| 00:03.26 | shmaltz | rhombus, thats a good question, it's 2.6.27 |
| 00:03.42 | shmaltz | I downloaded 1.4 now, ant it works, but I reaaly want 1.2 |
| 00:04.10 | rhombus | shmaltz: I'd be willing to bet that kernel is not supported in 1.2. |
| 00:04.16 | rhombus | shmaltz: Why do you want 1.2? |
| 00:04.53 | *** join/#asterisk coppice (n=chatzill@204.196.17.210.dyn.pacific.net.hk) |
| 00:07.01 | shmaltz | rhombus, because I'm comfortable with it. |
| 00:07.43 | rhombus | shmaltz: It might actually be time to contemplate getting comfortable with 1.4. I am still running 1.2 myself, but I think 1.4 is now stable enough that it merits switching, unless you have a specific issue with 1.4. |
| 00:08.27 | shmaltz | <PROTECTED> |
| 00:08.49 | rhombus | shmaltz: Have you had problems with SIP with 1.4 recently? |
| 00:08.56 | shmaltz | and when it does it will be simple SIP between PRI only, no fancy stuff like AMI or any app integration |
| 00:09.08 | shmaltz | rhombus, never tried 1.4 before |
| 00:09.26 | shmaltz | today was the first day that I tried 1.4 |
| 00:09.29 | shmaltz | works so far |
| 00:09.31 | rhombus | shmaltz: let me know how it goes :) |
| 00:09.46 | shmaltz | I was able to complet 2 calls in differnt direction |
| 00:09.53 | shmaltz | using the same PRI |
| 00:10.02 | shmaltz | evantualy it's going to use 7 PRIs |
| 00:10.09 | shmaltz | hopefully one day this week |
| 00:11.38 | shmaltz | does 1.4 support B channel transfer? |
| 00:15.28 | *** join/#asterisk ArchGT (n=ArchGT@190.149.125.27) |
| 00:15.46 | *** join/#asterisk securevoip (n=securevo@76.123.20.170) |
| 00:21.26 | *** join/#asterisk Iamnacho (i=Iamnacho@ip174-70-137-120.ks.ks.cox.net) |
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| 01:01.53 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-7dcd11efdd0165c1) |
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| 01:27.42 | hardwire | blah |
| 01:28.45 | PhunTelTek | yawns |
| 01:32.29 | *** join/#asterisk vegbox (n=kevinle@24-205-54-7.dhcp.gldl.ca.charter.com) |
| 01:49.00 | *** join/#asterisk viq (n=viq@unaffiliated/viq) |
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| 01:56.20 | jplank | is it possible to get the current date/time from inside the dial plan? Something like ${DATE} or something like that? |
| 01:58.53 | carrar | ${DATETIME} |
| 01:59.17 | carrar | ${STRFTIME(${EPOCH},,%d%mNaVH:NaVS)}) |
| 02:00.47 | carrar | DateTime([unixtime][|[timezone][|format]]) |
| 02:01.09 | jplank | thanks I'm looking at STRFTIME insdie the wiki right now |
| 02:01.14 | jplank | seems to be exactly what I'm looking for |
| 02:01.26 | carrar | anything else I can google for you? |
| 02:02.39 | *** join/#asterisk riddlebox (n=user@75-132-225-75.dhcp.stls.mo.charter.com) |
| 02:05.29 | *** join/#asterisk _bugz_ (n=bugz@99.129.29.102) |
| 02:07.47 | jplank | anyone have a fax machine near them? |
| 02:09.14 | riddlebox | sorry nope |
| 02:09.23 | Alfio | jplank me |
| 02:09.40 | jplank | can you try faxing something to 3472731219 ? |
| 02:09.50 | jplank | I want to see if it properly emails me |
| 02:09.57 | Alfio | its long distance for me, sorry |
| 02:10.00 | jplank | no |
| 02:10.03 | jplank | n/p |
| 02:10.11 | jplank | I think I found a free one online |
| 02:10.36 | Alfio | its long distance for me |
| 02:10.46 | Alfio | i cant dial that number |
| 02:13.07 | jplank | hmm lets see if this free online site actually sends the fax |
| 02:13.55 | carrar | install halifax |
| 02:14.05 | jplank | I'm using fax for asterisk |
| 02:14.25 | jplank | I got it working, I just wrote a little script to forward the faxes to my email when they arrive |
| 02:14.28 | carrar | hylafax |
| 02:15.00 | andresmujica | anyone knows where can i find the 2008 astricon pdf files for downlad? |
| 02:15.15 | jplank | that was a pain to find |
| 02:15.58 | jplank | I have a bunch of the PPT's downloaded if your looking for something specific |
| 02:16.47 | andresmujica | SellingFlexibility.pdf |
| 02:16.50 | jplank | grrrr the fax from this free website didn't go through, surprise surprise |
| 02:17.04 | jplank | ha I have it |
| 02:17.08 | jplank | the pdf |
| 02:17.17 | jplank | its about 5 mb |
| 02:17.24 | andresmujica | hehe |
| 02:17.26 | andresmujica | just found it |
| 02:17.29 | andresmujica | at sokoll |
| 02:17.44 | jplank | dont need it? |
| 02:17.46 | andresmujica | hmmm.. |
| 02:17.59 | andresmujica | give me a sec.. it seems to be at sokoll's site.. |
| 02:18.32 | andresmujica | http://www.sokol-associates.com/2008/glendale/web/presentations/ |
| 02:18.33 | andresmujica | yeap |
| 02:18.40 | andresmujica | thanks jplank !! |
| 02:18.44 | andresmujica | i've got it now |
| 02:19.03 | jplank | dont thank me, I didn't do anything |
| 02:19.16 | *** join/#asterisk OrNix (n=ornix@l151-249-47.static.cn.ru) |
| 02:19.18 | andresmujica | give me luck ;) |
| 02:19.25 | jplank | good luck! |
| 02:29.30 | S2AnGeL | [TK]D-Fender: So making a custom extention that dials custom-call_cell,wwwwtheNumberIamTryingToGetItToDial,1 looks about right |
| 02:30.02 | *** join/#asterisk Defraz (n=T0tal@24-117-156-215.cpe.cableone.net) |
| 02:37.10 | stope | jplank, what are you using to do the actual emailing? mutt? |
| 03:01.35 | *** join/#asterisk drmessano (n=nonya@pdpc/supporter/active/drmessano) |
| 03:02.07 | *** join/#asterisk kamanashisroy (n=kamanash@119.30.36.79) |
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| 03:25.12 | jplank | stope: yea, mut |
| 03:25.15 | jplank | mutt* |
| 03:25.35 | *** join/#asterisk howie (n=howie@71-95-220-206.dhcp.mtpk.ca.charter.com) |
| 03:26.41 | vegbox | i love asterisk |
| 03:26.49 | vegbox | lets make bumper stickers that say that |
| 03:26.50 | vegbox | lol |
| 03:32.11 | [TK]D-Fender | Slogan : "There's more than one way to crash Asterisk!" |
| 03:32.21 | stope | are you running mutt from a cronjob? |
| 03:32.23 | [TK]D-Fender | or "One more way!" |
| 03:34.15 | rob0 | I tried installing Halifax, but I ended up with too many Nova Scotians. |
| 03:35.01 | [TK]D-Fender | rob0: That definitely necessitates an Echo Canceller :p |
| 03:35.45 | rob0 | On the plus side, there were more of them than there were Newfies. |
| 03:45.52 | [TK]D-Fender | rob0: There was this Torontonian who wanted to become a Newfie and finally foudn a doctor that had a procedure for it. He warned the man that it was risky and involved removing 80% of his brain. The man agreed and awoke 3 days after the operation. Still groggy he heard the doctor say "I'm so terribly sorry, but there was an accident and we accidentally removed 95% of your brain". The man... |
| 03:45.54 | [TK]D-Fender | ...looked up and said "C'est rien, c'est pas grave!". :D |
| 03:46.14 | rob0 | haha yes I know that one |
| 03:46.19 | jplank | stope: I was using System to call it directly from asterisk |
| 03:46.52 | stope | cool, I'll give it a whirl |
| 03:47.57 | rob0 | BTW I'm down here in the deep South, Digium land. But it gets old joking about rednecks all the time, so I figured I'd pick on Canada. |
| 03:49.02 | [TK]D-Fender | rob0: Oh we're plenty capable of laughing at ourselves... |
| 03:49.36 | [TK]D-Fender | rob0: You're in a war with the Newfies and there are explosions all around. They're throwing grenades at you, what do you do? |
| 03:49.53 | [TK]D-Fender | rob0: Pull the pins out and throw them back! |
| 03:50.01 | [TK]D-Fender | Ba-dum-bomm! |
| 03:50.22 | *** join/#asterisk AussieGuy (n=AussieGu@r220-101-170-252.cpe.unwired.net.au) |
| 03:50.58 | AussieGuy | I can connect a netcomm v300 directly into an ethernet adsl2+ modem, then connect the netcomm to my linux machine for internet access right? |
| 03:51.29 | S2AnGeL | http://pastebin.com/d29ca5b9 ugh I seem to be going no where.. It just hangs up on me. I have some thing terribly wrong I am sure. I wish I understood zap flash better.. |
| 03:52.03 | AussieGuy | its a 3 port voip router one port to connect to the modem, one to the pc and one to the analog phone line |
| 03:52.57 | S2AnGeL | Hey I'm up in canada got my snow vehicle and my gun for them pesky polar bears |
| 03:54.56 | S2AnGeL | Has anyone done a zap flash transfer of a call before using the telco transfer option |
| 03:56.02 | S2AnGeL | if somene has a moment and could look over maybe give me a tip on what I am doing wrong http://pastebin.com/d29ca5b9 or has done it and could show me how.. |
| 03:56.52 | [TK]D-Fender | S2AnGeL: First are you SURE your telco supports flash TRANSFERS? This is extremely rare |
| 03:57.01 | S2AnGeL | Oh yes |
| 03:57.09 | S2AnGeL | with a manual phone I can do one |
| 03:57.15 | [TK]D-Fender | S2AnGeL: And noramlly you need a SECOND flash to confirm the tranfer much lie an Attended transfer |
| 03:57.24 | [TK]D-Fender | like* |
| 03:57.28 | S2AnGeL | oh |
| 03:57.41 | S2AnGeL | you got something there.. I need to flash again don't I |
| 03:58.36 | [TK]D-Fender | S2AnGeL: You should know the answer to that... |
| 03:58.41 | S2AnGeL | oh G wait a min.. no I though the hangup did but no you have something there.. |
| 03:58.57 | S2AnGeL | lemmie try that |
| 04:13.04 | S2AnGeL | nope |
| 04:13.27 | S2AnGeL | how do I read or see whats going on.. the error tells me crap |
| 04:13.51 | S2AnGeL | or the cli and logs just sorta that its hungup.. |
| 04:14.30 | S2AnGeL | does how I pass it look right its almost as if it sorta ignors it then heads to the net piority and hangs up |
| 04:14.44 | S2AnGeL | next not net |
| 04:15.28 | S2AnGeL | I added a exten => s,4,Flash() and increased the hangup to 5 |
| 04:17.07 | [TK]D-Fender | S2AnGeL: perhaps you should WAIT. |
| 04:17.16 | [TK]D-Fender | S2AnGeL: And perhaps even TEST this process by hand. |
| 04:17.32 | S2AnGeL | maybe a wait of some sort after the flash.. but flash is set in my zapata.conf rxwink=300 flash=750 wink=150 prewink=50 preflash=50 debounce=600 rxflash=1250 |
| 04:17.32 | [TK]D-Fender | S2AnGeL: `I might recommend doing each didig separately |
| 04:17.45 | S2AnGeL | yah |
| 04:17.51 | vegbox | Man a four port FXO card is 120 bucks |
| 04:17.55 | vegbox | thats straight cash |
| 04:17.57 | vegbox | :( |
| 04:18.57 | S2AnGeL | does thous for zapata.config look about right for canada toronto Bell Canada |
| 04:19.13 | S2AnGeL | vegbox: sounds like a good deal |
| 04:19.28 | S2AnGeL | but I ask whats wrong with it |
| 04:19.32 | [TK]D-Fender | vegbox: 120 for a 4-port card? Which? |
| 04:19.41 | [TK]D-Fender | vegbox: thats WAY too cheap |
| 04:19.49 | vegbox | TDM400p |
| 04:19.54 | vegbox | or something, a cheapie off ebay |
| 04:19.56 | S2AnGeL | where? |
| 04:20.01 | [TK]D-Fender | vegbox: an EMPTY TDM400p perhaps |
| 04:20.03 | rob0 | that's what I was thinking, cheap |
| 04:20.10 | S2AnGeL | lol |
| 04:20.35 | rob0 | modules are still running ~$70-80 last I saw |
| 04:21.54 | AussieGuy | the Linksys SPA3102 voip router has a lan port to connect to the pc and a wan port to connect to the modem, does that mean it will send internet access to the PC? or will I need to buy a router with multiple ports? |
| 04:22.05 | [TK]D-Fender | rob0: 41/44 |
| 04:22.27 | [TK]D-Fender | AussieGuy: It can act like a router if you want it to. |
| 04:22.32 | rob0 | oh is that all? Damn, my TDM400P was stolen. :( |
| 04:22.41 | AussieGuy | ah k cool, that saves alot of trouble in buying a router |
| 04:22.47 | AussieGuy | since I only have one pc |
| 04:22.58 | [TK]D-Fender | AussieGuy: Not necessarily |
| 04:23.09 | *** join/#asterisk obnauticus (n=obnautic@about/windows/regular/obnauticus) |
| 04:23.32 | [TK]D-Fender | AussieGuy: Unless you can configure it to keep off 5060 for both ports so * can work behind it |
| 04:23.44 | *** join/#asterisk war9407 (i=war@liquidswords.org) |
| 04:24.17 | AussieGuy | dhcp port? |
| 04:24.20 | *** join/#asterisk frk2 (n=frk2@zivios/member/fkhan) |
| 04:24.38 | [TK]D-Fender | AussieGuy: SIP <- |
| 04:25.20 | [TK]D-Fender | AussieGuy: Actually worse still.... that device only speaks SIP on its WAN jack |
| 04:25.30 | [TK]D-Fender | AussieGuy: No, I think you're pretty much going to need a router |
| 04:26.10 | AussieGuy | mainly I just want to forward all incoming calls to an overseas mobile number |
| 04:26.33 | AussieGuy | and have internet on my pc |
| 04:27.00 | [TK]D-Fender | AussieGuy: And where does * come into play? |
| 04:27.13 | AussieGuy | *? |
| 04:27.20 | [TK]D-Fender | ......... |
| 04:27.22 | [TK]D-Fender | ASTERISK |
| 04:27.31 | [TK]D-Fender | AussieGuy: Do you not know where you are? |
| 04:27.53 | AussieGuy | well asterisk is one possible solution |
| 04:27.58 | *** join/#asterisk CrazyTux[m]1 (n=Brandon@99-53-97-235.lightspeed.cyprtx.sbcglobal.net) |
| 04:28.18 | AussieGuy | so basically your saying I cant put asterisk with that router |
| 04:28.40 | [TK]D-Fender | AussieGuy: Not in that combination. |
| 04:28.55 | AussieGuy | Im likely to set up an asterisk server later |
| 04:29.09 | AussieGuy | for more advanced things |
| 04:29.37 | CrazyTux[m]1 | Hey guys -- quick question. I've a simple dial plan I'm working on, and I'm using MixMonitor to record the call, after a successful channel open/close (i.e. on hangup of the actual call) -- I want to do some database operations, can I simply do this after the Dial() application? |
| 04:29.37 | CrazyTux[m]1 | i.e. exten => x,1,MixMonitor() ---- exten => x,2,Dial() ---- exten => x,3,MYSQL stuff here............... |
| 04:29.41 | [TK]D-Fender | AussieGuy: You might be able to get the SPA to do that for you.... |
| 04:29.52 | AussieGuy | so if I wanted to set up an asterisk server with that, id need to get a router as well as the SPA |
| 04:30.03 | [TK]D-Fender | AussieGuy: Yes |
| 04:31.14 | AussieGuy | well its more a question of when rather than if, but the spa will do fine for now just giving my pc internet and forwarding off calls to a single mobile number |
| 04:32.16 | CrazyTux[m]1 | [TK]D-Fender: sorry to bug you -- but any input on my question all mighty? |
| 04:32.22 | AussieGuy | asterik will come in later when I start forwarding to multiple mobiles off a single line |
| 04:32.45 | [TK]D-Fender | CrazyTux[m]1: Do you not know how Dial works? |
| 04:33.05 | CrazyTux[m]1 | [TK]D-Fender: refresh my memory |
| 04:34.23 | [TK]D-Fender | CrazyTux[m]1: Go place some calls and relearn |
| 04:35.05 | CrazyTux[m]1 | [TK]D-Fender: I have no one to call at 11:34pm :P -- (totally kidding) ... ok let me go back to the drawing boards. |
| 04:35.40 | CrazyTux[m]1 | actually, let me do a quick test with noOp |
| 04:37.59 | CrazyTux[m]1 | [TK]D-Fender: looks like I want option g |
| 04:38.52 | CrazyTux[m]1 | [TK]D-Fender: perhaps not.... |
| 04:40.58 | S2AnGeL | ugh I am done |
| 04:41.02 | S2AnGeL | for the nite |
| 04:41.25 | S2AnGeL | maybe some chance It will come to me when I sleep |
| 04:41.51 | *** part/#asterisk S2AnGeL (n=S2AnGeL@74.12.50.212) |
| 04:47.08 | jplank | anyone with a fax near them? |
| 04:53.00 | *** join/#asterisk Topdeck (n=Topdeck@oxcoda.safenetbox.biz) |
| 05:04.09 | vegbox | Can I post a link here? |
| 05:04.12 | *** join/#asterisk shido6 (n=shido6@74-132-202-71.dhcp.insightbb.com) |
| 05:04.37 | vegbox | http://cgi.ebay.com/FXO-card-TDM400P-asterisk-card-with-4-FXO-S-ChinaRoby_W0QQitemZ150355814638QQcmdZViewItemQQptZLH_DefaultDomain_0?hash=item2301e7a8ee&_trksid=p3286.c0.m14&_trkparms=65%3A12|66%3A2|39%3A1|72%3A1234|293%3A1|294%3A50 |
| 05:04.44 | vegbox | Thats 100 bucks plus 20 bones for shipping |
| 05:04.48 | vegbox | comes with four fxo ports |
| 05:07.19 | [TK]D-Fender | vegbox: Chinese knock-off |
| 05:11.29 | vegbox | hmmmm |
| 05:11.31 | vegbox | lol |
| 05:11.34 | vegbox | that means nothign to me |
| 05:11.35 | vegbox | lol |
| 05:12.00 | [TK]D-Fender | vegbox: Also means that support is non-existant |
| 05:12.19 | vegbox | i did okay with the chinese knock off of the 100p card |
| 05:12.19 | vegbox | lol |
| 05:12.23 | [TK]D-Fender | vegbox: Go for it, good luck and if you get burned, consider this a pre-emptive "We ttold you so" |
| 05:14.17 | *** join/#asterisk ruben23 (n=AGENT@122.55.48.243) |
| 05:14.29 | rob0 | Bastards even went so far as to put an Asterisk logo on the card! |
| 05:14.50 | vegbox | ol |
| 05:14.51 | vegbox | lol |
| 05:14.58 | vegbox | well given this card/box is used for VOICEMAIL only |
| 05:15.05 | vegbox | i think i can win on this knock off |
| 05:15.23 | vegbox | if i was using it was a switch for both calls and voicemail id be cautious in getting this |
| 05:15.36 | CrazyTux[m]1 | [TK]D-Fender: I'm sincerely still unsure on how to do this, just spent some time googling, not coming up with much. |
| 05:15.41 | [TK]D-Fender | rob0: I've already notified the PTB's |
| 05:15.43 | CrazyTux[m]1 | [TK]D-Fender: and any of the asterisk flags im trying arent working |
| 05:15.48 | CrazyTux[m]1 | s/arent/aren't/ |
| 05:17.17 | vegbox | I wanted to start a business that offers unlimited calls to vietnam, but how would do that on the cheap. I would have to get a server and adapters in vietnam. Have customers connect to a tunnel here in the USA to server in vietnam. Then from there use the local lines to translate the calls. |
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| 05:19.51 | toughmarketing | Hey guys I am using odbc with mysql and storing the extensions in my database in the format of: id,context,exten,priority,app,appdata and this is working great! The only issue is at the top of my context for my ivr I have include => ivr1-day,09:00-16:59,mon-fri,*,* and if it is during those hours and days it goes to ivr1-day context... Is there a way I can include this in the database as well? |
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| 05:33.27 | [TK]D-Fender | checkout time, later all |
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| 06:45.41 | *** join/#asterisk orangeservice (n=onepoint@118.142.4.226) |
| 06:47.19 | orangeservice | hello all - quick CDR question "${CDR(channel)}" returns "SIP/blucows_hk-08f22350" - is the last bit of hex the actual number that was called? (I am trying to get a list of destination telephone numbers out of CDR) |
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| 06:57.35 | creativx | orangeservice: no |
| 07:00.39 | orangeservice | anybody know an easy to extract the number dialed in a channel from the logs? |
| 07:01.29 | mbrevda | orangeservice: its not hex, most likely a randome id |
| 07:02.12 | creativx | orangeservice: if its not part of the logs columns.. then probably not easy |
| 07:06.33 | *** join/#asterisk icyValk77 (n=icyValk7@gateway.ash.thebunker.net) |
| 07:07.33 | pa | hi |
| 07:07.45 | pa | does anyone have a thomson telecom 3s55 voip phone? |
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| 07:34.26 | *** join/#asterisk trymi1 (n=please@dip5-237.myantel.net.mm) |
| 07:34.58 | trymi1 | hi all |
| 07:35.17 | trymi1 | how to asterisk to Mediant 2000? |
| 07:43.49 | *** join/#asterisk KermitTheFragger (n=KermitTh@118-197.bbned.dsl.internl.net) |
| 07:44.50 | trymi1 | hi all |
| 07:44.53 | trymi1 | there? |
| 07:45.39 | trymi1 | hello |
| 07:45.44 | trymi1 | everybody there? |
| 07:47.14 | *** join/#asterisk Iamnacho (i=Iamnacho@ip174-70-137-120.ks.ks.cox.net) |
| 07:50.46 | *** join/#asterisk trymi1 (n=please@dip5-237.myantel.net.mm) |
| 07:50.53 | trymi1 | hi all |
| 07:50.54 | *** part/#asterisk orangeservice (n=onepoint@118.142.4.226) |
| 07:52.19 | trymi1 | hello |
| 08:05.53 | *** join/#asterisk war9407 (i=war@liquidswords.org) |
| 08:19.35 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-16fa6484432d0a55) |
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| 08:56.06 | *** join/#asterisk xrmx__ (n=rm@87.2.254.129) [NETSPLIT VICTIM] |
| 08:56.06 | *** join/#asterisk reg (n=reg@2a01:240:fe29:1:0:0:101:dead) [NETSPLIT VICTIM] |
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| 08:56.06 | *** join/#asterisk hardwire (n=hardwire@216-67-98-253.static.acsalaska.net) [NETSPLIT VICTIM] |
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| 08:56.07 | *** join/#asterisk seanbright (n=sean@asterisk/contributor-and-bug-marshal/seanbright) |
| 08:56.07 | *** mode/#asterisk [+oo angler Deeewayne] by irc.freenode.net |
| 08:59.05 | *** join/#asterisk devyll (n=paul@89.36.24.2) |
| 08:59.31 | devyll | is there anyway to adjusting the music on hold volume ? (without editing the wav files) |
| 09:01.05 | *** join/#asterisk aurax (n=aurax@bzq-179-76-199.static.bezeqint.net) |
| 09:01.27 | aurax | hi folks |
| 09:01.40 | aurax | short question, can canreinvite=yes may cause one way audio ? |
| 09:03.37 | *** join/#asterisk setunado (n=fabien@setuns.fr.nf) |
| 09:03.47 | *** join/#asterisk Iamnach0 (i=Iamnacho@ip174-70-137-120.ks.ks.cox.net) |
| 09:15.20 | *** join/#asterisk yahh (n=root@122.169.93.71) |
| 09:15.41 | yahh | <PROTECTED> |
| 09:15.53 | yahh | I am having some issue with RTP |
| 09:20.22 | kaldemar | aurax: yes, depending on network setup |
| 09:26.41 | tzafrir_laptop | aurax, it can change the way rtp flows, so the answer is: "yes, it can" |
| 09:27.01 | aurax | hmm, because i have random one way audio issue (on outgoing audio) |
| 09:27.50 | *** join/#asterisk skirmisha (n=asd@79-100-41-71.btc-net.bg) |
| 09:28.01 | yahh | well i am seding garbage RTP packet from asterisk1 to asterisk2 in between of Actual RTP flow |
| 09:28.12 | aurax | my nat is very simple, i configured nat with externip=... localnet=... and port forwarding (dnat) 5060-65384 on my SIP IP to my asterisk server. |
| 09:28.47 | skirmisha | guys |
| 09:28.58 | skirmisha | why insecure=invite is not working correctly |
| 09:29.16 | aurax | btw, i'm using 1.4.25 |
| 09:29.33 | skirmisha | could it be something related to asterisk realtime? |
| 09:29.38 | yahh | not sure about insecure=invite |
| 09:29.48 | yahh | but insecure=port is working for me |
| 09:30.24 | skirmisha | could it be that username has priority over ip |
| 09:30.59 | skirmisha | as call is coming from peer that is set with insecure=invite, but user already exist in system |
| 09:31.50 | skirmisha | so it just ingnore it |
| 09:37.13 | *** join/#asterisk MrNaz (n=mrnaz@ppp118-208-191-191.lns10.mel4.internode.on.net) |
| 09:42.05 | *** join/#asterisk frk2 (n=frk2@zivios/member/fkhan) |
| 09:42.50 | skirmisha | any ideas |
| 09:50.47 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 09:52.14 | tzafrir_laptop | skirmisha, how can you tell it is not working properly? |
| 09:54.26 | skirmisha | ok i set on my peer insecure invite |
| 09:54.34 | skirmisha | and asterisk still wants to auth the invite |
| 09:54.41 | skirmisha | and send 407 first |
| 09:55.05 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-74cb29f838ef4bd2) |
| 09:55.33 | yahh | If you do not set "secret=" |
| 09:55.52 | yahh | then it wont send 407 |
| 09:59.00 | yahh | skirmisha are you there? |
| 09:59.07 | skirmisha | yes |
| 09:59.25 | skirmisha | there is no secret |
| 09:59.31 | skirmisha | i have host |
| 09:59.34 | skirmisha | insecure |
| 09:59.36 | skirmisha | context |
| 09:59.44 | skirmisha | and that is |
| 10:00.13 | skirmisha | and still i see asterisk answer with 407 first |
| 10:01.09 | yahh | ohh |
| 10:01.59 | skirmisha | i tried everything and i think that this option does not work properly |
| 10:20.55 | *** join/#asterisk qdk (n=qdk@81.7.168.130) |
| 10:21.56 | *** join/#asterisk proute (n=AnthonyC@LMontsouris-152-63-16-150.w217-128.abo.wanadoo.fr) |
| 10:22.23 | proute | Hello |
| 10:23.32 | proute | I work on * 1.4.25. Sometimes, When I have a call with somebody, I can hear "beep" like I push a button.... but I don't push button.... |
| 10:24.03 | proute | I think that is a dtmf problem |
| 10:24.15 | proute | my setting about dtmf is rfc2833 |
| 10:24.41 | proute | Does anyone have already meet this problem? |
| 10:24.49 | proute | and How can I fixe it? |
| 10:24.51 | proute | thanks |
| 10:25.38 | proute | This problem appear, with voip (sip) via Internet or with PRI or isdn connection |
| 10:27.37 | *** join/#asterisk ice_croft (n=nolan@213.132.86.246) |
| 10:30.25 | *** join/#asterisk tamiel (n=tamiel@213.30.183.226) |
| 10:31.25 | *** join/#asterisk jgoo (n=r3rman@94.70.56.43) |
| 10:31.40 | aurax | I'm having this weird one way audio again, an anyone assist? |
| 10:31.50 | jgoo | can I get a call group to call all phones in the list AND let them all answer / conference, rather than hangup the ones that aren't first to answeR? |
| 10:32.34 | kaldemar | jgoo: what version of asterisk are you using? |
| 10:32.36 | jgoo | I also have a question similar to the esteemed gentleman above |
| 10:33.58 | jgoo | trixbox1*CLI> core show version |
| 10:33.58 | jgoo | Asterisk 1.4.22-2 RPM by vc-rpms@voipconsulting.nl built by root @ revisor.trixbox.com on a i686 running Linux on 2008-12-04 05:36:44 UTC |
| 10:33.58 | jgoo | trixbox1*CLI> core show version |
| 10:33.58 | jgoo | Asterisk 1.4.22-2 RPM by vc-rpms@voipconsulting.nl built by root @ revisor.trixbox.com on a i686 running Linux on 2008-12-04 05:36:44 UTC |
| 10:33.59 | jgoo | trixbox1*CLI> core show version |
| 10:34.01 | jgoo | Asterisk 1.4.22-2 RPM by vc-rpms@voipconsulting.nl built by root @ revisor.trixbox.com on a i686 running Linux on 2008-12-04 05:36:44 UTC |
| 10:34.04 | jgoo | trixbox1*CLI> core show version |
| 10:34.06 | jgoo | Asterisk 1.4.22-2 RPM by vc-rpms@voipconsulting.nl built by root @ revisor.trixbox.com on a i686 running Linux on 2008-12-04 05:36:44 UTC |
| 10:34.12 | jgoo | 1.4.22 |
| 10:34.14 | jgoo | ffffffffffs fucking lag |
| 10:34.16 | jgoo | lol |
| 10:34.48 | kaldemar | core show application Page <-- use option d for two-way audio |
| 10:36.15 | *** join/#asterisk Router222 (n=CK@212.98.141.199) |
| 10:36.18 | Router222 | hi ppl |
| 10:36.42 | Router222 | i an trying to load module res_fax.so |
| 10:36.56 | Router222 | but its fails with Error loading module 'res_fax.so': /usr/lib/asterisk/modules/res_fax.so: undefined symbol: ao2_lock |
| 10:38.46 | *** join/#asterisk pif (n=ldm@zenon.apartia.fr) |
| 10:40.06 | Router222 | what do it means |
| 10:42.17 | Chainsaw | Router222: It suggests that chan_iax2.so is required but not loaded. |
| 10:42.55 | Chainsaw | Router222: Make sure you haven't prevented it from loading. If you haven't, try to preload it. |
| 10:45.26 | *** join/#asterisk pa (n=pa@unaffiliated/pa) |
| 10:47.22 | Router222 | Chainsaw i will check |
| 10:48.52 | pa | why my asterisk was working. now i rebooted the server and i keep getting |
| 10:48.55 | pa | app_dial.c:1210 dial_exec_full: Unable to create channel of type 'Zap' (cause 66 - Channel not implemented) |
| 10:50.06 | kaldemar | pa: you don't have chan_zap.so loaded for some reason. |
| 10:50.14 | aurax | I don't get it... for 10 minutes i have no outgoing audio and then suddently it's back to normal... like nothing happend... |
| 10:50.40 | pa | mmh.. |
| 10:50.47 | pa | i loaded zaphfc and zaptel modules |
| 10:50.54 | pa | am i missing something else? |
| 10:52.45 | kaldemar | yes, chan_zap. the asterisk module. |
| 10:54.00 | pa | ah ok |
| 10:54.07 | pa | but that is not a kernel module |
| 10:55.37 | aurax | can anyone help me debug and find the cause for this problem ? |
| 10:57.36 | pa | i get this: |
| 10:57.39 | pa | [Jul 20 12:56:52] WARNING[12362]: chan_zap.c:1082 zt_open: Unable to specify channel 1: No such device or address |
| 10:57.46 | pa | mmmh.. |
| 10:58.06 | kaldemar | pa: it is an asterisk module. |
| 10:59.06 | *** join/#asterisk ccesario (n=ccesario@189-19-9-100.dsl.telesp.net.br) |
| 10:59.31 | pa | mby my hardware is fucked up? |
| 11:00.56 | pa | mmh.. |
| 11:01.18 | pa | but when i modprobe zaphfc, i get the card recognized, in /var/LOG/messages |
| 11:01.23 | pa | maybe it's misconfigured? |
| 11:03.00 | kaldemar | probably. compare the contents of /proc/zaptel/* (those files tell you how the card(s) are configured) to zapata.conf channel definitions. |
| 11:03.28 | *** join/#asterisk lou_gr (n=lou@static062038221130.dsl.hol.gr) |
| 11:03.31 | pa | ok thank |
| 11:03.32 | pa | s |
| 11:04.23 | pa | kaldemar, it seems to be configured in master mode |
| 11:04.36 | pa | and i modprobed zaphfc, and dmesg showed: |
| 11:04.37 | pa | [ 3948.625627] zaphfc: Card 0 configured for TE mode |
| 11:04.37 | pa | [ 3948.625632] zaphfc: Card 0 configured for master mode |
| 11:04.39 | pa | so |
| 11:04.45 | pa | seemingly it changed the mode |
| 11:05.21 | *** join/#asterisk mikkel (n=mikkel@84-238-113-66.u.parknet.dk) |
| 11:15.48 | aurax | does rtp.conf relates to incoming or outgoing rtp ? |
| 11:18.21 | kaldemar | pa: the opposite of TE mode would be NT mode, not master mode. channel numbering seems to be your problem, not the mode. |
| 11:18.53 | pa | i see.. |
| 11:21.53 | pa | kaldemar, ok. apparently it now works. i havent changed anything tho.. |
| 11:22.14 | pa | is it possible that channel numbering varies randomly when i load the module? |
| 11:35.20 | *** join/#asterisk skirmisha (n=asd@79-100-41-71.btc-net.bg) |
| 11:35.32 | skirmisha | can someone explain me how insecure works |
| 11:36.18 | kaldemar | pa: no, it takes the numbering for zaptel from /etc/zaptel.conf. |
| 11:38.27 | *** join/#asterisk Victor_Yure (n=victor@unaffiliated/victoryure/x-837844) |
| 11:41.31 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 11:52.25 | pa | kaldemar, then there must be something strange if just rerunning asterisk makes it work |
| 11:59.12 | redax | hi, |
| 11:59.25 | redax | where can I find libpri-1.6.* ? |
| 12:00.30 | *** join/#asterisk Alfio (n=amunoz@75.112.88.200.m.sta.codetel.net.do) |
| 12:00.48 | kaii | http://www.asterisk.org/downloads |
| 12:01.10 | beek | It would be quite a trick as libpri is at 1.5.10.1 |
| 12:01.19 | beek | s/1.5/1.4/ |
| 12:01.55 | kaii | "We are not planning on releasing a 1.6 version of libpri, but instead you need to use the recently released version 1.4.4 of libpri with Asterisk 1.6. There were not big enough changes for Asterisk 1.6 to require a major ABI change release of libpri, so instead most of the 1.6 specific functions were back ported to the 1.4 branch of libpri" |
| 12:04.06 | redax | uh.. |
| 12:04.40 | redax | I've interconnected an * 1.6.1.1 using digium te122 (E1) and a siemens Hicom 300 |
| 12:04.50 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 12:05.06 | beek | [TK]D-Fender is in the room. |
| 12:05.26 | beek | notes the sounds of thunder |
| 12:05.47 | [TK]D-Fender | Can you hear, can you hear the thunder! |
| 12:06.12 | beek | [TK]D-Fender: Good morning. |
| 12:06.21 | redax | hicom calls -> asterisk OK |
| 12:06.28 | [TK]D-Fender | beek: y0 |
| 12:06.30 | redax | asterisk -> hicom [ no sound] |
| 12:06.45 | redax | if I press a '4' on the telephone the sound arrives :D |
| 12:06.49 | redax | what the hell is it? |
| 12:07.10 | *** join/#asterisk Dksaarth (n=Dks@dsl-145-216-83.telkomadsl.co.za) |
| 12:07.25 | redax | [TK]D-Fender: do you rememeber this hicom vs asterisk problem from the last week, I had? |
| 12:07.40 | [TK]D-Fender | rebOnly that you had one |
| 12:07.49 | Dksaarth | Hi guys - I am currently developing a sip phone, and am running into the problem of asterisk stopping to send rtp packets to my sip phone ? |
| 12:08.01 | [TK]D-Fender | redThere is an RC that addresses a "no audio until DTMF bug" <---- |
| 12:08.02 | redax | seems like when I press the digit '4' nothing transmitted at pri debug level, but the voice goes ok |
| 12:08.08 | [TK]D-Fender | redax: There is an RC that addresses a "no audio until DTMF bug" <---- |
| 12:08.22 | redax | hm. |
| 12:08.33 | [TK]D-Fender | Dksaarth: and that's a QUESTION? |
| 12:08.59 | Dksaarth | Okay, my question would be what is causing asterisk to stop sending rtp packets ? |
| 12:09.11 | [TK]D-Fender | Dksaarth: MoH <- |
| 12:09.40 | [TK]D-Fender | Dksaarth: Or more specifically.... to the person PUTTING someone on hold. |
| 12:10.15 | redax | [TK]D-Fender: hm. what does 'RC' really means btw? |
| 12:10.32 | [TK]D-Fender | redax: Release Candidate |
| 12:10.45 | redax | ah. |
| 12:10.47 | Dksaarth | I call my test phone phone (A) from another soft phone (phone B). A answers immediatly, and I can hear the audio from A to B, but the audio from B to A never arrives (only between 37 and 64 packets are sent) |
| 12:11.07 | Dksaarth | Nobody has put anybody on hold yet ? |
| 12:11.38 | [TK]D-Fender | Dksaarth: That isn't RTP stopping, that's RTP **FAILING** to start in the first place. |
| 12:12.02 | Dksaarth | okay - I saw a couple of rtp packets so I thought it was started okay |
| 12:12.31 | [TK]D-Fender | Dksaarth: And don't end statements with a "?". And you haven't described the working envirenment or pastebin-d a failed call with SIP dubeg for us to examine. |
| 12:12.33 | [TK]D-Fender | ~pb |
| 12:12.34 | 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/, or apt-get install pastebinit |
| 12:12.35 | [TK]D-Fender | ^^^^^^^^ |
| 12:12.55 | redax | [TK]D-Fender: found a bug #15420 which is addressed for * v1.4.25.1, but I'm using * v1.6.1.1 |
| 12:16.31 | aurax | [TK]D-Fender, mind helping me out with my random outgoing oneway audio problem ?!, i got all sip & rtp debugs. |
| 12:17.53 | *** join/#asterisk propellerhead (n=yogurt2u@190.136.235.36) |
| 12:18.57 | [TK]D-Fender | redax: Below others with possibly related issues, that I'm very sure this will fix, including my bug 0015389, but this bug report has progressed further. Fixed 0015420 1.4.25.1 asterisk -> Nortel reported by scottbmilne, tested by scottbmilne Fixed 0015416 1.4.25.1 asterisk -> Avaya reported by avinoash, tested by avinoash Fixed 0015389 1.6.1.0 asterisk -> Fujitsu, reported bu... |
| 12:18.59 | [TK]D-Fender | ...alecdavis, tested by alecdavis 0015205 1.6.1.0 awaiting response. This patch is confirmed to fix the NoAudio problem, by both ScottMilne and myself. |
| 12:19.01 | proute | Re-Hello |
| 12:19.17 | [TK]D-Fender | redax: applicable to 1.6.1 |
| 12:19.35 | proute | I work on * 1.4.25. Sometimes, When I have a call with somebody, I can hear "beep" like I pressing a button.... but I don't press button.... |
| 12:19.43 | [TK]D-Fender | aurax: ... |
| 12:19.45 | [TK]D-Fender | ~pb |
| 12:19.46 | 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/, or apt-get install pastebinit |
| 12:20.12 | aurax | sure |
| 12:20.15 | aurax | http://pastebin.com/d273499f |
| 12:20.17 | proute | I think that is a dtmf problem, my setting for dtmf is rfc 2833 and this problem appear randomly |
| 12:20.25 | redax | [TK]D-Fender: Thanks. |
| 12:20.33 | aurax | [TK]D-Fender, loosing my hair here.. HELP! |
| 12:20.52 | [TK]D-Fender | aurax: You enabled SIP debug too late. do another call. |
| 12:20.56 | proute | and How can I fixe it? (I think that is dtmf problem) But Why I hear a press touch without to push a touch? |
| 12:21.02 | *** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
| 12:21.08 | aurax | i did called after enabling sip debug |
| 12:21.09 | proute | thanks |
| 12:21.15 | aurax | i got another debug, i will upload it to pb |
| 12:21.42 | aurax | http://pastebin.com/d5b527207 |
| 12:21.47 | aurax | think that the second one is better. |
| 12:22.11 | *** join/#asterisk Dovid (n=annon@tony09-118-62.inter.net.il) |
| 12:23.19 | *** join/#asterisk eliyahu (n=eliyahu@77.126.64.188) |
| 12:23.31 | aurax | sup dovid |
| 12:26.25 | *** join/#asterisk malcolmd (n=malcolmd@pdpc/sponsor/digium/malcolmd) |
| 12:30.12 | Dksaarth | [TK]D-Fender: I have pasted the sip debug log at http://pastebin.com/m881588 - it is (Asterisk 1.4.22-4 RPM) |
| 12:30.18 | Dksaarth | Is there anything else I can tell you that might help ? |
| 12:30.23 | [TK]D-Fender | aurax: And there are 2 sides to this call and you only bothered showing me the second. And no configs. |
| 12:31.01 | Dksaarth | (the RTP to destination port 9898 is the one that doesn't arrive) |
| 12:31.27 | [TK]D-Fender | Dksaarth: Do another call. No RTP debug, just verbose 10, SIP debug enabled. |
| 12:31.34 | [TK]D-Fender | Dksaarth: No core debug |
| 12:31.45 | [TK]D-Fender | Dksaarth: And check your firewalls |
| 12:35.20 | aurax | i don't have access to the other side |
| 12:35.23 | aurax | it's my provider |
| 12:37.00 | aurax | [TK]D-Fender - my isp conf http://pastebin.com/m751bf4ba |
| 12:47.21 | [TK]D-Fender | auurYes you di.. the end that is STARTING THE CALL |
| 12:47.24 | [TK]D-Fender | do* |
| 12:47.58 | *** join/#asterisk malaiwah (n=mbelleau@64.47.115.5) |
| 12:48.10 | [TK]D-Fender | aurax: And providers should never be "nat=yes" |
| 12:50.18 | Dksaarth | [TK]D-Fender: I turned off my firewall, did core set verbose 0, then asterisk -vvvvvvvvvvr (to set it to 10) - new log is at http://pastebin.com/db6c8f9d |
| 12:50.40 | Dksaarth | I'm very new to asterisk, so maybe I'm not setting the debug level's correctly. |
| 12:52.10 | [TK]D-Fender | Dksaarth: ENTIRE CALL, disable your core dbug "debug 0" |
| 12:52.15 | [TK]D-Fender | "set debug 0" |
| 12:53.29 | *** join/#asterisk kamanashisroy (n=kamanash@119.30.36.83) |
| 12:57.22 | Dksaarth | I did core set debug 0 |
| 12:57.26 | Dksaarth | and then the log is at http://pastebin.com/d1ccc807d |
| 12:57.36 | Dksaarth | from the invite to the bye/200 ok |
| 12:57.47 | Dksaarth | Is there something else I am leaving out ? |
| 12:57.52 | aurax | [TK]D-Fender even if the server is behind NAT? |
| 12:59.27 | [TK]D-Fender | aurThe other side should take care of itself <- |
| 12:59.32 | *** join/#asterisk leifmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 12:59.32 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 13:00.34 | aurax | ah ok |
| 13:02.02 | [TK]D-Fender | Dksaarth: [Jul 20 14:54:12] VERBOSE[2790] logger.c: Retransmitting #1 (NAT) to 10.10.10.38:5060: <-- shouldn't be NAT on a private IP, and I said THE ENTIRE DAMN CALL.that includes the CALLING PHONE's debug. and your configs. |
| 13:04.39 | ice_croft | hi mates |
| 13:04.53 | ice_croft | anybody usin thomson st 2030 ipphone? |
| 13:04.55 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 13:05.52 | *** join/#asterisk KavanS (n=KavanS@71.117.242.28) |
| 13:08.04 | *** join/#asterisk s14ck (n=s14ck@ccscliente156.ifxnetworks.net.ve) |
| 13:08.36 | *** join/#asterisk afink (n=afink@204.26.87.226) |
| 13:09.20 | *** join/#asterisk Iamnacho (i=Iamnacho@ip174-70-137-120.ks.ks.cox.net) |
| 13:10.27 | *** join/#asterisk DelphiWorld (n=Miranda@41.201.71.63) |
| 13:10.32 | DelphiWorld | hello |
| 13:11.16 | DelphiWorld | Please give me the SVN Checkout URL to checkout asterisk 1.6.X |
| 13:11.17 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 13:11.21 | DelphiWorld | latest please |
| 13:11.29 | Dksaarth | [TK]D-Fender - Would the calling phone's debug be in the asterisk log file ? That is where i am pulling this information from, and hence thought I was giving you the entire call - the phone I am using is sitting on my desk, ip phone - I can give you a wireshark dump of the sip traffic from my soft phone on my laptop if required. |
| 13:12.56 | [TK]D-Fender | Dksaarth: http://pastebin.com/d1ccc807d <-- I want to see the debug from with SIP/100 started the damn call. |
| 13:13.28 | [TK]D-Fender | Dksaarth: YES Asterisk sees this. Please wake up. this is PART of the equation. You are only focussing on * -> YOUR new SIP phone. |
| 13:13.39 | [TK]D-Fender | Dksaarth: The OTHER phone may be part of the problem. |
| 13:14.42 | afink | Is there any way to show what time asterisk thinks it is? |
| 13:15.29 | DelphiWorld | [TK]D-Fender: SVN URL for asterisk 1.6.X please |
| 13:16.23 | afink | I am having a problem with the GotoIfTime application. http://pastebin.com/m7ee98019 It is 8:17 where I am and it still goes on to the next priority |
| 13:16.28 | [TK]D-Fender | DelphiWorld: Go search asterisk.org yourself, and don't target pople for support like that. |
| 13:17.34 | kaldemar | DelphiWorld: http://svn.digium.com/svn/asterisk/ |
| 13:17.34 | Dksaarth | As far as I understand, that log is from when the SIP/100 starts the call - it has the invite from 100 (normal phone) to 150 (test phone). there is nothing above it that is related to these two phones except for the register and option messages. |
| 13:17.53 | *** join/#asterisk x86 (n=porteb1@p3m/member/x86) |
| 13:17.55 | [TK]D-Fender | Dksaarth: No, it isn't Where si the SIP debug for the call as it arrives to *? |
| 13:18.08 | DelphiWorld | kaldemar: thanks |
| 13:18.25 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 13:20.25 | *** join/#asterisk eliyahu (n=eliyahu@77.126.64.188) |
| 13:21.57 | Dksaarth | Does it matter that these are both local phones behind asterisk ? as far as I can see there is no INVITE message above what I have pasted - Executing [150@from-internal:1] Macro("SIP/100-08a39478", "exten-vm|novm|150") in new stack seems to be the start from what I understand |
| 13:22.07 | Dksaarth | all on one network. |
| 13:23.23 | [TK]D-Fender | Dksaarth: then you didn't enable GLOBAL SIP debug and only targeted your peer. and I STILL don't see your configs. |
| 13:24.01 | Dksaarth | True - that is my mistake: i only am debugging for the one ip. My apologies. |
| 13:24.24 | Dksaarth | Which config files are relevant for me to pastebin ? |
| 13:25.28 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
| 13:25.46 | *** join/#asterisk ricko73 (n=dhartman@wilug/newlug/ricko73) |
| 13:25.50 | [TK]D-Fender | Dksaarth: Just get the call debug to start |
| 13:27.18 | *** join/#asterisk coppice (n=chatzill@204.196.17.210.dyn.pacific.net.hk) |
| 13:36.34 | afink | would anybody please be willing to take a look at these GotoIfTime statements and tell me if there is something that I am missing? http://pastebin.com/m31a44d93 |
| 13:37.12 | *** join/#asterisk Anth8708 (n=Anth8708@client105.jdcc.edu) |
| 13:37.54 | Anth8708 | morning everybody |
| 13:37.55 | Anth8708 | :) |
| 13:40.17 | Anth8708 | I have a question that I think is fairly simple and common. We currently have a Nortel CS1000M. I need to setup secretary/boss relationships in asterisk. |
| 13:40.35 | Anth8708 | Example: Boss is ext 1000, Secretary is 1001. I need to have 1000 appear as a separate line on Secretary's phone (fake label is fine, easy enough). |
| 13:41.54 | Anth8708 | Both can ring at the same time (easy enough). Secretary needs to be able to put 1000 on hold and have Boss pick up ext 1000 to get the call. Should I be attempting to use SLA, meetme or something else? |
| 13:43.19 | *** join/#asterisk gigman (n=ian@96-32-123-171.static.oxfr.ma.charter.com) |
| 13:43.41 | gigman | Hello, is anyone in here? |
| 13:43.58 | _ShrikE | Anth8708: call park? |
| 13:44.51 | Anth8708 | _ShrikE: hmm. that makes sense I suppose. Can there be a visible indicator that a call is parked? |
| 13:45.03 | *** join/#asterisk SirThomas (n=tomc@mail.kendeco.com) |
| 13:45.25 | Dksaarth | [TK]D-Fender: I am having trouble seperating out the sip we are interested in from the other traffic on this network - there are a number of phones in use here. That is why I was only logging traffic from ip. Do you know of a way to use sip set debug ip with more than one ip address ? |
| 13:45.39 | [TK]D-Fender | Dksaarth: There is none. |
| 13:46.02 | Dksaarth | ouch okay thanks |
| 13:46.07 | [TK]D-Fender | Dksaarth: Test another soft-phone on that PC. If that works, then its yours |
| 13:46.08 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 13:47.05 | gigman | Hi everyone, Im a pretty big novice when it comes to * but I am having an issue starting it |
| 13:47.07 | Dksaarth | I have tested with a soft phone on the same pc as my test phone, same result. Double checking it now. |
| 13:47.32 | _ShrikE | Anth8708: I have never done it personally, but there is a good bit written up on subscribing to parking lots. |
| 13:47.44 | gigman | whenever I attempt to start * I get "unable to connect to remote asterisk" |
| 13:47.49 | gigman | can anyone help me? |
| 13:48.06 | Anth8708 | _ShrikE: rgr. Thanks. Trying to narrow down my reading a bit. Thanks again. |
| 13:48.17 | Anth8708 | gigman: are you using the command asterisk -r? |
| 13:48.22 | _ShrikE | Anth8708: np |
| 13:48.55 | gigman | @ Anth8708 Yes, I am, and Im still getting this error, any ideas? |
| 13:49.07 | Anth8708 | gigman: That reconnects to a running * console. Try running the command "asterisk" (without quotes) first, THEN running asterisk -r |
| 13:49.37 | Anth8708 | gigman: The -r is "reconnect to console" basically |
| 13:49.40 | gigman | @ Anth8708 I just tried that... no dice.. |
| 13:49.53 | gigman | oh, good to know |
| 13:50.05 | Anth8708 | gigman: What happens when you run the command asterisk without the -r? What output do you get? |
| 13:50.32 | gigman | the full error is "Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) |
| 13:50.45 | yahh | hello.. |
| 13:51.21 | gigman | Anth8708, if I run just asterisk the console just returns a new line |
| 13:51.31 | gigman | anything else I should try? |
| 13:51.47 | yahh | I am having some issue with RTP |
| 13:51.56 | Anth8708 | gigman: First, check to see if * is running. Some variation of ps ax |grep asterisk |
| 13:52.13 | yahh | Normal rtp is being sent from asterisk1 to asterisk2 |
| 13:52.27 | yahh | <PROTECTED> |
| 13:52.41 | yahh | from asterisk1 to asterisk2 |
| 13:53.12 | yahh | that's why voice on asterisk2 's connnected phone is not coming proper |
| 13:53.21 | *** join/#asterisk ariel_ (n=chatzill@63.214.236.169) |
| 13:53.24 | gigman | Anth8708, grep asterisk is returning nothing |
| 13:53.36 | Anth8708 | gigman: WHen you run asterisk without the -r, you should be getting no output and just the next line. This is what starts *. After you run that command, you should be able to run asterisk -r and get a console. However, if "asterisk" without -r is having a problem, then your "asterisk -r" will continue having problems |
| 13:53.41 | Anth8708 | gigman: Try asterisk -c |
| 13:53.46 | *** join/#asterisk jy (n=jy@cm179.zeta200.maxonline.com.sg) |
| 13:54.26 | Anth8708 | gigman: That will start * in the foreground I think .. and you will be in the console. |
| 13:54.54 | gigman | ok I just ran that "asterisk -c" |
| 13:55.01 | gigman | it did not return an error |
| 13:55.06 | yahh | can someone help in this |
| 13:55.21 | gigman | is it now started? |
| 13:55.53 | gigman | Anth8708, if I run asterisk -r after I run asterisk -c I still get this same error |
| 13:56.09 | jy | Hi guys, anyone has any idea how to do dual forking of calls on Asterisk? (Each extension has a soft client + hard ip phone registered) |
| 13:56.25 | ariel_ | gigman: what is the error your getting |
| 13:56.35 | Anth8708 | gigman: If you run asterisk -c and don't get straight to the console, you likely have some sort of configuration problem. |
| 13:56.37 | ariel_ | yahh: did you ask a quetion |
| 13:56.46 | yahh | yes.. |
| 13:56.55 | yahh | let me come again |
| 13:57.05 | gigman | ariel_, Ithe full error is "Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl |
| 13:57.12 | yahh | <PROTECTED> |
| 13:57.14 | Anth8708 | gigman: is this a new installation? do you know what version you have installed? |
| 13:57.22 | yahh | Normal rtp is being sent from asterisk1 to asterisk2 |
| 13:57.24 | gigman | yes, I have the latest version |
| 13:57.30 | ariel_ | gigman: that means it 's not running |
| 13:57.33 | yahh | now i have modified and sent some garbage udp packets in between |
| 13:57.40 | yahh | from asterisk1 to asterisk2 |
| 13:57.53 | yahh | that's why voice on asterisk2 's connnected phone is not coming proper |
| 13:57.54 | gigman | there is a file in that dir however |
| 13:58.02 | ariel_ | gigman: do asterisk -vvvgc and see what error it fails on |
| 13:58.17 | yahh | asterisk should ignore those dummy packets |
| 13:58.29 | yahh | but it is not happning |
| 13:59.59 | yahh | how can i ignore those packets? |
| 14:00.35 | gigman | ariel_, I just lost my connection.. But I ran the command you said, no errors, just multiple entries of Managed Service Stared |
| 14:00.52 | Dksaarth | [TK]D-Fender - I am still sifting apart his call log, but it doesn't seem to be a problem here as it is the same with other phones - i ran a wireshark trace, and found that there are actually 5 rtp legs instead of 4. |
| 14:00.53 | Dksaarth | http://pastebin.com/d2e193f99 |
| 14:01.26 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 14:01.49 | gigman | ariel_, is this where I would find errors? In that command? |
| 14:01.56 | Alfio | afink |
| 14:01.58 | [TK]D-Fender | Dksaarth: This is beyond my ability to assist you further, sorry. |
| 14:02.54 | afink | Alfio: yes? |
| 14:04.08 | Alfio | why do you have a "_" in the extensions |
| 14:04.08 | Alfio | ? |
| 14:04.36 | aurax | Can i limit the rtp port range that i'm sending with to my provider? |
| 14:04.47 | aurax | like rtp range per sip trunk |
| 14:04.48 | *** join/#asterisk putnopvut (n=putnopvu@asterisk/master-of-queues/mmichelson) |
| 14:04.48 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:04.59 | gigman | ariel_, Im back in my server, any more ideas? |
| 14:05.02 | Alfio | aurax YES |
| 14:05.06 | aurax | Alfio, how ? |
| 14:05.08 | Alfio | sorry about caps |
| 14:05.12 | aurax | np :) |
| 14:05.36 | [TK]D-Fender | aurax: No. |
| 14:05.42 | *** join/#asterisk moy (n=moy@bas1-unionville55-1177733615.dsl.bell.ca) |
| 14:05.49 | [TK]D-Fender | AFAIK |
| 14:06.04 | aurax | so what happend if i have two providers one that uses 10k-20k and another one that uses 16k-65k ? |
| 14:06.09 | Alfio | i think you could in rtp.conf |
| 14:06.21 | Alfio | dosent work i really never try it |
| 14:06.23 | Alfio | ? |
| 14:07.08 | [TK]D-Fender | aurax: Thats not how it works |
| 14:07.29 | [TK]D-Fender | aurax: * has no control over what port THEY use, only of what * itself uses for inbound |
| 14:07.53 | aurax | yes of course.. |
| 14:08.09 | aurax | i just thuoght that there's might be a chance that i'm sending data to provider on wrong port |
| 14:11.08 | jplank | is there any harm in moving back from dadhi to zaptel? |
| 14:11.56 | jplank | essentially "downgrading" back to zaptel |
| 14:13.14 | *** join/#asterisk moy (n=moy@bas1-unionville55-1177733615.dsl.bell.ca) |
| 14:15.48 | afink | would you guys/girls take a look at this and tell me if I have something screwed up? http://pastebin.com/m43c5bfdf |
| 14:16.39 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 14:18.32 | paulgm | Last try :) Anyone have any config files for Cisco 7941G-GE phones? |
| 14:19.09 | [TK]D-Fender | afink: That isn't your problem. |
| 14:19.40 | *** join/#asterisk ehsjoar (n=ehsjoar@c-24-9-91-203.hsd1.co.comcast.net) |
| 14:20.11 | afink | [TK]D-Fender: When I change that one line it works. What else could it be? |
| 14:20.38 | [TK]D-Fender | afink: You show us 1 puny little line. no sense of context, no failed call to examine. Seriosly... |
| 14:20.58 | afink | I showed the entire dialplan earlier. |
| 14:21.17 | [TK]D-Fender | afink: I don't see the failed CALL |
| 14:21.36 | [TK]D-Fender | afink: Who says that dialplan means anything at all? How do I know what * and your call is looking for? |
| 14:21.48 | Kobaz | epic fail |
| 14:22.19 | [TK]D-Fender | Kobaz: Not yet... plenty of room to dig though ;) |
| 14:22.46 | afink | failed call: http://pastebin.com/m377eb925 |
| 14:23.21 | DelphiWorld | thanks, i'm building my asterisk 1.6 now |
| 14:23.23 | Kobaz | heh |
| 14:24.33 | [TK]D-Fender | afink: I sure don't see "This one says 'all circuits busy' when I call" |
| 14:24.59 | DelphiWorld | asterisk 1.6 support SIP over TCP? |
| 14:25.05 | [TK]D-Fender | DelphiWorld: Yes |
| 14:25.14 | afink | [TK]D-Fender: on the phone a recording |
| 14:25.18 | *** join/#asterisk ingenius (n=alektro@186.136.6.218) |
| 14:25.36 | [TK]D-Fender | afink: What do we care about a phone talking to you? We see what * is doing. |
| 14:25.54 | [TK]D-Fender | afink: Your gotoif did not match and you ran out of priorities to execute |
| 14:26.00 | [TK]D-Fender | +time* |
| 14:26.05 | DelphiWorld | finally |
| 14:26.32 | DelphiWorld | and what about GTalk integration? |
| 14:26.45 | afink | [TK]D-Fender: I see what your saying but it should match. Asterisk goes off of system time correct? |
| 14:27.20 | [TK]D-Fender | afink: I also don't see any attempt from you to show your system time, have * output it so you can see, etc |
| 14:28.02 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 14:28.47 | Anth8708 | Hey guys, everytime I call extension 7611 (while it is in use), it rings directly to extension 7611 as a second call on the same line and doesn't ring to 76112? extension config: http://pastebin.com/d12c1218f |
| 14:31.10 | *** join/#asterisk gigman (n=ian@96-32-123-171.static.oxfr.ma.charter.com) |
| 14:32.11 | DelphiWorld | ~pb 9785 |
| 14:32.33 | afink | [TK]D-Fender: -- Executing [3100@default:1] NoOp("DAHDI/1-1", "20090720-073042") in new stack : so my * thinks it is 7:30:42? |
| 14:32.55 | [TK]D-Fender | AnthPriority jumping? Holy shit, this isn't 2004... |
| 14:33.02 | [TK]D-Fender | Anth8708: Priority jumping? Holy shit, this isn't 2004... |
| 14:33.38 | [TK]D-Fender | Anth8708: That was toast in 1.2.... |
| 14:33.53 | paulgm | ok |
| 14:34.01 | Anth8708 | [TK]D-Fender: sorry. The wiki talks about priority jumping. I'll dig into the book then. It should have current info, right? |
| 14:34.04 | *** join/#asterisk Airozi (n=user@200-171-41-207.dsl.telesp.net.br) |
| 14:34.11 | *** join/#asterisk CrazyTux[m] (n=Brandon@216-110-94-230.static.twtelecom.net) |
| 14:34.11 | [TK]D-Fender | AnthBook is for 1.4 |
| 14:34.44 | CrazyTux[m] | [TK]D-Fender: ok so I figured out how to get it working last night, I'm still using MixMonitor, but it seems to not record the entire duration of the call, only about 30-40 seconds, of i.e. 5 minutes or whatever, is there a maxduration setting or default? |
| 14:34.52 | CrazyTux[m] | [TK]D-Fender: if I dont "specify" one |
| 14:34.54 | [TK]D-Fender | Anth8708: And 16.0 replaced it. And you're on 1.6.1. Congratulations on umping 2 major releases |
| 14:35.26 | [TK]D-Fender | CrazyTux[m]: This doesn't sound right... |
| 14:35.38 | Anth8708 | [TK]D-Fender: well, I'm a new install and just reading what I can find:). Wiki doesn't always specific what version the documentation is for and makes for some confusion. |
| 14:35.38 | CrazyTux[m] | [TK]D-Fender: whys that lol |
| 14:35.42 | [TK]D-Fender | DelphiWorld: 1.4 had this.... go look. |
| 14:36.01 | Anth8708 | [TK]D-Fender: Thanks though. I'll keep reading and see what I can find. |
| 14:36.05 | [TK]D-Fender | Anth8708: WIKi has a lot of 1.0 and 1.2 stuff and is seriously outdateed |
| 14:36.19 | [TK]D-Fender | Anth8708: "core show application dial" <- |
| 14:36.31 | Anth8708 | [TK]D-Fender: Rgr, thanks again:). |
| 14:36.42 | [TK]D-Fender | Anth8708: There are some rather obvious channel variables to look at... |
| 14:37.21 | CrazyTux[m] | [TK]D-Fender: perhaps I'm using the wrong application? |
| 14:37.44 | [TK]D-Fender | CrazyTux[m]: It should work. Perhaps you should be showing something. |
| 14:38.01 | Anth8708 | [TK]D-Fender: Got it. Copied out and documented as 1.6 extension basic info. As always, thanks a million. |
| 14:40.41 | *** part/#asterisk DelphiWorld (n=Miranda@41.201.71.63) |
| 14:40.42 | CrazyTux[m] | [TK]D-Fender: http://pastebin.com/maa525c8 |
| 14:41.15 | [TK]D-Fender | CrazyTux[m]: CALL <- |
| 14:42.22 | *** join/#asterisk delphus (n=delphus@unaffiliated/delphus) |
| 14:43.31 | delphus | (cause 34 - Circuit/channel congestion) asterisk 1.4.25.1 / dahdi 2.1.0.2 / pri 1.4.10.1 / sangoma 3.5.4 with a104 euroisdn link status OK any ideas ? |
| 14:43.58 | CrazyTux[m] | [TK]D-Fender: http://pastebin.com/d1042d65b |
| 14:44.01 | afink | thanks [TK]D-Fender |
| 14:46.04 | [TK]D-Fender | CrazyTux[m]: I don't see a bridge notice there. |
| 14:47.35 | CrazyTux[m] | [TK]D-Fender: well its definitely recording? just like 30 seconds of the call however. |
| 14:47.45 | [TK]D-Fender | delphus: Some switches throw back 34 for when the target # is busy. You should be looking at a complete failed call with PRI debug |
| 14:47.49 | CrazyTux[m] | [TK]D-Fender: should not the tg <- bridge the call |
| 14:48.33 | [TK]D-Fender | CrazyTux[m]: I don't see the file either... |
| 14:49.51 | CrazyTux[m] | [TK]D-Fender: ah left that bit out, but there is no bridge |
| 14:50.15 | [TK]D-Fender | CrazyTux[m]: You do get audio and the call seems normal minus the recording? |
| 14:50.16 | CrazyTux[m] | [TK]D-Fender: it does show this: -- Executing [call-record-8005558355@default:2] MixMonitor("SIP/SIP_PROXY-8000a5e0", "/home/storage/PHONE_NUMBER/8005558355_20090720-144704.wav") in new stack |
| 14:50.30 | CrazyTux[m] | [TK]D-Fender: correct, it records both ways... just not the whole duration |
| 14:50.39 | *** join/#asterisk UQlev (n=yuriy@nb11-125.static.cytanet.com.cy) |
| 14:51.24 | [TK]D-Fender | CrazyTux[m]: Also, stop masking things... makes me not trust whats being passed. |
| 14:51.33 | [TK]D-Fender | CrazyTux[m]: Things like ILLEGAL CHARS, etc... |
| 14:51.46 | CrazyTux[m] | [TK]D-Fender: alrighty |
| 14:54.37 | *** join/#asterisk theHub (n=theHub@69.177.93.21) |
| 14:54.37 | Anth8708 | ok, what would the DIALSTATUS be of an idle phone? core show application dial doesn't tell me that.:( More detail: http://pastebin.com/d495727cd |
| 14:54.49 | *** join/#asterisk propellerhead (n=yogurt2u@host36.190-136-235.telecom.net.ar) |
| 14:54.58 | delphus | [TK]D-Fender: thanks, wiil check. |
| 14:56.02 | [TK]D-Fender | Anth8708: there is no DIALSTATUS of an idle phone. This is the result of your CALL |
| 14:56.02 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 14:56.49 | *** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler) |
| 14:58.04 | Anth8708 | [TK]D-Fender - rgr. looking for more documentation on manipulating extensions. Thanks |
| 14:58.30 | *** join/#asterisk Failrar (n=Failrar@coffee.ipv6.goatse.be) |
| 15:00.51 | CrazyTux[m] | [TK]D-Fender: so very odd right --- look like anything I'm doing wrong? |
| 15:01.33 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 15:04.28 | *** join/#asterisk CunningPike (n=CunningP@S01060014bf81366b.vc.shawcable.net) |
| 15:05.14 | [TK]D-Fender | AnthAnd go look at CrazyTux[m] Not sure. test straing Monitor + M |
| 15:05.19 | [TK]D-Fender | CrazyTux[m]: ^^ |
| 15:07.19 | *** join/#asterisk RoyCrowder (n=roy@net-216-37-64-130.in-addr.worldspice.net) |
| 15:08.27 | CrazyTux[m] | [TK]D-Fender: will do |
| 15:10.43 | *** join/#asterisk CunningPike (n=CunningP@204.239.8.97) |
| 15:11.52 | *** join/#asterisk xpot-mobile (n=james@static-66-182-88-85.bbsc.net) |
| 15:13.45 | *** join/#asterisk marv[work] (n=timr@24.214.206.254) |
| 15:13.50 | *** join/#asterisk encbladexp (n=stefan@p5495AACB.dip.t-dialin.net) |
| 15:13.56 | encbladexp | hello |
| 15:14.14 | encbladexp | i have some Problem with Audio Quality and Asterisk :-( |
| 15:14.58 | encbladexp | Example: I call my Extension 50 (Playback demo-echotest) from my Ekiga Phone with GSM Codec » All Fine |
| 15:15.24 | encbladexp | But, if i use PCMA (alaw) Codecs to call this extension the sound qualitiy ist ... horrible :-( |
| 15:15.39 | *** join/#asterisk qdk (n=qdk@0x573d8dd1.bynqu1.dynamic.dsl.tele.dk) |
| 15:16.25 | [TK]D-Fender | ~gsmbug |
| 15:16.26 | infobot | [~gsmbug] there is a bug compiling Asterisk with GCC 4.2 where optimization errors cause GSM transcoding to distort heavily. Compile with GCC 4.1 or lower, or follow this patch : http://bugs.digium.com/view.php?id=11243 Also please read : http://forums.digium.com/viewtopic.php?p=116731&sid=3c65e49ec840380ed20f1c8426382b39 |
| 15:16.28 | [TK]D-Fender | encbladexp: ^^^^^^^^^6 |
| 15:16.33 | *** join/#asterisk ickmund (n=ickmund@ada-bcn-fw01.adamoeurope.com) |
| 15:16.55 | [TK]D-Fender | encbladexp: GSM transcoding optimization issue |
| 15:16.59 | *** join/#asterisk mbrevda (n=mbrevda@unaffiliated/mbrevda) |
| 15:17.01 | encbladexp | nice |
| 15:17.11 | coppice | [TK]D-Fender: you got that backwards |
| 15:17.22 | [TK]D-Fender | coppice: Ok/fine/sure :) |
| 15:17.38 | [TK]D-Fender | coppice: Right idea, presented the other way. |
| 15:17.45 | coppice | he said GSM is OK, and A-law is bad |
| 15:17.47 | [TK]D-Fender | or right pieces anyway |
| 15:17.56 | encbladexp | lol, i need only set DONT_OPTIMIZE with GCC 4.2 and all get okay? |
| 15:18.07 | encbladexp | alaw ist ok if i call to another alaw Target |
| 15:18.22 | [TK]D-Fender | coppice: Playback demo-echotest <--- he's transcoding the GSM recording back to ALAW. |
| 15:18.25 | encbladexp | but whenever Asterisk must Transcoding » Sound ist bad |
| 15:18.33 | encbladexp | [TK]D-Fender: exactly |
| 15:18.35 | [TK]D-Fender | encbladexp: How does your actual "echo " test sound? |
| 15:18.37 | encbladexp | but the same on speex |
| 15:18.48 | yahh | I am having some issue with RTP |
| 15:18.50 | encbladexp | [TK]D-Fender: i have a file for you if you want? |
| 15:18.59 | coppice | just apply the patch that fixes the GSM codec. its very small |
| 15:19.37 | yahh | <PROTECTED> |
| 15:19.49 | yahh | from asterisk1 to asterisk2 |
| 15:19.56 | [TK]D-Fender | coppice: Sure, if you can advise on some smaller-scale fix for this I'm sure he'd appreciate. I'm jsut aware of the issue mor gloablly. |
| 15:19.59 | yahh | so voice on asterisk2 's connnected phone is not coming proper |
| 15:20.07 | encbladexp | ehm, it alow happens from e.g. SPEEX»GSM as i know |
| 15:20.16 | yahh | but asterisk should ignore those dummy packets |
| 15:20.17 | encbladexp | i never tried SPEEX»ALAW |
| 15:20.25 | encbladexp | i will try it |
| 15:20.28 | encbladexp | hope these helps |
| 15:21.01 | yahh | please advise |
| 15:21.01 | encbladexp | 5 Hours google » no resolution, 1 Minute #asterisk » an (hopely) helpfull link :-D |
| 15:21.08 | coppice | [TK]D-Fender there's nothing very global about it. there was an incorrect constraint on some embedded assembly language, that didn't cause problems until the more aggressive optimisation in 4.<something> |
| 15:21.21 | [TK]D-Fender | coppice: I'm talking about my awareness of this. |
| 15:21.59 | CrazyTux[m] | [TK]D-Fender: ok im an idiot |
| 15:22.04 | CrazyTux[m] | [TK]D-Fender: :) |
| 15:22.12 | [TK]D-Fender | coppice: that I know there is an issue witht he way GSM may be compiled that leads to this result when transcoding. I'm by no means even decently versed in compiling 9or optimization) or the down & dirty bits about codecs |
| 15:22.30 | CrazyTux[m] | [TK]D-Fender: my script is emailing these wav files prematurely -- before the call is "over" |
| 15:22.36 | CrazyTux[m] | [TK]D-Fender: thus the reason for the short durations. |
| 15:22.41 | [TK]D-Fender | CrazyTux[m]: SMRT :p |
| 15:23.22 | CrazyTux[m] | [TK]D-Fender: so now I have two questions..... A) How can I make it so it runs my macro AFTER the call is over? B) Which one is better to use Monitor() with m or MixMonitor() |
| 15:24.24 | [TK]D-Fender | coppice: Believe me, on just about anything everything telephony related you have any firm advice on you can expect that I'd defer to you |
| 15:24.51 | [TK]D-Fender | CrazyTux[m]: ..M() is when the call CONNECTS. You should already know what bits of dialplan can execute after a call... |
| 15:25.28 | coppice | [TK]D-Fender: well, the * fanboys used to get very abusive when it was pointed out that this was a bug in * |
| 15:25.37 | Katty | http://www.tasteofhome.com/Recipes/Cheddar-Tomato-Dumplings <- lunch. |
| 15:26.07 | [TK]D-Fender | coppice: Think I'm going to argue about DSP's and Codecs with YOU? That's insane :P |
| 15:26.35 | CrazyTux[m] | [TK]D-Fender: I dont use asterisk day to day |
| 15:26.46 | CrazyTux[m] | [TK]D-Fender: so I kind of relearn it everytime I touch it |
| 15:27.52 | [TK]D-Fender | Katty: nomNOMnomNOMnomNOMnomNOMnomNOMnomNOMnomNOM |
| 15:27.54 | carrar | sounds HOT |
| 15:28.11 | jaytee | mmmm, that sounds yummy |
| 15:29.15 | CrazyTux[m] | [TK]D-Fender: so are you essentially telling me its not possible? Without something like an AGI per say |
| 15:30.08 | [TK]D-Fender | CrazyTux[m]: No I'm saying that you should already know what happens to calls after a Dial ends.... |
| 15:30.25 | carrar | goes to heaven? |
| 15:30.32 | CrazyTux[m] | [TK]D-Fender: well, I see the variable ${DIALSTATUS} so I would assume.... something / some call back method |
| 15:30.58 | [TK]D-Fender | carrar: Yup, with a minor pit-stop ;) |
| 15:31.04 | CrazyTux[m] | lol |
| 15:31.22 | Katty | [TK]D-Fender: i'm making a double tomatoey batch...you are welcome to join me |
| 15:31.34 | CrazyTux[m] | [TK]D-Fender: Also still any word on MixMonitor() vs Monitor w/ m option |
| 15:32.44 | [TK]D-Fender | Katty: I would, but I'm afraid it'd be SENTIENT by the time I got there :p |
| 15:32.50 | Katty | there is a conspiracy with my voss water. it will not open. |
| 15:33.23 | [TK]D-Fender | CrazyTux[m]: Answer : YES |
| 15:33.35 | [TK]D-Fender | Katty: load res_chainsaw.so |
| 15:34.41 | jaytee | hmmm, is res_chainsaw.so part of the 1.6 addons? I'm not seeing it in 1.4 |
| 15:34.47 | CrazyTux[m] | [TK]D-Fender: I've had 2 hours of sleep last night due to extreme insomnia so bare with me if I still am confused. |
| 15:35.15 | [TK]D-Fender | jayyIts an add-on (and a hack-off, conveniently) |
| 15:35.26 | jaytee | it's bear with me, not bare with me. if you really got bare [TK]D-Fender would have to hit you with a ClueBat for nudity in chat |
| 15:35.43 | *** join/#asterisk Joel (n=jjshoe@h69-129-142-83.mdsnwi.tisp.static.tds.net) |
| 15:36.10 | Chainsaw | Wait what? |
| 15:36.21 | Anth8708 | I HATE to even ask, but I'm stuck and not finding anything. If an extension's ${DIALSTATUS} doesn't return BUSY when it is (happening on both test phones), any idea what it could be? |
| 15:36.58 | [TK]D-Fender | Anth8708: O RLY? What is it returning? |
| 15:37.17 | Katty | [TK]D-Fender: keys worked. |
| 15:37.22 | *** join/#asterisk CrazyTux[m]1 (n=Brandon@216-110-94-230.static.twtelecom.net) |
| 15:37.29 | [TK]D-Fender | Anth8708: And you may want to reconsider what you believe a state is and how its reported... that's an awefully small box you're in... |
| 15:37.31 | Katty | [TK]D-Fender: just twist does NOT work :< |
| 15:37.35 | [TK]D-Fender | Katty: Suck-cess! |
| 15:37.48 | CrazyTux[m]1 | [TK]D-Fender: accidentally hit the wifi button, so if you sent me anything I didnt get it |
| 15:37.53 | Katty | [TK]D-Fender: well i don't have a chainsaw sitting around here at work! |
| 15:38.17 | CrazyTux[m]1 | [TK]D-Fender: since your last "YES" response. |
| 15:38.21 | Chainsaw | You should really. Preparation is everything these days. |
| 15:38.25 | [TK]D-Fender | CrazyTux[m]that was it. |
| 15:38.37 | Anth8708 | [TK]D-Fender - ok . you got me. Let me see if I can figure out how to see the ${DIALSTATUS} |
| 15:38.43 | Katty | jaytee: bear wwith you. |
| 15:38.48 | Katty | jaytee: is it a black bear or a brown bear |
| 15:38.58 | Katty | jaytee: i'll bring salmon. |
| 15:41.33 | Jumpie | mm |
| 15:41.37 | jaytee | Katty, well if it's this zoo then we're talking Alaskan brown bears. We used to have 2 Kodiak bears we got as cubs but they were about 23 years old and both died within a year of each other. |
| 15:41.42 | [TK]D-Fender | ~asterisktrademark |
| 15:41.43 | infobot | [~asterisktrademark] See http://www.digium.com/en/company/view-policy/5 for Digium's trademark policy. |
| 15:42.17 | encbladexp | [TK]D-Fender: recompiling Asterisk fixes my Problem with alaw»gsm |
| 15:42.21 | encbladexp | :-D |
| 15:42.23 | encbladexp | thx |
| 15:42.30 | [TK]D-Fender | encbladexp: You're welcome... |
| 15:42.33 | Katty | jaytee: :< |
| 15:42.46 | Katty | jaytee: i bet that was a sad sad day. |
| 15:42.47 | encbladexp | so, next Problem speex»alaw is also a Problem, but dont works after recompiling |
| 15:42.50 | encbladexp | :-( |
| 15:42.52 | jaytee | yeah |
| 15:43.02 | Jumpie | anybody have any idea why my dnd.php script works for everybody's extension except for just one person? and says not authorized to use this application |
| 15:43.06 | encbladexp | maybe, it is also a 4.2 Bug |
| 15:43.09 | Jumpie | when its the exact same application/syntax as all others |
| 15:43.59 | [TK]D-Fender | Jumpie: We don't see anything, and its your script, how should we know? |
| 15:44.15 | Jumpie | its aastra's script :P |
| 15:44.19 | jaytee | not as sad as when Amali passed. She was a baby African elephant born in 2000, The very first African Elephant born from artificial insemination. Our zoo was the first and since then we've had 3 more and another's on it's way in about 22 months |
| 15:45.00 | [TK]D-Fender | Jumpie: Yeah, that makes us care SO much more :) |
| 15:45.12 | *** join/#asterisk Woody2143 (n=Woody214@machine76.Level3.com) |
| 15:45.22 | Jumpie | hehe lemme try to get more info |
| 15:45.49 | [TK]D-Fender | Jumpie: s/more/any/ |
| 15:46.40 | Jumpie | i have aastra 55is, great phones so far |
| 15:46.53 | Jumpie | on their website there is a slew of standard php scripts, visual voicemail, call parking, etc |
| 15:46.56 | [TK]D-Fender | Aastra 5i = meh... |
| 15:46.58 | Jumpie | one of them is DND..its a toggle function |
| 15:47.01 | CrazyTux[m]1 | [TK]D-Fender: mind pointing me in the right direction for what im trying to do... |
| 15:47.13 | Jumpie | you press it...says dnd activated, press again, says dnd deactivated |
| 15:47.15 | CrazyTux[m]1 | [TK]D-Fender: I need to execute this AFTER hangup of the call |
| 15:47.24 | [TK]D-Fender | CrazyTux[m]1: "Asterisk Standard Extensions" <- JFGI |
| 15:47.37 | CrazyTux[m]1 | [TK]D-Fender: JFGI ? |
| 15:47.40 | Jumpie | <PROTECTED> |
| 15:47.47 | [TK]D-Fender | ~jfgi |
| 15:47.47 | infobot | http://www.google.com/search?q=jfgi |
| 15:47.52 | [TK]D-Fender | ^^ |
| 15:47.52 | Jumpie | and this extension has no other issues with authentication, registration, calling, etc |
| 15:48.11 | Jumpie | so im not sure if this is just a gremlin or not..because i see nothing in my logs/debugs to narrow this down maybe ill have to look harder |
| 15:49.06 | [TK]D-Fender | Jumpie: What does some random PHP not even called by * mean here? |
| 15:49.18 | CrazyTux[m]1 | [TK]D-Fender: ah h => |
| 15:49.22 | [TK]D-Fender | Jumpie: And I'm not seeing what it DOES |
| 15:49.28 | Jumpie | well..im not sure if it is a phone/script thing, or somethin funky on the server |
| 15:49.33 | Jumpie | lol ya..i know what you mean |
| 15:49.42 | Jumpie | just wondering if this flagged any prior issue you may have seen tucked into your brain |
| 15:49.43 | Jumpie | :D |
| 15:50.35 | [TK]D-Fender | I don't really deal with Aastra. the only SIP phone from them I have from them (57i CT, deluxe even) sits in my warehouse collecting dust and I touch it as little as possible. |
| 15:50.45 | Jumpie | well so far..all i can see is on the phone screen "not authorized to use this application' yet other php scripts work fine |
| 15:50.53 | CrazyTux[m]1 | [TK]D-Fender: thanks! :) |
| 15:50.54 | Jumpie | fender..why not an aastra fan on purpose? |
| 15:51.00 | Jumpie | i like the polycoms..but they are more of a pain to setup |
| 15:51.02 | CrazyTux[m]1 | [TK]D-Fender: totally flew over my head forgot about those |
| 15:51.41 | [TK]D-Fender | Jumpie: Setup doesn't slow me down in the slightest. More complex yes, but more stable and a better physical phone. |
| 15:51.59 | Jumpie | the ip 6000 i have seems to strugle tremendously on any extended time off the network |
| 15:52.00 | [TK]D-Fender | Jumpie: I have half a dozen or more real gripes about Aastra's |
| 15:52.20 | *** join/#asterisk timeshell_atwork (n=chatzill@gw.lusi.on.ca) |
| 15:52.36 | Jumpie | when i do network maintenance it just will not work or synch back up or anything unless i reboot the phone, and even sometimes it takes 2 or 3 times |
| 15:52.44 | Jumpie | but other than that its great hehe |
| 15:56.05 | *** join/#asterisk errotan (n=errotan@62.201.123.242) |
| 15:58.32 | *** join/#asterisk icyValk77 (n=icyValk7@host81-153-254-0.range81-153.btcentralplus.com) |
| 15:59.43 | *** join/#asterisk ruben23 (n=RPL@122.55.48.243) |
| 16:00.50 | *** join/#asterisk mykhyggz (n=mykhyggz@evolone.org) |
| 16:06.27 | Anth8708 | OK Fender. I'm not getting BUSY from the extension while it's off hook (at least as far as I can tell), but I don't know why. http://pastebin.com/d5d34aab4 |
| 16:07.16 | Anth8708 | line 44 in the pastebin is really where I'm looking and saying that there is no BUSY |
| 16:07.49 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 16:10.09 | *** part/#asterisk delphus (n=delphus@unaffiliated/delphus) |
| 16:11.49 | pif | hi, can I reboot a polycom phone from its web interface? |
| 16:11.55 | leifmadsen | yes |
| 16:12.34 | Anth8708 | pif, you can also do that from the * console |
| 16:12.40 | pif | how? |
| 16:13.05 | leifmadsen | sip notify |
| 16:13.11 | pif | just that? |
| 16:13.13 | leifmadsen | no |
| 16:13.23 | leifmadsen | help sip notify |
| 16:13.38 | leifmadsen | or, sip notify <TAB><TAB> |
| 16:13.57 | pif | ok what <type> should I use? |
| 16:14.05 | leifmadsen | polycom... |
| 16:14.17 | *** join/#asterisk thansen (n=thansen@76.27.110.194) |
| 16:14.21 | leifmadsen | use the <TAB> trick to see the format |
| 16:14.29 | Anth8708 | biggest thing when doing this is that you will want to increment the revision number in one of the files the phone pulls: https://www.trixbox.org/forums/trixbox-forums/trixbox-endpoints/possible-remote-restart-polycom-phones |
| 16:14.43 | pif | leifmadsen: polycom-check-cfg ? |
| 16:15.01 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 16:15.13 | leifmadsen | pif: please use some common sense... of all the options, which one is the most closely related to what you're doing? |
| 16:15.14 | Anth8708 | pif: that's it |
| 16:15.36 | *** part/#asterisk kiwi_uk (n=jonathan@hq.mobell.com) |
| 16:16.20 | pif | thx |
| 16:17.27 | timeshell_atwork | Is there any gizmo that can play flash based radio stream for moh? |
| 16:19.48 | Joel | timeshell_atwork anything that utilizes the soundcard |
| 16:19.56 | timeshell_atwork | eh?? |
| 16:19.57 | Joel | timeshell_atwork however seems far easier to rip the flash and convert it to mp3 |
| 16:20.01 | timeshell_atwork | NO |
| 16:20.03 | timeshell_atwork | live stream |
| 16:20.16 | timeshell_atwork | play mpg123 http://38.99.208.186/chfi/ |
| 16:20.25 | timeshell_atwork | Except some stations are switching to flash players |
| 16:20.32 | timeshell_atwork | And aren't providing URL's for the stream. |
| 16:20.37 | Joel | so configure asterisk to use the sound card |
| 16:20.44 | Joel | run line in to a box with a browser playing your audio |
| 16:20.46 | timeshell_atwork | Don't have a sound card |
| 16:20.52 | timeshell_atwork | bah |
| 16:20.54 | Joel | good luck! |
| 16:20.54 | timeshell_atwork | no |
| 16:21.19 | Joel | let me know when you finish your real time flash audio ripper and the module to hook it into asterisk. |
| 16:21.43 | timeshell_atwork | My original question was does such a gizmo exist. |
| 16:21.52 | timeshell_atwork | By your answer, I can assume you really meant no. |
| 16:22.34 | Joel | timeshell_atwork what does your google searching reveal? |
| 16:22.55 | timeshell_atwork | Not much. But I may just be searching with the wrong search words |
| 16:23.02 | timeshell_atwork | That's why I'm asking |
| 16:23.53 | *** join/#asterisk Iamnach0 (i=Iamnacho@ip174-70-137-120.ks.ks.cox.net) |
| 16:24.28 | Joel | not in real time, no. |
| 16:26.28 | proute | re-hello |
| 16:26.56 | proute | during a call, sometimes I can hear a beep like a dtmf but I don't press any touch |
| 16:27.10 | proute | in my sip debug when the "beep" appear I have this: |
| 16:27.11 | proute | SIP/2.0 401 Unauthorized^M |
| 16:27.11 | proute | Via: SIP/2.0/UDP 192.168.0.24:5060;branch=z9hG4bK312846b4c5f5a621f;received=192.168.0.24^M |
| 16:27.11 | proute | From: <sip:55@192.168.0.254:5060>;tag=d3a263eeba^M |
| 16:27.11 | proute | To: <sip:55@192.168.0.254:5060>;tag=as5ab8b218^M |
| 16:27.11 | proute | Call-ID: deff8a8c19518953^M |
| 16:28.01 | proute | Why this problem appear.? |
| 16:28.06 | Alfio | ~ pastebin |
| 16:28.06 | infobot | [~pastebin] 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 |
| 16:28.17 | Alfio | proute thats fro you |
| 16:28.29 | proute | yes ok thanks |
| 16:28.37 | proute | I post my log .... |
| 16:29.23 | proute | My log is here: http://pastebin.com/d5fc39065 |
| 16:29.49 | proute | When the call is answered, I have a beep . this beep appear randomly in call... |
| 16:30.14 | proute | And I saw an "401 Unauthorized" in my log when i'm in call... |
| 16:31.01 | proute | Someone can help me about this "beep" :@ |
| 16:31.06 | proute | thanks |
| 16:32.10 | [TK]D-Fender | Anth8708: the phone is not busy because it can accept the call. Also it is psycho to dial the phone for only 2 SECONDS. |
| 16:32.55 | [TK]D-Fender | proute: that's a register failure, not a call failure |
| 16:33.18 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 16:34.18 | proute | [TK]D-Fender: yes, but this failure is my phone that I use during call. When this failure appear (may be is a coincidence) but I hear a beep :( |
| 16:34.23 | *** join/#asterisk Maxxed (n=max@216.215.95.114) |
| 16:34.39 | [TK]D-Fender | proIt is. |
| 16:34.46 | [TK]D-Fender | proute: It is. |
| 16:37.10 | proute | yes it is... |
| 16:43.47 | *** join/#asterisk vgster (n=vgster@94-194-190-189.zone8.bethere.co.uk) |
| 16:44.33 | *** join/#asterisk many (n=many@dslb-188-098-002-093.pools.arcor-ip.net) |
| 16:46.13 | *** join/#asterisk jicksta (n=jicksta@c-98-210-108-28.hsd1.ca.comcast.net) |
| 16:47.11 | zr0 | off topic: is there a term used to describe a pots line that can only receive incoming calls, not make outgoing calls besides 911? |
| 16:47.28 | *** join/#asterisk CrazyTux[m] (n=Brandon@99-53-97-235.lightspeed.cyprtx.sbcglobal.net) |
| 16:50.34 | [TK]D-Fender | zr0: No |
| 16:50.47 | jetsnoc | a sort-of bat phone |
| 16:50.56 | jetsnoc | but that term usually means it auto answers |
| 16:51.20 | [TK]D-Fender | "batphone"'s don't auto answer... they auto DIAL |
| 16:54.31 | *** join/#asterisk Darkmadda (n=root@c-76-27-95-83.hsd1.ut.comcast.net) |
| 16:57.18 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 16:59.35 | *** join/#asterisk brutuz (n=brutuz@66.151.192.103) |
| 17:10.10 | *** join/#asterisk farkus (i=chatzill@cpe-72-225-212-219.nyc.res.rr.com) |
| 17:12.17 | hesco | what, pray tell, does this mean? And what should I / could I do about this? |
| 17:12.19 | hesco | pbx.c:2055 pbx_find_extension: Maximum PBX stack exceeded |
| 17:12.26 | *** join/#asterisk jameswf (n=james@unaffiliated/jameswf-home) |
| 17:12.35 | hesco | I see this on a `dialplan reload` |
| 17:15.11 | Anth8708 | [TK]D-Fender just fyi, 2 seconds was for testing:) might be a "call waiting" type thing on the polycom? looking |
| 17:15.21 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 17:20.23 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 17:25.22 | *** join/#asterisk troy- (n=troy@worldnet.tauri.ca) |
| 17:27.01 | *** join/#asterisk tdg911 (n=tdg911@75-131-246-217.static.slid.la.charter.com) |
| 17:28.09 | *** join/#asterisk luismm75 (n=lm@ip-200-003-170-146.coopvgg.com.ar) |
| 17:28.28 | luismm75 | Hello |
| 17:29.31 | [TK]D-Fender | Anth8708: Typically a phone won't return BUSY unless its incapable of accepting a call or was told to refuse it |
| 17:31.27 | luismm75 | Hello, I want to know if its posible to use a panasonic tx-tes824 with asterisk. thanks |
| 17:31.45 | Anth8708 | [TK]D-Fender so I'm guessing. Must be something I changed on the Polycom config since they used to return BUSY when the line was off hook. Rethinking that this could be a good thing however. |
| 17:32.29 | [TK]D-Fender | Anth8708: Nope. |
| 17:33.31 | [TK]D-Fender | luismm75: Its an analog phone system. What would this have to do with *? |
| 17:34.34 | luismm75 | i hoped i could at least record incoming calls |
| 17:35.57 | luismm75 | thank you very much [TK]D-Fender |
| 17:36.09 | [TK]D-Fender | luismm75: this will add answering delay and an odd double-ring for your callers, but its possblie. |
| 17:36.16 | [TK]D-Fender | luismm75: How many lies do you have on it? |
| 17:36.19 | [TK]D-Fender | lines* |
| 17:36.41 | *** join/#asterisk Keizer (n=Keizer@64.238.20.94) |
| 17:37.26 | beek | Hmmmm.... Freudian slip [TK]D-Fender ? |
| 17:37.43 | luismm75 | 4 lines |
| 17:39.34 | *** join/#asterisk ingenius (n=alektro@host95.190-229-175.telecom.net.ar) |
| 17:39.45 | [TK]D-Fender | luismm75: OK, well this also increases the cost of your setup as you'll need 8 FXo on your server for this. |
| 17:39.51 | [TK]D-Fender | luismm75: How many phones does it have? |
| 17:42.13 | luismm75 | i see. i'm not sure how many phones.. about 15..maybe |
| 17:42.37 | *** part/#asterisk kaii (n=kai@ciphron.de) |
| 17:43.19 | [TK]D-Fender | luismm75: I might very likely consider replacing the entire system with *... |
| 17:44.10 | *** join/#asterisk Deeewayne (n=dwayne@75.76.254.162) |
| 17:44.10 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 17:44.26 | luismm75 | You are right. Thank you very much [TK]D-Fender |
| 17:45.49 | *** join/#asterisk thansen (n=thansen@76.27.110.194) |
| 17:53.17 | brah | Does hangup() terminate only the call, or the whole extension script? |
| 17:57.22 | gigman | hi |
| 17:57.26 | gigman | maybe someone can help me |
| 17:58.11 | leifmadsen | brah: dialplan execution always stops after a call is hung up, other than in the 'h' extensions |
| 18:01.33 | *** join/#asterisk Defraz (n=T0tal@63.228.246.229) |
| 18:03.21 | brah | ok, cool |
| 18:05.03 | Katty | my brain hurts. |
| 18:09.03 | *** join/#asterisk twanny796 (n=chatzill@85.232.203.93) |
| 18:09.09 | *** join/#asterisk bhodder (n=blake@142.166.111.151) |
| 18:09.54 | gigman | im getting this error |
| 18:09.56 | gigman | unable to access the running directory (Permission dined). Changing to / for compatibility |
| 18:09.56 | gigman | unable to open pid file '/var/run/asterisk.pid' : permission denied |
| 18:10.11 | brah | Which user are you running asterisk as? |
| 18:10.15 | gigman | root |
| 18:10.32 | brah | It should have permission |
| 18:10.35 | brah | But double check, anyway |
| 18:10.41 | bhodder | Hi, I am trying to record a message using audacity and then place it on the asterisk server for use, but asterisk will not play the file? anyone have suggestions on the best method of doing this |
| 18:11.26 | leifmadsen | bhodder: make sure it is 8khz, 8bit, mono |
| 18:12.07 | *** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim) |
| 18:12.44 | *** join/#asterisk kiwi_ (n=kiwi@ks359129.kimsufi.com) |
| 18:13.00 | bhodder | In audacity what would be the extension type for that |
| 18:13.03 | [TK]D-Fender | gigman: really? that may be the user starting a daemon, but it doesn't mean thats the user it runs as... |
| 18:13.05 | *** join/#asterisk Iamnacho (i=Iamnacho@ip174-70-137-120.ks.ks.cox.net) |
| 18:13.48 | gigman | can I change the user that runs it to root? |
| 18:13.58 | gigman | That would avoid all these problems correct? |
| 18:14.37 | [TK]D-Fender | gigman: You could. not suggesting this is what you want to do... |
| 18:15.05 | gigman | there are adverse effects running * as root? |
| 18:15.35 | [TK]D-Fender | gigman: Yes, if * is comprimised then they can execute things as root. |
| 18:15.52 | bhodder | I am trying to record it from my laptop and I try saving as GSM 6.10 WAV (mobile) but it will not play back when I copy it to the server |
| 18:16.36 | beek | gigman: Google 'run asterisk as non-root' andyou'll see lots of recipes to ensure that your permissions are correct. |
| 18:17.41 | hesco | I'm getting an 'extension not found' error which defies what dialplan show context shows me, as in: http://paste.debian.net/42233/ |
| 18:17.51 | hesco | Can anyone here please help me understand this? |
| 18:17.58 | jaytee | nope |
| 18:18.41 | beek | hesco: Yeah, the extension is there. How about the context? |
| 18:18.46 | gigman | thanks you guys |
| 18:19.02 | beek | Afternoon jaytee |
| 18:19.12 | hesco | beek: the dialplan show <tab> reveals the context |
| 18:19.19 | jaytee | hesco, where are you located? |
| 18:19.40 | jaytee | afternoon, beek |
| 18:19.59 | hesco | Decatur Ga, at my desk in the dining room |
| 18:20.05 | *** join/#asterisk darkmadda (n=root@c-76-27-95-83.hsd1.ut.comcast.net) |
| 18:20.12 | beek | hesco: Are your SIP entries pointed to that context? We're not seeing enough here to tell. |
| 18:20.20 | jaytee | just wondering, I know a Richard Searcy that used to work here |
| 18:20.33 | hesco | an oracle DBA ??? |
| 18:20.44 | hesco | where is here? |
| 18:20.53 | jaytee | no, a zookeeper who now works for an emergency animal hospital |
| 18:20.59 | jaytee | Indianapols |
| 18:21.03 | hesco | different guy then |
| 18:21.07 | jaytee | yep |
| 18:21.30 | hesco | my Richard is from Detroit, transplanted to Atlanta about 8 or 10 years ago |
| 18:22.20 | jaytee | anyways, we can't tell why the extension isn't found since we don't see your dialplan, i.e. extensions.conf and sip.conf but the most likely cause is neither context can "see" the other |
| 18:22.29 | hesco | my extensions.conf includes: '#include ymd.conf' and that file includes this context. |
| 18:22.38 | hesco | what does sip.conf have to do with it? |
| 18:22.41 | bhodder | what is a good software for recording the sound files for an IVR |
| 18:22.47 | jaytee | is it a sip phone? |
| 18:23.11 | hesco | yes, I'm using an HT-486 (sip based) |
| 18:23.19 | jaytee | well, there ya go! |
| 18:23.23 | hesco | bhodder: use asterisk |
| 18:23.31 | beek | sip.conf tells asterisk what context to start with. It's probably in the default context right now. |
| 18:23.38 | hesco | check out the Record() app |
| 18:23.58 | hesco | oh yeah, let me take a look then |
| 18:24.08 | bhodder | do I have to setup an extension for that |
| 18:24.31 | bhodder | or is it possible to use that from the console |
| 18:25.25 | jaytee | hesco, instead of you taking a look how about we take a look? that way we don't have to waste time going back and forth with questions. |
| 18:26.46 | gigman | beek, is there a way that I can run * as root just for now, I see the config stuff online (thanks) but I want to test one quick thing first, how do I set it to run as root in the config files? |
| 18:30.29 | beek | gigman: I don't know where you have asterisk located but just log on as root and use asterisk -c to run it. |
| 18:31.09 | gigman | ok thanks |
| 18:31.29 | beek | It'll run from the console that you've signed into. |
| 18:31.56 | [TK]D-Fender | gigman: Go look at how your start * in the first place |
| 18:32.43 | beek | Of course, any files (such as logs) that are created will be owned by root, thus making the conversion to non-root all that much more difficult for a non Unix user. |
| 18:39.14 | *** join/#asterisk bwat (n=IceChat7@atlantis.kanobe.com) |
| 18:44.28 | hesco | sorry, jaytee, on phone, will be with you in a moment |
| 18:44.56 | jaytee | wanders off to write his memoirs |
| 18:50.07 | *** join/#asterisk mgroman (n=miles@unaffiliated/mgroman) |
| 18:50.15 | *** part/#asterisk mgroman (n=miles@unaffiliated/mgroman) |
| 18:52.36 | bhodder | Is there any way to record IVR prompts on a laptop then copy them to the asterisk server that will work? |
| 18:53.23 | KavanS | bhodder, yes |
| 18:53.42 | KavanS | bhodder, you need to convert to the correct format....I recorded in wav, then converted to gsm |
| 18:54.17 | bhodder | ok, what did you use to convert it to gsm |
| 18:55.27 | KavanS | bhodder, asterisk will work |
| 18:55.46 | KavanS | bhodder, google it too....you can use asterisk to perform sound file conversions |
| 18:56.11 | bhodder | oh ok thanks. |
| 18:56.11 | jaytee | bhodder type help file convert at the CLI |
| 18:57.25 | *** join/#asterisk DelphiWorld (n=Miranda@41.201.71.63) |
| 18:57.28 | DelphiWorld | hello all |
| 18:57.41 | timeshell_atwork | Hello. Does mpg123 play wma streams? |
| 18:57.46 | DelphiWorld | please, how i can enable AMI to listen to all Interface using Asterisk consol no file edition? |
| 19:02.00 | DelphiWorld | debian users, welcome to #debian-voip |
| 19:02.11 | *** join/#asterisk bbryant (n=brett@c-68-59-22-114.hsd1.sc.comcast.net) |
| 19:03.10 | [TK]D-Fender | DelphiWorld: AMi listens on all interfaces with the sample configs. |
| 19:04.46 | bhodder | when trying to convert a .wav to .ulaw I get an error cannot open .wav and fails.convert it |
| 19:04.58 | bhodder | any ideas anyone? |
| 19:05.00 | *** join/#asterisk jeffgus (n=jeffgus@2002:ad33:b504:0:0:0:0:1) |
| 19:05.55 | beek | bhodder: it's 8Khz, mono? |
| 19:06.24 | beek | e.g. Asterisk can play the .wav file, right? |
| 19:06.57 | bhodder | no it will not play the file and I'm recording it with audacity |
| 19:07.28 | beek | It has to meet the specs for Asterisk. 8Khz, mono. |
| 19:07.47 | leifmadsen | bhodder: I had already mentioned the above earlier |
| 19:08.00 | beek | bhodder: http://lists.digium.com/pipermail/asterisk-users/2003-October/015583.html |
| 19:08.18 | beek | leifmadsen: He didn't like that answer. |
| 19:08.26 | leifmadsen | beek: sounds right |
| 19:08.55 | leifmadsen | bhodder: or use 'sox' to convert the wav file to the appropriate format |
| 19:09.01 | leifmadsen | (again, google will help you get the answer) |
| 19:09.20 | bhodder | ok |
| 19:09.27 | beek | It's in that thread that I just sent to you. |
| 19:10.06 | bhodder | I was using google but the sites were directing me to use the record() app in asterisk |
| 19:11.14 | beek | bhodder: That does have the advantages of 1) Already in the correct format and 2) Phone will filter out superfluous grap. |
| 19:11.33 | [TK]D-Fender | graps of wrath! |
| 19:11.40 | Katty | oh man |
| 19:11.44 | Katty | i just did, 162 squats |
| 19:11.46 | Katty | dies. |
| 19:11.56 | beek | s/grap/crap/ |
| 19:12.04 | beek | craps of wrath? |
| 19:12.06 | [TK]D-Fender | Katty: I do squat all day :p |
| 19:12.21 | Katty | that's *pantpant* nice *pantpant* |
| 19:12.29 | [TK]D-Fender | mmmmmm pnats |
| 19:12.32 | [TK]D-Fender | pants* |
| 19:12.34 | [TK]D-Fender | :p |
| 19:12.43 | Katty | yes. |
| 19:12.47 | Katty | yes they are. |
| 19:12.57 | Katty | especially if they're lose rise, flare leg, faded denim |
| 19:13.04 | Katty | lose? |
| 19:13.05 | Katty | low |
| 19:13.11 | bhodder | true the only thing is I would like to be able to record several prompts on my on computer and load them to the server after |
| 19:13.13 | Katty | lowes. |
| 19:14.03 | bhodder | using sox gave me this error: sox soxio: Failed reading `/var/lib/asterisk/sounds/custom/intro.wav': unknown file type `auto' |
| 19:14.49 | Katty | sytax error |
| 19:15.15 | Katty | you snickerdoodled something up |
| 19:16.05 | beek | snickerdoodled? That's a new one on me. |
| 19:16.38 | Katty | it's a very common phrase at my house. |
| 19:16.46 | Katty | it'd be a good cream colored ferret name. |
| 19:16.56 | Katty | cinnamon colored. |
| 19:16.57 | beek | I used the shorter, more direct Four letter word. |
| 19:17.13 | *** join/#asterisk gr0mit (n=tim@2001:8b0:3d9:0:294d:6b9:3fce:625e) |
| 19:18.18 | *** join/#asterisk Breyer (n=Breyer@ool-43540592.dyn.optonline.net) |
| 19:18.29 | Breyer | ~ITSP |
| 19:18.30 | infobot | [~itsp] An ITSP is an Internet telephony Service Provider or "VoIP Telephone company". The allow you to either PLACE calls to the PSTN (called Termination), RECEIVE calls from the PSTN (called Origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs |
| 19:18.39 | Breyer | ~itsplist-us |
| 19:18.40 | infobot | somebody said itsplist-us was Here are some popular ITSPs (USA) starting with the more respected ones : http://www.teliax.com , http://www.voicepulse.com/connect/ , http://www.broadvoice.com, http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net |
| 19:20.03 | leifmadsen | Breyer: next time, just do, "infobot tell breyer about itsp" |
| 19:20.21 | Breyer | ok, thanks, sorry about that |
| 19:20.25 | leifmadsen | np, fyi |
| 19:21.00 | beek | <PROTECTED> |
| 19:21.03 | leifmadsen | yes |
| 19:21.06 | leifmadsen | you can do that for anyone |
| 19:21.15 | leifmadsen | you don't always have to flood the channel |
| 19:21.16 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.static.bezeqint.net) |
| 19:21.20 | beek | Cool! I didn't know about that one. |
| 19:21.26 | leifmadsen | infobot: tell beek about asteriskversioning |
| 19:21.39 | beek | Very nice. |
| 19:22.16 | Alfio | infobot tell Alfio about itsp |
| 19:22.27 | Alfio | nice :) |
| 19:22.29 | beek | I'd be REALLY impressed if I could use 'me' instead of 'beek' |
| 19:22.36 | beek | ;-) |
| 19:22.56 | leifmadsen | infobot: tell me about stuff |
| 19:23.12 | leifmadsen | beek: funny enough -- it DOES work |
| 19:23.21 | beek | no shit? That's really great. |
| 19:23.21 | leifmadsen | ACTION is stuffing George* |
| 19:23.30 | leifmadsen | infobot: tell me about beek |
| 19:23.32 | [TK]D-Fender | EW! |
| 19:23.37 | beek | Now.... |
| 19:23.40 | leifmadsen | i dunno what is 'beek'. |
| 19:23.52 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 19:24.15 | leifmadsen | infobot: tell me about thebirdsandthebees |
| 19:30.12 | *** join/#asterisk DelphiWorld (n=Miranda@41.201.122.80) |
| 19:33.11 | beek | leifmadsen: Are you still waiting for your answer or is infobot that long-winded |
| 19:33.32 | leifmadsen | nope, it doesn't know anything about sex |
| 19:33.35 | leifmadsen | it's a bot... |
| 19:33.52 | leifmadsen | I got an answer long ago |
| 19:33.56 | leifmadsen | it just msg's it to me |
| 19:34.24 | beek | I just wondered if it had some sage advice. |
| 19:36.10 | *** join/#asterisk malcolmd (n=malcolmd@pdpc/sponsor/digium/malcolmd) |
| 19:39.40 | *** join/#asterisk [netman] (n=netman@158.Red-88-24-108.staticIP.rima-tde.net) |
| 19:42.36 | [TK]D-Fender | ~SEX |
| 19:42.37 | infobot | methinks sex is alias sex "updatedb; locate; talk; date; cd; strip; look; touch; finger; unzip |
| 19:42.47 | [TK]D-Fender | Sures eems to ;) |
| 19:43.06 | beek | [TK]D-Fender: That was yours, I presume. |
| 19:43.26 | [TK]D-Fender | nope |
| 19:44.02 | [TK]D-Fender | fixed now :) |
| 19:44.04 | [TK]D-Fender | ~sex |
| 19:44.05 | infobot | [~sex] updatedb; locate; talk; date; cd; strip; look; touch; finger; unzip; uptime; gawk; head; emerge --oneshot condom; mount; fsck; gasp; more; yes; yes; yes; more; umount; emerge -C condom; make clean; sleep |
| 19:44.08 | *** join/#asterisk Dickie_Workie (i=cf6fa01c@gateway/web/freenode/x-70a16f4c0e589c39) |
| 19:44.09 | [TK]D-Fender | :p |
| 19:44.16 | Dickie_Workie | Whoops |
| 19:44.23 | Dickie_Workie | Must've taken a wrong turn in Albequerque |
| 19:44.34 | [TK]D-Fender | Silly wabbit |
| 19:46.21 | beek | Cripes... |
| 19:48.42 | howie | is there a way to connect a cell phone to my asterisk box? |
| 19:49.08 | [TK]D-Fender | howie: Duct tape. |
| 19:49.14 | [TK]D-Fender | howie: LOTS of duct tape |
| 19:49.39 | howie | lol |
| 19:49.46 | howie | duck tape is the cure all isnt it |
| 19:50.56 | [TK]D-Fender | howie: That and WD-40 |
| 19:51.20 | [TK]D-Fender | howie: If it moves and shouldn't = duct tape. If it should and doesn't = WD-40 |
| 19:51.26 | howie | [TK]D-Fender: any way to get my box to recieve make calls with a cell? |
| 19:52.06 | [TK]D-Fender | howie: Google : chan_mobile |
| 19:52.53 | *** join/#asterisk jicksta_ (n=jicksta@c-98-210-108-28.hsd1.ca.comcast.net) |
| 19:54.47 | afink | Anyone know where I can find a good tutorial on how to block 000-000-0000 numbers from coming in? |
| 19:55.24 | hesco | jaytee, off the phone now. Thanks. that was the ticket. I added 'include => ymd_partners' to the default context listed in sip.conf and my missing extension started working (though its giving me a funky callerID, but I have other priority issues before I get to that). Thanks for the pointer. |
| 19:57.00 | dwery | with * 1.6.0. I've got 'iax2_trunk_queue: Maximum data space exceeded ...' . There were only one or two calls on the trunk at that time. I did a bit of research and found an old bug report thats closed. Any suggestion? thanks. |
| 19:58.05 | *** join/#asterisk howie (n=howie@71-95-220-206.dhcp.mtpk.ca.charter.com) |
| 19:59.12 | *** join/#asterisk batphone (n=wclayton@66.219.32.14) |
| 19:59.14 | [TK]D-Fender | afink: "core show application gotoif" <- |
| 19:59.19 | batphone | why dont i have a 'sip debug' command? |
| 19:59.32 | [TK]D-Fender | batphone: Checked to see it chan_sip even loaded? |
| 19:59.39 | dwery | batphone: sip set debug |
| 19:59.44 | batphone | ahh |
| 19:59.52 | [TK]D-Fender | batphone: Syntax helps too... |
| 20:02.04 | *** join/#asterisk xpot-mobile (n=james@204-228-153-210.ip.xmission.com) |
| 20:02.18 | batphone | hmm |
| 20:02.40 | afink | Should I be getting these if i have hard echo can? Unable to enable echo cancellation on channel 20 (No such device) |
| 20:06.57 | *** join/#asterisk nightrid3r (n=kvirc@78-20-232-172.access.telenet.be) |
| 20:07.02 | *** join/#asterisk blackest_mamba (n=blackest@c-71-239-160-143.hsd1.il.comcast.net) |
| 20:07.11 | [TK]D-Fender | afink: .... "no such device"..... |
| 20:07.20 | [TK]D-Fender | afink: Can't EC a port you don't have. |
| 20:09.54 | batphone | my phone rebooted over the weekend and now it wont register |
| 20:10.01 | batphone | i cant see why |
| 20:10.13 | *** join/#asterisk kmem (n=email@cpe-66-25-166-104.austin.res.rr.com) |
| 20:10.18 | batphone | it would appear that the phone is not responing to registration request acks from the pbx |
| 20:10.29 | batphone | username and password are fine |
| 20:10.33 | batphone | the phone's config file hasnt change |
| 20:11.38 | *** join/#asterisk ingenius (n=alektro@imhotep.toptech.com.ar) |
| 20:17.03 | *** join/#asterisk lesouvage (n=lesouvag@82.73.69.76) |
| 20:17.42 | batphone | i used sipsak to test the connectivity |
| 20:17.50 | batphone | the phone returns sip stuff |
| 20:17.58 | [TK]D-Fender | batphone: And we see precisely nothing... |
| 20:19.12 | batphone | http://pastebin.com/d7b6e1cdb |
| 20:20.00 | batphone | thats the debug |
| 20:20.01 | batphone | here is the config |
| 20:20.03 | batphone | http://pastebin.com/d607f3094 |
| 20:20.33 | beek | Is there a quick way to see what meetme conferences Page() has created? |
| 20:20.37 | batphone | asterisk shows the phoen as being registered |
| 20:20.46 | batphone | the phone does not think it is registered |
| 20:20.49 | *** join/#asterisk sjobeck (n=Adium@host-198-236-32-82.gladstone.k12.or.us) |
| 20:20.59 | [TK]D-Fender | batphone: I don't think i'm seeing global SIP debug there.. |
| 20:21.04 | *** part/#asterisk sjobeck (n=Adium@host-198-236-32-82.gladstone.k12.or.us) |
| 20:21.22 | [TK]D-Fender | batphone: * isn't retransmitting and wouldn't spit out 200's constantly. there is no need for the remote end to respond. |
| 20:21.41 | *** join/#asterisk _mwoodj_ (n=MWoodJ@pdpc/sponsor/digium/hyper-eye) |
| 20:21.50 | [TK]D-Fender | batphone: You should be showing us the phone attempting to register |
| 20:22.26 | *** join/#asterisk kotis (n=kotis@192.68.183.170) |
| 20:22.38 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 20:22.49 | kotis | what OS is used for development of asterisk? |
| 20:23.09 | *** join/#asterisk TheOpenSourcerer (n=TheOpenS@81-178-65-1.dsl.pipex.com) |
| 20:23.13 | batphone | probably linux |
| 20:23.20 | Qwell | kotis: many different OSes. Linux is considered the "most" supported though |
| 20:23.28 | kotis | I know it's linux, but which one? |
| 20:23.33 | [TK]D-Fender | kotis: Any |
| 20:23.33 | Qwell | any |
| 20:23.43 | Qwell | Linux is Linux is Linux (unless it's SUSE) |
| 20:23.57 | Qwell | ((I'll just bite my tongue now)) |
| 20:24.02 | batphone | heh |
| 20:24.07 | *** part/#asterisk TheOpenSourcerer (n=TheOpenS@81-178-65-1.dsl.pipex.com) |
| 20:24.14 | batphone | !pants Qwell |
| 20:24.16 | [TK]D-Fender | kotis: Linux is a KERNEL, who cares that distro XYZ includes VLC and not Totem? |
| 20:24.32 | Alfio | you can use asterlinuxdevelopment 5.0 |
| 20:24.34 | Alfio | :) |
| 20:24.50 | kotis | I normally use Solaris, but asterisk doesn't play well on it, no dahdi drivers :( |
| 20:25.11 | batphone | [TK]D-Fender: http://pastebin.com/d31d14a9 |
| 20:25.17 | batphone | i removed all other peers and reset the phone |
| 20:25.24 | batphone | so this is a clean 'sip set debug on' |
| 20:26.17 | *** part/#asterisk encbladexp (n=stefan@p5495AACB.dip.t-dialin.net) |
| 20:26.33 | batphone | just keeps on going like that |
| 20:26.42 | [TK]D-Fender | batphone: what frequency? |
| 20:26.54 | *** join/#asterisk propellerhead (n=yogurt2u@host36.190-136-235.telecom.net.ar) |
| 20:27.21 | [TK]D-Fender | kotis: Well go pick some distro you're capable of managing (except SUSE, Qwell's tonge is swollen enough). |
| 20:27.46 | batphone | [TK]D-Fender: doesnt seem to be much of a pattern to it |
| 20:28.03 | batphone | may 3 messages in 10 seconds at irregular intervals |
| 20:28.09 | batphone | then three or four in a row |
| 20:28.47 | batphone | argh. some asshole upgraded the firmware on my cisco phone |
| 20:28.52 | batphone | im just finding this out... |
| 20:31.27 | batphone | this is load file SIP75.8-5-2S for the cisco 7975 |
| 20:31.33 | batphone | and asterisk 1.6.1.1 |
| 20:31.55 | batphone | anyone know of reports regarding newer cisco SIP firmware causing problems? |
| 20:32.36 | [TK]D-Fender | batphone: Plenty of history on old ones doing that... |
| 20:32.43 | [TK]D-Fender | batphone: but at least they're consistent! |
| 20:32.45 | [TK]D-Fender | :p |
| 20:32.49 | [TK]D-Fender | ok, checkout time, BBIAB |
| 20:33.27 | batphone | lol |
| 20:33.29 | batphone | thanks man |
| 20:34.48 | batphone | hmm |
| 20:34.57 | batphone | Rx frames are not incrementing on my phone |
| 20:35.09 | batphone | the phone isnt seeing anything from the PBX |
| 20:37.18 | dwery | mmm udev does not load the firmware in my astribank. xpp.rules is present... any clue? |
| 20:37.47 | batphone | dwery: im fresh out of clues today man |
| 20:38.47 | dwery | :D |
| 20:50.28 | *** join/#asterisk TheOpenSourcerer (n=TheOpenS@81-178-65-1.dsl.pipex.com) |
| 20:51.54 | *** part/#asterisk TheOpenSourcerer (n=TheOpenS@81-178-65-1.dsl.pipex.com) |
| 20:56.38 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 20:56.42 | howie | is the asterisk-addons folder in etc/asterisk by default or do i need to create one? |
| 20:57.03 | leifmadsen | howie: it's a separate checkout |
| 20:57.16 | leifmadsen | what do you mean in /etc/asterisk/ ? |
| 20:57.29 | leifmadsen | there is no asterisk-addons subdir for configuration in /etc/asterisk |
| 20:57.41 | howie | is there one somewhere else? |
| 20:57.49 | leifmadsen | no |
| 20:57.55 | leifmadsen | the configs just go in /etc/asterisk |
| 20:58.04 | howie | ok |
| 20:58.18 | howie | ty |
| 20:58.34 | *** join/#asterisk luckyaba (n=Lucky@ip72-194-215-55.sb.sd.cox.net) |
| 20:59.18 | howie | also this patch im adding for bluetooth it text on a webpage so nano paster and save as what? |
| 20:59.24 | howie | is* |
| 20:59.44 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194) |
| 21:00.15 | *** join/#asterisk jicksta (n=jicksta@c-98-210-108-28.hsd1.ca.comcast.net) |
| 21:04.44 | *** join/#asterisk jong2 (n=chatzill@65.100.10.89) |
| 21:06.38 | batphone | anyone know what "processNodeName" is used for in the XML config for a SIP loaded Cisco phone? |
| 21:07.00 | batphone | i am seeing documentation on voip-info indicating that this is where the PBX ip can go |
| 21:07.04 | batphone | but there are other places for that |
| 21:07.54 | morko757 | hello |
| 21:08.17 | batphone | it looks to be for skinny only |
| 21:08.23 | morko757 | I am wondering, in Asterisk, if there are no NAT issues, do RTP packets travel through the asterisk server |
| 21:08.31 | leifmadsen | XML config for Cisco phone? I've never had great luck using any of the 79x1 series phones on Asterisk -- the XML configs are cryptic at best |
| 21:08.38 | leifmadsen | morko757: potentially |
| 21:08.51 | leifmadsen | morko757: see "canreinvite" in sip.conf |
| 21:08.57 | morko757 | or do they bypass the server and both sip clients send to each other |
| 21:09.08 | morko757 | ok, I'll have a look |
| 21:09.15 | *** part/#asterisk Breyer (n=Breyer@ool-43540592.dyn.optonline.net) |
| 21:09.23 | leifmadsen | morko757: both scenarios can happen depending on configuration |
| 21:09.28 | leifmadsen | (and scenario) |
| 21:09.47 | morko757 | the reason why I ask is that I have a SIP client inside a network that has different routing rules than the asterisk server connecting to it |
| 21:10.10 | leifmadsen | routing is not asterisk's issue :) |
| 21:10.31 | morko757 | accordingly, if the client is sending back RTPs to the asterisk server, all is good, but if sending directly to the originating client, packets may be taking a different direction |
| 21:11.04 | morko757 | to make things more clear, the client has two internet connections: a slow one for surfing and another for Voip |
| 21:11.17 | morko757 | the Voip connection attaches directly to an asterisk server |
| 21:11.42 | *** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir) |
| 21:15.23 | *** join/#asterisk blackest_mamba (n=blackest@c-71-239-160-143.hsd1.il.comcast.net) |
| 21:16.20 | batphone | no more dial command in 1.6? |
| 21:16.24 | batphone | chan_oss is loaded |
| 21:17.49 | *** join/#asterisk CrazyTux[m] (n=Brandon@99-53-97-235.lightspeed.cyprtx.sbcglobal.net) |
| 21:19.13 | *** join/#asterisk howie (n=howie@71-95-220-206.dhcp.mtpk.ca.charter.com) |
| 21:22.53 | hesco | any ideas how to configure audacity to handle .gsm files? |
| 21:23.19 | *** join/#asterisk jdnWEST (n=jdn@mx1.westparkcom.net) |
| 21:24.17 | [TK]D-Fender | batphone: console dial <--- read the UPGRADE docs |
| 21:28.57 | aurax | [TK]D-Fender, what's up ? |
| 21:30.40 | aurax | Can anyone tell me please if nat=yes in sip trunk can cause broken connect string in sip header? |
| 21:31.03 | aurax | i was debugging my sip trunk and i saw that it sending internal ip in connect string (instead of externip) |
| 21:31.08 | [TK]D-Fender | aurax: You might want to consider looking. If not perhaps showing. |
| 21:31.17 | aurax | alright |
| 21:33.48 | aurax | http://pastebin.com/d37513b3e |
| 21:34.07 | aurax | 212.199.157.154 is the rtp server. |
| 21:36.02 | aurax | see, Reliably Transmitting (NAT) to 212.199.157.154:5060: <- this is wrong, transmitting nat to external ip ? |
| 21:37.09 | *** join/#asterisk tripps (n=tripps@76.31.197.242) |
| 21:37.42 | *** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose) |
| 21:38.44 | tripps | I have some 7960 cisco phones with recent SCCP firmware and can't get them to load SIP on them no matter what I try. I've done this many times without problems, but these never load the code or even attempt to download the OS79XX.TXT file, etc. Even on a standalone network with my laptop as DHCP/TFTP and the phone as the only other node. ideas?. |
| 21:39.30 | [TK]D-Fender | aurProbably. Whats at that IP? |
| 21:40.05 | [TK]D-Fender | aurax: Looks like an ITSP at which point yes, it IS bad to have that reporting NAT |
| 21:41.05 | aurax | This is my isp's IP. (Sip) |
| 21:41.33 | aurax | it means that my server sends to this ip invite with NAT information instead of the internal ip? |
| 21:41.38 | aurax | external ip* |
| 21:41.43 | [TK]D-Fender | auxaSo go fix your configs |
| 21:42.34 | aurax | I'm asking if i'm right |
| 21:42.44 | aurax | that might be the reason why i cannot send audio ? |
| 21:42.57 | [TK]D-Fender | [17:40]<[TK]D-Fender>aurax: Looks like an ITSP at which point yes, it IS bad to have that reporting NAT <-------------------- |
| 21:43.02 | aurax | ok ! |
| 21:43.02 | aurax | :) |
| 21:43.03 | aurax | thx mate |
| 21:43.04 | [TK]D-Fender | [17:41]<[TK]D-Fender>auxaSo go fix your configs <----------- |
| 21:43.14 | [TK]D-Fender | reaches for his ClueBat (tm) |
| 21:43.22 | aurax | eh |
| 21:43.32 | aurax | so just nat=no will fix me up? |
| 21:43.57 | [TK]D-Fender | aurax: At least that part |
| 21:44.38 | aurax | beside that canreinvite=no, any other configuration that i have to consider? |
| 21:45.29 | *** join/#asterisk wonderworld (n=w@ip-62-143-16-28.unitymediagroup.de) |
| 21:49.04 | *** join/#asterisk howie (n=howie@71-95-220-206.dhcp.mtpk.ca.charter.com) |
| 21:49.36 | [TK]D-Fender | aurax: Go fix all that and come back |
| 21:49.47 | howie | anyone advise me on installing chan_mobile for asterisk? |
| 21:51.34 | [TK]D-Fender | howie: http://www.voip-info.org/wiki/view/chan_mobile |
| 21:53.11 | *** join/#asterisk Iamnach0 (i=Iamnacho@ip174-70-137-120.ks.ks.cox.net) |
| 21:53.53 | aurax | [TK]D-Fender fixed :) |
| 22:04.29 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 22:11.44 | hesco | any ideas how to configure audacity to handle .gsm files? I used the Record() app to record some ivr clips which I'd like to trim and save back as gsm. I found some advice to use File->Import->RawData in audacity, but I'm not sure what format I should use on the import. |
| 22:12.13 | beek | hesco: If nothing else, use sox to convert it to wav, edit in audacity, then convert it back. |
| 22:12.31 | hesco | ok, trying that, then |
| 22:14.54 | hesco | I'm getting: 'sox soxio: Can't open input file `asterisk-recording23.gsm': unknown file type `gsm'' |
| 22:15.00 | hesco | any ideas? |
| 22:16.45 | aurax | hesco, sound forge? |
| 22:17.20 | beek | hesco: show me the command line you typed |
| 22:21.25 | *** join/#asterisk ehsjoar (n=ehsjoar@c-24-9-91-203.hsd1.co.comcast.net) |
| 22:22.07 | leifmadsen | hesco: sox probably needs to be compiled to support the GSM format |
| 22:26.30 | KavanS | hesco, use wave with audacity....then use asterisk to convert from wav to gsm |
| 22:26.34 | KavanS | hesco, that's what I used...and it worked fine |
| 22:27.00 | KavanS | I got the wav files from a friend who recorded them for me....then I cleaned them up a little bit in audacity...then when I was ready I performed the final conversion with asterisk |
| 22:32.06 | *** join/#asterisk pheller (n=pheller@173.48.205.77) |
| 22:32.52 | pheller | anyone have any hints for how to deal with out-of-order RTP frames carrying rfc2833 dtmf data? |
| 22:33.54 | hesco | ok, I'm now working my way through the ./configure --help to compile sox from source. |
| 22:34.15 | hesco | I tried sox my_file.gsm my_file.wav |
| 22:34.23 | beek | hesco what distribution? |
| 22:34.26 | hesco | just guessing on that one, |
| 22:34.30 | hesco | debian Lenny |
| 22:35.20 | beek | I find that hard to believe that they don't have gsm enabled. CentOS5 does. |
| 22:35.30 | beek | (the package I got from CentOS does) |
| 22:36.19 | hesco | did I misuse the command? I did not really fully read the man page, just tried it from memory |
| 22:36.51 | *** join/#asterisk SkramX (n=SkramX@cpe-70-112-25-138.austin.res.rr.com) |
| 22:38.13 | beek | that looks fine. |
| 22:38.47 | beek | hesco: want to email me one of the files and I can see if it converts here? |
| 22:38.58 | *** join/#asterisk bbryant (n=brett@c-68-59-22-114.hsd1.sc.comcast.net) |
| 22:39.54 | hesco | sure, share an email addr and I'll send it your way |
| 22:48.14 | FreakGuard | I suppose pbxes is running asterisk too, so I may ask SIP questions here too - I'm trying to phone an external SIP via my extention, but I just get a small tone and nothing else |
| 22:49.00 | FreakGuard | and a hangup afterwards |
| 22:54.55 | *** join/#asterisk ArchGT (n=ArchGT@190.149.21.190) |
| 23:07.04 | *** join/#asterisk Iamnacho (i=Iamnacho@ip174-70-137-120.ks.ks.cox.net) |
| 23:13.09 | *** join/#asterisk SaiSoma (n=SaiSoma@74.167.136.30) |
| 23:23.13 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 23:25.15 | *** join/#asterisk sjobeck (n=Adium@208-151-246-203.dq1sn.easystreet.com) |
| 23:31.44 | *** part/#asterisk sjobeck (n=Adium@208-151-246-203.dq1sn.easystreet.com) |
| 23:32.38 | *** join/#asterisk serph (n=serph@CPE001f5b006e5e-CM001371144daa.cpe.net.cable.rogers.com) |
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| 23:41.07 | *** join/#asterisk SaiSoma (n=SaiSoma@74.167.136.30) |
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| 23:57.48 | *** join/#asterisk vegbox (n=kevinle@24-205-54-7.dhcp.gldl.ca.charter.com) |
| 23:58.11 | vegbox | if i wanted to use the same house phones i am using now, what kind of an adapter can i use to hook up to a asterisk box |
| 23:58.27 | *** join/#asterisk propellerhead (n=yogurt2u@host113.190-230-234.telecom.net.ar) |