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| 00:10.18 | Micc_ | does 1.8.6 support multi-tenant parking yet? |
| 00:11.55 | WIMPy | What the heck is that again? |
| 00:17.20 | Kobaz | different parking lots i suppose |
| 00:17.25 | Kobaz | you could always simulate i |
| 00:17.26 | Kobaz | t |
| 00:18.35 | Kobaz | tenant 1 is 11000-12999 but when they pick up a parked call they enter 1000 and you do for tenant 1 XXXX, goto parkedcalls 1${EXTEN} |
| 00:18.47 | Kobaz | and then tenant 2 would be XXXX, goto parkedcalls 2${EXTEN} |
| 00:21.30 | *** join/#asterisk droemel (~droemel@p4FCACC5E.dip.t-dialin.net) |
| 00:23.35 | p3nguin | I thought at least one branch prior to 1.8 supported that. If so, 1.8.6.0 surely does. |
| 00:26.27 | Micc_ | p3nguin, 1.6 does, but its discontinued. |
| 00:27.28 | p3nguin | 1.6.x branches are discontinued, or multi-tenant parking is discontinued? Because if you mean what I think you mean, See my previous statement. |
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| 01:34.35 | psykon | I can't seem to record a meetme room. Every time I try I get this: file.c:1159 ast_writefile: No such format ''. What gives? |
| 01:34.56 | p3nguin | How are you trying to record it? |
| 01:37.47 | psykon | penguin, exten => _X.,n,MeetMe(${EXTEN}, r) I tried to force the format and file with Set also to no avail. |
| 01:38.41 | p3nguin | The space does not belong there, but I don't know if that has anything to do with the error you've provided. |
| 01:39.06 | psykon | p3nguin, k removing the space. |
| 01:40.28 | psykon | p3nguin, must not be the root cause. Same thing. |
| 01:40.38 | p3nguin | I figured it wasn't. |
| 01:41.00 | psykon | wacky, I wonder if I had everything configured properly before I compiled. |
| 01:41.44 | psykon | asterisk 1.8.5 btw. |
| 01:42.31 | p3nguin | If you didn't try to set anything before running MeetMe, what happens then? |
| 01:45.29 | psykon | p3nguin, same thing |
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| 02:03.08 | psykon | p3nguin, oh. It only throws that error on a dynamic realtime conference. |
| 02:03.57 | p3nguin | If you use a conference that you have configured in meetme.conf, does it record? |
| 02:04.07 | psykon | p3nguin, like a champ... |
| 02:04.36 | p3nguin | Could be a bug. |
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| 02:10.31 | psykon | p3nguin, hmm. I tried 1.8.6 and got the same error. |
| 02:11.18 | gladier | hey guys - do you have any examples of a dialplan where you go to a queue, but after an arbitrary timeout it goes to the next priority in the context? |
| 02:11.25 | gladier | i'm not keen on setting the n option |
| 02:13.38 | p3nguin | What's odd is that I don't use the n option for queues, and it still goes to the next priority when the timeout is reached. |
| 02:14.01 | *** join/#asterisk vinhdizzo (~vinh@pool-173-60-112-55.lsanca.fios.verizon.net) |
| 02:16.47 | gladier | are you using the timeout in the cmd or in one of the queue options? |
| 02:17.50 | p3nguin | Queue(myQ,,,,10) after 10 seconds, it goes to the next line in dial plan. |
| 02:18.36 | p3nguin | I don't know if that's how it is supposed to work or not, since there is the n option which I thought was related. |
| 02:19.28 | gladier | rihgt... i screwed up my context a little bit |
| 02:19.54 | gladier | i had a "check if option was set" and if it wasn't just set it to 5 minutes |
| 02:23.17 | Smirker | howdy. Asterisk 1.8. have a sip 'friend' set up to place outbound and receive inbound calls. insecure=port,invite this works 100% outbound, works well inbound, but for some reason for some incoming requests (from the same host but depending on the callerid in the From header), asterisk goes back and says UNAUTHORISED |
| 02:24.09 | Smirker | i've done a SIP debug and the incoming INVITEs look identicaly besides from From callerid |
| 02:24.10 | p3nguin | Did you set trustrpid=yes for that peer? |
| 02:24.42 | Smirker | no i did not. i'll give that a go (and look up what exactly it is) |
| 02:25.38 | Smirker | hmm, still Unauthorised :/ |
| 02:26.42 | Smirker | is there any other reason Asterisk would send Unauthorised out, given the IP matches and the insecure setting? |
| 02:29.26 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
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| 03:23.00 | beta2k | Anyone around familiar with dundi? I'm able to do a dundi query, but not a lookup |
| 03:32.10 | *** join/#asterisk UD (~steve@unaffiliated/underdawg) |
| 03:36.25 | UD | anyone got asterisk working with google phone? |
| 03:40.40 | p3nguin | Where do I buy a Google phone? |
| 03:41.10 | UD | the google talk thing |
| 03:41.20 | UD | im not really sure |
| 03:43.16 | UD | i dont want to buy a phone i want a cheap way to make it work with real sip program |
| 03:46.48 | p3nguin | You can use Asterisk with Google Talk/Voice stuff. |
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| 03:52.41 | *** join/#asterisk CartoonCat (~Nope@66.78.86.109) |
| 03:53.12 | CartoonCat | Is there a quick test app for sending a call (or mp3 or other sound file) to an extention? |
| 03:54.05 | Kobaz | i'm finding more about my moh problem |
| 03:54.15 | Kobaz | it's something to do with res_timing_dahdi |
| 03:54.42 | Kobaz | if i don't load res_timing_dahdi, and i load either res_timing_timerfd or res_timing_pthread, i get music via mode=custom |
| 03:54.44 | p3nguin | cartooncat: Uhm, extensions run apps... so I don't know what you're trying to do. |
| 03:55.59 | CartoonCat | I am away from home atm. I have a friend with a pbx and just figured out that it might be publicly available. I have no ATAs with me (or even a softphone, need to install it) |
| 03:56.22 | p3nguin | okay |
| 03:56.34 | CartoonCat | Id like to fire off a registration to his pbx to see if its taking the call (it should be internal only) |
| 03:57.04 | p3nguin | You've just said two completely separate things: registration, and call. Which one are you trying to accomplish? |
| 03:57.34 | CartoonCat | both and then some |
| 03:57.42 | p3nguin | If you don't have a peer configured, you can't register. |
| 03:57.48 | CartoonCat | its late, i shoulda got more sleep lol |
| 03:58.24 | p3nguin | If you have a sip phone that will dial a SIP URI, you might be able to call him if you know which extension(s) he accepts anonymous calls on. |
| 03:59.03 | p3nguin | But if you do have a peer configured on that host already, you might be able to use it. |
| 03:59.19 | kaldemar | CartoonCat: channel originate tech/something application Playback tt-weasels |
| 03:59.33 | CartoonCat | i have nothing with me, all my gear is someplace else. have any softphones you recomend? |
| 03:59.49 | p3nguin | Windows? Zoiper classic. |
| 03:59.56 | CartoonCat | linux |
| 04:00.01 | CartoonCat | i can fire up vbox if needed |
| 04:00.02 | p3nguin | I like Twinkle. |
| 04:00.20 | kaldemar | replace tech/something with SIP/exten@ipaddress for example |
| 04:00.27 | p3nguin | People complain about its qt3 dependency, though. |
| 04:01.20 | CartoonCat | not an issue, already have qt3 |
| 04:04.23 | CartoonCat | ok, so hes taking all inbound calls ext@ip |
| 04:04.30 | CartoonCat | explains a lot about whats going on lol |
| 04:08.52 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 04:12.54 | *** join/#asterisk Fritz09 (~Adium@pop1-3802.catv.wtnet.de) |
| 04:20.27 | beta2k | Ok, got lookups working through dundi |
| 04:21.04 | beta2k | Now onto the iax part that doesn't seem to want to work |
| 04:22.21 | beta2k | Would pasting the last three lines be ok? This is where it tries to call the extension then hangs up |
| 04:33.31 | beta2k | Meh, here we go |
| 04:33.32 | beta2k | <PROTECTED> |
| 04:33.35 | beta2k | <PROTECTED> |
| 04:33.37 | beta2k | <PROTECTED> |
| 04:33.59 | beta2k | I don't get why it just hangs up |
| 04:48.37 | beta2k | Ahhhh, got it |
| 04:48.45 | beta2k | It's because I was trying to call a ring group |
| 04:48.57 | beta2k | I can call extensions fine |
| 04:49.47 | beta2k | Anyone know what ext- I need to use for a ring group? ext-local and ext-queues didn't work |
| 04:50.29 | p3nguin | You'll use whatever extension you create that dials a group of phones. |
| 04:59.13 | beta2k | Found it, ext-group |
| 04:59.36 | beta2k | All working now, yay :) |
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| 07:43.31 | Nasga | does anyone know how to soft hangup an agent during the agentlogin ? |
| 07:44.07 | Nasga | i want to kill the channel if the agent is already logged. |
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| 08:30.15 | carrar | moo |
| 08:41.29 | dandre | hello, |
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| 08:45.10 | dandre | I am trying to use @ characters in the dial string for a sip peer. My version of asterisk is now 1.6.2.9 and I have tried |
| 08:45.10 | dandre | Dial(SIP/peer/foo@bar), Dial(SIP/peer/foo\@bar), Dial(SIP/peer/foo%40bar) but none of these work. Is there anyway to excape the @ char so that it is not taken as host separator by the sip channel? |
| 08:45.27 | *** join/#asterisk anny__ (~chatzilla@77.42.241.114) |
| 08:47.42 | wdoekes2 | dandre: with pedantic=no you should be able to receive messages extension foo%40bar if you dial like that second example |
| 08:47.52 | wdoekes2 | but you might need a newer 1.6.2 |
| 08:48.32 | wdoekes2 | s/messages/invites at |
| 08:48.42 | wdoekes2 | *third example |
| 08:51.07 | dandre | wdoekes2: do you know wich version I should use? |
| 08:51.36 | dandre | Is there a patch backporting this featur to 1.6.2? |
| 08:51.55 | anny__ | hi all, anyone aware of an open source web softphone that uses IAX |
| 08:58.43 | wdoekes2 | dandre: I said a newer 1.6.2, not 1.8 (although that would work too) |
| 08:58.57 | *** join/#asterisk kwk (~kleine@carbon.gonicus.de) |
| 08:59.29 | dandre | pk |
| 08:59.55 | kwk | Hi. I'm experiencing a strange behavior with my asterisk. Could someone take a look at the problem please? I've wrote a few more details (about 20 line) about the problem here: https://gist.github.com/1202976 |
| 09:01.40 | wdoekes2 | dandre: https://issues.asterisk.org/jira/browse/ASTERISK-16597 |
| 09:05.31 | *** join/#asterisk droemel (~droemel@p4FCAD3EE.dip.t-dialin.net) |
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| 09:09.21 | kaldemar | kwk: peer is not an extension nor a hint. |
| 09:10.22 | kaldemar | kwk: extension is a part of dialplan, peer is a device you define in sip.conf. when you have [1666] in sip.conf, you'll see output with sip show peer 1666. |
| 09:11.26 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 09:11.28 | dandre | wdoekes2: Ok I have seen the issue but if I understand correctly, this is related to inbound. My issue is with outbound |
| 09:12.29 | wdoekes2 | if you have pedantic=no, then foo%40bar will create a sip:foo%40bar@peer R-URI |
| 09:17.46 | *** part/#asterisk sekil (~sekil@78.24.104.73) |
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| 09:23.07 | BenC[UK] | Hi Guys, I am trying to use realtime sip peers and queues in aserisk 1.8... I can login, but they're always showing the status as unavailable |
| 09:23.31 | BenC[UK] | and they dont show in sip show peers |
| 09:24.37 | *** join/#asterisk carloimperia (~carloimpe@85.40.123.0) |
| 09:25.28 | wdoekes2 | realtime sip peers will never show up in 'sip show peers' unless you have rtcachefriends=yes |
| 09:25.55 | wdoekes2 | you can show individual peers by doing 'sip show peer PEERNAME load' |
| 09:28.06 | *** join/#asterisk gmaruzz (~gmaruzz@2-225-249-20.ip178.fastwebnet.it) |
| 09:28.45 | BenC[UK] | ok, added the cache and that works |
| 09:28.55 | BenC[UK] | still shows unavailable in the queue though |
| 09:29.38 | *** part/#asterisk gmaruzz (~gmaruzz@2-225-249-20.ip178.fastwebnet.it) |
| 09:30.39 | BenC[UK] | hmm, after the load, I logged out and back in |
| 09:30.46 | BenC[UK] | and it shows Not in use now - not unavailable |
| 09:30.52 | BenC[UK] | but the queue isnt routing any calls |
| 09:32.48 | BenC[UK] | ignore that |
| 09:32.50 | BenC[UK] | was the member delay |
| 09:35.30 | BenC[UK] | == Using SIP RTP CoS mark 5 |
| 09:35.30 | BenC[UK] | <PROTECTED> |
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| 09:48.45 | *** join/#asterisk schmidts (~schmidts@Lo20-rt01.lm33.sil.at) |
| 09:48.46 | schmidts | hello |
| 09:51.52 | kwk | kaldemar: sadly, in our asterisk extensions and peers carry the same name. *arrgh*. We fixed the problem by modifying our realtime configs in sip.conf. (rtcachefriends=yes, rtupdate=yes, rtautoclear=yes, ignoreregexpire=no) |
| 09:55.51 | kaldemar | kwk: what exactly are you seeing as an issue? |
| 09:58.29 | schmidts | hey folks does anyone know a good software for fraud detection? |
| 09:59.37 | kwk | kaldemar: We have no issue anymore. It had something to do with peers that were previsouly registered and remembered by the Asterisk. When we've called the *unregistered* peer's extension, the Asterisk recalls to which IP address this peer was bound and updates the peer's and extension's state to "Ringing" which cannot be by definition, since the peer is not registered at the moment. Through the sip.conf settings, we've fixed this issue. Asterisk will now |
| 10:01.03 | kwk | kaldemar: When you call the extension of an unregistered peer the asterisk now says that the peers is not available and leaves the peer's and extension's state untouch (e.g. Unavailable) |
| 10:08.45 | dandre | ok I have traced the sip frames on a working phone and on asterisk. I think that the issue is that I register my peer to some host and must invite on another host: |
| 10:08.45 | dandre | say that I must register on sip.provider.com and use sip:foo@bar.proviver.com, both ip address beeing different. |
| 10:08.45 | dandre | When I directly use my phone everything is fine. |
| 10:08.49 | dandre | How can I do? |
| 10:12.20 | *** part/#asterisk kwk (~kleine@carbon.gonicus.de) |
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| 10:24.32 | kaldemar | dandre: use different hosts in the register statement and in dialplan. |
| 10:24.54 | kaldemar | or in the peer. |
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| 10:44.47 | *** join/#asterisk kaushal (~kaushal@49.248.16.122) |
| 10:44.49 | kaushal | Hi |
| 10:45.28 | kaushal | what is SS7 and does digium or Sangoma Card manufacturer Card to support SS7 ? |
| 10:45.51 | kaushal | or is it E1 ISDN PRI Line |
| 10:46.10 | kaushal | can someone please help me understand |
| 10:46.31 | singler | SS7 is a protocol |
| 10:46.38 | singler | it works on top of E! |
| 10:46.41 | singler | E1 |
| 10:46.53 | singler | sangoma supports it, not sure about digium |
| 10:48.06 | singler | you can check this out http://wiki.sangoma.com/Netborder-SS7-Voip-Media-Gateway |
| 10:49.13 | kaushal | singler: is it a separate card altogether ? |
| 10:49.46 | kaushal | since i have A108 E1 Pri Card |
| 10:50.07 | *** join/#asterisk k3asd` (~k3asd@static-94-32-127-180.clienti.tiscali.it) |
| 10:50.16 | singler | it is same card |
| 10:50.19 | k3asd` | hi |
| 10:50.45 | kaushal | singler: same card ? |
| 10:51.08 | singler | same card as for PRI |
| 10:51.12 | singler | A10x |
| 10:51.32 | kaushal | singler: ok |
| 10:51.37 | *** part/#asterisk mjordan (~mjordan@user-69-1-6-49.knology.net) |
| 10:51.46 | kaushal | so do i need to enable something ? |
| 10:52.02 | kaushal | I usually configure wanpipe for it |
| 10:52.11 | singler | you need special software for it (and license) |
| 10:52.12 | kaushal | on asterisk apps |
| 10:52.49 | kaushal | singler: can you please point me to the wiki or some docs ? |
| 10:53.33 | singler | http://wiki.sangoma.com/Netborder-SS7-Voip-Media-Gateway |
| 10:53.33 | kaushal | singler: also SS7 is it on 7th layer ? |
| 10:54.10 | kaushal | what does digit 7 denote in SS7 |
| 10:54.18 | singler | version :) |
| 10:54.22 | kaushal | oh ok |
| 10:54.40 | singler | http://en.wikipedia.org/wiki/Signaling_System_7 |
| 10:58.10 | kaushal | singler: ok |
| 10:58.54 | kaushal | so i need to use openssl7 instead of libpri while configuring Sangoma 8 port PRI A108E cards ? |
| 10:59.03 | *** join/#asterisk MarKsaitis (~MarKsaiti@94-30-69-205.xdsl.murphx.net) |
| 11:05.49 | singler | no |
| 11:10.13 | singler | installation instructions are in that wiki link. Also it seems that you missed the point that license is needed (you need to buy it), it comes with support contract, so sangoma's techsupport will be able to help you out with installation and configuration |
| 11:15.17 | kaushal | ok |
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| 11:22.25 | kaushal | singler: Thanks |
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| 12:16.52 | dandre | kaldemar: How could I do that? |
| 12:18.07 | dandre | I have my peer correctly registered |
| 12:19.03 | dandre | then the dial command should look like something Dial(SIP/peer/foo@bar.provider.com) ? |
| 12:23.46 | kaldemar | dandre: how do you not do that? register => user:secret@host_a. then [peer]\host=host_b, and Dial(SIP/peer/number) |
| 12:25.14 | dandre | Ok I will try |
| 12:25.36 | *** join/#asterisk oej (~olle@ns.webway.se) |
| 12:26.22 | dandre | I don't see how the register and the peer statement are tied togother |
| 12:28.37 | *** join/#asterisk luckman212 (~irc@2001:470:1f07:1225:ec3d:ac06:21a2:ec00) |
| 12:28.47 | kaldemar | they aren't. so what is the problem? |
| 12:29.53 | kaldemar | if i got you right, you want to register to address A and send invites to B. put A in the register line and B in a peer that you use in dial commands. |
| 12:31.01 | *** join/#asterisk chuckf (~chuckf@ubuntu/member/chuckf) |
| 12:31.27 | otwieracz | Hello. |
| 12:31.34 | dandre | yes but I must send the invite to B thru A |
| 12:31.43 | otwieracz | I need to record calls â how to? |
| 12:32.00 | kaldemar | dandre: does A not handle it? |
| 12:32.30 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 12:32.30 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 12:32.56 | kaldemar | dandre: in that case what i said earlier still applies, you just use foo@A as the extension and escape the @. |
| 12:33.18 | dandre | the invite is directly passed to B without contacting A |
| 12:33.31 | otwieracz | I'm trying to use MixMonitor, but the record file isn't created. |
| 12:33.47 | otwieracz | exten => s,1,Set(CALLFILENAME=${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)}_${CALLERID(num)}_${CALLERID(dnid)}_${UNIQUEID}) |
| 12:33.48 | leifmadsen | where are you looking? does asterisk have permissions to write there? |
| 12:33.50 | otwieracz | exten => s,2,MixMonitor(/var/spool/asterisk/monitor/${CALLFILENAME}.wav,b) |
| 12:34.10 | kaldemar | dandre: when you do what? |
| 12:34.14 | otwieracz | From logfile: |
| 12:34.15 | otwieracz | <PROTECTED> |
| 12:34.18 | otwieracz | <PROTECTED> |
| 12:34.27 | otwieracz | And after disconnect: == End MixMonitor Recording SIP/203-00000003 |
| 12:34.42 | otwieracz | ls -l /var/spool/asterisk/monitor/ |
| 12:34.42 | otwieracz | total 0 |
| 12:34.59 | leifmadsen | does the call every get bridged? |
| 12:35.08 | otwieracz | drwxr-xr-x 2 asterisk asterisk 4096 Sep 7 16:02 monitor |
| 12:35.22 | otwieracz | Hmm⦠|
| 12:35.30 | otwieracz | I think that it wasn't bridged. |
| 12:35.42 | otwieracz | I was the only one person in conference made with ConfBridger. |
| 12:36.18 | dandre | ok but the invite looks like number%40B@A and with a working configuration the invite looks like number@B but sent to A |
| 12:36.52 | dandre | I have try to escape but this dosn't work |
| 12:37.24 | kaldemar | dandre: show the dialplan line and what you see in sip debug. |
| 12:37.49 | otwieracz | Well? |
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| 12:40.23 | singler | dandre: did you try to use outboundproxy=host_a? |
| 12:40.28 | kaldemar | dandre: at least with 1.8.6.0 you don't even have to escape anything, just use SIP/peer/foo@bar and it sends an invite for foo@bar to the address that peer has. |
| 12:41.00 | dandre | but I am using 1.6.2 |
| 12:41.34 | dandre | and I con't upgrade now, will certainelly do it later |
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| 12:42.26 | dandre | singler: Didn't know about thi |
| 12:47.25 | wdoekes2 | oh, I dandre, I thought you wanted %40 in the URI. misunderstanding. the outboundproxy peer option is what you want indeed |
| 12:47.30 | dandre | kaldemar: http://pastebin.com/JQbLE67d |
| 12:49.49 | kaldemar | dandre: looks to me that the invite is going where it's supposed to. try something else instead of %40. plain @ for example. |
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| 12:55.47 | dandre | kaldemar: with a plain @: http://pastebin.com/9tfdaSte |
| 12:57.36 | dandre | If I have correctly understood, I can't use outboundproxy because I must send invite number@hostB thru hostA, hostA beeing the registrar and the proxy |
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| 13:03.16 | kaldemar | dandre: try \@ |
| 13:03.57 | dandre | I had already done that |
| 13:07.05 | leifmadsen | braces |
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| 13:11.00 | chuckf | unbraces |
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| 13:57.55 | ixyd_ | hi guys, i would like to split an ael statement over multiple lines like i can do by using a backslash in bash....can some tell me if/how that is possible? :) |
| 13:59.20 | leifmadsen | not possible |
| 13:59.37 | leifmadsen | lines of a dialplan must be written on a single line |
| 14:00.04 | leifmadsen | Asterisk 1.8.7.0-rc1 is now ready for testing! More information about this release candidate is available in the release announcement at http://www.asterisk.org/node/51673 |
| 14:00.34 | *** join/#asterisk bmint (~bmint@h90.87.19.98.static.ip.windstream.net) |
| 14:01.54 | ixyd_ | i was afraid you would say something like that ;) |
| 14:01.59 | ixyd_ | but thanks! :) |
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| 14:04.59 | anonymouz666 | leifmadsen: I was waiting for this, thanks |
| 14:05.05 | leifmadsen | :) |
| 14:05.33 | bmint | We recently started using g729 as the preferred codec on our phones. Everything works fine except for when we try to use the page function. It tries to transcode from SLIN (I think) and uses up all the g729 licenses so the page doesn't work. Can the codec be set within the dialplan? |
| 14:06.40 | *** join/#asterisk sourcode (~code@ppp-58-11-105-220.revip2.asianet.co.th) |
| 14:08.11 | malcolmd | https://wiki.asterisk.org/wiki/display/AST/chan_sip+Channel+Variables SIP_CODEC, SIP_CODEC_OUTBOUND you'll want to call the outbound w/ __ for inheritance |
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| 14:12.05 | bmint | malcolmd: Thanks I'll give that a shot! |
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| 14:19.59 | *** join/#asterisk otwieracz (~gonet9@v6.gen2.org) |
| 14:20.00 | otwieracz | Hello. |
| 14:20.06 | otwieracz | I have problem with Asterisk behind NAT. |
| 14:20.29 | otwieracz | When I'm trying to make outgoing call with my SIP provider (ovh) I'm getting âLoop detectedâ. |
| 14:21.27 | *** join/#asterisk bchia (~Adium@nat/digium/x-myifvflqmizneuxw) |
| 14:22.35 | otwieracz | That's after I moved Asterisk from router to machine behind nat. |
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| 14:28.33 | otwieracz | Anyone? |
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| 14:35.26 | p3nguin | Show us the dial plan and possibly the sip peer entry of the peer that is giving problems. |
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| 14:36.17 | kaldemar | otwieracz: while you're at it, pastebin a sip debug of a call. |
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| 14:42.14 | otwieracz | http://pastebin.com/1qRK8mX0 |
| 14:42.19 | Kobaz | what's a good console snmp explorer? |
| 14:42.21 | otwieracz | sip.conf and extensions. |
| 14:42.27 | leifmadsen | Kobaz: snmpwalk? |
| 14:42.38 | scubes13 | using polycom 601 phones, if a user presses transfer, types the other person's extension and presses transfer again... the system just puts the caller into an infinite ring and the receiving extension never rings at all |
| 14:43.18 | leifmadsen | more information required |
| 14:43.28 | Kobaz | leifmadsen: mm, k. i'll check it out |
| 14:43.31 | p3nguin | otwieracz: If sip.ovh.co.uk is not behind nat, their setting should be nat=no |
| 14:44.04 | leifmadsen | localnet and externhost/externip would be used for your Asterisk box that is living behind NAT |
| 14:44.35 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 14:44.54 | otwieracz | My asterisk box is living behind NAT. |
| 14:45.44 | p3nguin | otwieracz: You haven't configured it to live behind nat, and you did misconfigure the peer, because they are not behind nat. |
| 14:45.51 | p3nguin | otwieracz: You also probably don't need dial option r. And if you don't want the other end of the call to be able to transfer calls, you shouldn't use dial option t. |
| 14:46.05 | otwieracz | What about that nat? |
| 14:46.32 | p3nguin | You didn't configure nat=yes for your system. It goes in general section. |
| 14:46.36 | p3nguin | ~sipnat |
| 14:46.37 | 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. |
| 14:47.26 | p3nguin | If you are behind nat, nat=yes need to be in general section. If a peer is behind nat, nat=yes goes in that peer's section. If the peer is not behind nat, nat=no belongs in the peer's section. |
| 14:47.53 | otwieracz | But i'm still getting Loop detected. |
| 14:48.00 | leifmadsen | nat={yes,no} is whether the other end is behind NAT |
| 14:48.03 | otwieracz | I put nat = yes in sip's.conf [general]. |
| 14:48.09 | kaldemar | otwieracz: which end says loop detected? |
| 14:48.13 | leifmadsen | you have to configure you system to live behind nat with the localnet and externip/externhost options |
| 14:48.19 | otwieracz | Router. |
| 14:48.21 | leifmadsen | otwieracz: you need to read more carefully |
| 14:48.35 | kaldemar | router? |
| 14:48.35 | otwieracz | leifmadsen: I have externip, externhost and localnet configured. |
| 14:49.02 | p3nguin | Also, how are you calling anything when you don't have any phones configured on that system? |
| 14:49.18 | otwieracz | That wasn't whole sip.conf |
| 14:49.18 | p3nguin | You have only one peer -- the ITSP. |
| 14:49.24 | otwieracz | #include "users/vts.conf" |
| 14:49.24 | p3nguin | moves on |
| 14:49.24 | otwieracz | #include "users/devs.conf" |
| 14:49.56 | kaldemar | otwieracz: feel free to pastebin the sip debug of a call. |
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| 14:51.30 | otwieracz | Hmm. |
| 14:51.49 | otwieracz | I must filter it from confidential data. |
| 14:52.58 | p3nguin | ...and useful tidbits used for debugging. |
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| 15:14.55 | *** topic/#asterisk is 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 (2011/08/08), dahdi-tools 2.5.0 (2011/08/08), libpri 1.4.12 (2011/07/06) -=- Visit the official Asterisk wiki: wiki.asterisk.org -=- http://xkcd.com/806/ |
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| 16:13.06 | tbac | hi! i'm looking for a way to set up asterisk in a way that incoming calls (matching certain criteria) would be forwarded via sip to other sip servers |
| 16:13.51 | tbac | so far this is easy, but i'd like to have a failover mechanism: when call forwarding fails via one context, use another context |
| 16:14.10 | p3nguin | That's not what contexts are for. |
| 16:14.25 | p3nguin | Just use another dial after the first one. |
| 16:14.50 | p3nguin | If the first one fails, the next will run. |
| 16:15.36 | tbac | no conditions needed whatsoever? this is "built-in logic"? |
| 16:15.47 | p3nguin | That's how extensions work. |
| 16:16.06 | tbac | ok, i'll try that out, thanks! |
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| 16:17.00 | p3nguin | If you have a Dial() in priority 2, and it fails, it'll move on to priority 3. |
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| 16:17.29 | tbac | is a finer definition of "failing" possible? e.g. consider sip 503 response as fail, not 404 however? |
| 16:17.39 | tbac | ah, yes, the priorities.... indeed |
| 16:17.48 | p3nguin | You could set up conditions to check DIALSTATUS after the first one, and then determine what to do based on the status. |
| 16:19.06 | tbac | sounds good, thanks a LOT for the hints! |
| 16:19.35 | tbac | this way i guess even a timeout based fail-over will be possible |
| 16:19.43 | carrar | or use a sip routing server more suitable for what you want to do |
| 16:20.00 | carrar | opensips or kamilio |
| 16:20.30 | tbac | carrar: sure, that might happen in the long run, for now a quick and easy solution would suffice |
| 16:20.53 | carrar | asterisk is quick and easy for limited number of concurrent calls |
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| 17:10.08 | irroot | evening folks |
| 17:10.49 | navaismo | hi |
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| 17:30.11 | Vi0L0 | hi, does anybody tried to build active-active asterisk cluster? |
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| 17:30.39 | leifmadsen | Vi0L0: you can't |
| 17:31.03 | leifmadsen | there is no way for asterisk to track the audio between the two servers so that if a server fails, the audio keeps flowing |
| 17:31.08 | leifmadsen | Vi0L0: see Asterisk SCF |
| 17:31.09 | Vi0L0 | leifmadsen: on web i've found many people trying, are you sure i can't |
| 17:31.13 | leifmadsen | ~asterisk-scf |
| 17:31.30 | chazzam | ~asteriskscf |
| 17:31.31 | infobot | Asterisk SCF is the Asterisk Scalable Communications Framework. More information about Asterisk SCF is available on the wiki at https://wiki.asterisk.org/wiki/display/TOP/Asterisk+SCF+Home |
| 17:31.32 | chazzam | ? |
| 17:31.39 | leifmadsen | Vi0L0: it depends on what you mean by "active-active" I guess -- I'm assuming you mean, "one asterisk dies, and the other system takes over and no calls are dropped" |
| 17:31.55 | chazzam | load balancing? |
| 17:32.08 | Vi0L0 | so calls on node1 will be disconnected... but still node2 will work |
| 17:32.13 | leifmadsen | yes |
| 17:32.18 | leifmadsen | that's possible |
| 17:32.30 | leifmadsen | look into LinuxHA |
| 17:32.44 | Vi0L0 | ok, i know this site ;) |
| 17:33.01 | Vi0L0 | i was just wondering, cuz many people failed on this + asterisk |
| 17:33.11 | leifmadsen | I didn't |
| 17:33.17 | leifmadsen | not sure where you came up with "many people" |
| 17:33.17 | Vi0L0 | ahhh :) |
| 17:33.36 | Vi0L0 | i mean 2 first sites of google search ;) |
| 17:33.44 | *** part/#asterisk tbac (~tbac@p5DE85B7E.dip.t-dialin.net) |
| 17:33.55 | leifmadsen | that's not a very good sampling average |
| 17:34.03 | Vi0L0 | yes, i know |
| 17:34.30 | Vi0L0 | and i was pretty sure i will find answer here |
| 17:35.03 | Vi0L0 | thanks for the good news leifmadsen |
| 17:39.21 | *** part/#asterisk Vi0L0 (~vi0l0@unaffiliated/vi0l0) |
| 17:39.35 | *** join/#asterisk hardwire (~spencersr@12.17.188.86) |
| 17:40.14 | hardwire | asdf1234 |
| 17:42.40 | pabelanger | nice password |
| 17:42.55 | jaytee | back in the old days qwerty was also 12345 on a teletype machine and it had figs and ltrs buttons on either side of the spacebar to switch between modes, all upper case letters. hated it running in half-duplex cuz you could never see what you typed, only what the distant end typed. |
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| 17:51.45 | carrar | the OLD DAYS!! |
| 17:52.00 | hardwire | pabelanger: hehe |
| 17:52.09 | hardwire | I'm sick and my brain hurts.. that's all that I could type at the time. |
| 17:52.30 | hardwire | trying to figure out why dropped packets = asterisk eats eatself and the CPU. |
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| 17:55.41 | ijpalmer | hi all, I'm running * 1.8.5.0 with call queues, I have 2 queues setup one weighted higher than another. I have no agents available, then 2 calls come in the first to the lowest weighted queue, the second to the higher weighted. When an agent becomes available the call which came in first gets presented, not the highest weighted queued call which came in second. Is this right. |
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| 18:04.42 | irroot | ijpalmer it does not sound right but there reasons this could happen |
| 18:05.39 | ijpalmer | irroot: such as? |
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| 18:20.36 | hardwire | wonders why fop2 decided to include an old rusty perl in their binary. |
| 18:20.50 | hardwire | but no support libraries. |
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| 18:56.25 | *** join/#asterisk luckman212 (~irc@2001:470:1f07:1225:ec3d:ac06:21a2:ec00) |
| 18:57.36 | luckman212 | does anyone know the correct way to call mpg123 from a shell script specified in musiconhold.conf e.g. application=/etc/asterisk/mohstream.sh ? when I try that I get dead air. |
| 18:58.00 | luckman212 | ast1.8.6 + mpg123 1.13.4 |
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| 19:02.20 | luckman212 | steps out of the way of oncoming tumbleweeds |
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| 19:41.29 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 19:41.29 | *** topic/#asterisk is 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 (2011/08/08), dahdi-tools 2.5.0 (2011/08/08), libpri 1.4.12 (2011/07/06) -=- Visit the official Asterisk wiki: wiki.asterisk.org -=- http://xkcd.com/806/ |
| 19:45.08 | luckman212 | jumps out of the way of another tumbleweed |
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| 19:55.56 | pabelanger | luckman212: Interesting, but no. You cannot modify the SDP like that in asterisk. |
| 19:56.02 | pabelanger | Unless you modified the source |
| 19:56.43 | raden | Naikrovek, ??????? |
| 19:56.54 | Naikrovek | whut |
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| 20:02.27 | raden | Naikrovek, do u understand cisco config reg ? |
| 20:03.15 | Naikrovek | eh a little |
| 20:03.18 | Naikrovek | answering you in private |
| 20:03.23 | carrar | conf t |
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| 20:03.33 | carrar | 0x2103 |
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| 20:14.41 | *** join/#asterisk kaushal (~kaushal@14.97.13.213) |
| 20:14.44 | kaushal | Hi |
| 20:15.20 | kaushal | I am referring to https://wiki.asterisk.org/wiki/display/AST/Asterisk+Manager+Interface+%28AMI%29 |
| 20:16.09 | kaushal | Can some one please help me understand in using AMI to initiate Outbound Call to run a campaign for 240 phone Numbers. |
| 20:16.20 | kaushal | I am comfortable in BASH Scripting |
| 20:20.55 | kaushal | Please let me know if anyone needs additional information about my setup |
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| 20:24.58 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 20:25.13 | *** part/#asterisk russellb (~russellb@asterisk/contributor-and-cool-guy/russellb) |
| 20:25.54 | paulc | kaushal: you could do it with callfiles, which may be easier/simpler than interfacing via AMI |
| 20:26.29 | kaushal | paulc: ok |
| 20:27.05 | kaushal | paulc: but that doesnot show the status of the calls or events captured in case of failures ? |
| 20:28.01 | paulc | kaushal: Depends on how you do it - you can read the files afterwards and see the outcome.. or use some dialplan magic to update a database.. all down to how you implement it really.. AMI works too, you'll need to track call IDs etc. |
| 20:28.37 | *** topic/#asterisk by sruffell -> #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/ |
| 20:29.15 | kaushal | paulc: ok |
| 20:29.27 | kaushal | paulc: Any docs regarding callfiles ? |
| 20:30.12 | paulc | kaushal: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Call+FIles |
| 20:30.39 | paulc | You might also find this helpful: http://www.voip-info.org/wiki/view/Asterisk+auto-dial+out |
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| 20:32.51 | talntid | lets say I wanted in the dialplan to increment a variable every time something happened... how would you guys recommend doing that? |
| 20:33.16 | kaushal | paulc: thanks |
| 20:33.27 | kaushal | reading the wiki |
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| 20:33.44 | paulc | talntid: Set(SomeVar=$[${SomeVar}+1]) ? |
| 20:33.56 | leftist | how can i determine the version of asterisk? is there a simple command i can execute? |
| 20:34.05 | bchia | core show version |
| 20:34.07 | paulc | leftist: core show version I think |
| 20:34.12 | leftist | ok |
| 20:34.35 | *** part/#asterisk ijpalmer (~IceChat7@host81-137-172-233.in-addr.btopenworld.com) |
| 20:38.26 | Luyt | leftist: I'm on FreeBSD, I did yesterday something like: pkg_info | grep -i aster |
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| 20:38.47 | Luyt | and whoppa, asterisk18-1.8.5.0_1 An Open Source PBX and telephony toolkit |
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| 20:53.21 | *** mode/#asterisk [+o sruffell] by leifmadsen |
| 20:53.37 | Bipul | can any one give me some code to start contributing my work to asterisk |
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| 20:54.42 | kaushal | paulc: can i pastebin my call.py files ? |
| 20:54.48 | leifmadsen | Bipul: code? |
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| 20:55.26 | Bipul | idk but i want's to contribute something to this project and i know C and Java (at some extend) |
| 20:56.04 | paulc | Bipul: I'd love it if I could press CTRL-D when I'm in *CLI> console and have it exit to shell (just like MySQL does!) |
| 20:56.07 | Bipul | is there any 0 level to start |
| 20:56.12 | paulc | kaushal: Sure - I'll take a look if you like |
| 20:56.16 | kaushal | sure |
| 20:57.11 | Bipul | ? |
| 20:57.24 | Luyt | paulc: I noticed that too. I have to press ^C followed by an Enter. |
| 20:57.31 | leifmadsen | Bipul: cool! ok so go to https://issues.asterisk.org/jira and sign the license agreement (link at the top). After you've done that, when you login you'll be able to attach code to issues you open (or to existing issues if you find things you want to fix) |
| 20:57.53 | Bipul | wow |
| 20:58.32 | leifmadsen | Bipul: you might want to start here as well for some overviews: https://wiki.asterisk.org/wiki/display/AST/Development |
| 20:58.52 | *** part/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
| 20:58.55 | kaushal | paulc: http://pastebin.ubuntu.com/685575/ |
| 20:59.00 | paulc | Luyt: Never tried that.. although CTRL-C is sometimes seen as "badder".. I like CTRL-D x 3 to get me out of MySQL, my sudo -i, my initial login.. funny the things we like eh :) |
| 20:59.20 | kaushal | paulc: I am not sure about the line 26 bases on the dialplan |
| 20:59.25 | kaushal | based* |
| 21:00.13 | paulc | kaushal: nor me - I'm not too familiar with Python - I'd be more interested in seeing the actual call file it produces to see if it's grammatically correct |
| 21:00.23 | kaushal | ok |
| 21:00.25 | Luyt | paulc: Just a plain 'quit' command will also work for leaving the CLI ;-) |
| 21:00.50 | kaushal | paulc: ok |
| 21:01.32 | paulc | Luyt: Yeah, I know.. but why press 5 keys when 1 will do? ;) |
| 21:02.26 | Luyt | paulc: You have a point, but maybe people never feel the need to quit a CLI session? Perhaps they leave it running for months in their screen session |
| 21:03.10 | paulc | Luyt: Good point.. s'funny, because I run irssi in a screen session every day, but I'd never really given much thought to running *CLI in one.. |
| 21:03.44 | Luyt | I keep Konversation (an IRC client) also running 24/7, so it collects logs of my fav channels. |
| 21:03.56 | kaushal | paulc: http://pastebin.ubuntu.com/685583/ |
| 21:04.16 | kaushal | paulc: line 2 is incorrect |
| 21:04.40 | kaushal | my dialplan is here -> http://pastebin.ubuntu.com/685575/ |
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| 21:07.23 | paulc | kaushal: your call looks ok.. it'll dial that number, then playback(hello-world).. I usually call somewhere, then pass to a Local extension in a specific context to handle "what to do if the remote end answers" |
| 21:07.53 | kaushal | paulc: i do not have SIP |
| 21:08.26 | paulc | kaushal: So what do you have, telephony/outside world connectivity-wise? |
| 21:08.28 | kaushal | paulc: channel originate DAHDI/g3/9930963298 Application MP3Player /home/kaushal/obd-demo.mp3 |
| 21:08.35 | kaushal | it works fine |
| 21:08.53 | kaushal | so not sure how would it look like in call files |
| 21:09.06 | paulc | so translate those to your callfile Destination, Application and Data fields |
| 21:09.17 | paulc | give me a second and I can show you what one of my call files looks like |
| 21:09.24 | kaushal | paulc: sure |
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| 21:13.32 | paulc | kaushal: See http://pastebin.com/fKGQqjef - I set Caller ID and make an outbound call to +1604... once connected, the dialplan at ono-agent-greeting-management runs, exten s priority 1. I make a variable available for that dialplan to act on. |
| 21:14.26 | kaushal | paulc: ok |
| 21:15.13 | kaushal | still not clear about Channel: SIP/trunkname/9930963298 |
| 21:15.33 | kaushal | I use CLI > channel originate DAHDI/g3/9930963298 Application MP3Player /home/kaushal/obd-demo.mp3 |
| 21:15.46 | paulc | kaushal: Where the call should be placed to. So in your case "Channel: DAHDI/g3/9930.." |
| 21:16.01 | kaushal | ok |
| 21:16.03 | kaushal | got it |
| 21:16.34 | kaushal | so it would look like Channel: DAHDI/g3/9930963298 ? |
| 21:16.49 | kaushal | for example |
| 21:18.01 | kaushal | paulc: Thanks |
| 21:18.10 | kaushal | let me run it by hand |
| 21:19.14 | paulc | kaushal: Yeah, that looks right.. try it by hand first, then get your script to do the repetitive work for you :-) |
| 21:19.38 | paulc | kaushal: One note - when you do it by hand.. create the file somewhere else (like /tmp) then mv it to /var/spool/asterisk/outgoing |
| 21:19.49 | kaushal | paulc: sure |
| 21:19.57 | paulc | Good luck :) |
| 21:20.00 | kaushal | so place those lines in .call file |
| 21:20.17 | kaushal | run by hand ? |
| 21:20.25 | kaushal | not sure how do i invoke it |
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| 21:23.18 | kaushal | paulc: got it |
| 21:23.44 | paulc | kaushal: does it work for you? |
| 21:23.53 | kaushal | so create hello.call file under /tmp and do cp /tmp/hello.call /var/spool/asterisk/outgoing/ ? |
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| 21:24.10 | kaushal | am i understanding you correctly ? |
| 21:24.25 | kaushal | let me test it |
| 21:25.35 | p3nguin | If /tmp and /var are on different filesystems, there is a slight chance that it could get corrupted. |
| 21:25.39 | dr0ck | better to cp hello.call hello.call.1 and mv /tmp/hello.call.1 /var/spool/asterisk/outgoing |
| 21:26.00 | p3nguin | When that is the case, I put the files in /var/tmp/ first. |
| 21:26.34 | p3nguin | Will asterisk still run a file without .call as the file ext? |
| 21:26.41 | p3nguin | .call.1 per your suggestion |
| 21:27.05 | kaushal | it worked |
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| 21:27.11 | kaushal | paulc: thanks |
| 21:27.18 | kaushal | let me further test |
| 21:28.07 | paulc | yeah, it doesn't care about extension |
| 21:28.27 | paulc | my file is actually called something like GrtRec-43e8fe0912 or something |
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| 21:31.37 | kaushal | paulc: i tried this http://pastebin.ubuntu.com/685606/ |
| 21:31.45 | kaushal | did not worked this time |
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| 21:32.44 | paulc | kaushal: What did you see on the asterisk console? (did the phone even ring?) |
| 21:33.13 | kaushal | it rang but got disconnected immediately |
| 21:35.19 | kaushal | paulc: let me pastebin the logs |
| 21:35.20 | paulc | Hmm.. can you use Playback for an MP3 file? |
| 21:35.46 | paulc | I think Playback prefers "path+filename" with no extension... you'd be better off using .ulaw or something - less work for the CPU to do |
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| 21:37.05 | kaushal | paulc: ok |
| 21:38.31 | kaushal | paulc: http://pastebin.ubuntu.com/685610/ |
| 21:38.37 | paulc | kaushal: Set up a DID that does Answer, then Record(testfile.wav), then Hangup. Then use "testfile" as your Data: line in your call file. Whatever you recorded when you dialed in, should be played when you dial out. |
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| 21:39.34 | paulc | kaushal: yeah - it's complaining about not finding the file in a valid format. Do what I just said ^^^^ and you should be fine :-) |
| 21:39.52 | kaushal | sure |
| 21:40.21 | kaushal | so convert .mp3 to .ulaw ? |
| 21:40.53 | frem_ | I have a simura trunk device that disconnects on every call before it goes through. |
| 21:41.14 | frem_ | *sipura |
| 21:41.47 | kaushal | paulc: Thanks |
| 21:41.51 | kaushal | and would update |
| 21:41.56 | kaushal | much appreciated |
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| 21:42.54 | paulc | kaushal: Yes, either convert .mp3 to .ulaw (8 bit, 8KHz, headerless PCM/ulaw) or to prove a point, use the dialplan Record() app to record a new file in the right format. |
| 21:43.05 | paulc | frem_ what model? |
| 21:43.28 | frem_ | paulc: SPA 3000 2.0.13 |
| 21:43.41 | frem_ | here are some logs i collected from it http://pastebin.com/3B6BpWRM |
| 21:43.53 | kaushal | paulc: ok |
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| 21:46.02 | paulc | frem_: Great logs, but a bit too much for me to digest right now. Can you explain simply what you're trying to do? (PSTN --> SPA3000 --> Asterisk? or Asterisk --> SPA3000 --> PSTN outbound?) |
| 21:46.14 | frem_ | outbound |
| 21:47.16 | frem_ | inbound is also probably broken, but i don't think the logs will have any of that. :-P |
| 21:47.37 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 21:49.09 | frem_ | an example of an attempted call is around the line that looks like "2011-09-08 16:03:49Local3.Debug192.168.1.252FXO:Off Hook" |
| 21:51.11 | *** join/#asterisk billmania (~bill@38.98.130.98) |
| 21:52.22 | frem_ | FreePBX shows that the Sipura device is online and that we're not registered with it. Supposedly, asterisk don't need to send a registration string to use it. |
| 21:54.01 | frem_ | when dialing an outbound number, you hear a single "phone on the other end is ringing tone" and then it hangs up. the target phone doesn't actually ring. |
| 21:54.25 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 21:57.36 | *** join/#asterisk mangala (mangala@antenora.aculei.net) |
| 22:02.06 | frem_ | Oh, man. It was apparently already configured for another pbx and pots line. no wonder the howto that said to "start from factory defaults" wasn't working. >_< |
| 22:02.29 | *** join/#asterisk leftist (~dizzy@c-71-204-14-19.hsd1.ga.comcast.net) |
| 22:03.35 | frem_ | I'll reset it and come back later when that also fails. :P |
| 22:04.14 | *** join/#asterisk techknowlust (~Patrick@li257-211.members.linode.com) |
| 22:04.29 | techknowlust | is there a way to make asterisk not block when it's running a CURL command in a dialplan? |
| 22:04.40 | techknowlust | At the moment it doesn't even give a dialtone while the curl is still executing |
| 22:08.31 | *** join/#asterisk freeedrich| (~eeePC@hansaserver.de) |
| 22:12.45 | dijib | hey how do i have asterisk load iax modules |
| 22:17.04 | *** join/#asterisk catphish (~charlie@2001:9d8:2005:2::3) |
| 22:17.52 | catphish | what is the best way to play a message to a user but instruct the CDR that the call failed? |
| 22:20.44 | catphish | i could use the noanswer option to playback but not sure how well that will work over sip |
| 22:20.48 | simplydrew | does anyone have experience with installing chan_sscp? I'm running into some install errors |
| 22:21.03 | simplydrew | *sccp |
| 22:21.52 | *** part/#asterisk Bob_Pierce (~WESTMAN\p@216.36.132.162) |
| 22:25.08 | catphish | simplydrew: pastebin any errors |
| 22:25.37 | catphish | is it valid simple to answer then set CDR(disposition)=FAILED |
| 22:25.57 | simplydrew | catphish: well, so far I'm at the point of even having make errors such as "make: *** No targets specified and no makefile found. Stop.", when I've already installed asterisk-devel |
| 22:29.00 | catphish | i see it as "chan_skinny" |
| 22:29.28 | catphish | oh, there's 2 versions |
| 22:29.50 | Brixius | catphish: Ya I think the noanswer option is also refered to early media and I don't think many carrier's will allow it as it could allow for toll avoidance, ie, you call a LD #, and carry on a conversation without saying the call was answered to your upstream switch. I know we don't allow it where I work. I'm not even sure that's possible with SS7 or ISDN as I don't believe there's a media path until the call is actually answered. |
| 22:30.23 | catphish | Brixius: in this case i am the carrier |
| 22:30.48 | catphish | so if sip supports it, i could permit it downstream |
| 22:31.08 | catphish | simplydrew: which sccp driver are you actually trying to use? |
| 22:31.11 | catphish | where did you get it? |
| 22:31.47 | simplydrew | catphish: I'm a little foggy on what I need to be using. I currently have a 7960 that I'm using on SIP, but picked up a 7965 and was wanting to use sccp with it |
| 22:31.50 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
| 22:31.58 | WIMPy | Early media is unidirectional. |
| 22:31.59 | Brixius | catphish: rfc 3960 explains it and I believe CISCO Phones support it. |
| 22:32.16 | catphish | simplydrew: use chan_skinny |
| 22:32.28 | catphish | its included with asterisk |
| 22:32.28 | simplydrew | catphish: what's the difference between the two, just out of curiosity? |
| 22:32.33 | simplydrew | ah gotcha |
| 22:32.40 | catphish | simplydrew: http://www.voip-info.org/wiki/view/Asterisk+SCCP+channels |
| 22:33.01 | p3nguin | simplydrew: I use chan_sccp-b on asterisk 1.4 branch on my cisco phones. The support for asterisk 1.8 branch is still spotty. |
| 22:33.20 | p3nguin | chan_skinny barely works at all. |
| 22:33.32 | p3nguin | The last time I tried it, I couldn't even make a call with it. |
| 22:33.51 | simplydrew | hmm. yeah, I'm trying to make the transition to this 7965 and am a little in the dark |
| 22:33.58 | catphish | i once used sccp flawlessly around 1.2 but i forget how |
| 22:33.59 | dijib | p3nguin, i think while installing * i chose not to install iax, can i still have asterisk add those moduels? |
| 22:34.07 | catphish | sccp-b seems better |
| 22:34.13 | catphish | but i dont know how to install it |
| 22:34.41 | p3nguin | dijib: If you still have your build tree, it would be easy to go back to it and make menuselect to turn on iax, then make install again. |
| 22:34.55 | p3nguin | simplydrew: What asterisk version are you using? |
| 22:35.17 | dijib | i think i had it in /tmp |
| 22:35.19 | dijib | :S |
| 22:35.24 | dijib | not /usr/src |
| 22:35.29 | p3nguin | dijib: If it is still there, go back to it. |
| 22:35.38 | simplydrew | p3nguin: 1.6 |
| 22:35.41 | catphish | basically i just want to play "you have no credit remaining" messages to sip users when they attempt to make calls |
| 22:35.48 | p3nguin | simplydrew: That's a branch; I asked what version. |
| 22:35.52 | dijib | most likely not, i power cycled |
| 22:35.59 | catphish | and obviously i dont want the calls to appear answered in the cd |
| 22:36.01 | catphish | *cdr |
| 22:36.09 | simplydrew | p3nguin: 1.6.0.26-FONCORE-r78 |
| 22:36.17 | p3nguin | dijib: Are you saying that your /tmp is not persistent? |
| 22:36.17 | dijib | wait it is. |
| 22:36.35 | dijib | no i have my tree. should i be backing up my conf's? |
| 22:37.41 | p3nguin | Always back up your confs. |
| 22:37.47 | p3nguin | And don't run "make samples" |
| 22:37.56 | p3nguin | simplydrew: Are you comfortable using svn? |
| 22:38.17 | simplydrew | p3nguin: I've used it once or twice, a little shaky. but I can give it a try |
| 22:38.31 | p3nguin | svn co https://chan-sccp-b.svn.sourceforge.net/svnroot/chan-sccp-b/branches/V3.0/ chan-sccp-b_V3.0 |
| 22:38.49 | p3nguin | Then ./configure, make, and make install as usual. |
| 22:38.49 | dijib | do i need to core stop now? |
| 22:38.55 | p3nguin | dijib: no |
| 22:39.09 | dijib | k\ |
| 22:39.11 | p3nguin | dijib: Go to the build directory and make menuselect. |
| 22:39.21 | dijib | im in it |
| 22:39.30 | p3nguin | Go to the channel drivers section and enable iax2. |
| 22:39.43 | Bipul | for(i=1;i<=4;i++) can any one tell me what does & means? here exactly in C we have Double && for AND operator |
| 22:42.10 | dijib | p3nguin, iax2 is already selected, but when i do a core show help i see no iax2 stuff |
| 22:42.51 | p3nguin | dijib: Do you have the iax2 channel driver module in your modules directory? /usr/lib/asterisk/modules |
| 22:43.38 | simplydrew | p3nguin: okay, configured, make, and make install completed. do I need to download the specific scup firmware for my 7965 and put it in a directory somewhere before going any further? I'm used to dealing with sip, so wasn't sure |
| 22:44.09 | dijib | yeh its in there |
| 22:44.13 | dijib | i guess its not loaded |
| 22:44.20 | dijib | module load chan_iax2.so ??? |
| 22:44.24 | p3nguin | simplydrew: The sccp firmware will load onto the phones the same as sip firmware. I put a tftpd on my asterisk box for that. |
| 22:44.52 | p3nguin | If it isn't chan_iax2.so, it's chan_iax.so |
| 22:45.15 | dijib | ya k ive got it loaded now |
| 22:45.21 | dijib | howcome that wasnt automatic? |
| 22:45.32 | dijib | how are you today BTW |
| 22:46.05 | p3nguin | If you are using autoload, it could be that you have a noload for it or that you didn't have a conf for it when asterisk started. |
| 22:46.54 | p3nguin | I finally fixed that damn brake. I replaced the hose and it still hung up, so I replaced the caliper, too. The problem seems to be solved. |
| 22:47.20 | p3nguin | I test drove it before changing the caliper and the brake rotor was so hot that it was smoking. |
| 22:47.49 | p3nguin | I changed the caliper and test drove it again, and then I was able to touch the rotor with my bare fingers without getting burned. |
| 22:48.25 | dijib | then your caliper was seized |
| 22:48.42 | p3nguin | So it just needed a couple hours worth of work and about $100 thrown at it. |
| 22:48.48 | p3nguin | And it does use DOT 3 fluid. |
| 22:48.55 | dijib | if you didnt see any pinches in that brake line then hydrolics should be good |
| 22:49.19 | p3nguin | I changed the hose first, since it was cheapest, but that didn't fix it. |
| 22:49.39 | dijib | you could lose an eye if you get hit with that hydrolic oil from a line brake |
| 22:49.43 | p3nguin | So now I have new pads, new caliper, new hose, and new rotor on that corner. |
| 22:50.01 | dijib | the rubber hoses wont collapse as i see them |
| 22:50.16 | dijib | i need to do some major work b4 winter |
| 22:50.34 | p3nguin | Sometimes they do inside and you can't tell from the outside, so I changed it hoping that was the problem. |
| 22:50.39 | dijib | im still having issues getting iaxmodem to work |
| 22:50.43 | p3nguin | I should have stuck to my first feeling and just changed the caliper. |
| 22:50.52 | p3nguin | Why do you care about iaxmodem? |
| 22:51.07 | dijib | i want an email to fax option for lagal crud |
| 22:51.21 | p3nguin | I suppose you're going to want hylafax next. |
| 22:51.31 | dijib | eff no |
| 22:51.32 | dijib | y? |
| 22:51.47 | p3nguin | Seems like something else you'd want but don't need. |
| 22:51.58 | dijib | whats it do? |
| 22:52.09 | p3nguin | Google didn't know? |
| 22:52.29 | dijib | urg |
| 22:53.32 | dijib | i dont want it |
| 22:53.56 | *** join/#asterisk pdtpatrick_ (~pdtpatric@mainstwan.farheap.com) |
| 22:54.03 | pdtpatrick_ | Question .. what does this mean ? |
| 22:54.05 | pdtpatrick_ | [Sep 8 15:52:52] WARNING[9822]: app_dial.c:1747 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown) |
| 22:54.19 | dijib | next i want an auto wifi encryption cracking auto connecting router firmware |
| 22:55.55 | pabelanger | pdtpatrick_: you dialled something that didn't exist in Asterisk. EG: An unregistered peer |
| 22:55.55 | p3nguin | pdtpatrick_: Dial() was not able to create a channel to the device you wanted to dial. |
| 22:58.03 | pdtpatrick_ | Thanks |
| 22:58.16 | catphish | wow allison recorded some hilarious sounds |
| 22:58.31 | pdtpatrick_ | are you using cepstral ? vxml ? |
| 22:58.57 | p3nguin | Some of the sounds involving Karl are amusing. |
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| 23:10.38 | *** join/#asterisk prgmrchris (~chris@66.9.61.162) |
| 23:11.55 | prgmrchris | anyone in northeast US, and know if its possible to resell verizon pots or if theres any alternate pots providers? i have a friend whos just fed up with verizon and wants to do a hybrid pstn/itsp setup but he really doesnt want to deal with verizon at all for anything |
| 23:12.25 | prgmrchris | i have a feeling verizon is the only game and you cant resell :( |
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