use them for outbound calling from time to time, but don't have a phone number on them. |
| 23:10.34 | *** join/#asterisk dijib (~nobodysho@bas10-kitchener06-1176138825.dsl.bell.ca) |
| 23:10.50 | blitzrage | p3nguin: good to know -- I use Unlimitel exclusively right now, but with the 3 outages I've experienced lately, it looks like I'm going to be (at the very least), adding another provider to the mix. voip.ms is used by a company a buddy of mine works for in Vancouver, and I've had some good success with them in the little bit of trials I've done |
| 23:11.09 | p3nguin | I like 'em. |
| 23:11.16 | p3nguin | Good rates, decent quality. |
| 23:11.36 | p3nguin | I have both regular and toll-free DIDs with them. |
| 23:11.55 | p3nguin | And they support IAX2 if you're into high volume calling and want to trunk. |
| 23:12.35 | rotten777 | of course voip.ms has local did's on backorder... |
| 23:12.36 | rotten777 | lol |
| 23:12.42 | rotten777 | *sigh* |
| 23:12.57 | p3nguin | Another good thing I like about VoIP.ms is that they offer a refund of unused funds if you are not satisfied. |
| 23:14.11 | dijib | & more then 2half months only has cost me $14.69 |
| 23:14.25 | dijib | 2 and half |
| 23:14.43 | p3nguin | You must have paid for the 3000 minute package. |
| 23:14.52 | dijib | nope. toll free did |
| 23:14.54 | dijib | thats it.. |
| 23:15.13 | dijib | i dont use the phone much.. ive got more testing calls then any real ones |
| 23:15.20 | p3nguin | Good grief. I don't spend that much in 6 or more months. |
| 23:15.20 | dijib | p3nguin, feeling better today? |
| 23:15.32 | dijib | how do u do it then? |
| 23:15.38 | p3nguin | I wasn't feeling poorly before today. |
| 23:15.52 | dijib | i thought i had unlimited incomming but it doesnt look like it |
| 23:15.58 | dijib | or unlimited outgoing... or something |
| 23:16.08 | p3nguin | My DID is $0.99 per month, so that's just $12 per year. |
| 23:16.16 | p3nguin | Plus inbound calls. |
| 23:16.58 | p3nguin | In their toll-free numbers, they only do pay-per-minute. |
| 23:17.05 | p3nguin | same for outgoing calls. |
| 23:17.39 | p3nguin | Only regular DIDs have unlimited (limited to 3000 minutes) calling. |
| 23:25.58 | p3nguin | I guess the way I get by so cheap is by having several DIDs, some of which are free, and don't have a lot of inbound calling on the pay-per-minute numbers. |
| 23:26.15 | p3nguin | Also, a lot of my calling is to toll-free numbers, so those don't cost me anything. |
| 23:27.36 | *** join/#asterisk diemos (~root@2600:3c01::f03c:91ff:fe93:dabe) |
| 23:33.08 | rotten777 | well i'll take a toll free did but i'm on manual activation from voip.ms |
| 23:33.13 | rotten777 | so i have to wait for a human |
| 23:38.39 | p3nguin | pewp |
| 23:38.48 | rotten777 | flowroute looks good though |
| 23:38.50 | rotten777 | pretty slick interface |
| 23:38.53 | rotten777 | decent prices |
| 23:39.10 | p3nguin | slightly higher than voipms, but still very reasonable. |
| 23:39.27 | p3nguin | Actually they might be less on per-minute rates. |
| 23:39.32 | p3nguin | But the DIDs cost more. |
| 23:39.37 | rotten777 | 1.2 cents/m |
| 23:40.18 | p3nguin | for what? |
| 23:40.27 | rotten777 | DID |
| 23:40.29 | rotten777 | flowroute |
| 23:43.05 | p3nguin | Okay, so they are slightly higher on DIDs and minutes. |
| 23:43.16 | p3nguin | Maybe it's the termination rate that is slightly lower. |
| 23:43.30 | rotten777 | in comparison to using my cell 100% of the time, they're exponentially cheaper |
| 23:43.40 | rotten777 | and seeing as i'm on sprint, more reliable |
| 23:44.40 | *** join/#asterisk godmachine-x6 (~godmachin@unaffiliated/godmachine-x6) |
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