| 00:03.50 | *** join/#asterisk Aiatek (n=munoz@190.6.143.194) |
| 00:08.35 | *** join/#asterisk juanIMP (n=Juancho@200.26.152.222) |
| 00:10.40 | *** join/#asterisk mchou (n=mchou@unaffiliated/mchou) |
| 00:10.57 | *** join/#asterisk coppice (n=chatzill@46.166.17.210.dyn.pacific.net.hk) |
| 00:18.41 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 00:19.22 | *** join/#asterisk elitecoder (n=liq@apollo.bullethost.com) |
| 00:19.35 | elitecoder | At what point do call files get removed from the outgoing folder? |
| 00:21.40 | tfrew | when you make a cron job to remove them |
| 00:21.59 | elitecoder | no that's wrong |
| 00:22.22 | elitecoder | asterisk removes them on it's own... but I was wondering when exactly |
| 00:24.47 | *** join/#asterisk MaliutaLap (n=biteme@203.171.192.191) |
| 00:29.33 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
| 00:33.10 | deeperror | What could cause a channel instance to show the wrong number? "Building conference on call on Zap/38-1 and Zap/38-1" instead of one being 38-1 |
| 00:33.23 | deeperror | 38-2 |
| 00:34.09 | *** join/#asterisk autojack (n=owen@nerdnetworks.org) |
| 00:34.55 | autojack | I have a string of messages like this in my log: NOTICE[6277]: chan_sip.c:13952 handle_request_invite: Call from '' to extension '011442033935118' rejected because extension not found. |
| 00:35.10 | elitecoder | heh that's long |
| 00:35.20 | autojack | I suspect this is someone trying to use my PBX, unsuccessfully. |
| 00:35.34 | autojack | the requests try a bunch of different variations on that number. |
| 00:35.51 | elitecoder | When that happened to me I just blocked their IP |
| 00:36.17 | elitecoder | (Using the server's firewall) |
| 00:36.21 | autojack | not sure how they're connecting though. well, I guess trying to use SIP. |
| 00:36.31 | autojack | but there's only one SIP account and it has a password. |
| 00:36.55 | elitecoder | I don't think inbound calls require a password but I'm a newbie |
| 00:37.16 | autojack | the only inbound trunk is via a DID. |
| 00:37.34 | autojack | but when you actually dial the DID I get a lot of other data. |
| 00:37.53 | elitecoder | you could show me your sip conf really quick if you want |
| 00:37.56 | elitecoder | maybe I can help |
| 00:38.24 | autojack | nah it's OK |
| 00:38.31 | deeperror | autojack, seems like they would be going to default context could try to answer them see who it is |
| 00:38.33 | elitecoder | maybe you should look for variables to put in the general section that may require some kind of authentication |
| 00:38.45 | elitecoder | haha "STOP CALLING MY BOX" |
| 00:38.49 | autojack | haha |
| 00:38.51 | elitecoder | record that and play it back |
| 00:38.55 | autojack | these were from a few days ago. |
| 00:39.00 | autojack | so it's not active at the moment. |
| 00:39.15 | autojack | if you call my DID it dumps them into a specific extension to handle stuff. |
| 00:39.16 | deeperror | i think it's anonymous sip traffic? |
| 00:39.22 | autojack | so I don't know how else they could be connecting. |
| 00:39.24 | *** join/#asterisk Steve_J-obs (n=Chris123@pool-96-233-110-127.bstnma.fios.verizon.net) |
| 00:39.39 | Steve_J-obs | Hello everybody!!! |
| 00:39.50 | elitecoder | it would still go into the same context but fail because the extension they're trying isn't there |
| 00:40.54 | autojack | yeah |
| 00:40.57 | elitecoder | if you can find out exactly where they ended up and work backwards maybe you can figure out how they got there |
| 00:40.57 | Aiatek | hi, i have to configure a te207p, i have experience configuring te110p |
| 00:41.05 | elitecoder | with your error messages |
| 00:41.09 | Aiatek | what do i need to add |
| 00:41.24 | Aiatek | i used span=1,0,0,d4,ami |
| 00:41.30 | Aiatek | to the first one |
| 00:42.03 | Steve_J-obs | I am logging in here to see if I can learn from any of you geniouses |
| 00:42.04 | Aiatek | what do i need to add to configure the second T1 |
| 00:43.38 | Aiatek | ? |
| 00:52.13 | Aiatek | anybody can help me configuring a te207p |
| 00:52.28 | Aiatek | i just need a few lines |
| 00:57.15 | *** join/#asterisk ingenius (n=alektro@host57.190-138-60.telecom.net.ar) |
| 00:59.39 | *** join/#asterisk zamba (i=marius@flage.org) |
| 01:12.10 | *** join/#asterisk umpc (n=Justin@unaffiliated/umpc) |
| 01:20.19 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 01:21.08 | *** join/#asterisk stevetotaro (n=Steve@c-69-243-124-5.hsd1.md.comcast.net) |
| 01:22.17 | *** join/#asterisk stevetotaro (n=Steve@c-69-243-124-5.hsd1.md.comcast.net) |
| 01:23.06 | *** join/#asterisk stevetotaro (n=Steve@c-69-243-124-5.hsd1.md.comcast.net) |
| 01:27.18 | *** join/#asterisk maxagaz (n=maxagaz@soho2.i-xanadu.com) |
| 01:28.38 | *** join/#asterisk moy (n=moy@CPE001cdfec4cee-CM00080dab8485.cpe.net.cable.rogers.com) |
| 01:41.07 | *** join/#asterisk blaxthos (n=blaxthos@onyx.krisp.com) |
| 01:41.23 | *** part/#asterisk blaxthos (n=blaxthos@onyx.krisp.com) |
| 01:45.38 | elitecoder | I have a question about sip trunks. If I get a line and make several calls at once over that line what happens? |
| 01:46.01 | crunge | elitecoder: they go over the trunk |
| 01:46.42 | elitecoder | so I can get a single line and have 5 outgoing calls at once? |
| 01:47.06 | crunge | elitecoder: a trunk is not a line |
| 01:47.21 | elitecoder | I don't need 5 sip trunks to have 5 simultaneous calls then? |
| 01:47.29 | elitecoder | Ok. |
| 01:50.16 | deeperror | elitecoder, the provider may call it channels |
| 01:50.48 | crunge | elitecoder: trunk sends a call over a line rather than just sound data |
| 01:50.55 | *** join/#asterisk Kumbang (n=unknown@125.163.83.153) |
| 01:55.47 | *** join/#asterisk notjoo00 (n=hexman@mx01.anric.com.au) |
| 01:56.44 | notjoo00 | can someone help me with setting multiple sip providers in asterisk? would like to use 2 providers, primary with only 1 simultaneous call, and 2nd for rest of calls |
| 01:58.53 | *** join/#asterisk Globettrotter (n=eric@ool-457a1c8a.dyn.optonline.net) |
| 01:59.06 | *** join/#asterisk digitalirony (i=digitali@shellium/member/digitalirony) |
| 01:59.54 | Globettrotter | hi guys,, im trying to make calls out via DAHDI.. i get this error/// No translator path exists for channel type DAHDI (native 0x4c) to 0x100 |
| 02:00.07 | Globettrotter | Unable to create channel of type 'DAHDI' (cause 58 - Bearer capability not available |
| 02:00.17 | elitecoder | Thanks crunge and deeperror. |
| 02:05.21 | deeperror | Globettrotter, how you trying to dial those calls out? |
| 02:09.31 | Globettrotter | im using eyebeam,, |
| 02:09.53 | Globettrotter | DAHDI/trunk_1/8xxxxxxx |
| 02:12.40 | joobie | guys i have two sip providers.. problem is, the first sip provider only accepts 1 concurrent call at a time |
| 02:12.53 | *** join/#asterisk Octothorpe (n=Octothor@pdpc/supporter/professional/octothorpe) |
| 02:12.56 | joobie | is there a way i can setup the dialplan so that after that one call is made, it pushes the call to the next provider? |
| 02:13.17 | joobie | i tried playing with return codes, but unfrotunately the provider doesnt give me a busy tone, they just start playing a message saying i cant have more than 1 call at a given time |
| 02:13.47 | joobie | so i think i need to track the call use on my end and push through that way.. was thinking astdb could be used (ie. set and check a db entry when the line is in use), but duno if that's the best way to achieve this.. |
| 02:13.50 | *** join/#asterisk egypcio (n=egypcio@unaffiliated/egypcio) |
| 02:21.30 | KyleK | that reminds me, i need to find a sip provider for some outgoing calls |
| 02:28.30 | *** join/#asterisk devhen (n=devhen@c-67-169-245-6.hsd1.ut.comcast.net) |
| 02:33.33 | *** part/#asterisk juanIMP (n=Juancho@200.26.152.222) |
| 02:37.04 | *** join/#asterisk arthax0r (i=arthax0r@85.8.24.245.static.se.wasadata.net) |
| 02:42.57 | [TK]D-Fender | joobie: "core show application chanisavail" |
| 02:43.17 | [TK]D-Fender | [22:09]<Globettrotter>DAHDI/trunk_1/8xxxxxxx <- not valid |
| 02:49.12 | Globettrotter | D-Fender for testing i have this/// exten = _983352663,1,Dial(DAHDI/${EXTEN:1}) |
| 02:49.54 | Globettrotter | now i get this // Unable to create channel of type 'DAHDI' (cause 0 - Unknown) |
| 02:49.58 | [TK]D-Fender | Globettrotter: Nope. |
| 02:51.09 | [TK]D-Fender | Globettrotter: Dial(DAHDI/[enter the specific channel or GROUP like "g1", "g2", etc here minus any braces or quotes]/[the number to dial without braces]) |
| 02:52.08 | Globettrotter | ok,, im going to try that now |
| 02:59.02 | Globettrotter | i got this.. exten = _983352663,1,Dial(DAHDI/g0/83352663) i get this error: Unable to create channel of type 'DAHDI' (cause 0 - Unknown) |
| 03:00.46 | *** join/#asterisk lanning (n=lanning@173.8.187.197) |
| 03:06.09 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 03:07.06 | deeperror | Globettrotter, exten => _98 |
| 03:11.15 | Globettrotter | use _98 instead of _983352663 ? |
| 03:11.29 | deeperror | use a => instead of = |
| 03:12.07 | *** join/#asterisk jsgoecke (n=jsgoecke@c-67-180-102-94.hsd1.ca.comcast.net) |
| 03:12.31 | Globettrotter | ok |
| 03:12.36 | Globettrotter | tryng now |
| 03:13.59 | Globettrotter | that did not seem to make a difference |
| 03:14.22 | Qwell | it doesn't make a difference |
| 03:14.39 | [TK]D-Fender | Globettrotter: You have to have defined channels as belonging to group 0 |
| 03:14.56 | [TK]D-Fender | Globettrotter: I do not assume this is actually valid. SHOW me that you did do it |
| 03:16.20 | Globettrotter | i can paste my config file |
| 03:17.38 | Globettrotter | D-Fender,, how do you want me to show you? |
| 03:19.11 | Globettrotter | under /etc/asterisk/dahdi-channels.conf groups are set to 0 |
| 03:20.42 | [TK]D-Fender | Globettrotter: PASTEBIN is your friend... |
| 03:20.45 | [TK]D-Fender | ~pb |
| 03:20.45 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste , or , http://bin.cakephp.org/ |
| 03:20.47 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^ |
| 03:25.34 | Globettrotter | http://pastebin.com/d931dd3a |
| 03:25.50 | Globettrotter | this is my extensions.conf |
| 03:28.20 | *** join/#asterisk MrNaz (n=mrnaz@118.208.154.153) |
| 03:40.52 | *** part/#asterisk MIR100 (n=mir100@24-148-89-37.frg-bsr1.chi-frg.il.cable.rcn.com) |
| 03:42.31 | *** join/#asterisk nicoAMG (n=superunk@201.203.50.42) |
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| 03:58.48 | *** part/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 04:11.59 | kuku1 | [TK]D-Fender: moho = music on hold |
| 04:12.32 | kuku1 | [TK]D-Fender: any troubleshooting steps I can do to track down this loss of sound and repeat of sound in sip to sip ? |
| 04:12.41 | kuku1 | canreinvite=no |
| 04:14.57 | joobie | [TK]D-Fender, sorry man just got back |
| 04:15.07 | joobie | i got your msg on 'core show application chanisavail' |
| 04:15.21 | joobie | is there a way i can automate this from the diaplan? like a variable i can check? |
| 04:16.01 | joobie | sorry my bad :) just tried the command |
| 04:16.03 | joobie | cheers :) |
| 04:25.43 | *** join/#asterisk ultrav1olet (n=ultrav1o@94.180.29.50) |
| 04:27.00 | ultrav1olet | We've got a major problem after upgrading Asterisk 1.4 to 1.6 - it calls "wrong" numbers. What can be done about that? |
| 04:29.58 | [TK]D-Fender | ultrav1olet: Sorry, could you be a little more vague please.... |
| 04:40.12 | *** join/#asterisk werdan7 (n=w7@freenode/staff/wikimedia.werdan7) |
| 04:41.07 | ultrav1olet | OK, When I call 2123456 the station says either "Number doesn't exist" or someone else answers (like a completely different number). |
| 04:41.24 | *** join/#asterisk GameGamer43 (n=GameGame@cpe-66-67-181-247.rochester.res.rr.com) |
| 04:41.56 | *** join/#asterisk Kumbang (n=unknown@125.163.83.153) |
| 04:43.04 | [TK]D-Fender | ultrav1olet: pastebin your dialplan and your failed call attempt at verbose 10 |
| 04:43.06 | [TK]D-Fender | ~pb |
| 04:43.06 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste , or , http://bin.cakephp.org/ |
| 04:43.08 | [TK]D-Fender | ^^^^^^^^666 |
| 04:44.46 | ultrav1olet | what kind of configs would you like me to post? |
| 04:44.58 | ultrav1olet | aaaah, I see, sorry |
| 04:47.35 | ultrav1olet | http://pastebin.ca/1405749 |
| 04:48.05 | ultrav1olet | Ooops, the write extension is: exten => _8.,1,Dial(DAHDI/1/${EXTEN}) |
| 04:48.16 | ultrav1olet | right :) |
| 04:48.31 | ultrav1olet | should sleep more ;) |
| 04:49.09 | joobie | TK.. |
| 04:49.14 | ultrav1olet | [TK]D-Fender: I suppose there's something wrong with DAHDI or asterisk DAHDI configuration. Would you like to see them? |
| 04:49.19 | joobie | how do u think i should use ChanisAvail? |
| 04:49.45 | joobie | i see it sets variables.. reacon i should just run that before i dial, then check the variable .. then dial the number? |
| 04:50.07 | [TK]D-Fender | ultrav1olet: exten => _82XXXXXX,1,Dial(DAHDI/1/${EXTEN:1}) |
| 04:50.15 | [TK]D-Fender | ultrav1olet: -- Executing [83422444273@i_am_sane:1] Dial("IAX2/birdie-16293", "DAHDI/1/83422444273") in new stack |
| 04:50.26 | [TK]D-Fender | ultrav1olet: The dialplan you showed me is NOT what is being processed |
| 04:50.32 | [TK]D-Fender | ultrav1olet: completely different |
| 04:51.03 | [TK]D-Fender | ultrav1olet: and [i_am_sane] sure doesn't look like [outgoing_calls] to me |
| 04:51.34 | ultrav1olet | [TK]D-Fender: The problem is not in dialplan :) OK, Wait a sec |
| 04:53.19 | ultrav1olet | http://pastebin.ca/1405752 |
| 04:54.50 | [TK]D-Fender | [00:50]<[TK]D-Fender>ultrav1olet: -- Executing [83422444273@i_am_sane:1] Dial("IAX2/birdie-16293", "DAHDI/1/83422444273") in new stack <-- look at the bloody # and context. Damn right its dialplan, none of those patterns start with 83 with a length like that! |
| 04:54.54 | [TK]D-Fender | ^^^^^^^^ |
| 04:55.19 | ultrav1olet | exten => _8.,1,Dial(DAHDI/1/${EXTEN}) |
| 04:55.32 | [TK]D-Fender | ugh |
| 04:55.34 | ultrav1olet | [TK]D-Fender: you probably haven't slept enough as well :) |
| 04:55.45 | [TK]D-Fender | ultrav1olet: Fair enough, I missed that one |
| 04:55.54 | ultrav1olet | anyway, 50/50% I hit a wrong number |
| 04:56.02 | [TK]D-Fender | ultrav1olet: Working with customers right now... multi-task failure |
| 04:56.27 | ultrav1olet | [TK]D-Fender: it happens. Do you want to see my dahdi configuration and chan_dahdi.conf? |
| 04:56.53 | [TK]D-Fender | ultrav1olet: Ok, so that dialplan line is doing what it says. You dial a number starting with 8 and it pumps it out "as-is" |
| 04:57.00 | [TK]D-Fender | ultrav1olet: what don't you like about this? |
| 04:57.06 | *** join/#asterisk mintos (n=mvaliyav@nat/redhat-in/x-071688d6af3e7f2d) |
| 04:57.59 | *** join/#asterisk b14ck (n=blacky@cpe-98-151-210-28.socal.res.rr.com) |
| 04:58.02 | b14ck | hi |
| 04:58.10 | b14ck | quick question--anyone here use festival? |
| 04:58.36 | b14ck | im a festival noob, and im trying to get it to work. my dialplan does an answer, then: Festival("test test test") ; but it isnt working |
| 04:58.41 | b14ck | any ideas? |
| 05:00.42 | ultrav1olet | I don't like the fact that in 25-50% of cases when I dial, my Wildcard TDM400P REV E/F Board 5 dials a wrong number :( |
| 05:00.48 | joobie | TK.. i tried 'ChanIsAvail(SIP/pennytel)' .. then i do NoOp(${AVAILSTATUS}) and that's showing me always '0' |
| 05:00.55 | joobie | + even if i use the SIP account |
| 05:02.26 | [TK]D-Fender | joobie: read the instructions. You want to see if its INUSE so you can coose to do something else if it is |
| 05:03.03 | [TK]D-Fender | ultrav1olet: well * is dialing it out. Odds are your analog line loses the first digit, or maybe 2. add "www" before the number |
| 05:03.16 | [TK]D-Fender | ultrav1olet: 1.5 second waith for dialtone. |
| 05:03.32 | joobie | [TK]D-Fender, it always returns 0 |
| 05:03.36 | joobie | if it's in use or not....... |
| 05:03.40 | [TK]D-Fender | joobie: READ THE DAMN INSTRUCTIONS |
| 05:03.47 | [TK]D-Fender | :p |
| 05:04.07 | joobie | TK |
| 05:04.14 | joobie | the problem is, chanisunavail() aint great with SIP |
| 05:04.28 | joobie | i read the instructions.. it always returns 0.. it never returns taht the channel is in use |
| 05:04.54 | [TK]D-Fender | joobie: I told you to use that function because I know for a fat it can do the job if you CALL IT PROPERLY. Now go read the f-ing instructions again :) |
| 05:05.00 | joobie | i put in another hack, not so clean.. but i set a call-limit to 1 on the peer and i check ${DIALSTATUS} variable.. which gives me correct behavior to reroute on |
| 05:05.23 | joobie | fuk ok |
| 05:05.25 | joobie | gona re-read again |
| 05:07.52 | joobie | sry TK |
| 05:07.58 | joobie | it's not performing as expected *hides* |
| 05:08.33 | joobie | TK.. i see what you're saying.. the problem is, some write-up's say ChanIsAvail() is not accurate on SIP channels |
| 05:08.38 | [TK]D-Fender | joobie: Your expectations and ability to read, understand, and then follow instructions have no relationship with each other. |
| 05:08.56 | joobie | i even tried appending the |s option to the end.. which returns '6' (AST_DEVICE_RINGING) if it's in use |
| 05:09.03 | ultrav1olet | [TK]D-Fender: OK, Let's try www ;) |
| 05:09.05 | joobie | i can't get it to return IN USE |
| 05:09.07 | [TK]D-Fender | joobie: It WILL work, and I know how to do this myself with that samn damn function and have done this dozens of times before. |
| 05:09.33 | [TK]D-Fender | joobie: And I'm not seeing much. |
| 05:09.39 | joobie | exten => _0[8-9]XXXXXXX,n,ChanIsAvail(SIP/pennytel|s); exten => _0[8-9]XXXXXXX,n,NoOp(${AVAILSTATUS}) |
| 05:09.41 | joobie | that is all i'm doing man |
| 05:09.44 | joobie | literally |
| 05:09.51 | [TK]D-Fender | ultrav1olet>exten => _8.,1,Dial(DAHDI/1/www${EXTEN}) |
| 05:09.52 | joobie | and looking at what NoOP() returns.. |
| 05:10.14 | joobie | i removed the 's' option from ChanIsAvail, and I always get a return code of 0 .. if i use the 's' option, i get a return code of 6 if the line is in use |
| 05:10.30 | [TK]D-Fender | joobie: Sounds like you can tell if its in use then <- |
| 05:10.49 | joobie | yes, but it's returning 6 AST_DEVICE_RINGING - "Ringing"; ring, ring, ring. |
| 05:10.52 | [TK]D-Fender | joobie: 0, or NON 0 |
| 05:10.56 | joobie | as opposed to 2 AST_DEVICE IN USE - "In use"; channel is in use. |
| 05:11.26 | [TK]D-Fender | joobie: CLOSE ENOUGH |
| 05:11.29 | joobie | haha |
| 05:11.34 | joobie | same same, but different |
| 05:11.38 | joobie | fuk it man.. i might just use ${DIALSTATUS} |
| 05:11.42 | joobie | it looks cleaner |
| 05:11.53 | [TK]D-Fender | joobie: Does the f-ing job, don't whine about the colour of the check when its for $1,000,000 |
| 05:12.02 | [TK]D-Fender | :p |
| 05:12.04 | joobie | actually, i guess ${DIALSTATUS} needs to actually dial.. which isnt as clean as doing a check and not dialling if we get a non-0 |
| 05:12.10 | joobie | lol |
| 05:12.17 | joobie | kk |
| 05:12.17 | joobie | ta |
| 05:14.47 | ultrav1olet | [TK]D-Fender: Is there a way to "hear" the signal from the phone station and to hear how my asterisk dials a number? |
| 05:14.59 | [TK]D-Fender | ultrav1olet: Nope |
| 05:15.11 | ultrav1olet | pity |
| 05:15.13 | [TK]D-Fender | ultrav1olet: Well... unles you plug a real phon in parallele and pick it up... |
| 05:15.29 | ultrav1olet | [TK]D-Fender: a nice idea :) |
| 05:16.12 | b14ck | any of you familiar with festival? |
| 05:16.35 | ultrav1olet | [TK]D-Fender: http://pastebin.ca/1405771 |
| 05:17.05 | ultrav1olet | could it be that some of chan_dahdi.conf settings affect the way my digium card dials a number? |
| 05:17.22 | ultrav1olet | or echocanceller= setting of dahdi? |
| 05:17.51 | ultrav1olet | because with asterisk 1.4 we didn't have any echo cancelling in zapata.conf |
| 05:18.04 | b14ck | http://pastebin.com/m4061e94e <=== my dialplan, and my asterisk log of what happens when i dial it. festival isnt working :( the festival log file isn't showing any useful info |
| 05:18.09 | [TK]D-Fender | ultrav1olet: Not really, looks fine... |
| 05:18.26 | ultrav1olet | old zapata.conf contained just: fxsks=1-4 |
| 05:18.27 | ultrav1olet | loadzone=fr |
| 05:19.34 | [TK]D-Fender | ultrav1olet: Its fine. Before you used to have to completely recompile Zaptel to pick your EC routine to use. Now yo can do it in your configs |
| 05:20.39 | [TK]D-Fender | oops |
| 05:23.32 | ultrav1olet | Calling "DAHDI/1/www8342212345" - "You've dialed a wrong number. Please, check if the number you're calling is right" |
| 05:23.35 | ultrav1olet | damn |
| 05:24.01 | *** join/#asterisk drmessano^ (n=nonya@pdpc/supporter/active/drmessano) |
| 05:24.49 | drmessano | gives the finger to card.freenode.net |
| 05:24.50 | [TK]D-Fender | ultrav1olet: is the # part of that legit? |
| 05:27.19 | joobie | TK |
| 05:27.59 | joobie | got a query RE encoding for MOH. I read somewhere that RAW format is good, because * doesn't need to decode it as its in its raw format |
| 05:28.06 | joobie | so saves I/O |
| 05:28.25 | [TK]D-Fender | joobie: forget raw, have it in the codec your call is in |
| 05:28.42 | joobie | converted an MP3 to raw and was just about to put it on the box when I thought this MOH will go out an ISDN channel.. which is alaw i think... so it would take raw, then re-encode it to alaw to put it down the channel, presumably |
| 05:28.48 | joobie | bam |
| 05:28.55 | joobie | you read my mind |
| 05:29.43 | joobie | TK, is there a way to confirm what codec my ISDN uses? |
| 05:29.47 | joobie | pretty sure it's alaw |
| 05:32.13 | [TK]D-Fender | joobie: good guess |
| 05:32.53 | joobie | Fender, anyway to confirm in console / configs? |
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| 05:35.41 | [TK]D-Fender | joobie: uearo ISDN uses ALAW |
| 05:35.47 | [TK]D-Fender | checkout time, later all |
| 05:39.27 | ultrav1olet | [TK]D-Fender gone? |
| 05:54.36 | *** join/#asterisk grEvenX (n=even@apb9hb.ip.ssc.net) |
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| 06:08.39 | joobie | gone with the wind |
| 06:09.18 | joobie | guys anyway to get asaterisk to alter the dialplan based on time of day? I want to put like a "night mailbox" feature in.. so calls dojn't go to queues at certain timeframes of the day |
| 06:10.15 | kaldemar | joobie: yes, includes can take different time definitions. |
| 06:11.13 | kaldemar | joobie: http://www.voip-info.org/wiki/view/Asterisk+tips+openhours |
| 06:13.58 | joobie | kaldemar, is that 1.4 supported? |
| 06:14.02 | kaldemar | yes |
| 06:14.29 | joobie | cool :) crap thing is i need to convert all my dialplans to these contexts now! argh :P thanks |
| 06:14.40 | joobie | hey one more Q |
| 06:15.54 | joobie | I'm writing an application that updates the microbrowser file, showing on the phone how many people are in a specific queue |
| 06:16.17 | joobie | just wondering if there's an easy way to pull this info out of asterisk.. i dont have mysql integration so it will have to pull it out of asterisk itself somehow |
| 06:16.30 | joobie | are there good interfaces for this sorta stuff? |
| 06:17.06 | kaldemar | check into AMI if there's a suitful command to check queues. if not, there's always the CLI. |
| 06:18.35 | kaldemar | there seem to be manager interface (AMI) commands for queues. you should be able to use those. |
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| 07:02.23 | k4mi1 | hi all |
| 07:05.55 | k4mi1 | does anyone know something about T.38 implementation on ASTERISK? :] |
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| 07:34.16 | k4mi1 | anyone? |
| 07:38.43 | *** join/#asterisk cjk (n=cjk@vodsl-9494.vo.lu) |
| 07:39.07 | cjk | hi, does anyone know a way to show the called number on a snom phone? |
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| 07:44.17 | angryuser | cjk hello |
| 07:44.27 | cjk | hi angyuser :) |
| 07:45.04 | angryuser | you can SET(${CALLERID(name)) to a number you want before calling that snom phone |
| 07:45.16 | cjk | yes but i was looking for a cooler solution |
| 07:45.23 | cjk | this would also affect cdrs |
| 07:45.30 | cjk | missed calls list |
| 07:45.31 | cjk | etc... |
| 07:46.07 | angryuser | cjk, for example i have an agent who is 2 queues, SUPPORT and COMMERCIAL, i do SET this name so the agent see for what reasong and from which queue he is called |
| 07:46.22 | cjk | thats what i do at the moment |
| 07:46.23 | angryuser | very usefull |
| 07:46.31 | cjk | but i thought maybe something using sip headers |
| 07:47.19 | angryuser | cjk, well, i dopnt know what snom shows or how tom modify sip header with asterisk |
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| 07:51.12 | Urthwhyte | ~sip |
| 07:51.12 | infobot | sip is, like, http://www.cs.columbia.edu/sip/ X11 PPP dialer interface written in gtk+. URL: http://www.geocities.com/SiliconValley/Campus/3104/sip/ Session Initiation Protocol (see RFC 3261) |
| 07:51.17 | Urthwhyte | ~nat |
| 07:51.17 | infobot | nat is, like, Network Address Translation Usable in Asterisk sip.conf file with externip, localnet, and localmask setup properly. See docs. |
| 07:56.18 | *** join/#asterisk sulex (n=sulex@pdpc/supporter/professional/sulex) |
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| 08:30.16 | proxium | Hi to all, "DIAL_TRUNK_OPTIONS=rTto" what does it mean or is there any technical document to follow? |
| 08:31.05 | *** part/#asterisk iamy_china (n=iamy_chi@222.128.3.30) |
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| 08:51.27 | tzafrir_laptop | proxium, core show application dial |
| 08:52.54 | proxium | tzafrir_laptop, thank you a lot, so all is hidden there :p |
| 08:53.21 | tzafrir_laptop | and specifically 'r' may not be the greatest idea |
| 08:56.47 | *** join/#asterisk oej (n=olle@cust-IP-10.data.tre.se) |
| 08:57.17 | *** join/#asterisk yokobr (n=jeff@189.12.127.43) |
| 08:57.28 | yokobr | hey guys |
| 08:58.33 | yokobr | is there any way to install graphical interface on asterisknow? |
| 08:59.03 | yokobr | and, by off, there some guys here in brazil selling "asterisk servers"... |
| 08:59.12 | yokobr | that's not fair |
| 08:59.28 | yokobr | they're using asterisk name to sell hardware. |
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| 09:02.31 | MrParity | hello :-) |
| 09:04.05 | jbjapan | hello |
| 09:04.39 | MrParity | i have a question about queues. i want to write an external application which is able to show the callers in the que (and maybe the agents), but i don't know how to get this information |
| 09:05.05 | MrParity | does anyone know how i figure out who is in the queue? |
| 09:05.31 | MrParity | hi jbjapan :-) |
| 09:08.50 | MrParity | i know there are some applications (like queuemetrics) which are doing it, but i don't know how |
| 09:09.44 | freh | I'm having trouble getting call parking to work. The extension where the call is parked does not get announced to me by asterisk |
| 09:10.02 | *** join/#asterisk MaliutaLap (n=biteme@kiev.lusan.id.au) |
| 09:10.49 | freh | I can see it getting parked on the console, and I can connect back to it. It's just not getting announced. |
| 09:11.02 | freh | I'm using an attended transfer to park the call |
| 09:20.36 | *** join/#asterisk postel (n=jp@wikimedia/Postel) |
| 09:26.26 | yang | I am wondering about the FAX (hylafax) error, about uncompatible codecs, it used to go with alaw, now it doesn't work any longer ... http://pastebin.ca/1405899 |
| 09:26.42 | yang | coming in over SIP then forwarding to IAX |
| 09:29.43 | *** join/#asterisk phix (n=threat@123-243-44-131.tpgi.com.au) |
| 09:29.51 | *** join/#asterisk genin (n=phrame@ANice-252-1-47-95.w82-122.abo.wanadoo.fr) |
| 09:29.51 | k4mi1 | when I want to send fax over T.38 from Device_1 (T.38 capable) to Device_2 (T.38 NOT capable) I have an error 488 and then asterisk stops a call but he should inform Device_1 that second device does not support the T.38 protocol - is it proper behaviour or maybe this feature is not implemented in ast 1.6? |
| 09:29.54 | genin | mornin folks |
| 09:31.01 | genin | anyone alive? |
| 09:31.18 | freh | morning |
| 09:31.25 | genin | hows it going? |
| 09:31.37 | freh | ok.. |
| 09:31.41 | genin | :/ |
| 09:31.42 | genin | heh |
| 09:31.48 | k4mi1 | hi |
| 09:31.56 | genin | you know anything about a diguim TE420 E1 card? |
| 09:32.08 | genin | T1 if your american |
| 09:32.11 | genin | heh |
| 09:32.15 | freh | I'm afraid not |
| 09:32.21 | k4mi1 | no, I do not use it |
| 09:32.26 | genin | hrm |
| 09:32.29 | k4mi1 | and no I am not from america |
| 09:32.34 | genin | yeah me nitehr really |
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| 09:32.50 | k4mi1 | what is your problem? |
| 09:32.53 | genin | we have one installed but by an admin who left and i just got another one to put in a new machine |
| 09:32.55 | k4mi1 | with this card |
| 09:32.58 | k4mi1 | ? |
| 09:33.12 | genin | the E1 has 4 lines coming in and i know we use tw oof them but im not sure which ones |
| 09:33.30 | genin | when i put this new card in a need to plug in a line from the other E1 card for testing.. |
| 09:33.33 | genin | so the problem is |
| 09:33.38 | genin | how can i see in asterisk |
| 09:33.43 | genin | which spans are being used |
| 09:34.04 | genin | im sure there is some type of command in * |
| 09:34.34 | freh | which * version? |
| 09:34.58 | genin | asterisk-1.4.17 |
| 09:35.17 | freh | you are using zaptel? |
| 09:35.23 | genin | yeah |
| 09:35.35 | genin | ah wit on this machine |
| 09:35.44 | genin | in etc/asterisk i have zapata.conf |
| 09:36.03 | freh | try 'pri show spans' on the console |
| 09:36.10 | genin | cool ill try |
| 09:36.44 | genin | it says they are all active, in the zapata conf they are all configured to be up |
| 09:36.57 | genin | but i believe we are only passing traffic over 2 of them |
| 09:37.44 | genin | i have a list of phone numbers associated with the lines coming in so i mean i could always unplugg a line late at night and see if i got the right one, but i am sure there is a better way |
| 09:38.26 | *** join/#asterisk phix (n=threat@123-243-44-131.tpgi.com.au) |
| 09:39.02 | freh | You might be able to see which channels are used in the dialplan |
| 09:39.17 | freh | I'm not sure how to check it in the console |
| 09:39.22 | freh | maybe someone else |
| 09:39.41 | genin | yeah the previous admin didnt just put it in the dialplan he put tons of includes in it |
| 09:39.47 | genin | ill start digging through those |
| 09:40.24 | genin | i could always run ngrep during the day and grep for the numbers being called to see if i can determine what is going on |
| 09:43.01 | *** join/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 09:51.43 | freh | Anyone knows why call parking doesn't announce the extension to me where the call is parked? |
| 09:56.41 | pif | hi, I'm setting up agents, is there a way to test if an agent is logged in or not? |
| 09:56.43 | joobie | what function do u use to park the call freh? |
| 09:57.05 | pif | I'd like to use the same key for login and logout |
| 09:58.24 | joobie | pif.. how are u logging your agents in? |
| 09:58.40 | pif | they enter a key on their polycom |
| 09:59.07 | pif | press a key |
| 10:01.40 | joobie | what functions even |
| 10:01.42 | *** join/#asterisk UQlev (n=yuriy@87.228.199.125) |
| 10:01.54 | pif | AgentLogin() |
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| 10:04.18 | *** join/#asterisk tompaw (n=tompaw@slave20.tesserakt.eu) |
| 10:04.19 | tompaw | hello |
| 10:04.36 | tompaw | are there any reasons not to use 64bit architecture with asterisk? |
| 10:04.46 | joobie | not sure man |
| 10:04.52 | tompaw | like it being less stable, causing problems with codecs or anything? |
| 10:05.01 | joobie | i know when you add them to the queue, you can use 'queue show' and it will show u agents in the queue |
| 10:05.09 | joobie | duno about just agent logins tho |
| 10:06.06 | tamiel | Hello, when doing a Dial() with Dahdi/g1, is Dial() iterating over Dahdi/g1 group looking for an available channel or is Dial() getting the next channel on Dahdi/g1 and stop here ? |
| 10:06.21 | genin | allo |
| 10:06.38 | genin | i have to unplug one of the four cables of my TE420 card |
| 10:06.46 | genin | i was doing a sip set debug and i see that |
| 10:07.03 | genin | ZAP 23-1 and ZAP 13-1 are being used |
| 10:07.19 | genin | is it safe to say that the only line we are currently using is the first one |
| 10:07.25 | genin | on span 1 |
| 10:07.40 | pif | tompaw: 64bit works fine |
| 10:07.46 | *** join/#asterisk ITguru (n=ITGuru@82.70.253.90) |
| 10:08.11 | genin | q |
| 10:11.22 | genin | with the E1 what does the "1" mean in this |
| 10:11.24 | genin | 30-1 |
| 10:11.27 | genin | span 1? |
| 10:11.45 | genin | for the TE420B |
| 10:13.21 | *** join/#asterisk florz (i=nobody@2001:1a50:503c:0:0:0:0:1) |
| 10:14.18 | *** join/#asterisk SparFux (n=raoul@85.182.23.137) |
| 10:14.38 | SparFux | Hello! What is the most elegant way to define aliases for telephone numbers, like shortcuts? |
| 10:15.41 | *** join/#asterisk oej (n=olle@cust-IP-10.data.tre.se) |
| 10:18.10 | freh | joobie: I do an attended transfer to the extension defined in features.conf |
| 10:19.11 | genin | anyone know about E1/T1 s in here |
| 10:19.13 | genin | ? |
| 10:21.18 | freh | joobie: Do I need to do anything specific in the dialplan except for including the parked calls context? |
| 10:21.50 | freh | I also already added TtKk in the Dial applications |
| 10:22.11 | tamiel | genin: http://en.wikipedia.org/wiki/E-carrier |
| 10:22.19 | genin | cool thanks |
| 10:22.40 | genin | im trying to figure out which cable i can unplug out of the 4 on my TE420 card |
| 10:22.46 | genin | i checked zapata.conf |
| 10:22.51 | genin | and looked on the cli |
| 10:23.11 | genin | everytime i make a call to the numbers we are using it shows always less than 31 |
| 10:23.16 | genin | like 31-1 |
| 10:23.19 | genin | 22-1 |
| 10:23.26 | genin | 4-1 |
| 10:23.56 | genin | and those correnspond to channel=>1-15,17-31 under [channels] |
| 10:24.13 | genin | so i assume that that is the cable in port 1 that i dont want to touch |
| 10:25.08 | SparFux | Is it a sane way to have an alias by simply defining an extension: exten => <aliasno>,1,Dial(local/<realno>) ? |
| 10:26.01 | tamiel | genin: yes this is port 1 |
| 10:26.19 | genin | okay cool |
| 10:26.51 | genin | is it possible that we only have 30 channels all on port 1 and i am getting cunfused thinking each port has 30 chans? |
| 10:27.40 | genin | i have a paper that says 4 acces T2 of 30 channels and on other side of paper i have some other phone numbers and at bottom says 2 access T2 of 30 chans |
| 10:27.48 | tamiel | each port have 30 chans on E1 |
| 10:28.04 | genin | okay cool |
| 10:28.05 | tamiel | (E1/T2) |
| 10:28.15 | tamiel | T2 == french name |
| 10:28.25 | genin | ah ok |
| 10:28.47 | genin | so if i see 22-1 |
| 10:28.54 | genin | the "1" means port one |
| 10:29.03 | *** join/#asterisk oej_ (n=olle@fw01d.snowmen.se) |
| 10:29.05 | genin | if it was on port 2 i would get 22-2? |
| 10:29.33 | tamiel | genin: 22 is alaways in port 1 |
| 10:29.40 | genin | ah right |
| 10:29.41 | tamiel | /ala/al/ |
| 10:29.41 | genin | i mean |
| 10:29.43 | genin | 37-2 |
| 10:29.49 | tamiel | yep |
| 10:29.51 | genin | if it was on port 2 for example |
| 10:30.16 | genin | cool sorry for asking such redundant questions but our admin left and i have to go unplugg things on the production at the datacenter tonight |
| 10:30.17 | genin | heh |
| 10:30.48 | joobie | freh, is the parking / unparking functions working ok, bar no extension announcement> |
| 10:30.54 | joobie | -> +? |
| 10:31.04 | *** join/#asterisk oej_ (n=olle@fw01d.snowmen.se) |
| 10:31.04 | tamiel | genin: I think -1 or -2 is not important |
| 10:31.06 | joobie | afaik, there's no option to disable the extension announcement |
| 10:31.14 | tamiel | genin: 1-30 is on port 1 |
| 10:31.17 | joobie | so i'd be thinking you havent implemented it correctly |
| 10:31.30 | tamiel | genin: this is the channel number |
| 10:31.36 | genin | when i do a sip set debug i see it always as |
| 10:31.43 | genin | ah okay |
| 10:31.54 | genin | ZAP 4-1 etc |
| 10:31.56 | tamiel | and you can safely check in zap conf to check mapping between port and channel number :) |
| 10:32.07 | genin | zaptel or zapata conf |
| 10:32.32 | freh | joobie: I can see the call getting parked on the console (also the extension weher it gets parked). When I dial that extension I get connected to the call again. The parked call also gets the moh while it's parked |
| 10:38.53 | *** join/#asterisk Ng (n=cmsj@nurukipa.tenshu.net) |
| 10:39.55 | Ng | I want to have a Queue that escalates the calls if they're not answered, would someone have a suggestion of how to do that? afaics the options are to have the caller timeout from the first Queue into another, but that could mean their queue position changes and would seem a little confusing |
| 10:40.22 | joobie | freh, http://www.voip-info.org/wiki/view/Asterisk+cmd+ParkAndAnnounce .. are you configured the same as example 4? |
| 10:40.36 | joobie | that's what i used previously and it worked ok |
| 10:40.42 | *** part/#asterisk joobie (n=joobie@203-217-64-151.dyn.iinet.net.au) |
| 10:40.46 | Ng | I looked at autopause, but I don't want to rely on the staff to unpause themselves, so the first level callers would inevitably just end up paused all the time ;) |
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| 10:40.50 | joobie | woops |
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| 10:42.23 | freh | joobie: No I'm not. I only have included the parkedcalls context in the dialplan |
| 10:43.43 | freh | shouldn't that be enough? In all the documentation I can find there's only the configuring of the features.conf file necessary |
| 10:43.56 | freh | and including of the parkedcalls context |
| 10:47.00 | SparFux | All of the sudden after an update of Debian, I get this message when trying to use DTMF tones with asterisk on a SIP ATA: chan_sip.c:11281 handle_request_info: Unable to parse INFO message from ccdd29a7-4509e131@192.168.118.96. Content |
| 10:50.18 | genin | 1 E1/T2 is 30 channels |
| 10:50.49 | genin | a TE420 card can take in 4 E1/T2 channels then |
| 10:54.18 | joobie | freh, did you include the parkedcalls context in your default context of extensionsc.onf? |
| 10:54.46 | joobie | im not too sure freh.. not overly experienced with parked calls.. just sorta worked for me |
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| 11:03.53 | joobie | hmm.. guys via the AMI, is there a way to see how many people are in a queue? there doesn't seem to be an event that shows this afaik |
| 11:04.26 | creativx | queuemembers |
| 11:04.32 | creativx | should list all queue members |
| 11:05.16 | joobie | i have to send a request to get that output right? |
| 11:05.26 | joobie | is there a way to have asterisk just show that info as a event? |
| 11:05.30 | genin | to install a TE420B card from digium do i need to install the machine in a special way locally or can i go to the datacenter and just pop the card in and set it up remotely? |
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| 11:06.48 | creativx | joobie: you send that via AMI and then read the outputted events |
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| 11:07.09 | Stese | Hey... Can anyone help me with a compling issue with Zaptel and CentOS 5 |
| 11:07.09 | Stese | ? |
| 11:08.04 | Stese | *Compiling even |
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| 11:10.33 | joobie | creativx, how to trigger queuemembers ? I tried 'Action: queuemembers' .. |
| 11:10.36 | joobie | ive logged in OK |
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| 11:12.08 | joobie | im running 1.4 btw.. dont think it supports queuemembers |
| 11:17.16 | creativx | im on 1.2 |
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| 11:20.28 | Stese | Woosh |
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| 11:24.29 | Stese | OK, I'm having a Compiling issue... using the o'reilly * book as a guide, i'm trying to compile Zaptel.... but when I ask it to do a '# make' it complains that i've not got the source installed. I've got this with yum, and the devel for it as well. I'm building it on CentOS 5.3, with the lastest version of * using wget |
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| 11:31.07 | *** join/#asterisk Dealer2mogette (n=Dealer2m@217.128.94.168) |
| 11:31.16 | Dealer2mogette | hi everybody |
| 11:34.02 | beek | hi |
| 11:34.07 | SparFux | Hi. |
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| 11:34.20 | joobie | guys anyone got any ideas on how to track the number of callers in a queue, as new callers are added? the AMI interface allows me to query the queuestatus, which reports this, but i have to issue an Action to do this.. Can't seem to get it showing the value as an Event when someone joins the queue.......... |
| 11:34.21 | freh | Stese: Have got zaptel installed? |
| 11:34.43 | Dealer2mogette | i have a strange problem with my asterisk server. Actually it's a virtualized debian by VMware. Yesterday, it was ok, i could phone. But today, my sip client can connect to the server but when i test for the "welcome message" or when i want to call an other client it doesn't works. ExpressTalk says : Error. Other side said : Not found. Do you have an idea ?? |
| 11:35.38 | jblack | Could be anything, but mucking up your dialplan or contexts is most likely |
| 11:36.21 | jblack | asterisk -r, set debug and verbose to 9, and take some time to figure out what the debug is telling you. |
| 11:36.56 | freh | Stese what asterisk version are you planning to use? Zaptel has been renamed to DAHDI so perhaps you want to use that |
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| 11:42.13 | defsdoor | anyone got a minute to help me configure a polycom soundstation ip6000 ? I can't seem to get it to register |
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| 11:43.57 | Dealer2mogette | an idea ? |
| 11:45.41 | Stese | freh > no it won't do it |
| 11:46.00 | Stese | freh > complains I don't have the sources |
| 11:47.09 | Stese | freh > I'm trying to use the latest version of *, hang on, i'll see what the ver. number is |
| 11:48.01 | Stese | freh > * 1.6.0.9 |
| 11:48.02 | Stese | I |
| 11:48.03 | *** part/#asterisk my007ms (i=master@botmaster.x86.be) |
| 11:48.25 | Stese | freh > I'm guessing I can wget DAHDI? |
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| 11:51.40 | *** join/#asterisk MrParity (n=Miranda@80.131.4.32) |
| 11:51.46 | MrParity | hi :-) |
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| 11:53.57 | MrParity | i found contradictory answers to this question by google. can i use the hfc-s with asterisk 1.6.0.9 and dahdi? |
| 11:57.48 | freh | Stese: Yes |
| 11:58.31 | freh | Stese: all you need can be found in http://downloads.digium.com/pub |
| 11:59.17 | freh | first download, compile and install dahdi-complete |
| 11:59.21 | freh | then asterisk |
| 11:59.36 | Stese | ok, thanks :) |
| 12:00.20 | freh | but do you need dahdi? Have you got a digium card? or other hardware? |
| 12:00.21 | Dealer2mogette | anybody know why i have this problem ? |
| 12:01.00 | Stese | Yes, I'll need ztdummy at the least, for timing in this machine, but eventually i'll be building another with a b410P in it |
| 12:01.29 | MrParity | freh: was the message for me? |
| 12:01.47 | freh | MrParity: no sorry :-) |
| 12:02.22 | MrParity | freh: np. :-) |
| 12:02.30 | Stese | Dealer2mogette > have you done what Jblack suggested? |
| 12:02.48 | freh | MrParity: best way to find out by yourself is installing dahdi and use the dahdi tools to see if it finds compatible hardware |
| 12:03.26 | Dealer2mogette | yes but for debug i only found "sip set debug", is it the right command ? |
| 12:03.34 | freh | MrParity: When it's installed use the dahdi_hardware command |
| 12:03.37 | MrParity | Asterisk:~/asterisk/asterisk-1.6.0.9# dahdi_hardware |
| 12:03.37 | MrParity | pci:0000:00:08.0 zaphfc- 1397:2bd0 HFC-S ISDN BRI card |
| 12:04.06 | MrParity | but i don't see the driver with lsmod or in the sources of dahdi |
| 12:04.13 | Stese | Dealer2mogette > I think so... i've not got a machine running I can check with |
| 12:04.27 | freh | then try running dahdi_genconf |
| 12:05.26 | Dealer2mogette | how can i know if clients are connected to the server ? |
| 12:05.35 | freh | this should generate some configuration files based on your hardware. It will generate /etc/dahdi/modules , /etc/dahdi/system/conf and /etc/asterisk/dahdi-channels-conf |
| 12:06.00 | freh | Dealer2mogette: sip show peers |
| 12:06.18 | Stese | Dealer2mogette > sip show peers will show registered users |
| 12:06.46 | Dealer2mogette | ok thanks so i have problem with the connection :( |
| 12:06.57 | MrParity | ; Span 1: DAHDI_DUMMY/1 "DAHDI_DUMMY/1 (source: HRtimer) 1" (MASTER) |
| 12:07.13 | freh | Stese: remember that with dahdi, ztdummy == dahdi_dummy |
| 12:08.16 | Stese | freh > Yeah, good point... the O'Reilly Book is good, i just need to keep in moi |
| 12:08.28 | Stese | *mind that it is a little out of date |
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| 12:20.33 | Stese | freh > I've downloaded dadhi, gone to the folder, run '# make clean' which is ok, |
| 12:21.08 | Stese | but it's not got a ./configure script, and 'make' still falls over, complaining about the source |
| 12:22.20 | freh | I'm not sure about centos but under debian you have to install build-essential and linux-headers-`uname -r` before compiling anything |
| 12:22.54 | Stese | Ok... i'll see if CentOS has similar packages in Yum |
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| 12:24.37 | Stese | Hmm, that doesn't tie up... build essential isn;t there... but I'm guessing thats the kernel source, and i've got the headers for my kernel |
| 12:27.04 | [TK]D-Fender | Stese: http://www.voip-info.org/wiki/view/CentOS+5+and+Asterisk+1.4.x+installation |
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| 12:32.01 | jblack | Day two of quitting. |
| 12:37.18 | jblack | whoah. They closed every signle restaurant in mexico city? |
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| 12:47.46 | Dovid | good morning ev1 |
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| 12:48.02 | Dovid | to enable HD on my asterisk box all i need to do is allow=g722 ? |
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| 12:48.44 | jblack | No. If you want the highest definition, use ulaw. |
| 12:49.20 | [TK]D-Fender | Dovid: Yes |
| 12:49.21 | tamiel | How many times take dahdi to really clear/release a channel after hangup ? |
| 12:49.34 | tamiel | Because I have some congestion issues |
| 12:50.13 | tamiel | but less than 50% of my channel capacity is busy |
| 12:52.07 | tamiel | And I receive : dial_exec_full: Unable to create channel of type 'Dahdi' (cause 34 - Circuit/channel congestion) |
| 12:53.03 | Stese | [TK]D-Fender > Thanks, |
| 12:53.28 | [TK]D-Fender | tamiel: Sometimes PRI's report that code back when the number you are calling is busy. |
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| 12:54.29 | xheliox | Is there any good way to produce a tone to the end user(s) when using the Page() application? |
| 12:55.55 | tamiel | [TK]D-Fender: ok. So the remote side is already busy (the remote side is an EuroISDN ivr) |
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| 12:59.10 | tamiel | [TK]D-Fender: thanks for the answer :) |
| 13:00.38 | *** part/#asterisk ITguru (n=ITGuru@82.70.253.90) |
| 13:00.46 | [TK]D-Fender | xheliox: dial a Local channel with A() or M() and have that call your Page |
| 13:01.09 | xheliox | Hmm. |
| 13:01.44 | xheliox | Good call. Thanks. |
| 13:04.10 | Stese | does anyone know of a guide/doc/HOWTO for setting up the latest version of * (1.6.0.9) |
| 13:04.30 | [TK]D-Fender | Stese: Same as that guide, jsut with 1.6 packages. |
| 13:04.38 | [TK]D-Fender | Stese: The requirements aren't any different |
| 13:04.40 | MrParity | stese: http://www.debian-resources.org/node/129 |
| 13:04.43 | Superbartt | ./configure; make; make install ? :p |
| 13:04.57 | [TK]D-Fender | Stese: This is just s completely boring compile job. It isn't magic |
| 13:07.16 | *** join/#asterisk anonymouz666 (n=anonymou@189.24.118.128) |
| 13:07.18 | Stese | Well, i'm new to doing it all, so I want to be doing it right :) |
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| 13:07.57 | Superbartt | http://www.digium.com/en/training/courses/?tab=fast-obj ? :p |
| 13:08.22 | Stese | Superbartt > I would love to... I can't afford it, and nor can my company |
| 13:09.52 | [TK]D-Fender | Stese: You've been here quite a while already, its 3 stupid tartballs, and a million guides for the prerequisites. |
| 13:10.06 | [TK]D-Fender | Stese: the instructions are in the bloody tarballs anyway |
| 13:10.28 | Katty | GOOD MORNING ALL YOU BEAUTIFUL PEOPLE! |
| 13:11.18 | Superbartt | I think i'm goanna kill junghanns ;X |
| 13:11.19 | [TK]D-Fender | Katty: Mew. |
| 13:11.25 | Katty | hugs [TK]D-Fender |
| 13:11.34 | Superbartt | their support telephone number says as much as "please email the support" in crappy english |
| 13:11.35 | Stese | [TK]D-Fender > I've interited a prebuilt system that i've been learning in bits and bobs when something has had to be changed, or fixed |
| 13:11.40 | Superbartt | en the support email doesn't respond usefully :X |
| 13:12.13 | anonymouz666 | Katty: we need to stop hug pigs from now on :D |
| 13:12.49 | Stese | Superbartt > Sounds like Digium.... we've got an ABE product that keeps hitting the max calls limit, for no reason... been waiting 4 hours for a call back |
| 13:12.59 | anonymouz666 | (this is not for our friend Fender) |
| 13:13.10 | anonymouz666 | it's just a joke :D |
| 13:13.20 | Katty | hugs anonymouz666 |
| 13:13.24 | anonymouz666 | :P |
| 13:13.33 | [TK]D-Fender | anonymouz666: That's why men will never cath mad-cow disease... because we're all pigs ;) |
| 13:13.38 | [TK]D-Fender | catch* |
| 13:13.47 | Stese | what about Swine Flu |
| 13:13.50 | anonymouz666 | heh :D |
| 13:13.52 | Superbartt | Stese i'm having random channel congestions :x |
| 13:14.08 | coppice | swine can't fly |
| 13:14.22 | Stese | but people with it can |
| 13:14.47 | [TK]D-Fender | coppice: I've got a trebuchet that begs to differ :D |
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| 13:14.54 | Katty | someone give me the scoop on this swine flu |
| 13:15.12 | Superbartt | omg... I remembe why i don't watch tv... there is a commercial for a "fart ringtone" you can sms fart on to get it :| |
| 13:15.29 | mocker | buys. |
| 13:15.33 | [TK]D-Fender | Katty: its something you have less statistical odds of dying from than a raging hippo attack |
| 13:15.36 | Stese | it's basically the new possible Pandemic.... remember Bird Flu??? well it's kinda like that |
| 13:15.52 | Katty | sounds exciting. |
| 13:15.57 | Katty | symptoms? |
| 13:17.07 | Superbartt | death |
| 13:17.21 | Stese | it's flu... i guess they are flu like... (someone got ill on a flight from Mexico to Manchester airport today.... flight almost got locked down!) |
| 13:18.33 | *** join/#asterisk Emrah86 (n=root@sip.slimvoip.nl) |
| 13:18.35 | Katty | scowls. |
| 13:18.36 | Katty | googles. |
| 13:18.50 | Emrah86 | hi all |
| 13:19.46 | Katty | oh nice. |
| 13:19.51 | Katty | so they stick you on an anti-biotic. |
| 13:19.59 | Katty | and 12 cases have been reported between 2005 and 2009 |
| 13:20.40 | genin | yeah so whut up with this |
| 13:20.43 | genin | pig flu |
| 13:20.45 | Stese | apparently there have been 150 odd in Mexico, and we've had 1 confirmed case in Scotland.... |
| 13:20.47 | genin | no cure? |
| 13:20.58 | *** join/#asterisk theHub (n=theHub@69.177.93.21) |
| 13:21.07 | Katty | antibotics |
| 13:21.24 | Katty | amantadine, rimantadine, oseltamivir, and zanamivir |
| 13:21.41 | Katty | take it for 2 weeks, you're good |
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| 13:23.10 | Superbartt | remains scary that shit |
| 13:23.31 | Superbartt | one day alot of people will die, cause they can't cure all' |
| 13:23.57 | pif | hi, should I use AddQueueMember or AgentCallbackLogin ? |
| 13:24.53 | genin | take it or all that |
| 13:24.55 | genin | at once |
| 13:24.55 | genin | heh |
| 13:27.42 | *** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
| 13:29.34 | *** join/#asterisk axisys (n=axisys@155.70.141.45) |
| 13:30.17 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
| 13:30.33 | *** join/#asterisk fbnts (n=root@213.106.147.11) |
| 13:31.11 | fbnts | Hi, I'm trying to compile cbmysql which is part of web-meetme but am getting: app_cbmysql.c:584: warning: initialization from incompatible pointer type |
| 13:31.15 | fbnts | any ideas? |
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| 13:41.36 | *** mode/#asterisk [+o jtodd] by ChanServ |
| 13:42.46 | *** join/#asterisk merlin8282 (n=merlin82@88.122.143.64) |
| 13:43.05 | merlin8282 | Hi. Is there any way to schedule jobs within asterisk ? |
| 13:43.12 | mocker | crontab |
| 13:43.14 | mocker | :) |
| 13:43.32 | *** join/#asterisk SkywaIker (n=pirch@58.147.17.166) |
| 13:43.49 | merlin8282 | yes, that's the point: _within_ asterisk. Because if I use a cronjob, I have to keep the data up to date in my extensions.conf too |
| 13:44.11 | [TK]D-Fender | merlin8282: "data up to date"? What "data"? |
| 13:44.14 | *** part/#asterisk axisys (n=axisys@155.70.141.45) |
| 13:44.33 | *** join/#asterisk putnopvut (n=putnopvu@asterisk/master-of-queues/mmichelson) |
| 13:44.33 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 13:45.04 | [TK]D-Fender | merlin8282: And what "jobs"? I don't remember * being some sort of batch processor |
| 13:47.03 | merlin8282 | I have a support GSM phone, that has to be called from 18:00 to 8:00. This is OK. But if within this time range the support GSM has to be another one (a more experienced admin) I send an SMS to both supporters. |
| 13:47.15 | merlin8282 | And |
| 13:47.26 | *** join/#asterisk wonderworld (n=ww@ip-62-143-16-28.unitymediagroup.de) |
| 13:47.55 | [TK]D-Fender | merlin8282: that is not a "scheduled event", that looks like a "time-based decision" |
| 13:48.15 | merlin8282 | I already did some things with IFTIME() |
| 13:48.32 | merlin8282 | Let me explain it more |
| 13:48.35 | [TK]D-Fender | merlin8282: That would be one of 2 primary comamnds that you should use. |
| 13:49.02 | wonderworld | hi, we have an asterisk 1.6.0.3 running. the machine freezes once a week or so randomly. i was wondering if the freezes could be asterisk related. are there known issues in the 1.6 versions that could cause such problems or should i rather begin searching somewhere else? |
| 13:49.27 | wonderworld | (i could upgrade to the most recent * if neccesary) |
| 13:49.42 | merlin8282 | the problem is that I want to send at a fixed time (when the support GSM becomes active, so he can receive support calls) an SMS to this GSM. |
| 13:49.55 | [TK]D-Fender | wonderworld: Upgrade |
| 13:49.56 | merlin8282 | Even if there is no call. |
| 13:50.27 | [TK]D-Fender | merlin8282: THAT scheduled event has nothing to do with * |
| 13:50.47 | [TK]D-Fender | merlin8282: That you would use something like crontab for or some other script you've create |
| 13:51.00 | fbnts | Is Web-meetme compatible with Asterisk 1.6? |
| 13:51.22 | wonderworld | [TK]D-Fender: i know that the latest version is generaly a good thing to have, but i don't want to do it if the current running version has got nothing to do with the freezes...... |
| 13:51.24 | [TK]D-Fender | fbnts: What do they say? Doesn't sound like a part of * to me... |
| 13:51.30 | merlin8282 | I know. But the problem is that if I do a cronjob for sending this SMS, and if I have to modify the Support-beginning time I have to do it in the * AND in the cronjob. |
| 13:51.58 | [TK]D-Fender | wonderworld: Well your description provides anything but.... sso "upgrade" is all you're going to get so far... |
| 13:52.34 | fbnts | When I try to compile the module I get: app_cbmysql.c:594:38: error: macro "ast_config_load" requires 2 arguments, but only 1 given |
| 13:52.36 | *** join/#asterisk fskrotzki_ (n=fskrotzk@cpe-74-74-245-250.rochester.res.rr.com) |
| 13:52.43 | sulex | Hello. Is there anyway within AGI(not AMI or CLI), to know if a queue has at least one "idle" agent in? What I want to avoid is to load socket libraries or asterisk::manager to use AMI within my agi script, thanks |
| 13:52.46 | [TK]D-Fender | merlin8282: Yes, 2 places, because * was not meant to be used as an alarm clock, a fridge-magnet, a can-opener, or a bicycle pump. |
| 13:52.57 | merlin8282 | hehe... ok. |
| 13:53.03 | wonderworld | [TK]D-Fender: yes, my description sucks, but i checked all the logs after rebooting the machine. nothing there.... i just haven't found out why it crashes and was wondering if * might be the cause |
| 13:53.25 | [TK]D-Fender | wonderworld: You have nothing to offer to help debug. Go upgrade. |
| 13:54.38 | [TK]D-Fender | wonderworld: Don't tell your auto mechanic "my car is bad, fix it" and walk away. he'll waste weeks tearing up the engine and never find that the problem you meant was that your favourite AM radio station gets too much static <- |
| 13:56.15 | jplank | sulex: whats wrong with passing a CLI command? |
| 13:57.02 | *** join/#asterisk awkfu (n=awkfu@66.162.90.56) |
| 13:57.32 | [TK]D-Fender | sulex: Depends what kind of members your queue has, and it'll be a bitchload of dialplan to do it. |
| 13:57.54 | [TK]D-Fender | sulex: It is a hell of a lot smarter to do it via CLI/AMI |
| 13:58.23 | pif | is it possible to test if an agent is logged in or not? |
| 13:58.26 | jplank | I could understand not wanting to hard code in AMI (somewhat) but whats wrong with CLI? |
| 13:58.36 | jplank | pif - queue show |
| 13:59.02 | pif | from the dialplan? |
| 13:59.35 | [TK]D-Fender | pif: Depends on your definition of "agent" |
| 14:00.10 | pif | Agent/XXXX |
| 14:00.47 | [TK]D-Fender | pif: "core show function AGENT" |
| 14:00.54 | pif | I'd trying to configure a key to let agents login/out |
| 14:03.02 | pif | thx |
| 14:04.35 | pif | and is there a way to display one's status on a polycom phone (login/out) ? |
| 14:06.49 | [TK]D-Fender | pif: Presence / MicroBrowser |
| 14:06.57 | *** join/#asterisk Circlefusion (n=brian@74-132-91-42.dhcp.insightbb.com) |
| 14:07.03 | pif | ok |
| 14:07.16 | creativx | hmm.. when the samba share goes down my bash scripts goes haywire |
| 14:07.58 | *** join/#asterisk ricko73 (n=dhartman@wilug/newlug/ricko73) |
| 14:08.21 | sulex | jpeeler, [TK]D-Fender: sorry, i was afk. basically I have a callcenter where agents can be reached with an ID before I queue() the user. this means I have to check if the agent is online and logged before I transfer the call to the queue(where this check is done automatically). Since a lot of things are done for paying the service in an AGI and so on, i'd like to avoid to use addtional modules in the agi itself... that's why I'm looking |
| 14:08.21 | sulex | <PROTECTED> |
| 14:08.43 | sulex | but as you stated above... maybe it's the only reasonable solution |
| 14:09.10 | [TK]D-Fender | sulex: "paying the service". Sorry I don't follow you. |
| 14:09.37 | *** part/#asterisk merlin8282 (n=merlin82@88.122.143.64) |
| 14:10.20 | [TK]D-Fender | sulex: And If your AGI is on the same server then you may not need AMI, just parse out a text queue dump. |
| 14:11.09 | pif | [TK]D-Fender: does that look good to your ? exten => 4001,n,GoSubIf($[AGENT(${CALLERID(num)}) = LOGGEDIN]?in:out) |
| 14:11.23 | pif | er, to you |
| 14:12.25 | jplank | sulex: I still don't know why doing it CLI style isn't going to work, you could just pass queue show and parse it in your script. |
| 14:13.34 | deeperror | [TK]D-Fender, from yesterday the 3way where it doesn't show the instance i have this for you www.pastebin.ca/1406119 it shows a call, flash for 3way, then flash again to bring back into call, then hangup |
| 14:13.46 | jplank | or just parse the queue log like fender said |
| 14:13.56 | *** part/#asterisk Superbartt (n=bart@ipd50a21c9.speed.planet.nl) |
| 14:14.01 | *** join/#asterisk Superbartt (n=bart@ipd50a21c9.speed.planet.nl) |
| 14:14.30 | deeperror | [TK]D-Fender, about line 77 |
| 14:14.43 | sulex | [TK]D-Fender: the client offers a consultant service and has 3 guys as agents giving the service by phone. Users pay directly online by credit card and get a PIN code that will be authenticated by the AGI. Some user may prefer an agent among the others and wants to talk directly to him, each agent is assigned with an ID the user knows. So when a call arrives the user is asked to enter the ID of the agent if he wants to talk to him direc |
| 14:14.43 | sulex | tly or go to the normal queue. Now, if I have no agent logged I want to hangup... but the only solution I'm experienced with is using socket for AMI and trap a "Action: Queues" result |
| 14:14.48 | sulex | sorry for the flood guys |
| 14:15.01 | fbnts | Otherwise, can anyone recommend which confrencing app to use with Asterisk 1.6? |
| 14:15.18 | *** join/#asterisk KermitTheFragger (n=KermitTh@217.149.197.118) |
| 14:15.25 | sulex | jpeeler: sorry but I never did it, can I send something to CLI from within an AGI... and do I avoid loading further libriaries? |
| 14:16.28 | [TK]D-Fender | sulex: Or just parse a CLI call. Same thing. This isn't Raw-Cat Science. |
| 14:16.33 | jplank | just pass the command to the console, something like asterisk -rx queue show |
| 14:16.53 | [TK]D-Fender | fbnts: Whats wrong with the one * comes with? |
| 14:17.49 | [TK]D-Fender | pif: I am not a pre-trial debugger. Go try it and see. |
| 14:18.03 | coppice | Raw-Cat is some kind of strange sashimi? |
| 14:18.20 | *** join/#asterisk jmacz (n=jmacz@190.144.75.22) |
| 14:18.28 | sulex | jpeeler++, [TK]D-Fender++ |
| 14:18.32 | sulex | so easy :D |
| 14:18.36 | sulex | thanks |
| 14:19.56 | fbnts | D-Fender: Webmeetme has a nice interface to configure dynamic conferences and its options |
| 14:20.50 | [TK]D-Fender | fbnts: That doesn't sound like a dialplan app to me, that jsut sounds like a FRONT-END |
| 14:21.04 | [TK]D-Fender | coppice: Pass the soy :) |
| 14:21.06 | *** join/#asterisk freh (n=freh@198.0-66-87.adsl-static.isp.belgacom.be) |
| 14:21.44 | fbnts | yep its a frontend, it stores the data in mysql and the cbmysql app handles the confrence |
| 14:21.49 | [TK]D-Fender | deeperror: Curious... that looks pretty whacked. What does "core show channels concise' say at the start of the 3-way, and one bonded? |
| 14:22.03 | [TK]D-Fender | fbnts: then the answer is "Meetme" |
| 14:22.13 | fbnts | the php frontend is working fine, but i need the app_cbmysql to interface with the DB |
| 14:22.41 | fbnts | does Meetme all the easy creation of conference rooms? |
| 14:23.07 | [TK]D-Fender | fbnts: HUH? |
| 14:23.21 | coppice | [TK]D-Fender: OK. have some 豉油 |
| 14:23.29 | [TK]D-Fender | fbnts: Go read its instructions. |
| 14:23.44 | [TK]D-Fender | fbnts: And "easy" depends on a certain point of view. |
| 14:24.22 | fbnts | well we have about 30 staff they would use it - the current webpage is easy for them to setup a conference, set option like anounce & record. |
| 14:25.15 | *** join/#asterisk mohsen-ece (n=ahmed@41.196.81.102) |
| 14:25.16 | *** part/#asterisk mohsen-ece (n=ahmed@41.196.81.102) |
| 14:25.16 | [TK]D-Fender | coppice: 谢谢 |
| 14:25.16 | *** join/#asterisk De_Mon (i=de_mon@fl-67-77-166-5.dyn.embarqhsd.net) |
| 14:25.44 | deeperror | [TK]D-Fender, i now have a lot of calls going on do you want the full output or should i cut everything but the channel i'm using? |
| 14:26.22 | [TK]D-Fender | "shi-shi"... thanks. the word so nice the Chinese say it twice. Kinda like "candy" in French "bonbon", which pretty much translates as "good good" |
| 14:26.25 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 14:26.41 | [TK]D-Fender | deeperror: grep "240" |
| 14:26.43 | [TK]D-Fender | 24- |
| 14:27.00 | [TK]D-Fender | deeperror: or "/24" |
| 14:27.42 | coppice | [TK]D-Fender: doubling characters is common in chinese. like 天 means day, and 天天 means daily |
| 14:28.14 | pif | I'd like my polycom phone to check several voice mailboxes, is that possible? |
| 14:28.21 | [TK]D-Fender | coppice: How muchof your life have you spent in HK? |
| 14:28.35 | *** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim) |
| 14:28.44 | [TK]D-Fender | pif: Works in reverse. * notifies phones about X boxes |
| 14:28.51 | coppice | 18 years |
| 14:29.36 | [TK]D-Fender | coppice: Travelled much aside from that? |
| 14:29.48 | coppice | sadly, yes |
| 14:29.59 | [TK]D-Fender | coppice: No place like 127.0.0.1 I guess. |
| 14:30.33 | [TK]D-Fender | coppice: Originally from England, weren't you? |
| 14:30.48 | coppice | yeah |
| 14:31.18 | deeperror | [TK]D-Fender, www.pastebin.ca/1406139 |
| 14:31.34 | [TK]D-Fender | coppice: Well that's already a substantial culture jump. I'm somewhat jealous |
| 14:32.11 | [TK]D-Fender | deeperror: Yeah, that seriously doesn't look right. I'd take it up on the tracker.. |
| 14:33.28 | deeperror | [TK]D-Fender, ok thanks |
| 14:34.08 | *** join/#asterisk tobias (n=tobias@user-0ce2hp1.cable.mindspring.com) |
| 14:40.26 | *** join/#asterisk n3hxs (n=HAMming@151.196.93.200) |
| 14:45.03 | *** join/#asterisk eppigy (n=Dave@216-139-245-58.aus.us.siteprotect.com) |
| 14:45.06 | eppigy | hello |
| 14:45.08 | eppigy | i am dave |
| 14:45.45 | *** join/#asterisk ddickenson (n=ddickens@67-198-0-5.static.grandenetworks.net) |
| 14:46.02 | xheliox | Affirmative, Dave, I read you. |
| 14:46.29 | [TK]D-Fender | eppigy: .... what are you doing Dave? </hal> |
| 14:46.51 | xheliox | [TK]: Don't steal my gimmick! :) |
| 14:48.13 | ddickenson | Any ideas why when trying to compile dahdi I keep getting "you do not appear to have the sources for the 2.6.18-128.1.6.el5xen kernel installed? I have yum installed "kernel" "kernel-devel" "kernel-headers" |
| 14:49.09 | xheliox | ddickenson: Are you using you're running the kernel that you installed the -devel for? |
| 14:49.39 | xheliox | yum install kernel-devel-`uname -r` |
| 14:49.41 | *** part/#asterisk gego (n=rick@213.39.238.85) |
| 14:49.53 | ddickenson | I believe so, although i didn't give it a kernel version it picked it for me. I am running whatever comes standard on centos 5.3 |
| 14:49.55 | ddickenson | ok |
| 14:50.52 | ddickenson | no pkg avail |
| 14:51.27 | ddickenson | I tried using the ver number give when I ran uname -r and still no go |
| 14:53.28 | *** join/#asterisk tamiel (n=tamiel@213.30.183.226) |
| 14:53.36 | freh | ddickenson: you are running xen? install the sources for the kernel |
| 14:53.43 | ddickenson | xheliox: the difference seems to be the xen at the end in the version I have installed and the version it's looking for. What's the difference in the two? |
| 14:54.15 | freh | what's the output of uname -r? |
| 14:54.17 | ddickenson | freh: I guess i am, although I'm not familiar with it or what it does. |
| 14:54.41 | ddickenson | 2.6.18-128.1.6.el5xen |
| 14:56.09 | *** join/#asterisk Dvyjones (i=dvyjones@unaffiliated/dvyjones) |
| 14:57.53 | freh | try yum install kernel-xen-devel.i686 |
| 14:59.58 | ddickenson | yeah that made it much happier, thanks |
| 15:00.10 | freh | np |
| 15:00.14 | ddickenson | so what's the xen thing and do I need it? |
| 15:00.28 | ddickenson | Everything I see mentions sometihng about virtualization |
| 15:00.35 | freh | xen is a hypervisor for running virtual machines |
| 15:00.50 | ddickenson | ahhh |
| 15:01.07 | freh | during centos install you probably checked the "virtualization" checkbox |
| 15:01.44 | Dvyjones | Do you think one can run an Asterisk server on a linode 360 that has other uses too? |
| 15:02.55 | ddickenson | Yeah I did... thought I might play with it at some point |
| 15:03.14 | freh | never hurts :-) |
| 15:03.49 | Dvyjones | Or, what are the system requirements of a really small Asterisk server (I will probably be the only one to use it) |
| 15:04.32 | [TK]D-Fender | Dvyjones: Depends how you'ew going to use it |
| 15:04.32 | freh | in xen the virtual machines are running the same kernel as the host. |
| 15:04.38 | [TK]D-Fender | you're* |
| 15:04.47 | ddickenson | Dvyjones: I've run asterisk on a vmware machine that was running on a REALLY crappy laptop, so it doesn't take much for a small install |
| 15:04.49 | rob0 | 1. It runs a Unix or Linux OS; 2. It runs asterisk; 3. It's really small. |
| 15:05.03 | ddickenson | ah, so that's why it needs different kernel src |
| 15:05.22 | *** join/#asterisk moy (n=moy@74.12.124.89) |
| 15:05.36 | freh | yup |
| 15:05.44 | ddickenson | cool |
| 15:05.54 | Dvyjones | Ok, I'll try to set it up then :) |
| 15:06.39 | *** join/#asterisk jameswf (n=james@unaffiliated/jameswf-home) |
| 15:07.44 | Dvyjones | Quoth the wiki: "Remember that Asterisk is very proccessor-intensive, so choose very carefully which packages you are installing." |
| 15:07.47 | Dvyjones | :O |
| 15:08.24 | freh | If you are the only one to use it, there won't be a problem |
| 15:09.23 | ddickenson | freh: So now I'm compiling asterisk (or actually just running the ./configure script) and getting C++ preprocessor "/lib/cpp" fails sanity check |
| 15:09.56 | ddickenson | I don't remember having so much stuff missing or broken on centos 5.2 install... Have I done something terribly wrong? |
| 15:12.43 | freh | are your development tools installed? |
| 15:12.53 | freh | yum groupinstall "Development tools" |
| 15:13.51 | [TK]D-Fender | ddickenson: http://www.voip-info.org/wiki/view/CentOS+5+and+Asterisk+1.4.x+installation |
| 15:14.05 | ddickenson | I was pretty sure that was another "checkbox" that I checked during install knowing that I would need it for this but who knows. maybe I slept through that part of the install |
| 15:15.52 | ddickenson | freh: yeah that's what I did (or rather didn't do). Thanks again |
| 15:17.39 | *** join/#asterisk marv[work] (n=timr@router.asteriasgi.com) |
| 15:18.47 | freh | np |
| 15:18.50 | freh | gotta go now |
| 15:19.50 | ddickenson | later |
| 15:20.47 | kuku1 | Does 1.4.24.1 have many problems ? |
| 15:21.08 | *** join/#asterisk KavanS (n=KavanS@static-71-117-242-28.ptldor.dsl-w.verizon.net) |
| 15:21.35 | Qwell | kuku1: bugs.digium.com - see for yourself |
| 15:23.11 | kuku1 | since I ugpraded to a new kernel, 2.6.9-78.0.17, I've been having issues with asterisk - can I can't find where the issue is. I tried 1.2, tried 1.4 - same quality issues. |
| 15:24.25 | jplank | did you need to upgrade your kernel? |
| 15:24.35 | kuku1 | yes |
| 15:24.38 | eppigy | WIN 20 |
| 15:24.49 | kuku1 | Memory problems with exim |
| 15:25.13 | [TK]D-Fender | kuku1: And I don't see any practical description of your "problems" |
| 15:25.30 | [TK]D-Fender | kuku1: and exim is a mail app last I checked... nothing to do with * |
| 15:26.10 | dni | Hello Room,. I am trying to integrate a CCM with my asterisk server and this is the particular error message coming back from the CCM,.. Got SIP response 400 "Bad Request - 'Malformed/Missing URL' ... I tried searchign google for ana nswer but no luck |
| 15:26.11 | kuku1 | [TK]D-Fender: jplank asked why I needed to update the kernel. |
| 15:27.15 | [TK]D-Fender | dni: Who cares about the fact it didn't like the request when we don't SEE THE REQUEST. Don't jsut say "It says its bad! HELP!!". Show us what that is in RESPONSE to. |
| 15:27.33 | [TK]D-Fender | dni: pastebin is yrou friend |
| 15:27.36 | kuku1 | Sound problems occur ( cutting in and out, sound breaking, sound repeating ) since I upgraded the kernel. Was running 1.2.16 before, no issues for a few years, tried 1.4.24.1 and same thing. |
| 15:27.36 | [TK]D-Fender | ~pb |
| 15:27.36 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste , or , http://bin.cakephp.org/ |
| 15:27.42 | dni | my apologies |
| 15:28.12 | dni | you know im familiar with the PB! :) |
| 15:28.48 | [TK]D-Fender | kuku1: Still no pertinent detail on what the call comes over, etc. |
| 15:29.38 | kuku1 | This problem occurs with sip to sip |
| 15:29.47 | kuku1 | it doesn't occur iax to sip |
| 15:30.01 | kuku1 | ( or at least I wasnt able to reproduce ) |
| 15:30.06 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 15:30.16 | kuku1 | tried disablep2p =1, didn't help |
| 15:30.25 | kuku1 | tak |
| 15:30.56 | kuku1 | I don't know what else to check, especially trhat I tried 1.2 and 1.4 and had similair sound issues. |
| 15:31.27 | kuku1 | Could it be timing ? ( I have no zap cards, just using sip and iax ) |
| 15:31.46 | Talkradio | maybe it's your firewall / router |
| 15:31.50 | [TK]D-Fender | kuku1: bandwitdh, jitter, or packet-loss |
| 15:34.08 | *** join/#asterisk nny_1 (n=scott@64.203.244.146) |
| 15:34.08 | kuku1 | [TK]D-Fender: I don't know, everything was working fine for over a year - until I updated the kernel |
| 15:34.08 | *** join/#asterisk rue_mohr (n=rue@24.207.122.10) |
| 15:34.19 | rue_mohr | why did you go and do that? |
| 15:34.26 | rue_mohr | you prolly need to recompile it all |
| 15:34.35 | rue_mohr | to match the kernel |
| 15:34.39 | kuku1 | like stated above, had problems with memory on exim |
| 15:35.08 | [TK]D-Fender | kuku1: What re you running now? |
| 15:35.27 | rue_mohr | ok, the hwec is muting the silence, anyhow know how I turn that off? |
| 15:35.35 | kuku1 | 1.4.24.1 |
| 15:35.41 | [TK]D-Fender | kuOS <- |
| 15:35.52 | kuku1 | Centos |
| 15:36.08 | pif | when pressing on the polycom "message" button how can I have a list of mailboxes to consult? |
| 15:36.11 | kuku1 | <PROTECTED> |
| 15:36.49 | rue_mohr | I know this because if I make a call zap->zap there is no muting, but if its via a sip phone, it does happen, its not the sip phones because it was happening between the zap channels when echocancelwhenbridged was turned on |
| 15:37.07 | nny_1 | quick Q. using asterisk-stat cdr analyzer right now with my boxes, but have a job coming up that is gonna want a nice way to create human readable reports for long distance call time on each person connected to the system. I could hack asterisk-stat, but prefer not to reinvent the wheel. |
| 15:37.10 | [TK]D-Fender | kuku1: Old... |
| 15:37.16 | Qwell | [TK]D-Fender: CentOS.. |
| 15:37.36 | nny_1 | wondering if anyone has any good opinions of what's out there |
| 15:37.53 | [TK]D-Fender | nny_1: MS Excel |
| 15:38.18 | nny_1 | [TK]D-Fender: hahaha |
| 15:38.20 | nny_1 | [TK]D-Fender: aye |
| 15:39.03 | nny_1 | the problem is most of the robust ones act as their own complete system, with a web interface for the dialplan etc. Just want a nice way to parse the cdr data in the mysql db |
| 15:39.04 | *** join/#asterisk GeminiDomino (n=C@fl-71-0-246-138.sta.embarqhsd.net) |
| 15:39.26 | jameswf | I should write up a swine flue alert hotline... when you call it plays tt-monkeys |
| 15:39.40 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.static.bezeqint.net) |
| 15:39.41 | rue_mohr | why not pigs? |
| 15:39.45 | kuku1 | Any ideas ? |
| 15:39.50 | nny_1 | looks like we may be hiring a good php developer to hack asterisk-stat, can always re release that updated eh? |
| 15:39.53 | jameswf | no default pig sound in asterisk |
| 15:40.17 | rue_mohr | baff, it'd only take a min of searching the internet |
| 15:40.25 | *** join/#asterisk saftsack (n=oliver@p4FC76D72.dip.t-dialin.net) |
| 15:40.28 | nny_1 | coughing pigs? |
| 15:40.32 | GeminiDomino | Using Hardy, Dahdi 2.1.0.4, Ast 1.4.24.1. Getting "detected a problem with your Dahdi config" when I try to start Ast, but dahdi seems to be workingm juding by dahdi_cfg and dahdi_tool. Any way to troubleshoot exactly WHAT problem asterisk is finding? |
| 15:41.00 | saftsack | hi, if i do a blind transfer and the side to which i transfer is busy, the call is hanged up. is it possible to get the call back? |
| 15:42.00 | rue_mohr | hmm whats DCS |
| 15:43.35 | *** join/#asterisk Titanous (n=titanous@unaffiliated/titanous) |
| 15:44.01 | kuku1 | [TK]D-Fender: any ideas on pinpointing the problem ? any default testing techniques ? |
| 15:44.29 | Titanous | Is there any way to set the CallerID for one channel in a multiple simultaneous channel dial? |
| 15:44.34 | *** join/#asterisk tryfan (n=tryfan@216.58.2.4) |
| 15:44.49 | [TK]D-Fender | look at system load, and we've gone all this time without says whats on each bloody end of the call and the connections betweent hem. |
| 15:44.57 | [TK]D-Fender | kuku1: ^ |
| 15:45.18 | tryfan | Hopefully a quick question, in order to use a single T1 card as a timing source, I have to be able to see it in 'zap show status' right? |
| 15:45.22 | saftsack | someone any idea for my transfer issue? |
| 15:45.28 | [TK]D-Fender | Titanous: have that 1 channel be a local channel instead that sets the calleriid prior |
| 15:45.46 | Titanous | [TK]D-Fender: thanks, just thought of that after I asked |
| 15:46.16 | Katty | devours burrito |
| 15:46.26 | tryfan | I'm just having a driver issue with a stupid setup I was left by another guy |
| 15:47.07 | Katty | are you SURE it was a GUY |
| 15:47.23 | [TK]D-Fender | :O |
| 15:47.30 | GeminiDomino | And asterisk starts without Dahdi. |
| 15:47.38 | pif | can I test if a voicemail box has messages from the dialplan? |
| 15:48.25 | eppigy | i am so hungry |
| 15:48.27 | kuku1 | wonders what fender wants him to test .... |
| 15:48.32 | [TK]D-Fender | GeminiDomino: dahdi_cfg and dahdi_tool <- neither test *'s channel definitions against them |
| 15:48.41 | [TK]D-Fender | pif: |
| 15:48.47 | [TK]D-Fender | pif: "help voicemail" |
| 15:48.49 | _Steve_ | anyone heard of problems with calls not hanging up properly? call remote, remote answers, then hangs up, local thinks it's still connected? |
| 15:49.07 | rue_mohr | and the dahdi monitor code is broken, the http://www.voip-info.org/wiki/view/Asterisk+zapata+gain+adjustment the arrows dont come out right anymore, if I can fix that, shall I send in a patch? |
| 15:49.17 | [TK]D-Fender | _Steve_: who is this call from? |
| 15:49.23 | GeminiDomino | [TK]D-Fender: Ok. So what should I be running to test *'s chan defs? |
| 15:49.26 | _Steve_ | from me |
| 15:49.32 | pif | help voicemail |
| 15:49.46 | [TK]D-Fender | GeminiDomino: * itself. of course if you want a sanity check yous hould pastebin the whole mess for us to look at. |
| 15:49.48 | [TK]D-Fender | ~pb |
| 15:49.48 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste , or , http://bin.cakephp.org/ |
| 15:49.50 | [TK]D-Fender | ^^^^^^^^^^^^^^^^ |
| 15:50.02 | pif | ~voicemail |
| 15:50.03 | GeminiDomino | I can't get asterisk to even start, though. That's the problem. = |
| 15:50.07 | *** join/#asterisk Victor_Yure (n=victor@unaffiliated/victoryure/x-837844) |
| 15:50.13 | [TK]D-Fender | pif: * CLI silly |
| 15:50.24 | GeminiDomino | In the past whenever I had a problem with the drivers, it was always on the zaptel side so this is kind of new territory |
| 15:50.28 | [TK]D-Fender | GeminiDomino: provide everything else. |
| 15:50.43 | rue_mohr | the right half of that arrows are missing, thats what it is |
| 15:50.47 | *** join/#asterisk tamiel (n=tamiel@213.30.183.226) |
| 15:50.51 | rue_mohr | how the heck did someone break that |
| 15:52.30 | _Steve_ | [TK]D-Fender: calls from me to remote... |
| 15:52.45 | [TK]D-Fender | ...... |
| 15:53.00 | _Steve_ | [TK]D-Fender: then, remote hangs up, my end still thinks it's connected... |
| 15:53.17 | *** join/#asterisk _Raptor_ (i=raptorbl@andariel.informatik.uni-erlangen.de) [NETSPLIT VICTIM] |
| 15:53.28 | _Steve_ | talked to sip trunk vendor, they confirm they are sending the BYE |
| 15:53.43 | [TK]D-Fender | _Steve_: go look at the SIP debug for the call yourself. |
| 15:53.58 | _Steve_ | how do i do that? |
| 15:54.33 | GeminiDomino | [TK]D-Fender: http://pastebin.com/d3a3e9ba9 |
| 15:54.40 | [TK]D-Fender | _Steve_: Enable SIP DEBUG at CLi and LOOK |
| 15:55.16 | [TK]D-Fender | GeminiDomino: You have no [channels] header in /etc/asterisk/chan_dahdi.conf |
| 15:55.20 | [TK]D-Fender | GeminiDomino: NOT sane |
| 15:55.45 | [TK]D-Fender | GeminiDomino: and /etc/asterisk/dahdi-channels.conf is worthless as it isn't included anywhere |
| 15:55.53 | [TK]D-Fender | GeminiDomino: Very busted configs. |
| 15:56.53 | [TK]D-Fender | GeminiDomino: and that file as a whole has bits alluding to configuring the same channel-rage in duplicate. |
| 15:57.34 | GeminiDomino | I saw that. That's why I didn't include it. The update.txt said that zapata.conf -> chan_dahdi.conf but I wasn't sure how I had to reconcile the two |
| 15:58.00 | [TK]D-Fender | GeminiDomino: Still missing the important [channels] header |
| 15:58.08 | GeminiDomino | adding it now |
| 15:58.27 | _Steve_ | Scheduling destruction of SIP dialog 'a8f30f24-e7d6b8ea-692d7b67@192.168.1.250' in 32000 ms (Method: REGISTER) |
| 16:01.07 | [TK]D-Fender | _Steve_: PASTEBIN. And that line is from *, not your provider and means very little right now. |
| 16:01.12 | _Steve_ | oops, wrong one |
| 16:01.16 | [TK]D-Fender | ~pb |
| 16:01.16 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste , or , http://bin.cakephp.org/ |
| 16:01.19 | _Steve_ | wrong IP anyway... |
| 16:01.28 | GeminiDomino | Ok. Added channels header. Still no dice. :P |
| 16:02.57 | [TK]D-Fender | goes to lunch |
| 16:03.48 | *** join/#asterisk chiwawa_42 (n=jerome@can59-3-82-233-175-214.fbx.proxad.net) |
| 16:05.50 | *** join/#asterisk BobPierce (n=BobPierc@216.36.132.162) [NETSPLIT VICTIM] |
| 16:06.47 | *** join/#asterisk Failrar (n=Failrar@coffee.ipv6.kaufmann.tc) |
| 16:07.08 | *** join/#asterisk astassistant (n=skip@65-126-63-1.dia.static.qwest.net) |
| 16:08.57 | GeminiDomino | Truly amazing that even with debugging on the error doesn't change. |
| 16:09.01 | *** join/#asterisk retentiveboy (n=pdugas@74-95-28-37-Atlanta.hfc.comcastbusiness.net) |
| 16:09.02 | *** join/#asterisk dni (n=dniz0r@adsl-074-169-015-252.sip.mia.bellsouth.net) |
| 16:09.08 | *** part/#asterisk retentiveboy (n=pdugas@74-95-28-37-Atlanta.hfc.comcastbusiness.net) |
| 16:09.20 | dni | could someone confirm that this will remove the first digit i am sending in the extension (i.e: the 9) exten => _9NXXNXXXXXX,1,Dial(SIP/callman01/${EXTEN:1}) ? |
| 16:09.47 | *** join/#asterisk CunningPike (n=CunningP@204.239.10.119) |
| 16:09.51 | *** join/#asterisk juanIMP (n=Juancho@200.71.41.22) |
| 16:10.24 | *** join/#asterisk tfrew (n=tfrew@office.neteasyinc.com) |
| 16:10.29 | tfrew | howdy asterisk |
| 16:11.38 | kuku1 | Do we need dahdi if we are just using sip ? |
| 16:12.16 | Qwell | kuku1: "just sip"? no other applications? |
| 16:13.42 | *** join/#asterisk saftsack (n=oliver@p4FC75F29.dip.t-dialin.net) |
| 16:15.55 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
| 16:17.14 | Dvyjones | Hello, I'm configureing an asterisk system using this: http://www.asterikast.com/show_notes/sn_4.txt |
| 16:17.50 | Dvyjones | I've managed to connect to it using Ekiga, and made a call to the extension "600", but it cuts off during the playback of the "weasels" thing. |
| 16:18.38 | Dvyjones | "Weasels have eaten our" |
| 16:19.53 | kuku1 | @Qwell: meaning, no zap. sip and iax, it has otehr modules if that is what you are asking |
| 16:21.40 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 16:22.29 | GeminiDomino | Ok, evidently it's a timer error... |
| 16:24.34 | *** join/#asterisk tobias (n=tobias@cpe-071-070-219-040.nc.res.rr.com) |
| 16:25.10 | *** join/#asterisk ChkDigit (n=mike@static24-72-71-175.regina.accesscomm.ca) |
| 16:28.04 | *** join/#asterisk putnopvut (n=putnopvu@asterisk/master-of-queues/mmichelson) |
| 16:28.04 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 16:28.10 | Qwell | kuku1: well, if you use things like meetme, you will need it |
| 16:28.58 | kuku1 | well, the new one is dahdi |
| 16:29.05 | kuku1 | is there ztdummy in there ? |
| 16:29.15 | Qwell | dahdi_dummy |
| 16:29.19 | kuku1 | and how do I get rid of the old zaptel, if I already had it installed |
| 16:29.29 | kuku1 | make dahdi_dummy ? |
| 16:29.44 | Qwell | just install dahdi like normal |
| 16:29.54 | kuku1 | and dahdi-tools ? |
| 16:30.16 | Qwell | not sure if you need dahdi-tools.. I would install it though |
| 16:30.26 | Qwell | it won't hurt anything to |
| 16:30.48 | kuku1 | so no makefile modifications do dahdi to have the dummy running correct ? And how do I get rid of zaptel / |
| 16:31.57 | Qwell | installing dahdi removes zaptel |
| 16:32.04 | kuku1 | ok |
| 16:33.28 | pif | I'm trying this "exten => 8501,n,GoToIf($[${VMCOUNT(5844)} > 0]?vm1)" but it always succeeds even if VMCOUNT returns 0 |
| 16:33.39 | kuku1 | @Qwell: and asterisk install would be: make clean;make; make install; ? |
| 16:33.48 | Qwell | kuku1: you'll have to re-run configure |
| 16:34.07 | kuku1 | so ./configure;make;make install; ? |
| 16:34.12 | Qwell | yep |
| 16:34.24 | kuku1 | beautiful |
| 16:38.03 | [TK]D-Fender | dni: yes, and asking took longer than trying |
| 16:38.39 | [TK]D-Fender | pif: PASteBIN |
| 16:38.46 | jameswf | LMFAO my pigflu-pandemic-hotline dialplan done... http://pastebin.com/f6551ebbe |
| 16:39.05 | Qwell | jameswf: DID? |
| 16:39.14 | *** join/#asterisk utahsaint_ (n=utahsain@64.190.142.58) |
| 16:39.33 | [TK]D-Fender | jameswf: ANTEDILUVIAN. WTF is with the 1.0 dialplan? |
| 16:39.40 | jameswf | Qwell: not yet still have to do two official soundingrecordings then I am going to switch my asticrapper did over |
| 16:40.32 | [TK]D-Fender | jameswf: And inefficient at that |
| 16:41.16 | jameswf | [TK]D-Fender: this is a joke not dialplan golf |
| 16:41.49 | [TK]D-Fender | FORE!!!!! |
| 16:42.00 | [TK]D-Fender | swats are jameswf with a 7-iron |
| 16:42.07 | Dvyjones | How do I make an extension forward to another SIP account on another registrar (ekiga.net) |
| 16:42.26 | [TK]D-Fender | Dvyjones: No such thing as "forward". You DIAL. |
| 16:42.37 | *** join/#asterisk viraptor (n=viraptor@awh178.internetdsl.tpnet.pl) |
| 16:42.38 | thedonvaughn | Dvyjones: exten => blah,1,Dial(SIP/ekiga_gateway/blah) ? |
| 16:43.17 | kuku1 | Now I get this error: [Apr 28 11:42:21] ERROR[23046]: utils.c:966 ast_carefulwrite: write() returned error: Broken pipe |
| 16:43.37 | *** join/#asterisk grEvenX (n=even@cB78D5AC1.dhcp.bluecom.no) |
| 16:45.34 | viraptor | is there any way to catch and process a failed transfer? for example if a dialog named in sip/REFER doesn't exist and I want to try some other specific dialog? |
| 16:48.02 | *** join/#asterisk mchou (n=mchou@unaffiliated/mchou) |
| 16:50.55 | jameswf | crap my home box is not playing nice :( |
| 16:51.09 | dni | [TK]D-Fender, i agree but the reason i asked is because it wasn't removing the digit,. here is the debug ,. http://pastebin.com/m4065b98a |
| 16:53.04 | [TK]D-Fender | dni: Looking for 92130398663 in int-phones (domain 172.16.0.59) SIP/2.0 404 Not Found <-- not FOUND. the line is never getting called in the first place |
| 16:53.40 | *** join/#asterisk UQlev (n=yuriy@87.228.199.125) |
| 16:55.17 | [TK]D-Fender | [12:09]<dni>could someone confirm that this will remove the first digit i am sending in the extension (i.e: the 9) exten => _9NXXNXXXXXX,1,Dial(SIP/callman01/${EXTEN:1}) ? <- and realize that number you dialed does NOT match the patter in that exten |
| 16:55.57 | [TK]D-Fender | dni: 0 does not match N |
| 17:01.22 | Dvyjones | Lol, asterisk is fun :) |
| 17:07.53 | GeminiDomino | [TK]D-Fender: Apparently it's a driver issue. The card is failing the timer-check. |
| 17:08.05 | [TK]D-Fender | GeminiDomino: fun..... |
| 17:08.36 | GeminiDomino | Hardly surprising. An ebay clone of an EOLed card... Must not murder boss... |
| 17:08.44 | watchy | tk: in extensions.conf does the extension on top of botton override? |
| 17:08.54 | Qwell | ~cheap |
| 17:08.55 | infobot | rumour has it, cheap is a bad idea. If you're setting up an Asterisk system, one of the wisest pieces of advice you can take is this: DON'T BE A CHEAPSKATE. |
| 17:08.56 | Qwell | GeminiDomino: ^^ |
| 17:08.58 | [TK]D-Fender | watchy: ....huh? |
| 17:09.57 | watchy | if i had 2 dialplans for exten => 501,blah which would work, the 1 on top of extensions.conf or the one on bottom |
| 17:11.05 | [TK]D-Fender | watchy: UH?!?! |
| 17:11.10 | watchy | hmm |
| 17:11.17 | watchy | if you have 2 dialplans that are the same |
| 17:11.19 | bmoraca | watchy: are they in the same context? if not, neither. if yes, redo your dialplan. |
| 17:11.22 | [TK]D-Fender | watchy: apstebin it, you're descriptions suck :p |
| 17:11.30 | watchy | ok i'llpaste bin |
| 17:11.38 | Qwell | [TK]D-Fender: so does you're grammar |
| 17:11.39 | [TK]D-Fender | watchy: and "dialplan" is the entire damn file :0 |
| 17:11.42 | Qwell | :P |
| 17:11.53 | [TK]D-Fender | Qwell: No, only the left-right synch of my typing :) |
| 17:12.15 | watchy | http://pastebin.com/m277fede2 |
| 17:12.21 | watchy | thats about as simple as I can make it |
| 17:12.25 | watchy | which is actually used? |
| 17:12.46 | [TK]D-Fender | watchy: 2nd IIRC and is stupid to do that in the same context |
| 17:12.52 | watchy | i agree |
| 17:12.55 | watchy | its a dumb thing |
| 17:13.06 | [TK]D-Fender | watchy: so why are you doing it? |
| 17:13.13 | *** join/#asterisk utahsaint_ (n=utahsain@64.190.142.58) |
| 17:13.20 | watchy | i'm just curious which would actually be used, in case of duplicates but with different functions |
| 17:13.23 | [TK]D-Fender | watchy: and the least you could do is make those 2 lines DIFFEREN |
| 17:13.54 | watchy | i wanted to know if ones at the top or bottom had priority |
| 17:14.18 | [TK]D-Fender | watchy: First come, first serverd... you should see some of the nifty "n" bugs that have come out lately :) |
| 17:14.21 | *** join/#asterisk lanning (n=lanning@nat/yahoo/x-5ba01d319d396a00) |
| 17:14.23 | *** join/#asterisk generalhan (n=asd@about/windows/staff/generalhan) |
| 17:14.29 | [TK]D-Fender | gah, can't type today |
| 17:14.58 | watchy | ah ok, i was just curious how it was processed |
| 17:15.59 | watchy | what do you think about keeping configs in sql? |
| 17:16.12 | [TK]D-Fender | watchy: Never cared to |
| 17:16.38 | watchy | you guys do it all manually or you have a gui you use to edit confs? |
| 17:16.44 | [TK]D-Fender | watchy: watchy Yes |
| 17:17.13 | watchy | so both? |
| 17:17.51 | generalhan | hey all, im trying to figure out the best way to get my faxes in reliably ... i have a PRI >> TE210 >> Asterisk >> TD40B >> Fax Server ... i have EC turned off on the TDM and that seemed to help a little, but faxes are still very unreliable. is there any other tweaks i can make to the PRI config to better this ? |
| 17:17.57 | [TK]D-Fender | watchy: Yes |
| 17:18.17 | watchy | custom gui or freepbx? :) |
| 17:18.33 | [TK]D-Fender | watchy: Yes |
| 17:18.47 | rue_mohr | ok, this barheader code for dahdi monitor is so simple, so whats going wrong |
| 17:19.56 | generalhan | since there are probably things that i CAN tweak with the PRI, but might not want to since there will be more calls than faxes, what if i had a seperate PRI that ONLY fax numbers resided on ... could i then make some better-for-faxing changes ? |
| 17:20.06 | watchy | i want to interface asterisk with a barmonkey, anyone done that |
| 17:20.34 | coppice | generalhan: there is no way to sync your PRI card with your TDM card, so don't expect reliable FAXing that way. Sangoma have provided a means to line cards by an extra cable to sync them |
| 17:21.43 | rue_mohr | t1 chnnelbank and your set |
| 17:21.59 | generalhan | coppice: so you are suggesting that i would have better luck working with sagnoma cards and faxing, than with digium hardware ? |
| 17:22.45 | [TK]D-Fender | generalhan: Ask your interconnector if tin cups & string are right for you! |
| 17:22.45 | coppice | that, or use a channel bank attached to a port on the same PRI card, so there is accurate syncing |
| 17:23.09 | generalhan | [TK]D-Fender: lol, they might actually say yes |
| 17:23.59 | coppice | cans and string is fibre to the home |
| 17:24.29 | [TK]D-Fender | insists on ony the highest quality poly/cotton blend |
| 17:27.17 | coppice | yes, you need to take care how you spin a yarn |
| 17:28.19 | [TK]D-Fender | coppice: Line and get your very own Asterisk doily! |
| 17:28.26 | eppigy | D: |
| 17:32.06 | generalhan | maybe im not fully understanding this channel bank concept, you have a good reference to learn what you are talking about ? lol |
| 17:33.23 | *** join/#asterisk ghento (n=ghento@99.254.47.47) |
| 17:35.01 | [TK]D-Fender | generalhan: http://www.telephonydepot.com/Catalog/Rhino-Channel-Banks;jsessionid=0a0106521f434185c6548b8e4c78a47af3bc424ea200.e3eSc34RbhyRe34Pa38Ta3aKb3b0 |
| 17:35.34 | generalhan | [TK]D-Fender: i found some ... but im more looking for information about how it works, and/or how to set it up |
| 17:35.44 | rue_mohr | IN DAHDI_MONITOR.C dahdi_copy_string _MUST_ be changed to memcpy, the dahdi_copy_string is copying a NULL that causes dahdi_monitor to not print its header properly! |
| 17:35.54 | rue_mohr | do _I_ have to submit the patch!? |
| 17:35.58 | [TK]D-Fender | generalhan: Its a dumb T1 device. |
| 17:36.03 | rue_mohr | please dont make me go though that |
| 17:36.49 | rue_mohr | is anyone in an easy position to submit that patch? |
| 17:37.48 | generalhan | [TK]D-Fender: it seems to me that this is more like taking my analog signal from my provider and turning it into a T1-type connection for my PBX ? |
| 17:37.58 | generalhan | not the other way around |
| 17:38.20 | [TK]D-Fender | generalhan: plug your FAXES into one. I didnt' say to take analog lines from your telco |
| 17:40.09 | generalhan | [TK]D-Fender: i know you didnt say that... i just dont really understand how this works |
| 17:40.23 | [TK]D-Fender | generalhan: 1 side in, other side out |
| 17:40.34 | [TK]D-Fender | generalhan: mix as appropriate |
| 17:41.07 | *** part/#asterisk juanIMP (n=Juancho@200.71.41.22) |
| 17:41.47 | generalhan | [TK]D-Fender: thats way to general ... so i have 2 T1 ports on this device ? one for the line from the telco, and one to go into my asterisk server, then 24 FXO/FXS ports to connect analog devices to ? |
| 17:42.23 | [TK]D-Fender | generalhan: Whatever kind of ports you want that you need synced,e tc |
| 17:42.34 | [TK]D-Fender | generalhan: I also never said 24 :) |
| 17:42.49 | generalhan | [TK]D-Fender: the page you linked has nothing but 24 port devices |
| 17:42.57 | eppigy | 48 PROTS |
| 17:43.25 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 17:43.26 | [TK]D-Fender | generalhan: Failure... try again |
| 17:43.48 | generalhan | ... |
| 17:44.09 | [TK]D-Fender | generalhan: the last one is MODULAR. |
| 17:44.28 | eppigy | reading comprehension |
| 17:44.57 | *** join/#asterisk horvath (n=horvath@74-51-45-200.telnetcommunications.com) |
| 17:45.23 | [TK]D-Fender | eppigy: http://tinyurl.com/496svm |
| 17:45.42 | horvath | Has anyone managed to get key system functionality aka SLA working with SPA942's ? |
| 17:46.39 | eppigy | [TK]D-Fender: lollin |
| 17:47.09 | [TK]D-Fender | horvath: First its only 1/2 of "SLA", and next thats 3 lines max. |
| 17:47.10 | *** join/#asterisk cesar_CR (n=cesar@201.192.86.29) |
| 17:47.32 | [TK]D-Fender | horvath: And thats assuming you like having only 1 call at a time |
| 17:47.56 | horvath | [TK]D-Fender: There are 4 line buttons on the SPA942's |
| 17:48.10 | horvath | [TK]D-Fender: and what do you mean 1 call at a time? |
| 17:48.22 | [TK]D-Fender | hovYes, and I also know how *'s "SLA" "works" |
| 17:48.34 | *** join/#asterisk voxter (n=voxter@76.77.95.2) |
| 17:48.46 | [TK]D-Fender | horvath: Go read up on how you implement it and you'll discover otherwise |
| 17:49.13 | horvath | [TK]D-Fender: Thats what I'm having a problem with theres the blog post and sla.pdf thats all the documentation I have found |
| 17:49.29 | [TK]D-Fender | horvath: 1 reserved for actual call appearance, other 3 for SPEED DiALS to fake it out. |
| 17:49.55 | *** join/#asterisk ingenius (n=alektro@host57.190-138-60.telecom.net.ar) |
| 17:50.23 | generalhan | [TK]D-Fender: ok so if i have a TE210 i would have my telco PRI coming into one port, and the other port going out to the channel bank, and then the analog lines to the fax server will plug into the channel bank ? sound like im getting closer ? |
| 17:50.41 | horvath | [TK]D-Fender: Is trying to implement key system functionality not work my time at this point? ie more trouble then its worth |
| 17:50.47 | [TK]D-Fender | generalhan: There you go... 3rd time's the charm |
| 17:51.08 | [TK]D-Fender | horvath: It only works for "lines", and in your case 3, if you're lucky |
| 17:51.29 | [TK]D-Fender | horvath: maybe you'll feel it is |
| 17:51.38 | *** join/#asterisk martyn-job (i=be18869a@gateway/web/ajax/mibbit.com/x-cc3ab7209fa44241) |
| 17:51.42 | martyn-job | hi |
| 17:51.51 | generalhan | [TK]D-Fender: and do you know of any channel banks that are WAY less than 24? my fax server only has a 4 line availability, so 24 is overkill, i think. |
| 17:52.21 | martyn-job | do you know if with asterisk can i do an ivr with accompaniment with the customer and agent? |
| 17:52.47 | *** join/#asterisk scurb (n=scurb@static-93.158.79.102.got.public.icomera.com) |
| 17:53.06 | [TK]D-Fender | generalhan: Well this usint is modular, in fact ALL that I've ever seen are 24 port because thats what T1. what its equiped for on a modular basis is another matter |
| 17:53.07 | coppice | generalhan: then consider the sangoma cards which allow the clock to be shared |
| 17:54.33 | horvath | [TK]D-Fender: Is this patch on the bugtrac required? http://bugs.digium.com/view.php?id=11688 |
| 17:55.48 | *** join/#asterisk tessier_ (n=treed@kernel-panic/sex-machines) |
| 17:56.40 | [TK]D-Fender | horvath: Required? More like another in-process alternative. |
| 17:56.55 | [TK]D-Fender | horvath: no idea of its suitability, stability,etc |
| 17:57.18 | horvath | [TK]D-Fender: So your saying I can do SLA with SPA942's without any patches with just following sla.pdf and the digium blog post? |
| 17:58.10 | horvath | [TK]D-Fender: Cause I think I'm missing something because when I press line2 on the spa942 It's not dialing asterisk and getting DISA |
| 17:58.11 | [TK]D-Fender | horvath: I'm saying you can use what * currently passes off as "SLA" if you want, or you can try one of these continually open patches if you want. |
| 17:58.46 | [TK]D-Fender | horvath: I also can't speak of your problems implementing the included "SLA' functionality because I don't see any configs, output, etc |
| 18:00.05 | horvath | [TK]D-Fender: Ok question tho just incase I'm missing something. The idea is that when I press line2 on the phone it dials asterisk and I get DISA for dialtone correct? |
| 18:00.31 | [TK]D-Fender | horvath: Yeah, something like that |
| 18:00.31 | kuku1 | Any ideas for: [Apr 28 12:58:45] ERROR[7291]: utils.c:966 ast_carefulwrite: write() returned error: Broken pipe ? |
| 18:00.36 | [TK]D-Fender | horvath: its all MeetMe'd up |
| 18:00.43 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 18:00.58 | horvath | [TK]D-Fender: Ok then its a problem with my SPA942 config |
| 18:02.41 | carrar | Sounds like a 3 step process |
| 18:02.58 | carrar | 1) FIX |
| 18:02.58 | carrar | 2) IT |
| 18:02.58 | carrar | 3) FIX IT |
| 18:03.35 | Superbartt | u sure about step 1 and 2? |
| 18:03.39 | carrar | no |
| 18:11.16 | *** join/#asterisk nkohh (n=justin@unaffiliated/kohh) |
| 18:21.57 | *** join/#asterisk ghento (n=ghento@CPE0014bfaf7d7a-CM00159a03312c.cpe.net.cable.rogers.com) |
| 18:22.18 | *** join/#asterisk Foriskak__ (n=Foriskak@69.38.211.98) |
| 18:23.00 | Foriskak__ | I want to dial a full number if a user enters in an extension, does this work: &dial-with-failover-NYC(646985${EXTEN}); |
| 18:23.29 | Foriskak__ | where ${EXTEN} is from a _49XX => { } |
| 18:25.26 | jameswf | no reply to our critical packet is natting issues? |
| 18:26.57 | [TK]D-Fender | jameswf: Or other networking. |
| 18:27.17 | [TK]D-Fender | jameswf: Why the facket isn't getting answered can vay. NAT issues are popular :) |
| 18:28.37 | jameswf | i had to redo my firewalls and havent played with asterisk since... now setting up the piggy flu thing and it drops the call in 10-15 seconds |
| 18:28.48 | jameswf | bah |
| 18:33.06 | [TK]D-Fender | jameswf: Your system must have caught it! |
| 18:33.14 | [TK]D-Fender | tightens his firewalls |
| 18:33.30 | *** join/#asterisk bionoid (i=terje@mesyah.com) |
| 18:35.28 | bionoid | Hello telephone gurus. Ages ago I set up an asterisk system that has a TDM400. It connects one single (remote) VOIP user to a PBX. So the mainboard died, and I'm migrating to newer Asterisk version. Q follows: |
| 18:36.38 | bionoid | Everything seems to work, I can call the PBX and am immediately connected with the VOIP endpoint, but when dialling, I see the incoming request w/number from SIP client, and yield this error: [Apr 28 20:11:39] WARNING[7296]: chan_zap.c:2010 zt_call: Unable to start channel: No data available -- Couldn't call 3/MYNUMBER |
| 18:37.18 | bionoid | And not much else to help make sense of what exactly is the problem. Could anyone point me in the right direction? |
| 18:37.29 | gr0mit | no incoming callier id? |
| 18:37.48 | *** join/#asterisk degrade (n=degrade@unaffiliated/degrade) |
| 18:37.54 | gr0mit | never really used analogue though |
| 18:38.10 | bionoid | Hm no callerid, but I also disabled everything related to it |
| 18:40.16 | Foriskak__ | Anyone think what I want to do works? |
| 18:40.23 | Foriskak__ | I want to dial a full number if a user enters in an extension, does this work: &dial-with-failover-NYC(646985${EXTEN}); |
| 18:40.38 | Foriskak__ | does it just concatenate the two? |
| 18:43.06 | [TK]D-Fender | forYes. |
| 18:43.15 | [TK]D-Fender | Foriskak__: rYes. |
| 18:43.17 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) |
| 18:43.18 | [TK]D-Fender | gah |
| 18:45.04 | Foriskak__ | [TK]D-Fender: so it should just work, if it occurs inside of a _49XX => {} block? |
| 18:46.11 | [TK]D-Fender | Foriskak__: I don't see a complete implementation so I can't say |
| 18:46.27 | [TK]D-Fender | Foriskak__: So far a macro call doesn' tell me what it does |
| 18:47.47 | *** join/#asterisk empiric (n=empiric@116.71.57.155) |
| 18:47.51 | empiric | hi all |
| 18:48.35 | empiric | i have dlink dvg-3004S 4 port FXO gateway i want to integrate with asterisk any idea? |
| 18:48.51 | defsdoor | can anyone help me with a polycom ip6000 - I cannot get it to register - I'm using the web interface to configure it |
| 18:49.05 | *** join/#asterisk unspin (n=unspin@96.49.129.159) |
| 18:49.53 | Superbartt | empiric can you let it talk sip or smthing for every port? |
| 18:50.14 | empiric | i hve web based panel for dlink |
| 18:50.43 | nkohh | one of my SIP providers is sending inbound calls to my Asterisk box with no extension ([s@usercontext]) -- i'm puzzled by this because it seems to allow me no way to send multiple DIDs to multiple extensions on one SIP user. is this the case? is it common for commercial PBXs to require a SIP user for every DID? there's gotta be something I'm missing. |
| 18:50.55 | Superbartt | well if the device just "passes trough" the FXO ports to sip accounts, then it's easy to make it use Asterisk |
| 18:50.55 | [TK]D-Fender | empiric: Go set up a peer for it like you would for an ITSP, point the gateway towards * and watch how it sends the call over. |
| 18:50.56 | empiric | in astersk i have 200 and 201 sip accounts how i configure with dlink to dial out outside my network |
| 18:51.49 | [TK]D-Fender | nkohh: they send the call to "s" because you didn't tell them otherwise in your REGISTER statement |
| 18:51.50 | Katty | hummm. |
| 18:52.41 | Superbartt | empiric get the dlink to register to asterisk, how... totally depends on the device, i'm not familiair with it |
| 18:53.59 | Katty | do i want: popcorn, ruffles, cheetos, doritos, or pretzles for my snack |
| 18:54.03 | empiric | hoow? |
| 18:54.22 | Katty | i also have strawberry and blueberry cereal bars, oatmeal, and granola bars. |
| 18:54.55 | [TK]D-Fender | empiric: Go look at sample configs for any other device. basic peer entry |
| 18:54.59 | *** join/#asterisk KavanS (n=KavanS@static-71-117-242-28.ptldor.dsl-w.verizon.net) |
| 18:55.01 | nkohh | [TK]D-Fender: Oh. |
| 18:55.06 | nkohh | thank you. |
| 18:55.32 | Katty | mmm |
| 18:55.33 | Katty | cheetos. |
| 18:55.39 | [TK]D-Fender | nkhWhat peer they auth as if any at all is another matter, and simply the way they work |
| 18:55.39 | eppigy | man I miss cereal |
| 18:55.50 | Superbartt | Katty do you have any asterisk knowledge? :p I've never seen you say anything asterisk related actually, just wondering |
| 18:55.52 | empiric | peer2peer i gave my asterisk server ip 192.168.1.1 |
| 18:55.55 | empiric | 15 |
| 18:56.08 | Katty | Superbartt: i'm just here for the stimulating conversation. |
| 18:56.10 | Foriskak__ | [TK]D-Fender: the macro calls that number with dial |
| 18:56.14 | Superbartt | lol |
| 18:56.37 | [TK]D-Fender | Foriskak__: Well I'm not going to sign off that your setup is fine, I've seen only a tiny portion of it |
| 18:56.41 | Superbartt | how uhmm, usefull :p |
| 18:57.00 | Foriskak__ | [TK]D-Fender: macro dial-with-failover-NYC( ext ) { } and the ${ext} gets use in a dial |
| 18:57.13 | [TK]D-Fender | Foriskak__: So... does it do what you expect? |
| 18:57.33 | Superbartt | does anyone actually is a dCap here? I'm trying to convince my boss to pay the training and the certificate... |
| 18:57.37 | Katty | Superbartt: also, that was extremely rude. |
| 18:58.02 | Foriskak__ | [TK]D-Fender: haven't tried yet, will ${ext} contain what I expect it will? |
| 18:58.24 | Superbartt | Katty: I would prefer sarcastic... |
| 18:58.38 | [TK]D-Fender | Foriskak__: dpends what you expect. How about you jsut go and try and tell us after |
| 18:58.42 | Katty | Superbartt: no, i think rude sums it up nicely. |
| 18:58.55 | Foriskak__ | That is, if macro called like this: &dial-with-failover-NYC(646985${EXTEN}); , will ${ext} be 64698599XX depending on extension? |
| 18:59.14 | Katty | but i am a nice person and don't usually tell people to piss off. |
| 18:59.21 | Katty | so you're in luck today! |
| 18:59.38 | Superbartt | how sweet of you... but even if you did I dont think I would have really bothered :p |
| 19:00.14 | Foriskak__ | [TK]D-Fender: the hope was that I could rely on experience/expertise in this channel to know ahead of time :) |
| 19:00.17 | [TK]D-Fender | Foriskak__: Should concaatenate |
| 19:00.25 | Foriskak__ | [TK]D-Fender: TIAS is an always easy answer |
| 19:00.34 | eppigy | http://imgur.com/27K39.jpg |
| 19:00.37 | nkohh | [TK]D-Fender: I see what you're saying... but no, I get that... the problem is that with many of my sip providers, the calls they send route through to, for instance, 8885555555@whatever-context, this allows me to put multiple DIDs on the same sip user and route them all differently. ... I assume this is because they are specifying this extension on their end, right? like I can call whoever@whatever. this one, however, does not. and I'm wondering if the |
| 19:00.37 | [TK]D-Fender | Foriskak__: Except it takes you 10 times as long to ask as to find out for yourself. |
| 19:00.42 | Katty | HI DAVE |
| 19:00.49 | eppigy | herro |
| 19:00.51 | eppigy | :D |
| 19:00.53 | Katty | i'm grumpy today |
| 19:00.56 | eppigy | :< |
| 19:01.03 | Katty | people keep pissing me off, like Superbartt |
| 19:01.08 | Katty | for like, no reason |
| 19:01.12 | eppigy | damn |
| 19:01.13 | Katty | ARE YOU THINKING WHAT I"M THINKING |
| 19:01.20 | eppigy | Superbartt: damn dog whats the deal |
| 19:01.21 | Katty | i'm thinking it's like doom, for the next week |
| 19:01.24 | [TK]D-Fender | nkohh: Ho look at the actual invite. Perhaps they send you ANOTHER important header with the actually dialed # <- |
| 19:01.39 | eppigy | I am thinking about foods :< |
| 19:01.41 | Superbartt | In my defense... I deny everything eppigy |
| 19:01.42 | nkohh | good idea. will do. thanks! |
| 19:01.51 | Foriskak__ | [TK]D-Fender: no it doesn't, since system is in production, have to schedule downtime, etc. |
| 19:01.54 | Katty | eppigy: cheetos cheetos cheetos |
| 19:01.58 | Katty | eppigy: mushroommushroom |
| 19:02.03 | eppigy | I liek cheetos |
| 19:02.04 | Katty | eppigy: ohhhh some cheese, get some cheese |
| 19:02.07 | *** join/#asterisk agx (n=badpengu@88-149-227-96.dynamic.ngi.it) |
| 19:02.14 | eppigy | except the disscoloration |
| 19:02.20 | eppigy | of your fingers |
| 19:02.25 | Katty | Superbartt: and yes. i do know asterisk. |
| 19:02.31 | eppigy | I have some hot pockets |
| 19:02.33 | [TK]D-Fender | Foriskak__: Don't have a single softphone on the side to test with without taking EVERYBODY down? very sad. |
| 19:02.35 | eppigy | with cheese in them |
| 19:02.35 | Katty | Superbartt: i have been here for nearly 3 years. |
| 19:02.45 | rob0 | chan_cheetos.c |
| 19:02.57 | nny_1 | http://vimeo.com/4294567 |
| 19:03.00 | [TK]D-Fender | robI heard that eats up a lot of resources |
| 19:03.01 | rob0 | Katty wrote that channel driver. |
| 19:03.06 | Katty | rob0: hot. |
| 19:03.22 | Katty | i'll write your channel driver in a minute. |
| 19:03.26 | Superbartt | ok Katty, then I got my answer ;) was just wondering |
| 19:03.40 | Katty | Superbartt: i am pissy today. i appologize. |
| 19:03.49 | Katty | presents Superbartt with a blueberry muffin as peace offering. |
| 19:04.05 | Superbartt | Don't think my stomach can hold muffins currently :p |
| 19:04.10 | Katty | :< |
| 19:04.16 | gr0mit | mmmmh muffins |
| 19:04.20 | Katty | well maybe you could just look at it. |
| 19:04.24 | Katty | and admire it from a safe distance. |
| 19:04.27 | Superbartt | but i'll accept your peace offer |
| 19:04.35 | watchy | i see you in the club showin thugs love |
| 19:04.48 | Katty | watchy: your face. |
| 19:04.58 | watchy | dont make me serial hug you. |
| 19:05.05 | Katty | watchy: hug your face. |
| 19:05.08 | jameswf | The piggy Flu hotline powered by asterisk: +1-253-243-1726 |
| 19:05.13 | KavanS | omg thug luv! |
| 19:05.14 | Katty | ( i have no idea where i'm going with this) |
| 19:05.26 | Katty | jameswf: does that go to rhino equipment |
| 19:05.27 | rob0 | jameswf: ipkall++ :) |
| 19:05.33 | watchy | katty: your a very mean girl and you should cry |
| 19:05.45 | Katty | watchy: i kinda feel like crying |
| 19:05.48 | Katty | watchy: i'm very emotional today |
| 19:06.02 | watchy | im on my peroid ive been crying all day |
| 19:06.02 | eppigy | judging from the hugged appearance of your face, you are suffering from fetal alchohol syndrome |
| 19:06.03 | rob0 | Have some cheetos. |
| 19:06.16 | Katty | i had some cheetos. |
| 19:06.18 | rob0 | I wonder if asterisk has been ported to CheetOS yet? |
| 19:06.19 | Katty | OH |
| 19:06.23 | Katty | i have monster in the server room. |
| 19:06.28 | Katty | woah. |
| 19:06.32 | eppigy | FIRED |
| 19:06.35 | Katty | cheetOS |
| 19:06.43 | Katty | eppigy: it's in a cardboard container. in a drawer. |
| 19:06.48 | eppigy | :D |
| 19:06.53 | Katty | eppigy: you can have some if you don't tell. |
| 19:07.08 | rob0 | CheetOS : Unix that turns your fingers orange |
| 19:07.16 | watchy | i wish i could eat cheetos |
| 19:07.21 | watchy | but im just to fat |
| 19:07.22 | eppigy | yeah dude I am down |
| 19:07.29 | jameswf | Katty: it is Ipkall via my house |
| 19:07.30 | *** join/#asterisk agallo (n=badpengu@88-149-227-96.dynamic.ngi.it) |
| 19:07.42 | Katty | jameswf: hot. |
| 19:07.43 | eppigy | i am like a vault of secrets |
| 19:07.57 | [TK]D-Fender | eppigy: ... stay sealed :p |
| 19:07.57 | eppigy | brb making hot pocket |
| 19:08.05 | eppigy | 8[] |
| 19:08.12 | *** part/#asterisk agallo (n=badpengu@88-149-227-96.dynamic.ngi.it) |
| 19:08.14 | Katty | i'm feeling a comedy skit quote coming on |
| 19:08.21 | watchy | tk: is there a standard in place for phone system extensions? for voicemail voicemenus etc? |
| 19:08.36 | *** part/#asterisk nny_1 (n=scott@64.203.244.146) |
| 19:08.39 | [TK]D-Fender | watchy: ...... |
| 19:08.54 | [TK]D-Fender | reaches for his ClueBat (tm) |
| 19:08.55 | watchy | like *8 = this and should always eb this? |
| 19:09.19 | *** join/#asterisk Assimilate (n=Assimila@72.22.242.66) |
| 19:09.22 | watchy | i havent eaten all day so i'm trying to not be stupid |
| 19:09.27 | jameswf | I should make it twitter... Swineflu hotline got a call from: Arizona |
| 19:09.29 | watchy | but like i said i'm to fat to eat |
| 19:09.40 | Pan3D | sighs |
| 19:09.49 | [TK]D-Fender | watchy: http://tinyurl.com/496svm |
| 19:10.06 | watchy | haha thats a nice url |
| 19:10.22 | watchy | i would just think there would be some kinda standard |
| 19:10.37 | Foriskak__ | [TK]D-Fender: how would a single softphone help? there is still only one PBX and one dialplan? |
| 19:10.46 | [TK]D-Fender | watchy: there might be... for some sanely reassembled fragment of your previous question. |
| 19:11.11 | [TK]D-Fender | Foriskak__: copy&paste the macro to another name and use it |
| 19:11.11 | watchy | ok, i thought of a better way to explain |
| 19:11.16 | Foriskak__ | [TK]D-Fender: you mean a spare PBX setup identically but with no contact with the lines/PRI with option to connect a softphone to it |
| 19:11.27 | watchy | is there a standard to what extensions certain features should be mapped to? |
| 19:11.31 | [TK]D-Fender | Foriskak__: No, I mean copy& paste 1 silly macro |
| 19:11.48 | [TK]D-Fender | watchy: Some people follow CLASS |
| 19:11.49 | Foriskak__ | [TK]D-Fender: of course I can do that, but asterisk needs restart to reload dialplan? |
| 19:11.50 | [TK]D-Fender | ~class |
| 19:11.50 | infobot | i heard class is over |
| 19:11.58 | Foriskak__ | [TK]D-Fender: it's not like sip.conf or voicemail changes? |
| 19:12.04 | jameswf | C has no class |
| 19:12.09 | [TK]D-Fender | forAnd what does that impact when no other devices uses the renamed copy of the macro? |
| 19:12.13 | watchy | class is over, so that doesnt exactly help :( |
| 19:12.26 | [TK]D-Fender | ~vsc |
| 19:12.26 | infobot | [vsc] Vertical Service Codes such as *67, *69, *72, and *82. These codes are generally reserved for specific uses, and it's a bad idea to conflict with the official assignments. A list of assigned VSCs for North America is at http://nanpa.com/number_resource_info/vsc_assignments.html and http://www.nanpa.com/number_resource_info/vsc_definitions.html |
| 19:12.28 | [TK]D-Fender | ^^^^^^^ |
| 19:12.42 | watchy | thanks, i love you. |
| 19:12.57 | watchy | when i visit quebec ill buy you beer and food |
| 19:13.01 | [TK]D-Fender | Foriskak__: What sip.conf changes? |
| 19:13.10 | rob0 | He won't go beyond a kiss on the first date, however. |
| 19:13.24 | [TK]D-Fender | Foriskak__: You telling me you're also changing all sorts of other configs and not applying those for fear as well? |
| 19:13.35 | watchy | rob: with ghb he might |
| 19:14.28 | [TK]D-Fender | watchy: First sign of anything wonky and I go into defensive "kill" mode :) |
| 19:14.52 | watchy | i believe that, your probably a non fat geek ninja, rare but they exist |
| 19:15.06 | [TK]D-Fender | watchy: I can then blame the outcome on being drugged and get off with nothing more than the dry-cleaning bill :) |
| 19:15.32 | [TK]D-Fender | watchy: Samurai, get it right serf :p |
| 19:15.57 | [TK]D-Fender | "Ninja".... lol... virtually no legit schools of this anywhere |
| 19:16.27 | [TK]D-Fender | Mind you ninjutsu (the true aspect of intelligence gathering) is an aspect of my curriculum |
| 19:16.58 | [TK]D-Fender | is enjoying our expansion towards the jiu-jutsu side right now |
| 19:18.03 | Katty | armed or unarmed? |
| 19:18.07 | Foriskak__ | [TK]D-Fender: nothing, except the restart of the server cuts off any current calls? I can do this at night with 'at', but will have to check on it that it restarted |
| 19:18.32 | [TK]D-Fender | Katty: Jiu-jutsu = unarmed |
| 19:18.35 | *** join/#asterisk goupil (n=goupil@2a01:e35:2f3d:7900:240:63ff:fedc:10e) |
| 19:18.40 | goupil | hi |
| 19:18.41 | [TK]D-Fender | Foriskak__: No, it doesn't |
| 19:19.20 | Katty | well at least we don't have to worry about you being sliced and diced then. |
| 19:20.42 | *** join/#asterisk Dealer2mogette (n=Dealer2m@16.104.80-79.rev.gaoland.net) |
| 19:20.54 | Dealer2mogette | hello |
| 19:22.13 | Dealer2mogette | i've a little problem with my asterisk server. when i connect to it with a client, it said "Register attempt for proxy sip: 555@192.168.1.25 failed" |
| 19:22.18 | eppigy | SUCCESS |
| 19:22.20 | Dealer2mogette | Do you know how can i resolve that ? |
| 19:23.43 | guax | why on sip trunking the callerid number is not the real callerid? =x, i got the real number as callerid name and no as numebr |
| 19:23.44 | guax | number |
| 19:24.07 | [TK]D-Fender | guax: Depends on your provider |
| 19:24.16 | guax | [TK]D-Fender, i provide to myself |
| 19:24.33 | [TK]D-Fender | Dealer2mogette: That doesn't say why it failed. Go look at * SIP debug to see what is going on. |
| 19:24.45 | [TK]D-Fender | guax: Do you talk to yourself often? |
| 19:24.54 | guax | its another asterisk that dials to the sip in another machine, callerid number in the other asterisk machine is right |
| 19:25.07 | guax | [TK]D-Fender, mostly for debuging purposes =P |
| 19:25.28 | [TK]D-Fender | guax: I'd go look at some SIP debug as well. |
| 19:25.48 | guax | i had look at it, but just receive the real number as callerid |
| 19:25.53 | guax | as id at all |
| 19:26.10 | [TK]D-Fender | guax: well we can't add anything until you show us. Pastebin is your friend |
| 19:26.13 | [TK]D-Fender | ~pb |
| 19:26.13 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste , or , http://bin.cakephp.org/ |
| 19:27.28 | guax | [TK]D-Fender, http://pastebin.com/d477c2cd |
| 19:28.29 | [TK]D-Fender | guax: From: "4891575933" <sip:8037@192.168.10.252>;tag=as0942d7db |
| 19:28.30 | Dealer2mogette | [TK]D-Fender: the command is : sip set debug ? |
| 19:28.43 | eppigy | Katty: how do you feel about anteaters |
| 19:28.50 | Katty | eppigy: cute nose. |
| 19:28.54 | guax | [TK]D-Fender, exactly |
| 19:28.59 | [TK]D-Fender | guax: indeed no name. No look from the OTHER EnD |
| 19:29.03 | [TK]D-Fender | now* |
| 19:29.13 | [TK]D-Fender | Dealer2mogette: yes |
| 19:29.17 | Dealer2mogette | ok |
| 19:29.51 | eppigy | i recently found out they can walk on hind legs |
| 19:29.59 | eppigy | they are my new favorite animal |
| 19:30.15 | [TK]D-Fender | eppigy: http://tinyurl.com/c4lkwd |
| 19:30.26 | Katty | eppigy: neat. |
| 19:30.33 | eppigy | YERS |
| 19:30.34 | eppigy | YES |
| 19:30.47 | bmoraca | someone emailed that pic to me yesterday |
| 19:30.59 | Katty | hehe |
| 19:31.19 | eppigy | http://i58.photobucket.com/albums/g245/eppigy/greg-anteater.jpg |
| 19:31.25 | Dealer2mogette | [TK]D-Fender: i've this http://pastebin.com/d2c0a59b5 |
| 19:32.19 | guax | [TK]D-Fender, http://pastebin.com/d45b15770 |
| 19:32.59 | bmoraca | Dealer2mogette: "No matching peer found" from your debug |
| 19:33.55 | Dealer2mogette | bmoraca: ok, but how can i resolve that ? |
| 19:34.15 | bmoraca | Dealer2mogette: create a peer in sip.conf that matches the credentials you're trying to register with |
| 19:34.52 | Dealer2mogette | it's what i've done |
| 19:35.20 | bmoraca | Dealer2mogette: why don't you pastebin your sip.conf and the credentials you're trying to use |
| 19:36.36 | *** join/#asterisk qdk (n=qdk@0x55816749.terminal.tdcmobil.dk) |
| 19:38.42 | [TK]D-Fender | guax: I've got a pretty good guess as to why it isn't coming through, and you'd have to look at your peer entry |
| 19:39.19 | [TK]D-Fender | Dealer2mogette: You have no peer for 555 |
| 19:39.28 | guax | its a friend, i tryed peer and user as well |
| 19:39.42 | [TK]D-Fender | guax: SAME THING. |
| 19:39.51 | guax | what you sugest? |
| 19:39.59 | [TK]D-Fender | guax: that you pastebin it |
| 19:40.01 | guax | let-me pastebin it |
| 19:40.07 | guax | =P |
| 19:41.16 | Dealer2mogette | [TK]D-Fender, bmoraca : i've add this at the default sip.conf file http://pastebin.com/d1f3124f7 |
| 19:41.39 | bmoraca | Dealer2mogette: well, there's your problem...neither of those are peer 555 |
| 19:41.47 | [TK]D-Fender | Dealer2mogette: I sure as hell don't see a [555] in there |
| 19:42.05 | *** join/#asterisk UQlev (n=yuriy@91.184.221.31) |
| 19:43.11 | guax | [TK]D-Fender, http://pastebin.com/d12a155ac |
| 19:43.21 | *** join/#asterisk SparFux (n=raoul@f050021136.adsl.alicedsl.de) |
| 19:43.46 | [TK]D-Fender | guax: fromuser=8037<- bad |
| 19:44.09 | [TK]D-Fender | guax: I see you've been fighting like hell jsut to get it to auth |
| 19:44.13 | SparFux | I am using Asterisk 1.4 with linux-call-router chan_lcr and I get random DTMF tones. Asterisk tells me: [Apr 28 21:42:37] NOTICE[24408]: chan_lcr.c:1059 lcr_in_dtmf: [call=6 ast=lcr/1] Recognised DTMF digit '5'. Anybody got an idea how to get rid of this? |
| 19:44.17 | guax | let-me try, in that server the entries are generated, not my work |
| 19:44.20 | nkohh | does asterisk have an implementation of switch/case? |
| 19:44.33 | [TK]D-Fender | guax: add "sendrpid=yes" and "trustrpid" to both |
| 19:44.36 | SparFux | I think not. |
| 19:44.42 | Qwell | nkohh: in AEL |
| 19:44.58 | nkohh | Qwell: thanks :-) |
| 19:44.58 | [TK]D-Fender | nkohh: "sorta" |
| 19:45.01 | guax | trustrpid=yes and sendrpid=yes added |
| 19:45.04 | guax | let me try |
| 19:45.47 | guax | it now works ? |
| 19:45.53 | guax | s/?/:}/g |
| 19:46.08 | guax | good bot |
| 19:46.41 | guax | [TK]D-Fender, what that sendrpid does? |
| 19:47.56 | guax | ok, i google it =P |
| 19:47.57 | bmoraca | browsing through pastebin is quite funny |
| 19:48.36 | Katty | it's very interesting to listen to glen miller and stare at pictures from the 1920s |
| 19:48.42 | [TK]D-Fender | guax: Remote Party ID. Auth as the user pass 33rd party callerID. Good for... oh I dunno... calling between 2 * boxes and wanting to see a devices specifics :) |
| 19:48.54 | [TK]D-Fender | 3rd |
| 19:48.57 | [TK]D-Fender | wow |
| 19:49.06 | [TK]D-Fender | 33 & 1/3rd! |
| 19:49.14 | [TK]D-Fender | Neison's out |
| 19:49.45 | nkohh | [TK]D-Fender: you were right, by the way. the called number was provided in a previous "To:" header that I was able to parse out. |
| 19:49.46 | bionoid | I'm SSHing to a box and doing asterisk -r -- is there a way to "force dial" a number on a zap channel?, I have no OSS or ALSA drivers in the kernel (or, obviously, in Asterisk) |
| 19:50.16 | bionoid | I just want to call my cell and see if the channel actually has outbound capabilities (beginning to doubt it..:\) |
| 19:50.20 | Qwell | bionoid: originate |
| 19:50.28 | [TK]D-Fender | nkohh: Never would have guessed :) |
| 19:50.41 | bionoid | Oh, thanks Qwell! |
| 19:51.03 | nkohh | two Cut commands and a rewrite of my dialplan into AEL later, I'm able to route by DID. |
| 19:52.37 | *** join/#asterisk sulex (n=sulex@pdpc/supporter/professional/sulex) |
| 19:52.45 | martyn-job | .. |
| 19:53.06 | [TK]D-Fender | nkohh: .... |
| 19:53.29 | [TK]D-Fender | nkohh: AEL .... highly unnecessary |
| 19:53.53 | nkohh | how would you recommend i switch the value for 14 different cases |
| 19:53.59 | martyn-job | Do you know how can i do to do an ivr with accompaniment feature ? . |
| 19:54.07 | martyn-job | That the agent do accompaniment to customer on ivr |
| 19:54.20 | watchy | tk: you use anything to graph call volume? |
| 19:55.48 | [TK]D-Fender | nkohh: 14 lines of extensions, and 1 variable GOTO. 15 lines total, and probably a lot cleaner looking :) |
| 19:56.24 | horvath | [TK]D-Fender: Spent the time... got it working and your right its quite ugly |
| 19:56.51 | [TK]D-Fender | watchy: http://tinyurl.com/ajx7yd |
| 19:57.18 | [TK]D-Fender | horvath: "FUGLY" |
| 19:57.26 | watchy | hahah :( |
| 19:58.45 | seanbright | how anyone could think straight dialplan is "cleaner looking" than AEL is beyond me |
| 19:59.10 | seanbright | we could just add line numbers to dialplan too |
| 19:59.15 | seanbright | and then i can stab myself in the eye |
| 19:59.30 | [TK]D-Fender | seanbright: Allow me to do it for you :) |
| 19:59.34 | seanbright | pass |
| 19:59.38 | horvath | [TK]D-Fender: No way to get around this using 2 line buttons for 1 call? |
| 19:59.54 | seanbright | just a shame that "because I don't use it" equates to "it blows" in here |
| 19:59.55 | [TK]D-Fender | horvath: Yes... invent real SLA :) |
| 20:00.12 | horvath | [TK]D-Fender: Ok... I'll get right on that for you :) |
| 20:00.33 | [TK]D-Fender | seanbright: No, it jsut blows. Your use of it is purely incidental :) |
| 20:00.42 | keith4 | I have a queue that has a Local/cellphone as a member. if they guy sent the call to voicemail on his cell, the queue would consider that an answer. so, I changed the outgoing context to use a call-screening macro (that is working fine for "find me" use), but it doesn't quite work with the queue |
| 20:01.16 | Dealer2mogette | [TK]D-Fender, bmoraca : now i've this error |
| 20:01.17 | Dealer2mogette | [Apr 28 22:00:28] WARNING[10512]: chan_sip.c:2921 create_addr: No such host: Client1 |
| 20:01.17 | Dealer2mogette | [Apr 28 22:00:28] WARNING[10512]: app_dial.c:1202 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) |
| 20:01.33 | [TK]D-Fender | Dealer2mogette: pastebin the whole mess |
| 20:01.55 | Dealer2mogette | ok |
| 20:03.06 | horvath | [TK]D-Fender: Do you know how to get the SPA942 to use the fake line4 button by default when you pickup the handset to make a call? It grabs from top to bottom I want from bottom to top :) |
| 20:03.31 | Dealer2mogette | [TK]D-Fender, bmoraca: http://pastebin.com/d377ea86d |
| 20:04.10 | keith4 | Dealer2mogette: DNS problem? |
| 20:04.24 | [TK]D-Fender | Dealer2mogette: show me that Client2 i registered |
| 20:05.03 | Dealer2mogette | yes : http://pastebin.com/d50d96827 |
| 20:05.48 | keith4 | heh |
| 20:06.02 | keith4 | Dealer2mogette: paste 'sip show peers' |
| 20:06.16 | *** join/#asterisk sulex (n=sulex@pdpc/supporter/professional/sulex) |
| 20:06.22 | sulex | j #linux-it |
| 20:06.25 | [TK]D-Fender | Dealer2mogette: you have no [client2] |
| 20:06.26 | sulex | sorry |
| 20:06.56 | [TK]D-Fender | Dealer2mogette: you are renaming everything and breaking something new for everything you fix |
| 20:07.48 | Dealer2mogette | so i must rename Client1 by 555 for example ? |
| 20:08.17 | *** join/#asterisk agallo (n=badpengu@88-149-227-96.dynamic.ngi.it) |
| 20:08.55 | *** part/#asterisk agallo (n=badpengu@88-149-227-96.dynamic.ngi.it) |
| 20:09.48 | [TK]D-Fender | 556 <- |
| 20:10.29 | Dealer2mogette | [TK]D-Fender: 556 is the second client (client2) |
| 20:11.01 | bionoid | I've successfully dialled out with my TDM400 (from CLI), if I dial in to Zap, SIP forward works perfectly; But I'm still stuck here; SIP client attempts to dial outbound results in WARNING[7296]: chan_zap.c:2010 zt_call: Unable to start channel: No data available -- Couldn't call 3/012345678 |
| 20:11.09 | [TK]D-Fender | Dealer2mogette: Sorry, yes... the peer you dial is the [thisthinghere] header from sip.conf |
| 20:11.32 | Dealer2mogette | yep |
| 20:12.00 | bionoid | If anyone has an idea, please do talk, and save me a trip tomorrow ;-) |
| 20:15.55 | tzafrir_laptop | bionoid, what is the exact Dial line in your dialplan? |
| 20:16.30 | Dealer2mogette | [TK]D-Fender: it's ok i've replace client1 by 555 and client2 by 556 in sip.conf and in extension.conf and it's works |
| 20:17.04 | Dealer2mogette | Thanks for your help :) |
| 20:17.22 | [TK]D-Fender | Dealer2mogette: You're welcome |
| 20:17.53 | Dealer2mogette | it's not easy to begin with asterisk (but it's an interesting voip server :D) |
| 20:18.30 | Qwell | ~book |
| 20:18.31 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 20:18.31 | *** join/#asterisk apocn (n=apo@unaffiliated/apocn) |
| 20:18.32 | Qwell | It's very easy. |
| 20:19.13 | bionoid | tzafrir_laptop: exten => _XXX.,1,Dial(${OUTBOUND}/0${EXTEN}) |
| 20:19.15 | apocn | Hello all, how can I see a specific channel without hitting TAB? right now to see a channel I first do: sip show channels then I get the ID and do sip show channel 2a94966e200 <TAB><ENTER> |
| 20:19.22 | jameswf | with the book + IRC + Common sense = easy asterisk |
| 20:19.24 | Dealer2mogette | hmm not very |
| 20:19.26 | bionoid | tzafrir_laptop: OUTBOUND=Zap/3 |
| 20:19.44 | *** join/#asterisk bgmarete (n=marebri_@196.201.208.129) |
| 20:19.49 | Qwell | apocn: huh? |
| 20:20.10 | jameswf | Dealer2mogette: what is hard? |
| 20:20.36 | jameswf | evaluates that question |
| 20:20.39 | apocn | Qwell: after hitting tab I get the full channel id (which is cutted to display sip show channels). |
| 20:22.29 | apocn | I'd like to get the full channel id without hitting tab, for example I want to get it from my shell script using -rx '...' |
| 20:23.57 | jameswf | apocn maybe asterisk -rx "show channels" | grep SIP |
| 20:24.06 | apocn | jameswf, nope |
| 20:24.14 | apocn | if you go to the CLI, execute sip show channels |
| 20:24.16 | Dealer2mogette | jameswf: to understand the mistake i have made |
| 20:24.51 | apocn | thensip show channel <the chan id it shows> then <tab> (to get the full channel id) then <enter> |
| 20:25.02 | apocn | I just want to get the full channel id without hitting tab |
| 20:25.43 | jameswf | show channels has all the channels in it |
| 20:26.11 | apocn | abreviated one (shown in sip show channels) = 75445e0a2f5 then after hitting tab it is 75445e0a2f54de90623acc8f42a2e7fa@mydomain.com |
| 20:26.29 | apocn | jameswf: all channel id are abreviated (cutted) in sip show channels |
| 20:26.41 | Qwell | the CLI isn't supposed to be used for scripting stuff like that.. |
| 20:26.43 | Qwell | use manager |
| 20:26.52 | apocn | Qwell: in the manager it will display it full? |
| 20:26.57 | Qwell | yes |
| 20:27.04 | apocn | ok, let me use it now |
| 20:27.06 | apocn | thanks Qwell |
| 20:27.51 | apocn | once I tried using the "command" action of AMI, maybe I need to look for the other action |
| 20:28.13 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 20:28.39 | bionoid | Is there a 'reverse-search' akin to bourne ^R in the CLI? |
| 20:29.16 | [TK]D-Fender | checkout time, BBIAB |
| 20:29.54 | bionoid | or is that bash only, I forget |
| 20:30.07 | jameswf | bash + AMI = kinda fugly... |
| 20:30.34 | apocn | jameswf: Perl + AMI actually |
| 20:30.45 | apocn | if I do it using a shell script I just use -rx |
| 20:30.48 | apocn | if I use AMI I prefer Perl |
| 20:31.10 | *** join/#asterisk agallo (n=badpengu@88-149-227-71.dynamic.ngi.it) |
| 20:32.39 | *** part/#asterisk agallo (n=badpengu@88-149-227-71.dynamic.ngi.it) |
| 20:33.45 | *** join/#asterisk umpc (n=Justin@unaffiliated/umpc) |
| 20:34.25 | *** join/#asterisk bbryant (n=brett@c-68-59-20-153.hsd1.sc.comcast.net) |
| 20:36.08 | *** join/#asterisk thehar (i=thehar@thehar.xmission.com) |
| 20:36.22 | *** part/#asterisk martyn-job (i=be18869a@gateway/web/ajax/mibbit.com/x-cc3ab7209fa44241) |
| 20:36.23 | thehar | russellb: you around? |
| 20:45.04 | *** join/#asterisk lesouvage (n=lesouvag@62.140.137.125) |
| 20:45.28 | Dealer2mogette | bye |
| 20:47.10 | apocn | another question, is it possible to start a MixMonitor with the call then start a second one only for a short period of time (without stopping the first one) |
| 20:50.17 | *** join/#asterisk SkywaIker (n=pirch@58.147.17.166) |
| 21:02.52 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 21:03.59 | Katty | eppigy: what's furrr dinner |
| 21:09.57 | *** join/#asterisk CrashSys (n=james@rrcs-24-173-156-170.se.biz.rr.com) |
| 21:10.12 | CrashSys | Anyone ever messed with global crossings PRI's in Asterisk? |
| 21:10.29 | CrashSys | I'm having an issue where when I dial out, they return cause code 99, and asterisk hangs-up |
| 21:12.14 | eppigy | Katty: htrmmmmm |
| 21:12.41 | eppigy | I am thinkign taco bell |
| 21:12.48 | eppigy | grilled stuffed burrito |
| 21:13.02 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194) |
| 21:13.13 | bionoid | Alright going on-site tomorrow I guess, thanks for your help everyone :) goodnight |
| 21:14.05 | *** join/#asterisk Maneta (n=daniel@77.228.135.212) |
| 21:22.28 | CrashSys | Anyone know if there is an option in the dial command to not send out caller id name? |
| 21:27.14 | [TK]D-Fender | CrashSys: not there. |
| 21:28.13 | *** join/#asterisk shinao1 (n=shinao1@41.219.234.124) |
| 21:28.26 | jblack | wtf. http://news.google.com, the word "influenza" isn't on the page. News blackout? |
| 21:29.46 | [TK]D-Fender | jblack: Why should it be? |
| 21:30.03 | jblack | um, because it's the start of a pandemic? |
| 21:30.22 | jblack | Though at this point it's only killing brown people, i hear. |
| 21:30.43 | [TK]D-Fender | jblack: look for FLU and realize there's over 1/2 dozen articles on that page |
| 21:30.44 | *** join/#asterisk shinao1 (n=shinao1@41.219.234.124) |
| 21:30.59 | [TK]D-Fender | jblack: Just because they didn't write it out in full... sheesh |
| 21:31.11 | jblack | Not even "flue" here. |
| 21:31.23 | jblack | heh. /me turns pink |
| 21:31.39 | [TK]D-Fender | jblack: FAIL |
| 21:32.01 | jblack | hard core. |
| 21:41.22 | rob0 | jpink |
| 21:43.11 | unpaidbill | anyone tried the new version of t38modem? |
| 21:43.25 | unpaidbill | should i get my hopes up that it doesnt suck and actually works? |
| 21:46.30 | *** join/#asterisk joako (n=joako@opensuse/member/joak0) |
| 21:48.18 | *** join/#asterisk batphone (n=wclayton@aus-pix-00-pat-01.corenap.com) |
| 21:48.43 | batphone | anyone seen this error on a polycom spectralink 8002: "TFTP ERROR(2):17" |
| 21:48.55 | batphone | i found documents defining some error codes, but not "17" |
| 21:49.16 | batphone | this is a brand new unit. im trying to put new firmware on it. |
| 21:50.08 | ghento | Hi all. I'm trying to understand how Transfer() works, was wondering if anyone can help. I have an outgoing call to a mobile phone via SIP. Once connected, I want to Transfer() the call and have it ring another mobile so the two phones are connected. I am trying this, but when I get to the Transfer(), the second phone rings once, and then quits. |
| 21:50.38 | *** join/#asterisk micols (n=mio@rlogin.dk) |
| 21:50.58 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 21:52.29 | jameswf | ~transfer |
| 21:54.06 | ghento | ~transfer |
| 21:57.50 | *** join/#asterisk freakazoid0223 (n=matt@pool-71-246-17-74.phlapa.fios.verizon.net) |
| 22:01.37 | *** join/#asterisk ecret (n=ecret@CPE001e9002348e-CM001225d8ab30.cpe.net.cable.rogers.com) |
| 22:08.43 | *** join/#asterisk jtodd (n=jtodd@180.sub-75-254-55.myvzw.com) |
| 22:08.43 | *** mode/#asterisk [+o jtodd] by ChanServ |
| 22:21.49 | rue_mohr | how do I know what is tx and what is rx on an phone interface? |
| 22:22.20 | *** join/#asterisk cp5 (n=samy@72.37.252.206) |
| 22:23.04 | jaytee | "it's like Deja Vu all over again" |
| 22:25.36 | cp5 | any of you know of a bug where asterisk (tested on 1.2 and 1.6) will crash when too many members exist in queues.conf (regardless of whether it's spread out over a lot of queues or a few) |
| 22:25.36 | cp5 | the crash either happens at startup or when calls come in, or when calls come in and hit any queues |
| 22:26.07 | cp5 | the core doesn't seem to properly reflect where the root cause of the problem is...happens on an ast_strdupa in 1.6 |
| 22:29.04 | *** join/#asterisk eliel (n=eliels@120-17-235-201.fibertel.com.ar) |
| 22:31.04 | seanbright | cp5: have you compiled with DONT_OPTIMIZE enabled in menuselect? |
| 22:31.19 | cp5 | seanbright: good call, i'll confirm i have that enabled |
| 22:32.11 | cp5 | going to test now |
| 22:39.47 | cp5 | seanbright: happens at the same location with DONT_OPTIMIZE and DEBUG_THREADS enabled |
| 22:40.06 | cp5 | valgrind says "Bad permissions for mapped region at address 0x53C1FF8" |
| 22:40.16 | seanbright | hmmm. blowing out the stack mayhaps. |
| 22:40.33 | seanbright | what version of asterisk and what line number? |
| 22:40.39 | cp5 | how would that happen? i haven't found any sort of maximums in the ao2 struct or queue module |
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| 22:41.06 | cp5 | 1.6.0.9, app_queue.c around line 850 -- specifically in the handle_statechange function: interface = ast_strdupa(curint->interface); |
| 22:43.00 | seanbright | ah. yes... |
| 22:43.08 | seanbright | as a test... change that line to: |
| 22:43.14 | seanbright | ast_strdup(curint->interface); |
| 22:43.32 | cp5 | ok, testing |
| 22:43.33 | seanbright | actually. hold that thought. i'll make a patch. |
| 22:43.35 | seanbright | no no |
| 22:43.37 | cp5 | k |
| 22:43.40 | seanbright | memory leak |
| 22:43.40 | seanbright | :) |
| 22:43.42 | cp5 | :) |
| 22:44.46 | seanbright | do you know how to apply patches? |
| 22:44.49 | cp5 | yeah |
| 22:45.05 | seanbright | cool |
| 22:46.14 | cp5 | gotta apply anti-pregnancy patches to my girlfriends all the time...you know, just in case |
| 22:47.18 | seanbright | http://pastie.org/461825.txt?key=i1ywkpow3ztjoqbxdunblw |
| 22:47.23 | cp5 | awesome, thanks, testing |
| 22:47.36 | cp5 | what's the diff between strdup and strdupa? |
| 22:47.44 | seanbright | strdupa allocates on the stack |
| 22:47.48 | seanbright | (really just moves the stack pointer) |
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| 22:48.33 | seanbright | malloc is to strdup as alloca is to strdupa |
| 22:49.28 | cp5 | ah |
| 22:50.02 | cp5 | so shouldn't strdupa auto-free the alloc'd memory? |
| 22:50.08 | seanbright | it does, yes. |
| 22:50.12 | seanbright | after the function returns |
| 22:50.45 | seanbright | which doesn't help in your case since ast_strdupa is running in a loop |
| 22:50.50 | seanbright | (this is my theory anyway) |
| 22:50.52 | cp5 | i see |
| 22:50.57 | cp5 | testing now |
| 22:51.07 | seanbright | cool. i'll go smoke in premature celebration. |
| 22:51.23 | cp5 | awesome |
| 22:54.23 | cp5 | seanbright: i think that fixed it! |
| 22:54.38 | cp5 | i'm going to run a few more tests here, but it's looking great. no crashing so far |
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| 22:54.47 | cp5 | where i was consistently crashing it before |
| 22:55.24 | cp5 | i'm curious why it would crash though...it doesn't seem like it could eat up that much memory in that function, even if it does have to permute through X thousand members. i was watching memory in top and didn't see it go crazy or anything |
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| 22:59.33 | seanbright | cp5: the stack is relatively small compared to the heap. how many 'interfaces' do you have? |
| 23:00.20 | cp5 | well in my big test, i have 20,000. that's unreasonable, in most cases i have a queue with a few hundred members, but i have enough queues that it becomes 5000+ members (even though it's the same member in most of the queues) |
| 23:00.31 | cp5 | the crash doesn't happen as easily in that scenario, it requires a number of calls before the problem occurs |
| 23:01.26 | seanbright | well, assuming a 4k stack size |
| 23:01.47 | seanbright | and 84 bytes per interface (80 for the name, 4 for the next pointer) |
| 23:01.58 | seanbright | yikes |
| 23:02.04 | cp5 | i see, makes sense |
| 23:02.16 | seanbright | i might be off |
| 23:02.21 | seanbright | but you get the general idea |
| 23:02.22 | seanbright | :) |
| 23:02.33 | seanbright | based on my math it should fail at ~50 |
| 23:02.49 | seanbright | which can't be right |
| 23:02.50 | cp5 | hah |
| 23:02.51 | cp5 | nice |
| 23:02.59 | cp5 | well thank you |
| 23:03.06 | seanbright | no problem. |
| 23:03.19 | seanbright | reminds himself to commit that fix at some point |
| 23:03.20 | seanbright | :) |
| 23:03.51 | cp5 | :) don't know if you guys still update 1.2, but it happens there too in the 'changethread' function |
| 23:04.05 | cp5 | i'll go ahead and patch that myself, similar enough |
| 23:04.07 | cp5 | but just fyi |
| 23:04.32 | seanbright | ulimit -s |
| 23:04.35 | seanbright | run that ^^ |
| 23:07.05 | seanbright | cp5: can you run 'ulimit -s' on the command line and tell me what it returns? |
| 23:07.48 | cp5 | seanbright: sure, 10240 |
| 23:07.54 | seanbright | ah |
| 23:08.10 | cp5 | does it hurt to increase it? |
| 23:08.13 | seanbright | ok, so 10240 / 80 is your max number of interfaces before crashing |
| 23:08.17 | cp5 | inside of safe_asterisk |
| 23:08.28 | cp5 | i see |
| 23:09.12 | seanbright | and honestly, i'm not sure if it hurts. might hurt performance. |
| 23:09.19 | seanbright | but i'm far from an expert. |
| 23:09.45 | cp5 | ok, i'll leave it then |
| 23:10.26 | seanbright | as for 1.2, that's not getting fixed. you'll just want to patch that locally. |
| 23:10.58 | seanbright | unless it was a security problem, which it isn't, since you control the members of your queues. interesting that it hasn't cropped up before with other users though. |
| 23:11.18 | seanbright | anyway |
| 23:11.21 | seanbright | good luck and god speed. |
| 23:11.22 | seanbright | :) |
| 23:14.06 | cp5 | i see |
| 23:14.29 | cp5 | seanbright: thanks again. one more question, is there a similar function to ast_strdup in 1.2? i don't see it. should i simply copy the macro from 1.6 and drop it in 1.2 without worry? |
| 23:14.32 | rue_mohr | OH is rx and tx all relative to ASTERISK aka, if a co line is reveiving audio its gain is set by rx and out to the sip phone (would be) tx? |
| 23:17.16 | seanbright | cp5: i actually have a better patch for you. give me a second to whip up a 1.2 version. |
| 23:18.10 | cp5 | seanbright: nice, thanks |
| 23:20.33 | seanbright | cp5: http://pastie.org/461856.txt?key=ihfwa8rccuadqeom6o4tsw |
| 23:20.36 | cp5 | k |
| 23:20.44 | seanbright | if you could test that and let me know, that would be great. |
| 23:21.04 | seanbright | can't commit to 1.2, but i can base 1.4, 1.6.x and trunk on it |
| 23:21.54 | rue_mohr | how can I make a mwi light come on, on a sip phone? |
| 23:22.04 | cp5 | seanbright: sure. i also see another place the copy should happen about 30 lines lower too |
| 23:22.14 | cp5 | inside: for (cur = q->members; cur; cur = cur->next) { |
| 23:22.41 | seanbright | ah, indeed. |
| 23:22.44 | seanbright | hold for new patch. |
| 23:22.44 | rue_mohr | it looks like the mwi thing can run a shall script |
| 23:23.55 | rue_mohr | as a joke, the recpetionist said to have her phone play camptown races if the co line mwi is active, she better be carefull or she'll get it |
| 23:25.18 | rue_mohr | how is paging done with sip sets, do i have a line set up with a auto answer? |
| 23:25.32 | cp5 | rue_mohr: does your phone support it? look at hints: http://www.voip-info.org/wiki/view/Asterisk+standard+extensions#StandardPriorities |
| 23:26.07 | rue_mohr | its a plycom601 |
| 23:27.04 | rue_mohr | oh I see, well, the polycom phone dosn't have enough buttons with lights for me to do that |
| 23:27.14 | rue_mohr | its why I dont like the polycoms |
| 23:27.32 | seanbright | cp5: http://pastie.org/461865.txt?key=aoudrusyaemcqm5l7licg |
| 23:27.51 | cp5 | rue_mohr: oops, mwi..got confused. do you have mailbox= set for the sip peers? |
| 23:27.59 | cp5 | seanbright: awesome, will test |
| 23:28.48 | rue_mohr | they dont have mailboxes |
| 23:28.56 | rue_mohr | its a pots line mwi |
| 23:29.16 | rue_mohr | in the chan_dahdi.conf you can have it detect the mwi and run a shell script |
| 23:30.02 | rue_mohr | wondering how I do paging, |
| 23:30.05 | rue_mohr | work bye |
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| 23:38.41 | cp5 | seanbright: works on 1.2 like a charm. no more reproducible crashing |
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| 23:40.26 | seanbright | cp5: splendid |
| 23:40.33 | cp5 | seanbright: YTMND |
| 23:41.09 | seanbright | heh |
| 23:41.17 | seanbright | worst. movie. ever. |
| 23:42.03 | seanbright | also patting myself on the back for figuring that one out. |
| 23:42.13 | seanbright | the YTMND thing |
| 23:42.14 | seanbright | heh |
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| 23:42.38 | cp5 | hah |
| 23:42.48 | cp5 | i cried at the end. |
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| 23:44.03 | StinkyJew | yo |
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| 23:46.22 | seanbright | cp5: you aren't compiling with LOW_MEMORY defined are you? |
| 23:46.38 | cp5 | seanbright: i haven't set that flag myself, but i'll double check |
| 23:46.55 | *** topic/#asterisk by leifmadsen -> Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.1.0 (2009/04/28), Asterisk 1.6.0.9 (2009/04/06), 1.4.24.1 (2009/04/02), *-Addons 1.6.1.0 (2009/04/28), 1.6.0.1 (2008/12/02), 1.4.8 (2009/04/28), dahdi-linux 2.1.0.4, dahdi-tools 2.1.0.3 (2009/02/03), Libpri 1.4.10 (2009/04/18) -=- Related channels: #asterisknow #asterisk-gui #switchvox #freepbx #asterisk-bugs #asterisk-dev #asterisk-commits |
| 23:47.10 | leifmadsen | Asterisk 1.6.1.0, and Asterisk-Addons 1.6.1.0; 1.4.8 are now available! |
| 23:47.15 | Qwell | OH NOES! |
| 23:47.22 | leifmadsen | OH YAH! |
| 23:47.33 | Qwell | leifmadsen: Did you build AsteriskNOW packages for me?! |
| 23:47.47 | leifmadsen | can't say that I did |
| 23:47.52 | Qwell | lame! you used to be cool, man. |
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| 23:49.49 | nauticalthinker | k |
| 23:54.03 | jameswf | can someone push a sip call to 1000@68.109.169.243 please thx |
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| 23:58.50 | IsUp | hello |
| 23:59.06 | *** join/#asterisk telnettech (n=telnette@cpe-71-74-91-116.insight.res.rr.com) |
| 23:59.06 | IsUp | any ideas about faxing on 1.4.24.1? |
| 23:59.31 | telnettech | good evening TK and jaytee |
