| 00:00.09 | *** part/#asterisk HockeyInJune (i=HockeyIn@pool-68-161-152-103.ny325.east.verizon.net) |
| 00:00.21 | allen_s | jbot: thanks for the link, always trying to learn! |
| 00:00.21 | jbot | allen_s: bitte |
| 00:00.22 | lerat | I have an asterisk server with a 4 slot PRI card so my 3 PRI are connected to it |
| 00:00.44 | lerat | but they came from different region |
| 00:00.50 | JT | lerat: yes we understand that bit, what we are unsure of, is if your 3 PRIs connect to the one telco switch, or 3 different ones |
| 00:00.51 | elriah | lerat: Digium's card? |
| 00:01.18 | JT | lerat: if they're from different exchanges, they're definately different switches |
| 00:01.44 | lerat | yeah different exchange and yes Digium |
| 00:03.04 | JT | lerat: they're not on the same switch then |
| 00:03.38 | Strom_C | lerat: figure out which PRI you would like to have the fewest frame slips on, and set that as primary :) |
| 00:04.08 | JT | the rest to 0 |
| 00:04.36 | Strom_C | no, the rest to secondary and tertiary |
| 00:04.46 | Strom_C | only set to 0 if you're going to be providing timing on that span |
| 00:04.49 | lerat | sorry but... frame slips ??? i m probably not as experience as you are :) |
| 00:04.51 | *** join/#asterisk ms345 (n=mike_sim@71.56.100.254) |
| 00:05.01 | Strom_C | lerat: do you know how T1 works? |
| 00:05.16 | JT | that makes sense |
| 00:05.18 | lerat | it s a 23 channel with 1 D |
| 00:05.37 | Strom_C | lerat: please please read the "telephony 101" document I linked to earlier :) |
| 00:06.59 | lerat | ok , is it really useful and understandable ??? |
| 00:07.05 | Strom_C | extremely |
| 00:07.08 | lerat | ok |
| 00:07.10 | lerat | :) |
| 00:07.16 | lerat | soory for asking |
| 00:07.57 | lerat | can you breif me a bit? |
| 00:08.04 | Strom_C | on what? T1? |
| 00:08.14 | lerat | on what it talk |
| 00:08.25 | JT | consulting ;) |
| 00:08.27 | lerat | it s only on T1 and PRI config |
| 00:08.42 | lerat | ??? |
| 00:08.52 | JT | lerat: you need to understand those |
| 00:08.53 | Strom_C | lerat: it sounds like you don't know much about T1, PRI, or traditional telephony in general. |
| 00:09.03 | Strom_C | I suggest you read that document. |
| 00:09.06 | thekidrio | anyone know a good proggie for recording gsm files? |
| 00:09.14 | thekidrio | or should i make a wav and convert to gsm? |
| 00:09.16 | Strom_C | thekidrio: um, Record()? :) |
| 00:09.22 | Strom_C | thekidrio: you can also use wav |
| 00:09.28 | Strom_C | do you have a good reason for using GSM? |
| 00:09.37 | thekidrio | Strom_C: i was thinking of something outside of asterisk to record |
| 00:09.49 | thekidrio | Strom_C: not really just the size of the file compared to wav |
| 00:10.01 | Strom_C | gsm sounds like crap compared to wav |
| 00:10.13 | thekidrio | yeah i agree heh |
| 00:10.21 | thekidrio | just nice and tiny |
| 00:10.30 | lerat | i know pretty much everything on the old sytem config like norstar |
| 00:10.41 | Strom_C | just use 8khz 16-bit mono wav files |
| 00:10.49 | JT | they're prompts... how much disk space can they take? |
| 00:10.54 | thekidrio | thanks Strom_C, thats what i wanted to hear actually |
| 00:10.55 | lerat | but i didn work much with PRI andf T1 |
| 00:10.58 | JT | i record all phone calls as wav |
| 00:10.58 | Strom_C | lerat: please stop protesting and just read the document |
| 00:11.44 | Strom_C | :) |
| 00:11.45 | lerat | ok :) |
| 00:11.45 | thekidrio | jt, haha i just dont like wasting space no matter what eh |
| 00:11.45 | thekidrio | old habits die hard |
| 00:11.45 | lerat | but i m not protesting :) |
| 00:11.45 | JT | thekidrio: even for no reason? wav is best for transcoding to different formats too |
| 00:11.53 | ms345 | could a bad PRI card cause asterisk to not start? " Junk at the beginning 49443303" crash on startup? |
| 00:12.22 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 00:12.28 | thekidrio | jt, well the reason is hardcoded in to me, i come from days with serious memory limits and working with embedded systems where you only had X memory no matter what |
| 00:12.41 | thekidrio | just an old habit that gets me in trouble sometimes heh |
| 00:12.45 | lerat | ok thk Strom_C |
| 00:13.14 | JT | thekidrio: fair enough |
| 00:13.19 | JT | i don't advocate waste |
| 00:13.30 | JT | but in this case, waste seems to be recording prompts as gsm :) |
| 00:13.54 | thekidrio | hehe yeah, wasting my time if wav is fine |
| 00:14.11 | thekidrio | that makes my life a lot easier as we already have stuff for recording to wav |
| 00:15.58 | *** join/#asterisk coppice (n=chatzill@249.193.17.210.dyn.pacific.net.hk) |
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| 00:27.47 | *** join/#asterisk tuan_modulis (n=chatzill@hvmoduli.enter-net.com) |
| 00:29.51 | tuan_modulis | I'm currently using asterisk-1.2.16 ... does call-limit work for u guys when doing dial with an extension that has call waiting activated? |
| 00:30.24 | tuan_modulis | for some reason, I can't get incominglimit/outgoinglimit to work either |
| 00:33.41 | BLAHsterC | PROGRESS with cause code 41 received |
| 00:33.47 | BLAHsterC | that is VM on zap ? |
| 00:36.55 | *** join/#asterisk Katty (n=Administ@66-189-236-10.dhcp.cpgr.mo.charter.com) |
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| 00:51.43 | PaulTech85 | Anyway when placing a outbound call via Zap inteface to detect you hit a voicemail and to wait for tone |
| 00:51.51 | PaulTech85 | or is it not sending dtmf tones to alert us? |
| 00:52.25 | Katty | sure is quiet tonight. |
| 00:52.30 | Katty | or is quiet every night now? |
| 00:52.47 | JT | it's the middle of the day here |
| 00:52.59 | tuan_modulis | it's usually quiet when i post my ambiguous problems |
| 00:53.01 | Katty | i see, i see. |
| 00:53.22 | PaulTech85 | thx |
| 00:53.43 | Katty | works been a bit slow lately (thank the powers that be!) so i've been playing wow lately ;) |
| 00:59.19 | ltdwk | playing wow at work... been there done that.... but those days are done |
| 01:00.07 | J4k3- | wow at work... sounds like the 90s (Except back then it was quakeworld @ work) |
| 01:00.14 | *** join/#asterisk shodan- (n=shodan@ip212.99-113-216.pppoe4.joliette.intermonde.net) |
| 01:00.41 | thekidrio | woot quake world |
| 01:00.47 | thekidrio | qwtf player here |
| 01:00.58 | thekidrio | as [n]l0b0 |
| 01:03.58 | *** join/#asterisk HockeyInJune (n=HockeyIn@pool-68-161-146-98.ny325.east.verizon.net) |
| 01:04.08 | PaulTech85 | How do you flush the manager eventq ? :P |
| 01:05.21 | *** join/#asterisk flenders (n=fserto@unaffiliated/flenders) |
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| 01:05.31 | *** part/#asterisk techie (n=techie@c-67-181-184-170.hsd1.ca.comcast.net) |
| 01:06.01 | flenders | hey, is there a "status" command on the CLI? |
| 01:07.26 | Qwell | status for/of what? |
| 01:08.12 | flenders | of the server itself |
| 01:08.40 | Qwell | that's kinda vague, don't you think? |
| 01:09.04 | flenders | yes, agree, so I guess that's a no? |
| 01:09.24 | *** join/#asterisk boch (n=fran@190.48.195.167) |
| 01:09.24 | Katty | <CLI> Show status |
| 01:09.28 | flenders | I'm writing a bunch of nagios scripts to monitor a few things on asterisk |
| 01:09.28 | Katty | <SERVER> I HATE YOU |
| 01:09.34 | boch | anyone using MYSQL() in 1.4.1 ? |
| 01:11.58 | *** join/#asterisk dseeb_ (n=dcb@CPE-58-169-73-237.vic.bigpond.net.au) |
| 01:13.42 | Qwell | boch: func_odbc kinda made it useless |
| 01:20.57 | Znuff | Katty, common conversation |
| 01:28.59 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 01:29.03 | tuan_modulis | I get something like: I'll give you 99% of all functionalities, and will withhold the last percent to keep things interesting |
| 01:29.25 | tuan_modulis | now im trying to negociate a bit with the source code |
| 01:36.30 | *** join/#asterisk litage (n=nick@203.220.55.70) |
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| 01:37.59 | *** join/#asterisk Hansin321 (n=Eric@67.174.180.163) |
| 01:38.05 | Hansin321 | #openpbx |
| 01:42.14 | tuan_modulis | woohoo problem solves |
| 01:42.16 | tuan_modulis | d |
| 01:42.31 | tuan_modulis | once again, just a stupid configuration mistake |
| 01:42.49 | Znuff | fix my problem! |
| 01:43.06 | tuan_modulis | what is it? |
| 01:43.12 | tuan_modulis | maybe I can pass on some of my luck |
| 01:43.20 | Znuff | Getting a 415 error from the SIP provider:) |
| 01:43.25 | Znuff | And it's not a codec related issue |
| 01:43.34 | tuan_modulis | which provider? |
| 01:43.45 | Znuff | own $custom provider |
| 01:43.53 | Znuff | (a harris PBX) |
| 01:44.08 | tuan_modulis | ack, i've only woirked with unlimitel and openface |
| 01:44.40 | Znuff | yeah, seems nobody has a clue on what my problem could be :) |
| 01:45.10 | tuan_modulis | I wish the best of luck to u though... or a healthy dose of reverse engineering |
| 01:45.34 | Znuff | debug gets me nowhere :-( |
| 01:50.26 | *** join/#asterisk shinux__ (n=shinux@195.166.241.139) |
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| 01:52.26 | dasenjo | Hi! I have a question, is there any way of configure allowed/disallowed codecs for a zap phone? |
| 01:54.39 | *** join/#asterisk Dovid (n=Dovid@l192-117-114-1.broadband.actcom.net.il) |
| 01:55.40 | PaulTech85 | Handling events in php sucks |
| 01:59.19 | *** join/#asterisk Mw3 (i=mw3@ip59934bd1.bp-1031.rubicom.hu) |
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| 02:07.25 | Znuff | JT, you still around? |
| 02:08.13 | *** join/#asterisk roguebug (n=roguebug@dip-109-055.bras.dsl.breisnet.com) |
| 02:08.16 | roguebug | hey |
| 02:08.36 | roguebug | just noticed a --with-zaptel switch in configure of asterisk 1.4. |
| 02:09.24 | JT | Znuff: yes |
| 02:09.25 | roguebug | i always built asterisk before zaptel (with asterisk 1.2.x anyways, i never tried a 1.4). any reason to first build zaptel? doesn't it need asterisk to build? |
| 02:09.50 | JT | roguebug: then you've always built it wrong |
| 02:10.06 | Znuff | JT, think my 415 response could be related to amaflags? |
| 02:10.11 | JT | zaptel is a dependancy of asterisk |
| 02:10.25 | JT | libpri is a dependancy of zaptel, only if you use pri |
| 02:10.32 | JT | dependancies should always be built first |
| 02:10.33 | roguebug | ok good to know (always worked anyways) |
| 02:10.42 | JT | roguebug: maybe you didn't use zaptel? |
| 02:10.46 | Znuff | ofcourse, if you don't have devices that require zaptel..... |
| 02:10.47 | JT | Znuff: no idea |
| 02:10.49 | roguebug | just ztdummy |
| 02:11.04 | JT | ztdummy shouldn't even work if you build asterisk first |
| 02:11.35 | *** join/#asterisk bjohnson (n=bjohnson@i209-195-93-223.cia.com) |
| 02:11.40 | roguebug | well, either it does or at least neither asterisk nor the build complained and i always had a ztdummy module that was loadable |
| 02:14.02 | JT | did you actually use ztdummy for anything? |
| 02:14.24 | Znuff | perhaps you installed asterisk again after building zaptel and figuring out it didn't work :) |
| 02:14.42 | roguebug | JT: just as a fake time ref i believe |
| 02:14.44 | Znuff | common "but it worked for me!" situation |
| 02:15.00 | JT | roguebug: ok, but, What did you use it for? |
| 02:15.09 | roguebug | Znuff: actually most everything except maybe moh |
| 02:15.16 | roguebug | would work without |
| 02:15.22 | JT | roguebug: do you use IAX2 trunking? |
| 02:15.26 | roguebug | nah |
| 02:15.32 | JT | roguebug: do you use MeetMe conferences? |
| 02:15.38 | roguebug | nah |
| 02:15.43 | JT | then you don't need ztdummy |
| 02:15.46 | JT | nothing uses it |
| 02:15.51 | JT | except those |
| 02:16.01 | JT | asterisk would not have had access to it |
| 02:16.02 | roguebug | i believe moh or mp3 or so asked for it |
| 02:16.22 | JT | ah but it did not work? |
| 02:16.47 | roguebug | sure it did, like a charm (after adding zaptel) |
| 02:17.08 | Znuff | liar :P |
| 02:18.14 | roguebug | i have a mp3 moh by default running as soon as i'm called, with the ring tones mixed on top, so if it didn't work, i would have noticed it quite quickly |
| 02:19.15 | roguebug | but hey maybe it wasn't moh |
| 02:19.18 | *** join/#asterisk juanjoc (n=juanjoc@248-32-235-201.fibertel.com.ar) |
| 02:19.24 | Znuff | like JT said, don't think you need zaptel for mp3 |
| 02:19.50 | roguebug | i don't exactly remember which one it was, just remember there was one that wanted it and that's why i always installed it... after asterisk |
| 02:20.23 | roguebug | but this time, as you told me i have to build zaptel first, i do just that |
| 02:20.36 | Znuff | you couldn't possibly build that module before zaptel |
| 02:21.05 | Znuff | if it required it |
| 02:21.08 | *** join/#asterisk anthonyl (n=anthonyl@r74-193-79-81.pfvlcmta01.grtntx.tl.dh.suddenlink.net) |
| 02:21.24 | roguebug | dunno, i just remember i wanted to get rid of a warning message asking for a ztdummy |
| 02:21.46 | roguebug | for time ref |
| 02:22.09 | roguebug | i never had any zaptel hardware, so i never used the zaptel module |
| 02:23.19 | roguebug | basically, i use chan_capi, chan_sip, some moh, wav playback and agi, and cdr_pgsql and that's about all. a rather simple little config |
| 02:24.00 | Znuff | probably the default configuration of asterisk required that :) |
| 02:24.19 | roguebug | might very well be |
| 02:24.46 | Znuff | I get warning everywhere about things |
| 02:25.15 | roguebug | yeah, the default config is a lil modules greedy |
| 02:34.57 | PaulTech85 | Haha |
| 02:35.00 | PaulTech85 | fucking event API |
| 02:35.05 | PaulTech85 | it does a HangUp for each command? |
| 02:35.32 | PaulTech85 | How do you tell a call is really over? |
| 02:35.33 | PaulTech85 | Ya dont? |
| 02:36.08 | *** join/#asterisk mcab (n=mb@66.195.193.151) |
| 02:36.33 | sbingner | you call hangup on it |
| 02:36.54 | JerJer | you will get a CDR |
| 02:37.08 | sbingner | otherwise it depends on what you were doing right before, right? |
| 02:38.43 | anonymouz666 | JerJer mobile! |
| 02:38.44 | anonymouz666 | hehe |
| 02:39.00 | JerJer | not mobile - in the bat cave |
| 02:41.20 | Znuff | so, if I make a call from the outside world to an voip phone, this is what I get: http://pastebin.ca/383077 |
| 02:41.32 | Znuff | phone rings but call dies as soon as I pick up |
| 02:42.30 | Znuff | is there a problem if the VoIP phone can't contact the SIP provider directly? |
| 02:44.17 | ltdwk | make sure if your sip phone is NAT'd that you dont do silly things with canreinvite=no |
| 02:44.47 | *** join/#asterisk techie (n=techie@c-67-181-184-170.hsd1.ca.comcast.net) |
| 02:45.19 | Znuff | uh |
| 02:45.27 | Znuff | my phone is not NAT-ed |
| 02:45.42 | ltdwk | okay, is it routable to the sip provider? |
| 02:45.46 | Znuff | but it can't communicate with the SIP provider directly, but it can communicate with Asterisk |
| 02:46.05 | ltdwk | if it's not routable, make sure you use canreinvite=no |
| 02:46.18 | Znuff | hmmm, lemme try making it routable |
| 02:48.15 | ltdwk | Not saying it's your problem, but asterisk will try to REINVITE the sip peers to avoid bridging if possible |
| 02:48.34 | ltdwk | and if there's no routing between them the audio stream will be lost |
| 02:49.47 | Znuff | I see |
| 02:49.59 | Znuff | oh, stupid me, I forgot to set a default gateway on the SIP provider |
| 02:50.37 | Znuff | aaaaargh The IP settings have been configured in the 20-20 editor and cannot be |
| 02:50.37 | Znuff | changed here. |
| 02:53.04 | Znuff | I've got canreinvite=no |
| 02:57.20 | *** part/#asterisk omarc55 (n=omar@dsl092-214-151.atl1.dsl.speakeasy.net) |
| 03:00.42 | Znuff | ltdwk, so, in other words, asterisk is trying to bridge, right? -- Packet2Packet bridging SIP/5060-08232388 and SIP/23003-08228400 |
| 03:00.57 | Znuff | and it doesn't work, right? |
| 03:03.47 | *** join/#asterisk HockeyInJune (n=HockeyIn@pool-68-161-165-186.ny325.east.verizon.net) |
| 03:04.10 | Znuff | or... does it try to do a RTP between asterisk and my sip? |
| 03:04.13 | Znuff | X-asterisk-Info: SIP re-invite (External RTP bridge) |
| 03:06.13 | ltdwk | with a re-invite the RTP is sent direct |
| 03:06.20 | ltdwk | there is no bridging |
| 03:06.32 | ltdwk | no internal bridging anyway |
| 03:07.03 | Znuff | I'm confused |
| 03:07.21 | Znuff | RTP basically means the peers connect straight with each other, right? |
| 03:07.39 | Znuff | Asterisk only does the initial "negotation" |
| 03:07.47 | Znuff | nociation |
| 03:07.49 | Znuff | damn |
| 03:07.51 | Znuff | negociation |
| 03:08.09 | Znuff | uh, whatever is it spelled |
| 03:08.39 | Znuff | ltdwk, but bridging means that the connection is done trough asterisk, right? |
| 03:09.05 | JT | rtp is the traffic |
| 03:09.11 | JT | the actual voip data goes over rtp |
| 03:09.15 | JT | sip only does signalling |
| 03:11.07 | Znuff | In this case -- Packet2Packet bridging SIP/5060-0822ef78 and SIP/23003-0823b6c0; means the VoIP traffic goes trough asterisk, right? |
| 03:11.27 | Znuff | ~thebook |
| 03:11.28 | jbot | methinks thebook is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 03:11.36 | *** join/#asterisk shinux__ (n=shinux@195.166.241.139) |
| 03:11.39 | Znuff | jbot, what are you? |
| 03:11.40 | jbot | I am a blootbot. For more info see <blootbot> |
| 03:11.46 | Znuff | heh, 100% sure |
| 03:11.53 | *** join/#asterisk freestyle (i=freestyl@2002:c951:8358:0:0:0:c951:8358) |
| 03:13.59 | *** join/#asterisk shinux__ (n=shinux@195.166.241.139) |
| 03:15.51 | JT | Znuff: umm i'm not sure, haven't seen that message before |
| 03:16.07 | JT | pastebin sip.conf already to pastebin.ca, censor passwords |
| 03:16.50 | Znuff | JT, should I remove comments too? |
| 03:17.00 | JT | no |
| 03:17.07 | Znuff | uh |
| 03:17.14 | Znuff | it's kind of the default sip.conf with bits added :> |
| 03:17.42 | JT | yeah |
| 03:17.46 | JT | and? :) |
| 03:18.03 | ManxPower | Znuff: if you have audio problems set canreinvite=no for the devices in sip.conf |
| 03:18.19 | ManxPower | and upgrade to 1.4.1 |
| 03:18.23 | Znuff | got 1.4.1 :) |
| 03:18.26 | ManxPower | I assume you are on 1.4.0 right now |
| 03:18.31 | ManxPower | ah, OK |
| 03:18.34 | Znuff | upgraded last night |
| 03:18.37 | Znuff | don't have audio problems |
| 03:18.40 | JT | he's been trying to solve these problems for days but hasn't shown us his sip.conf |
| 03:18.43 | JT | it's about time :) |
| 03:18.49 | Znuff | well, you never asked |
| 03:19.21 | JT | i still think it's something in the harris pbx set wrong |
| 03:20.15 | *** join/#asterisk Iamnacho (i=Iamnacho@ip68-103-153-140.ks.ok.cox.net) |
| 03:20.45 | Znuff | JT, http://pastebin.ca/383114 |
| 03:20.55 | Znuff | JT, thing is that it worked with a previous setup |
| 03:21.23 | Znuff | but don't ask me how it was done :) |
| 03:22.06 | *** join/#asterisk shinux__ (n=shinux@195.166.241.139) |
| 03:23.17 | Znuff | ofcourse, probably this doesn't work is because of my newbie-ness in asterisk... |
| 03:25.04 | JT | setup... with what? |
| 03:26.00 | Znuff | uh? |
| 03:27.51 | *** join/#asterisk shinux__ (n=shinux@195.166.241.139) |
| 03:28.16 | JT | what was the previous setup |
| 03:28.23 | JT | i assume it was not identical |
| 03:28.49 | Znuff | same stuff |
| 03:28.56 | Znuff | http://i5.tinypic.com/2wozspk.png <- same idea |
| 03:35.44 | JT | Znuff: umm, so was there a username and password set on the harris sip entry? |
| 03:35.44 | Znuff | nope |
| 03:35.45 | JT | and where is the sip phone entry? |
| 03:35.45 | *** join/#asterisk CrazyTux (n=CrazyTux@64.95.219.140) |
| 03:35.45 | Znuff | users.conf |
| 03:35.47 | JT | Znuff: are you sure it accepts unauthenticated requests? |
| 03:35.49 | JT | arrgh |
| 03:36.02 | *** join/#asterisk InHisName (n=Administ@c-68-38-105-1.hsd1.pa.comcast.net) |
| 03:36.08 | Znuff | JT, yes, I'm sure |
| 03:36.13 | Znuff | JT, want users.conf too? |
| 03:37.02 | Znuff | it it'd required username/password it wouldn't connect at all, right? |
| 03:37.16 | JT | it would provide an error |
| 03:37.30 | *** join/#asterisk mercestes (n=merceste@inet.hou.devry.net) |
| 03:37.30 | JT | have you tried connecting with a softphone? |
| 03:37.40 | Znuff | directly to the PBX? |
| 03:37.46 | Znuff | Harris I mean? |
| 03:37.58 | JT | yes |
| 03:38.05 | Znuff | uhm, nope =/ |
| 03:38.25 | Znuff | any console softphone clients? :-) |
| 03:38.34 | JT | not that i know of |
| 03:39.08 | Znuff | You mean instead of connecting trough asterisk, right? |
| 03:39.17 | JT | yes |
| 03:39.18 | PaulTech85 | sipsak |
| 03:39.26 | JT | otherwise what are the errors again? |
| 03:39.27 | *** join/#asterisk bmg505 (n=leon@196.209.250.100) |
| 03:40.10 | Znuff | JT, well, when I try to dial-out, I get a 415 error, media not supported, you know |
| 03:40.49 | ManxPower | Znuff: you really need to do a sip debug and look at the codecs the other side says it supports |
| 03:41.15 | JT | Znuff: how about a sip debug on ONLY the ip of the harris |
| 03:41.15 | Znuff | and when I try to call a sip phone behind asterisk from my mobile phone or any other outside network, I get a ring, but when I pick up I get disconnected |
| 03:41.32 | anonymouz666 | anyone in here already use SIPP? |
| 03:41.34 | JT | and also any other errors in console, with verbosity at least 10 |
| 03:41.35 | Znuff | JT, hmm, sure, just tell me how to do it :) |
| 03:41.48 | JT | help sip debug |
| 03:42.21 | Znuff | SIP Debugging Enabled for IP: 10.0.0.1 |
| 03:42.21 | Znuff | done |
| 03:42.22 | ThoMe | Good Morning! |
| 03:42.40 | anonymouz666 | good morning? |
| 03:42.42 | Znuff | let's try to dial-out |
| 03:42.49 | anonymouz666 | its still 00:42 |
| 03:43.05 | anonymouz666 | i need to sleep to wake up 8 am tomorrow |
| 03:43.36 | Znuff | it's 5:43 here |
| 03:43.48 | Znuff | AM |
| 03:44.09 | ThoMe | anonymouz666: here is 4:44 :-9 |
| 03:44.12 | ThoMe | AM. |
| 03:44.36 | ThoMe | Germany.. |
| 03:44.43 | Znuff | JT, this is when I try to dial my cellphone from a voip phone: http://pastebin.ca/383142 |
| 03:46.15 | Znuff | and this is from my cellphone to voip phone behind asterisk: http://pastebin.ca/383144 |
| 03:46.25 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 03:47.00 | *** part/#asterisk mmurdock (n=vircuser@mail.kimballequipment.com) |
| 03:47.22 | *** join/#asterisk mmurdock (n=vircuser@c-24-10-190-87.hsd1.ut.comcast.net) |
| 03:47.59 | JT | Znuff: line 72 and 73 in sip.conf |
| 03:48.01 | JT | replace |
| 03:48.02 | JT | # |
| 03:48.02 | JT | disallow=all ; First disallow all codecs |
| 03:48.02 | JT | # |
| 03:48.03 | JT | allow=alaw ; Allow codecs in order of preference |
| 03:48.08 | Znuff | yes |
| 03:48.09 | JT | with allow=all |
| 03:48.13 | *** join/#asterisk Giofe (n=opera@190.81.4.169) |
| 03:48.35 | Znuff | reloaded |
| 03:49.50 | Znuff | JT, same thing: http://pastebin.ca/383147 |
| 03:51.09 | anonymouz666 | I want FX Channel here! |
| 03:51.11 | anonymouz666 | grrrrr |
| 03:51.17 | *** join/#asterisk fakhir (i=fakhir@unaffiliated/fakhir) |
| 03:51.24 | Hmmhesays | what? |
| 03:51.31 | anonymouz666 | FX TV Channel |
| 03:51.38 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
| 03:52.13 | anonymouz666 | right now - the FX girl |
| 03:52.25 | Znuff | wscreenshots? :) |
| 03:53.06 | Hmmhesays | you want to watch "the transporter" |
| 03:53.29 | anonymouz666 | lol |
| 03:54.04 | Hmmhesays | good movie, but why would you want to watch it on a cable tv |
| 03:54.08 | Hmmhesays | with commercials n stuff |
| 03:54.37 | JT | the transporter is a good movie? :o |
| 03:54.54 | Hmmhesays | I thought so |
| 03:55.00 | Hmmhesays | big booms hot chicks |
| 03:55.16 | JT | it's action packed |
| 03:55.23 | JT | but not something you can watch again and again |
| 03:55.56 | Znuff | how the hell do you dial with that sipsak thingie? |
| 03:56.39 | PaulTech85 | You dont |
| 03:56.42 | PaulTech85 | well you could |
| 03:56.44 | PaulTech85 | but I wouldnt |
| 03:56.48 | Znuff | oh? |
| 03:56.56 | PaulTech85 | its just a test base |
| 03:57.36 | Znuff | all I get is sip 501 / not implemented |
| 03:57.36 | PaulTech85 | SIPp may be better |
| 03:57.39 | PaulTech85 | it wont dial either |
| 03:57.40 | PaulTech85 | hehe |
| 03:57.47 | PaulTech85 | actually SIPp might |
| 03:58.03 | Znuff | I'm trying ./sipsak -s sip:0748881658@10.0.0.1 |
| 03:59.08 | Znuff | ;et |
| 03:59.12 | Znuff | let's get sipp |
| 04:03.15 | Znuff | JT, any other ideas? |
| 04:06.43 | JT | yes |
| 04:06.50 | JT | get a big fat harris manual out |
| 04:06.55 | JT | start reading, and doing stuff on it |
| 04:06.56 | JT | :) |
| 04:07.25 | Znuff | heh :) |
| 04:08.38 | Znuff | wonder if I just slap the old config files over this... |
| 04:10.19 | *** join/#asterisk terlouw (n=FreePBX5@proxy.amsterdam.intruder.nl) |
| 04:10.37 | terlouw | people!! anyone know a good working click2call script for asterisk ? :D |
| 04:11.24 | *** part/#asterisk dasenjo (i=dasenjo@gateway/tor/x-e06aa8d30bfe11d2) |
| 04:11.39 | Strom_C | yes, people!! (as opposed to addressing a marauding band of stray cats) |
| 04:14.37 | Znuff | I've dialed my phone number _so_ many times that I've started to memorize the DMTF tones |
| 04:15.08 | Strom_C | you mean you haven't already learned to decode DTMF by ear? |
| 04:15.17 | Strom_C | lame. |
| 04:15.29 | ManxPower | Egads! Catch up, Znuff! |
| 04:15.29 | Strom_C | :) |
| 04:15.40 | Znuff | heh |
| 04:15.43 | ManxPower | Hell, Strom_C can whistle DTMF |
| 04:15.58 | Strom_C | woot, yes |
| 04:18.45 | Znuff | so, JT, having suggested to take out the big harris book, am I guess that my setup is okay-ish? my asterisk setup, I mean |
| 04:19.28 | Znuff | configuration |
| 04:19.47 | *** join/#asterisk dseeb_ (n=dcb@CPE-58-169-130-113.vic.bigpond.net.au) |
| 04:20.53 | *** join/#asterisk NZheretic (n=NZhereti@203-118-142-135.pppoe.ihug.co.nz) |
| 04:21.41 | NZheretic | This may be the wrong channel but, does anyone here have recent experience of getting soft modems to work under Linux? |
| 04:21.57 | *** join/#asterisk sbingner (n=sam@pdpc/supporter/sustaining/sbingner) |
| 04:22.09 | Znuff | wow, soft modems |
| 04:22.31 | NZheretic | Yes they suck |
| 04:22.39 | NZheretic | Yes I hate them too. |
| 04:22.43 | *** part/#asterisk BSDTech (n=RNeese@ppp-69-239-113-101.dsl.irvnca.pacbell.net) |
| 04:22.44 | NZheretic | But... |
| 04:22.56 | Znuff | no buts in here |
| 04:22.59 | JT | Znuff: seems roughly okay... but depends on the harris |
| 04:23.09 | NZheretic | External serial modems are getting rare on the ground in NZ. |
| 04:23.10 | Strom_C | for god's sake man, just go to the local computer junk store and get yourself a Hayes serial modem :) |
| 04:23.32 | Znuff | JT, good to know, I'll get the big guns / harris manual out |
| 04:23.44 | Znuff | tomorow |
| 04:23.46 | *** join/#asterisk lowlevel (n=Stuart@CPE000e0c057fad-CM000f9f7d6742.cpe.net.cable.rogers.com) |
| 04:24.17 | NZheretic | New External modem $63+ NZ ( $44 US ) and hard to find, Second hand ones getting much harder to find. |
| 04:24.27 | JT | Znuff: like the no logon thing is sus, but that's not what the error says |
| 04:24.42 | JT | NZheretic: hard? |
| 04:24.45 | Znuff | holy crap! o.O |
| 04:24.47 | Znuff | http://www.franklin800.com/F-Telecom.htm |
| 04:24.49 | JT | NZheretic: don't you guys have ebay/etc |
| 04:24.54 | Znuff | I'm guess that's a picture of the manuals |
| 04:25.01 | Znuff | and yes, it's that big |
| 04:25.23 | NZheretic | JT : www.trademe.co.nz , last two modems I got were junkers. |
| 04:25.42 | *** join/#asterisk djs_2_6 (n=djstillm@cpe-071-077-048-198.nc.res.rr.com) |
| 04:25.44 | JT | buy from ebay australia then :P |
| 04:25.45 | JT | man |
| 04:25.47 | Znuff | there are 8 bible-sized manuals |
| 04:25.53 | JT | modems are easy to get |
| 04:25.55 | *** join/#asterisk stkn__ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 04:26.01 | JT | i have 92 i think |
| 04:26.05 | JT | but none are good for you |
| 04:26.24 | NZheretic | Non-NZ modems not NZ telepermited. |
| 04:26.48 | JT | lol who cares about that bullcrap |
| 04:26.56 | JT | just makes money from the government for testing |
| 04:27.01 | NZheretic | It really sucks, but I need an interal modem chipset that works ( abet with drivers added ). |
| 04:27.14 | JT | why internal? no wonder they're junk |
| 04:27.28 | NZheretic | New External modem $63+ NZ ( $44 US ) and hard to find, Second hand ones getting much harder to find. |
| 04:27.45 | JT | err, but i reckon they're easy to get |
| 04:27.48 | Strom_C | NZheretic: dunno, i can find them here in the states for $5ish |
| 04:27.50 | JT | point is they don't suck |
| 04:29.00 | NZheretic | Not the case in NZ. Around 70% Net users still use Dial Up ( due to no per minute pricing on local calls ). |
| 04:29.28 | Strom_C | flat-rate local service is common in the states too |
| 04:29.31 | JT | NZheretic: i'm in australia, i can send you a modem if you want |
| 04:30.49 | NZheretic | JT, Thanks but I really need to find a source of telepermited modems that work with Linux. Unfortually the way the market is working that means some form of non-hardware modem. |
| 04:31.15 | JT | NZheretic: why does it need a permit? |
| 04:33.05 | NZheretic | JT : I have had customers who have brought in overseas modems in the past, even after a lighting strike ( nothing to do with the modem ) , the telecom repair man shook his head an presented them with a LARGE bill. It sucks. |
| 04:33.33 | Znuff | telepermited? |
| 04:33.34 | Znuff | wtf? |
| 04:33.48 | JT | hrm you need to rip some stickers off crappy internal modems and put them on the external one from overseas |
| 04:34.00 | JT | Znuff: NZ's telecommunications equipment approval |
| 04:34.07 | JT | Znuff: lots of countries have their own scheme |
| 04:34.12 | Znuff | Who cares about if they permit it or not |
| 04:34.27 | JT | Znuff: i think what he last said said the reasons why.. |
| 04:34.42 | Znuff | Bill me for what? I'll freakin' kick that guy out of my place if he presents a bill. |
| 04:34.52 | JT | yeah anyway |
| 04:34.56 | JT | onto the realworld ;) |
| 04:35.25 | Znuff | And I'd use a baseball bat too! |
| 04:38.59 | *** join/#asterisk snewpy (n=markl@ppp250-45.lns3.syd6.internode.on.net) |
| 04:39.00 | NZheretic | It is obvous that no one here is going be of help, so ... |
| 04:39.03 | *** part/#asterisk NZheretic (n=NZhereti@203-118-142-135.pppoe.ihug.co.nz) |
| 04:39.22 | Znuff | I still support my kick & baseball bat idea |
| 04:39.36 | JT | wow what a tightarse |
| 04:39.43 | JT | as we'd say in australia and new zealand |
| 04:39.44 | JT | :P |
| 04:40.19 | Znuff | What kind of country still uses dial-up? |
| 04:40.22 | Znuff | Except the US |
| 04:40.22 | Strom_C | so tight you could put a lump of coal in his ass, and in two weeks you'd have a diamond? |
| 04:40.33 | Znuff | Strom_C, 3 days |
| 04:40.50 | bkw_ | bueller? bueller? |
| 04:41.37 | JT | Znuff: new zealand apparently |
| 04:41.46 | Znuff | heh |
| 04:41.55 | Znuff | And they say Romania is not a civilized country :P |
| 04:42.07 | JT | they have broadband there |
| 04:42.11 | JT | clearly not everyone does... |
| 04:42.48 | Znuff | 80% of internet users in romania use broadband |
| 04:43.18 | Znuff | or more |
| 04:43.37 | Juggie | i had cable internet in 96 |
| 04:43.43 | Juggie | 1.5mbit even then |
| 04:43.53 | Znuff | I had 128kbps in 2001 ^_^ |
| 04:43.57 | Znuff | cable, ofcourse |
| 04:44.01 | Juggie | hah |
| 04:44.18 | Znuff | well, we've catched up pretty fast by now :P |
| 04:44.45 | Juggie | where i'm from was lucky |
| 04:44.57 | Juggie | we got cable tv so late that the coax was all great quality |
| 04:45.05 | Juggie | so they didnt have to upgrade much to roll out cable internet |
| 04:45.10 | Znuff | well we've had communism 'till the 1989 |
| 04:45.18 | Znuff | s/the// |
| 04:45.24 | JT | 2 way coax network? |
| 04:45.45 | Juggie | JT, you asking me? |
| 04:45.50 | Znuff | They'va had to upgrade everything for co-ax networks here =/ |
| 04:45.51 | JT | yeah |
| 04:46.06 | Znuff | damn, I'm typing like hell these days |
| 04:46.10 | Juggie | i'm sure it wasnt initially two way, but the cable was all of a higher grade then older installs. |
| 04:46.11 | Znuff | So many typos -_- |
| 04:46.39 | JT | ours was new enough that it was all 2-way capable from the start i believe |
| 04:46.42 | Juggie | we were late to have cable tv, but we were the first provider in canada maybe usa to roll out cable internet i believe |
| 04:46.43 | Znuff | Wonder if my network will support ipTV |
| 04:47.13 | Znuff | ipTV is multicast, right? |
| 04:49.23 | Juggie | JT, i cant find anything but i had cable modem in like 96, since rogers swallowed up our old provider i cant find any history on them |
| 04:49.42 | Juggie | i got it for my 16th birthday as i recall |
| 04:49.51 | Juggie | when i first got it i was using a pentium120 |
| 04:50.30 | Znuff | I used dial-up with my first pentium :( |
| 04:51.32 | Juggie | JT, looks like cable atlantic started in 75 |
| 04:51.48 | Juggie | which would mean much better quality coax then those started in the 50-60's |
| 04:52.39 | Znuff | pfft, we haven't even heard about cable in the 80's-_- |
| 04:53.19 | Znuff | our country introduced cable in '93-'94 |
| 04:53.23 | JT | ah ok |
| 04:53.34 | JT | lol 50-60s wtf |
| 04:53.42 | JT | yeah we didn't have cable till 90s |
| 04:54.13 | Znuff | damn US, they've kept everything for themselves :P |
| 04:55.19 | Juggie | hah |
| 04:55.39 | Juggie | yah, seems cable atlantic powered up their cable internet in Q4 96 |
| 04:56.10 | Juggie | there may be a few more companies who were ealier but not by more then a few months. |
| 04:59.22 | flenders | what's the easiest way to transfer a call on asterisk? |
| 04:59.28 | *** join/#asterisk CunningPike (n=CunningP@204.239.8.149) |
| 05:00.10 | flenders | I've been using the transfer buttons on the phone, but want to do it using an ATA |
| 05:00.58 | Strom_C | on the ATA? |
| 05:01.05 | Strom_C | hookflash, dial new number, hang up |
| 05:01.35 | flenders | is there a way to press "#5" for example? |
| 05:06.02 | *** join/#asterisk topping (n=topping@204.152.96.238) |
| 05:08.30 | Juggie | you mean like a speeddial? |
| 05:09.00 | flenders | also, I have a bunch of SPA-921s, and they have a button called 'xfer', which does attended transfers, can I change that to do blind transfers instead of attended transfers? |
| 05:09.42 | flenders | Juggie: no, I meant transfer a call. |
| 05:09.58 | flenders | #5, then number, and the call will be transfered |
| 05:10.04 | Juggie | this and more can be answered by looking at http://www.voip-info.org |
| 05:10.44 | *** join/#asterisk Bazy (n=bazy@eclipse.upcnet.ro) |
| 05:10.48 | Juggie | eg, http://www.voip-info.org/wiki-Asterisk+config+features.conf |
| 05:10.52 | Juggie | look @ blind transfer |
| 05:13.09 | *** join/#asterisk Avochelm (n=damien__@gw-morphett.koalatelecom.com.au) |
| 05:14.50 | *** join/#asterisk yidiyuehan (n=yidiyueh@58.185.253.70) |
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| 05:24.45 | *** join/#asterisk ShadowHntr (i=sentinel@wikipedia/Shadowhntr) |
| 05:29.32 | flenders | ok, I was missing the t/T on the Dial() app. |
| 05:31.36 | flenders | is there a way to know what my phone dials when I press the "xfer" button? |
| 05:32.01 | *** join/#asterisk tkup (n=tkup[2]@linux21.ece.utexas.edu) |
| 05:38.48 | *** join/#asterisk [TK]D-Fender (n=chatzill@64.235.216.2) |
| 05:43.52 | *** join/#asterisk ivan` (i=user@unaffiliated/ivan/x-000001) |
| 05:45.02 | [TK]D-Fender | Hey, running 1.2.15, which ap.txfax / app_rxfax should I be downloading. I'm gettin make errors off 2 versions I've pulled from soft-switch.org already... |
| 05:45.30 | JT | wait, [TK]D-Fender asking a question |
| 05:45.43 | JT | i thought they didn't work except with old version of asterisk |
| 05:46.34 | [TK]D-Fender | Like I said, 1.2.15. |
| 05:47.06 | [TK]D-Fender | I've done earlier installs on 1.2 series jsut fine... |
| 05:47.07 | [TK]D-Fender | and yeah, every once in a blue moon I can actually use a hand :) |
| 05:47.14 | [TK]D-Fender | If only coppice were here.... |
| 05:47.17 | JT | you know coppice is the right person to speak to |
| 05:47.37 | JT | but i thought asterisk operativity with that stuff was discontinued |
| 05:50.23 | [TK]D-Fender | not afaik.... |
| 05:51.07 | JT | or maybe that was spandsp |
| 05:53.26 | [TK]D-Fender | Well that part compiled fine... |
| 05:53.58 | [TK]D-Fender | I've jsut got a handful of nearly identical erros like this : app_rxfax.c:307: warning: passing arg 1 of `fax_rx' from incompatible pointer type |
| 05:54.18 | JT | i know spandsp is definately not supported to work with asterisk anymore |
| 05:54.25 | [TK]D-Fender | Actually.... a LOT more than jsut those |
| 05:54.32 | [TK]D-Fender | damnammit |
| 05:54.34 | JT | i'm not sure exactly how that intertwines with app_rxfax |
| 05:54.46 | JT | but maybe that's the bit that's not supported |
| 05:57.42 | JT | that sucks a bit if it's something you rely on |
| 05:59.47 | [TK]D-Fender | yeah.... this is going to blow... |
| 06:00.06 | *** part/#asterisk ivan` (i=user@unaffiliated/ivan/x-000001) |
| 06:00.08 | JT | so you either have to downgrade... or use something else.. |
| 06:00.09 | [TK]D-Fender | I think I'll waith till morning to catch word from the "Man" himself... |
| 06:01.46 | *** join/#asterisk IgorG (n=FeedomPa@195.162.32.126) |
| 06:01.51 | *** part/#asterisk techie (n=techie@c-67-181-184-170.hsd1.ca.comcast.net) |
| 06:02.43 | *** part/#asterisk Giofe (n=opera@190.81.4.169) |
| 06:03.23 | *** join/#asterisk Exhar (i=Ask@84-105-192-215.cable.quicknet.nl) |
| 06:06.17 | IgorG | Hello, all! Yesterday I have see a bug. In organisation VoIP system exists incoming queue with ringall strategy. Yesterday two operator answered in one time on one call. All of three peoples can talk and listen... |
| 06:09.03 | Snake-Eyes | thats a feature :P |
| 06:09.37 | IgorG | you think? |
| 06:09.58 | IgorG | :) I don't think so |
| 06:10.25 | budmang | I need a good provider |
| 06:10.37 | budmang | i dont like the softcap at teliax :-) |
| 06:11.06 | ltdwk | bug/features, the line is sometimes blurry |
| 06:11.59 | IgorG | Bug/feature difference than I can switch of using this feature |
| 06:12.18 | IgorG | ^) |
| 06:12.19 | IgorG | :) |
| 06:14.16 | ltdwk | =p |
| 06:15.11 | IgorG | i onlu think write or not to write to bugtracker about this |
| 06:15.14 | budmang | I need a good outbound provider recommend anyone? |
| 06:20.35 | fetcher | budmang: I use voipjet.com for light volume outbound... no major issues |
| 06:20.46 | budmang | umm |
| 06:20.50 | budmang | ill look into them |
| 06:21.00 | fetcher | budmang: they're based out of the US, though. Probably not a good choice for other parts of the world |
| 06:21.03 | budmang | any good inbound/outbound unlimits you know of? |
| 06:21.05 | [TK]D-Fender | budmang: VoicePulse Connect |
| 06:22.32 | budmang | I like both thanks |
| 06:22.37 | budmang | voice pulse looks cheap |
| 06:23.14 | [TK]D-Fender | budmang: I've had some clients running with them who've been quite happy |
| 06:23.37 | *** join/#asterisk aadilismail (i=zeeesh@202.38.55.125) |
| 06:23.37 | aadilismail | hi |
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| 07:22.58 | gfraysse | <PROTECTED> |
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| 07:37.52 | duco | Hello |
| 07:38.22 | duco | Can asterisk 1.4 use ogg vorbis for music on hold? |
| 07:38.22 | Nugget | heh. |
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| 07:40.21 | *** join/#asterisk X-Rob (n=rob-x@149.135.44.222) |
| 07:40.33 | duco | I have an icecast2 who streams oggs, and hope use it wit * |
| 07:41.16 | duco | Is someone ising icecast2 as streamer? |
| 07:49.37 | *** join/#asterisk kgx_home (n=karuna@port-60-234-196-160.jet.net.nz) |
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| 08:13.39 | angryuser | i have a problem, it is driving me mad....whatever i do, in about an hour mo sip peers become all unreachable(only outgoing peers) |
| 08:14.11 | angryuser | i can not turn off qualify! |
| 08:14.28 | angryuser | myscripts are base on qualify |
| 08:16.34 | *** join/#asterisk inspired (n=mikael@85.221.7.59) |
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| 08:43.27 | *** join/#asterisk fatred (n=fatred@87-194-32-104.bethere.co.uk) |
| 08:44.20 | fatred | morning everyone |
| 08:45.17 | *** part/#asterisk sergee (i=opera@195.94.224.197) |
| 08:45.22 | *** join/#asterisk Chris-NB (n=chris@ng1.kurtkrenn.com) |
| 08:45.39 | Chris-NB | hi |
| 08:45.40 | Chris-NB | what codecs are used on zap channels? |
| 08:45.49 | Chris-NB | or, can I set/define that codecs? |
| 08:47.07 | fatred | i stopped by for some help yesterday, but in fixing one thing i appear to have broken another...when doing an atxfer to the park extension, instead of parking the calling channel (e.g. misdn2-1) it creates a local/700@from sip and parks that. when you collect the call with 701 (as announced) you get connected to nothing, and the end user gets stuck listening to moh. when you have up the internal channel, the external channel drops moh, but |
| 08:47.08 | *** join/#asterisk IgorG (n=FeedomPa@195.162.32.126) |
| 08:47.12 | ilan | anyone available for a quick test of FWD? |
| 08:47.15 | fatred | is this something stupid? |
| 08:49.18 | af_ | has anyone run astertest? |
| 08:49.28 | af_ | the setup look far complicated |
| 08:50.46 | *** join/#asterisk Verifier (n=chatzill@dyn-13-cust150.netit.se) |
| 08:51.39 | fatred | <PROTECTED> |
| 08:51.47 | jeremy_g | :) |
| 08:52.20 | *** join/#asterisk Aurs (n=Aurs@c6870BF51.dhcp.bluecom.no) |
| 08:54.05 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 08:54.57 | *** join/#asterisk yassine (n=yassine@dsl.voicint.com) |
| 08:56.18 | *** join/#asterisk skirmisha (i=507e4918@gateway/web/cgi-irc/ircatwork.com/x-79ec0283c70c48bd) |
| 08:56.22 | skirmisha | hello guys |
| 08:56.38 | skirmisha | does anyone know if asterisk can handle fax over sip |
| 08:57.15 | skirmisha | means to send fax to asterisk over sip and asterisk to send it to emal |
| 08:57.18 | skirmisha | email |
| 08:57.39 | fatred | rx_fax and tx_fax? |
| 08:57.42 | fatred | worked for me |
| 08:59.00 | skirmisha | no i mean there is not ISDN card |
| 08:59.04 | skirmisha | just pure sip |
| 08:59.12 | fatred | you need to get the spandsp libs and install them, then i just made a macro to handle inbound faxing and called it from the fax extension |
| 08:59.19 | fatred | think there is an article about it somehere |
| 08:59.36 | skirmisha | fatred: how do u get faxes |
| 08:59.41 | skirmisha | from isdn or from sip |
| 09:00.16 | fatred | well i bit of both |
| 09:00.49 | skirmisha | tell me if i get fax over sip using G711 can asterisk handle it and send it to email? |
| 09:01.10 | skirmisha | what is nvfax for? |
| 09:01.55 | fatred | isdn -> asterisk -> ata is one method i use and ata -> asterisk -> ata for internal stuff (between sites) |
| 09:02.24 | fatred | http://www.cyber-cottage.co.uk/wiki/index.php?title=Asterisk_1.2.1_and_spandsp&redirect=no |
| 09:02.25 | skirmisha | ok let me give more details |
| 09:02.26 | fatred | try that |
| 09:03.33 | fatred | you dont neccessarily need the dbget stuff if you only need one or two fax extensions |
| 09:03.40 | skirmisha | i want something like virtual fax, means i get many faxes over sip to numbers that are register on asterisk. So what i want is asterisk to get those faxes and send them to email |
| 09:04.15 | fatred | follow the link - there is an example of what to do there. |
| 09:04.40 | skirmisha | ok let me see |
| 09:05.44 | *** join/#asterisk tr2x (n=alvar@80-218-185-55.dclient.hispeed.ch) |
| 09:06.14 | skirmisha | fatred: exten => h,1,system(/usr/local/sbin/mailfax ${FAXFILE} ${EMAILADDR} ${CALLERIDNUM}) |
| 09:06.23 | skirmisha | do i need to put this row in [fax] section |
| 09:06.53 | fatred | that goes at the bottom fo the fax context |
| 09:07.03 | skirmisha | ok i understand |
| 09:07.16 | fatred | it means once the call has ended (i.e. allthe fax has been recieved) then email the file |
| 09:07.37 | fatred | you need to install mime-construct |
| 09:07.50 | skirmisha | yes that is not a problem |
| 09:07.56 | fatred | you need to make sure your sip channels have good qos tho |
| 09:07.57 | skirmisha | i'll give it a try |
| 09:08.13 | fatred | fax doesnt really like coughs and splutters |
| 09:08.20 | skirmisha | is this works on 1.4 asterisk |
| 09:08.24 | fatred | no idea |
| 09:08.27 | fatred | im still on 1.2 |
| 09:08.30 | fatred | sorry |
| 09:08.37 | skirmisha | no problem |
| 09:08.39 | skirmisha | will try |
| 09:08.42 | fatred | good luck! |
| 09:09.11 | *** join/#asterisk darviria (n=darviria@194-105-181-29.ifb.co.uk) |
| 09:12.26 | skirmisha | thanks |
| 09:13.21 | *** join/#asterisk lorinc (n=ang@pool-1002.adsl.interware.hu) |
| 09:17.15 | *** join/#asterisk stimpie (n=michiel@ip565faf27.direct-adsl.nl) |
| 09:21.52 | *** join/#asterisk ball (n=ball@adsl-68-72-85-49.dsl.chcgil.ameritech.net) |
| 09:22.10 | ball | Do Nortel make any SIP phones that play nicely with asterisk? |
| 09:26.40 | dlynes_laptop | ball: define 'play nicely' |
| 09:27.38 | dlynes_laptop | ball: the early firmware i2004's will work using the Nortel channel driver |
| 09:27.47 | dlynes_laptop | ball: but the later firmwares on the i2004 won't |
| 09:27.48 | *** join/#asterisk toot (n=toot@84.19.255.123) |
| 09:27.55 | dlynes_laptop | ball: or at least not well |
| 09:27.56 | mendol | morningo |
| 09:28.06 | mendol | i got analog phone which can use 2 lines |
| 09:28.10 | ball | dlynes_laptop: doesn't sound like Nortel are tremendously cooperative then. |
| 09:28.14 | ball | hello toot. |
| 09:28.21 | mendol | when i connect it to spa 2102 when im trying to make a call i have 2 signals |
| 09:28.31 | mendol | spa 2102 is connected to asterisk |
| 09:28.32 | dlynes_laptop | ball: well, why would they want you to be able to use cheaper handsets with their phone systems? |
| 09:28.34 | *** join/#asterisk nuonguy (n=john@c-24-6-175-26.hsd1.ca.comcast.net) |
| 09:28.36 | mendol | what can be the reason? |
| 09:28.44 | dlynes_laptop | ball: they don't get to rip you off as badly then |
| 09:29.00 | ball | dlines_laptop: I was thinking more of using their phones on an Asterisk switch. |
| 09:29.05 | ball | hello nuonguy |
| 09:29.38 | ball | mendol: does the spa 2102 (analog telephone adaptor?) provide two lines? |
| 09:29.47 | dlynes_laptop | ball: try the i2002(?) softphone |
| 09:29.50 | IgorG | ball: yes it does |
| 09:30.01 | dlynes_laptop | ball: it apparently plays quite well with asterisk, using the nortel channel driver |
| 09:30.15 | mendol | yes it does, but i use spa 2102 with 2 analog phones |
| 09:30.21 | mendol | separate ones |
| 09:30.44 | mendol | when im using them with asterisk i have like double conversation |
| 09:30.53 | mendol | when i use spa 2102 with sip provider |
| 09:30.55 | mendol | it works fine |
| 09:31.08 | mendol | so curious what option is responsible for that in asterisk? |
| 09:31.25 | IgorG | mendol: it is simple |
| 09:31.30 | ball | Is the 2102 wired as RJ14? |
| 09:31.35 | IgorG | mendol: just edit sip.conf |
| 09:32.18 | mendol | Igor and? |
| 09:32.22 | IgorG | set progressinband=yes |
| 09:32.23 | mendol | ball: yes |
| 09:32.52 | IgorG | mendol: i do not sure, but it is bug in Linksys, I wrote about it to tech support |
| 09:32.57 | mendol | oh |
| 09:33.48 | IgorG | mendol: :) |
| 09:34.16 | mendol | hehehe |
| 09:34.20 | mendol | checking ;-) |
| 09:34.24 | mendol | dont go away Igor :-) |
| 09:35.00 | mendol | wow |
| 09:35.01 | mendol | it works |
| 09:38.16 | ball | Congratulations |
| 09:38.29 | ball | So can one of those little telephone adaptors generate a ring signal? |
| 09:38.40 | ball | ...something that'd shake the bells on a 2500 set? |
| 09:38.55 | mendol | hm |
| 09:39.10 | mendol | now we got terrible mess hehehe |
| 09:39.46 | mendol | or it works |
| 09:41.31 | *** join/#asterisk drray (n=drray@c-67-170-9-176.hsd1.wa.comcast.net) |
| 09:41.50 | ball | Is there a name for the standard 25-pair amphenol connector and its pinout? |
| 09:44.43 | ball | oh well, I should go to bed now. |
| 09:44.46 | ball | Goodnight everyone. |
| 09:45.00 | *** part/#asterisk ball (n=ball@adsl-68-72-85-49.dsl.chcgil.ameritech.net) |
| 09:45.59 | *** join/#asterisk DA_ (n=da@70.84.179.74) |
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| 09:55.41 | lately | Hi. Can anyone explain why I cannot get commands beginning with "console" in 1.4.1 ? Have I to load a module? |
| 10:00.33 | angryuser | can somebody tell me why my sip peer become unreachable in about un hour?, and if i shutdown * and reload the come back again... for an hour |
| 10:00.38 | *** join/#asterisk nasls_lsa (n=chatzill@athedsl-219219.otenet.gr) |
| 10:00.46 | angryuser | in any * version |
| 10:01.18 | angryuser | and only external ones |
| 10:02.31 | toot | any uk folk about this morning? |
| 10:02.45 | Makenshi | yes |
| 10:03.13 | *** part/#asterisk mjun007 (n=mjun007@221.221.159.208) |
| 10:03.26 | toot | if you's get a second i would love a 10 second review on http://adsldemo.tigercube.net/ user: ircdemo pass: ircdemo |
| 10:05.16 | *** join/#asterisk UlbabraB (n=salama@ip-204-57.sn2.eutelia.it) |
| 10:05.38 | angryuser | toot: really nice work, done by yourself? |
| 10:07.49 | toot | yar it is |
| 10:07.58 | toot | would love to know what people perceive to be missing |
| 10:08.07 | toot | yeah its commerical but we do plan on releasing a lot of it |
| 10:08.10 | toot | just not all of it ;) |
| 10:09.09 | Makenshi | The hide buttons on the extensions page exhibit odd behaviour in my firefox |
| 10:09.17 | *** join/#asterisk anthonyl (n=anthonyl@r74-193-79-81.pfvlcmta01.grtntx.tl.dh.suddenlink.net) |
| 10:09.29 | Makenshi | I like it so far |
| 10:09.59 | angryuser | toot: well you have the basics dialplan edit(patterns ect) but when it comes to more complex use (db ect) i would prefer to edit manually |
| 10:10.13 | toot | oh? odd what way? a few people have told me off for using the word Huntgroup and not Queue |
| 10:10.40 | Makenshi | When I click them, they jump to a different place, it takes a couple of clicks before they do the desired action |
| 10:10.58 | toot | i was thinking of adding a more advanced editor in - but am hoping for a really simple to use version for the average sme |
| 10:11.58 | angryuser | toot: On the status screen, it is written the phonesystem is (ok) when do you consider that it is ok?(if peers are down would it be ok?) |
| 10:12.50 | toot | i had just about completed a full AMI client that we will hook into the pc tray apps and also the pbx status pages - right up until i discovered the java nbio stuff sucked ;) |
| 10:13.01 | toot | so have reverted in short term - we are still far off a v1.0 |
| 10:13.15 | toot | so i agree - its lacking in that respect :) |
| 10:13.35 | DA_ | toot: looks nice |
| 10:14.23 | angryuser | toot: just the state of qualifyed peers/users woul be fine, and intermediate state (yellow) -system is up bur some peers are unreachable needed |
| 10:14.46 | toot | we do show a disabled icon for unreachable yeah |
| 10:14.55 | toot | although i do need to vastly improve the pbx status page |
| 10:15.08 | toot | in terms of the labels - all make sense to the uk market? or should i go for Queues, etc |
| 10:16.11 | angryuser | toot: do you designed your soft based on idead that there is only one context? |
| 10:16.16 | angryuser | *idea |
| 10:16.19 | toot | yeah |
| 10:16.41 | toot | we have the schema working to handle more than one but for this version (sme - 1-20 users) we have 1 context |
| 10:16.43 | DA_ | Makenshi, toot: ahhh i see what you mean, if you click one of the hide buttons any hide buttons below it dont change position |
| 10:17.01 | DA_ | therefore causing somewhat of an overlap |
| 10:17.38 | Makenshi | Yip, it's just cosmetic, it seems |
| 10:17.44 | DA_ | indeed |
| 10:18.08 | angryuser | toot: for example i have some users that receive call from pstn(fxo), i am in obligation to create a context for them |
| 10:18.17 | toot | ah no |
| 10:18.21 | DA_ | toot: very nice work, looks to be really feature-rich and functional |
| 10:18.33 | *** join/#asterisk active_si (n=AcTiVe@84-255-238-137.static.dsl.t-2.net) |
| 10:18.36 | toot | you can setup Huntgroups for them (if i understand you correctly) |
| 10:18.52 | toot | and either have a line a line group or an IVR menu go to that |
| 10:19.07 | toot | if that answer yer q ? :) (soz up till 5am with 4 week old kid) :) |
| 10:19.15 | toot | thanks DA_ :) |
| 10:19.23 | angryuser | toot: in my case i cnt use ivr, person just need to be ringed directly |
| 10:19.40 | toot | you would setup a line that would call their extension directly |
| 10:20.44 | Makenshi | I have quite a complicated dialplan, i wonder if you would be able to do it with the gui.. want to see what i mean? |
| 10:21.03 | Makenshi | it takes into account outgoing sip-sip calls and e164 |
| 10:21.09 | toot | would love to |
| 10:21.44 | toot | also - i know you's are all much better at asterisk than me - but do you see the app as a sellable product (once completed) to the average SME |
| 10:21.49 | active_si | I am using two HFC-S cards one for NT and one for TE mode, sometimes when I am talking with someone I hear some DTMF tones on the line, this happens totally randomly... what could be wrong? |
| 10:21.57 | Makenshi | http://pastebin.ca/327878 |
| 10:22.08 | Makenshi | toot, definitely |
| 10:23.32 | Makenshi | I like that interface a lot |
| 10:24.14 | toot | hmm we cant handle enum lookups at the minute - can handle inbound sip calling though. actually need the concept to DEFINES/GLOBALS in the dialplan - so no it could not handle your dialplan but i will make it do so :) |
| 10:24.39 | Makenshi | toot, nifty :) |
| 10:24.45 | toot | thanks Makenshi :) been a long and interesting/costly route so fingers crossed |
| 10:25.04 | toot | we will be releasing the AMI proxy and call manager apps |
| 10:25.22 | toot | hopefully in the next few weeks once i figure out what the hell is wrong with sun :) |
| 10:25.30 | toot | and probably rewrite in c++ :) |
| 10:25.34 | Makenshi | hah :) |
| 10:25.46 | Makenshi | do you have a mailing list? |
| 10:26.02 | toot | although given my experience with Boost -> will end up C |
| 10:26.24 | toot | na not yet - but fire me an email to chris@tigercube.net and i will keep anyone notified of how we are getting on :) |
| 10:26.45 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@ppp-70-255-106-4.dsl.hstntx.swbell.net) |
| 10:27.27 | tzafrir | toot, what license? |
| 10:27.29 | toot | thanks loads of the comments folks - appreciate it - we hope to repay the favours |
| 10:27.59 | toot | tzafrir - we have plans - but we need some money to cover what we have done to date so we need to agree that with some likely investors |
| 10:28.10 | angryuser | sorry for offtopic does yate support B410P? |
| 10:28.17 | toot | we will return the favours we have pinced from the os community - no worries in that resepect |
| 10:28.34 | tzafrir | angryuser, the question should probably be: does yate support misdn? |
| 10:29.04 | *** join/#asterisk roguebug (n=roguebug@dip-109-055.bras.dsl.breisnet.com) |
| 10:29.06 | roguebug | hey |
| 10:29.22 | angryuser | tzafrir: and the answer is? |
| 10:29.30 | tzafrir | I don't know |
| 10:30.05 | roguebug | could it be that asterisk's configure accepts a --prefix, but ignores it (i.e. still wants to install in its default dirs, without prepending the specified prefix) ? |
| 10:30.29 | angryuser | tzafrir: i think it doen not support it |
| 10:31.19 | jeremy_g | toot:lol, the blocked sites are porn.com, next is windowsmedia.com :D |
| 10:31.33 | Ch0Hag | Is there a way to list all the active calls Asterisk is part of? |
| 10:31.34 | toot | hehe ;) |
| 10:31.38 | Makenshi | toot, are you based in the uk as well? |
| 10:31.56 | toot | uk and ireland :) i'm in ireland where houses are cheaper ;) |
| 10:32.04 | toot | but the rest of us are Edinburgh based |
| 10:32.08 | Makenshi | ahh, must be nice over there |
| 10:32.13 | toot | freezing |
| 10:32.17 | Makenshi | erk |
| 10:33.32 | toot | any things people think we are clearly missing? other than stuff in dialplans? |
| 10:34.19 | Makenshi | toot, how about jabber functionality also? |
| 10:34.34 | Makenshi | looks like you're going for an all-in-one box, so that would be something i'd really like to have |
| 10:35.34 | Makenshi | The thing I love most about Google's implementation is that it can automatically archive your conversations.. for a business that could be useful for regulatory compliance |
| 10:35.37 | toot | thats a good call (and one of the developers is v jabber friendly) |
| 10:36.09 | skirmisha | guys |
| 10:36.14 | skirmisha | when i call rxfax |
| 10:36.19 | skirmisha | should i hear fax tone? |
| 10:36.48 | Makenshi | skirmisha, i don't think so, try whistling the fax tone and see if you hear something back |
| 10:36.50 | lately | Hi. Can anyone explain why I cannot get commands beginning with "console" in 1.4.1 ? Have I to load a module? |
| 10:37.24 | skirmisha | because i am calling from sip phone and when rxfax execute i did not hear anything |
| 10:37.49 | skirmisha | shouldn't i get fax signal when rxfax is executed |
| 10:37.51 | Makenshi | rxfax waits for the fax machine to send a tone, i believe |
| 10:38.13 | skirmisha | so i better try with real fax |
| 10:38.24 | Makenshi | toot, i dropped you an email, my email address is also my jabber address (and sip too) |
| 10:40.06 | roguebug | jeez, at least wit 1.2.xx i could just set a prefix in the makefile |
| 10:40.31 | roguebug | now there's a standard configure, but the --prefix doesn't work |
| 10:41.32 | roguebug | <PROTECTED> |
| 10:41.58 | roguebug | so how do i get asterisk 1.4 to install all in /my/custom/prefix/dir ? |
| 10:42.43 | *** join/#asterisk Pon`work (n=jamesm@ip-217.146.113.66.merula.net) |
| 10:44.40 | jeremy_g | hi |
| 10:44.59 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
| 10:45.08 | jeremy_g | can anyone suggest how can i archive all the discussions on this channel? what irc client would i need? is #asterisk archived anywhere? |
| 10:46.16 | *** join/#asterisk toot_ (n=toot@84.19.255.123) |
| 10:46.26 | toot_ | um. ireland. great power supplies. hmm |
| 10:47.01 | roguebug | jeremy_g: for privacy reasons, most irc channels don't keep logs. however, many irc clients can do that. xchat and chatzilla definitely can |
| 10:48.08 | roguebug | jeremy_g: i recommend xchat. the logs are activated in Settings -> Preferences -> Chatting -> Logging |
| 10:48.17 | Pon`work | ya, just leave X-Chat running all day ^_^ |
| 10:50.21 | roguebug | <PROTECTED> |
| 10:50.38 | roguebug | ahm back to my prefix prob: |
| 10:51.18 | roguebug | is there no way in 1.4 to specify an installation prefix directory ? |
| 10:52.18 | nasls_lsa | chatzilla i use , fantastic irc client ! |
| 10:58.22 | toot_ | okay folks - thanks a million for the help and advice - i'll be sure to keep you's in the loop over the coming weeks as we try and get a v1.0 ready |
| 11:02.08 | *** join/#asterisk kezza491 (n=opera@c211-28-159-50.brasd1.vic.optusnet.com.au) |
| 11:02.13 | kezza491 | what dose |
| 11:02.32 | Makenshi | thanks toot, look forward to it! |
| 11:03.03 | kezza491 | chan_iax2.c:6754 socket_process: Rejected connect attempt from 192.246.69.186, who was trying to reach '838764@' |
| 11:03.14 | kezza491 | why is it doing that an not accepting the call? |
| 11:06.23 | *** join/#asterisk anthonyl (n=anthonyl@r74-193-79-81.pfvlcmta01.grtntx.tl.dh.suddenlink.net) |
| 11:06.27 | kezza491 | is there any way you can see in more depth why the call was rejected? |
| 11:08.39 | RoyK | iax2 debug |
| 11:08.43 | kezza491 | thanks |
| 11:09.10 | kezza491 | theres no such command |
| 11:09.32 | *** join/#asterisk jm|work (n=jm@sentry.flags.co.uk) |
| 11:13.24 | *** join/#asterisk dj-fu (n=ajc@202-74-195-152.ue.woosh.co.nz) |
| 11:16.17 | *** join/#asterisk fadey (n=fadey@239.Red-80-36-91.staticIP.rima-tde.net) |
| 11:19.52 | RoyK | kezza491: no? |
| 11:19.53 | RoyK | hm |
| 11:20.25 | RoyK | kezza491: there is on my box |
| 11:22.38 | kezza491 | Mmm? |
| 11:22.43 | kezza491 | ohh foudn it |
| 11:22.46 | kezza491 | still no help |
| 11:22.54 | kezza491 | wrong kinda infoamtion for some one like me |
| 11:24.30 | *** join/#asterisk CrazyTu[m] (n=CrazyTu[@ppp-70-255-106-4.dsl.hstntx.swbell.net) |
| 11:25.55 | kezza491 | ahh k fixed it and got another error |
| 11:26.13 | kezza491 | How do you load Crypto support? |
| 11:27.33 | *** join/#asterisk tcseke (n=chatzill@22-36.adsl.etel.hu) |
| 11:39.01 | *** join/#asterisk HockeyInJune (n=HockeyIn@pool-68-161-165-186.ny325.east.verizon.net) |
| 11:40.02 | *** join/#asterisk uwe (n=uwe@dogbert.palnet.com) |
| 11:41.27 | *** join/#asterisk fatred (n=fatred@87-194-32-104.bethere.co.uk) |
| 11:42.11 | fatred | hi guys, im still having issues with call parking, with the park app parking the wrong channel. anyone able to help? |
| 11:42.39 | fatred | im sure its something stupid, but im running out of ideas |
| 11:45.50 | uwe | hello, im trying to store monitoring files in mp3 format, or anything less storage consuming than .WAV, ive tried MONITOR_EXEC = /usr/local/bin/2wav2mp3 and got the 2wav2mp3 , edited it to fit locations of executables on my system, reloaded extensions , still not recording , any idea? am i missing something? |
| 11:48.08 | uwe | running asterisk 1.2.13 |
| 11:48.32 | IgorG | uwe: may be bettertry to use MixMonitor and by cron start converting to mp3? |
| 11:49.26 | uwe | hmmm ... |
| 11:50.22 | uwe | well, maybe this is better, thank you IgorG |
| 11:50.31 | IgorG | uwe: ok |
| 11:53.33 | fatred | http://pastebin.com/894576 |
| 11:54.34 | fatred | anyone see anything there ^ that is glaringly obvious?? |
| 11:57.02 | *** join/#asterisk eltech (i=G00Ds@ool-457c93b6.dyn.optonline.net) |
| 11:58.27 | JT | uwe: yeah, IgorG's idea did seem better |
| 12:03.46 | *** join/#asterisk JoelSolanki (i=Joel@202.160.161.94) |
| 12:11.35 | JoelSolanki | Hi Any body awake ? |
| 12:13.07 | aadilismail | how to update asterisk?? |
| 12:14.07 | reber | what a sweet asterisk demo voice ... |
| 12:19.19 | reber | did any of you already used asterisk with voipbuster for outbound calls ? |
| 12:24.14 | *** join/#asterisk coppice (n=chatzill@249.193.17.210.dyn.pacific.net.hk) |
| 12:27.54 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 12:29.09 | *** join/#asterisk friedrich| (n=friedric@e177250074.adsl.alicedsl.de) |
| 12:35.30 | *** join/#asterisk oej (n=olle@apollo.webway.se) |
| 12:42.24 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 12:45.02 | *** join/#asterisk yonahw (i=yonahw@IGLD-83-130-228-138.inter.net.il) |
| 12:48.05 | uwe | hello, im getting extremely many "empty" wav monitoring files, ie, i have 11812 files and 10112 of them have the length of 44 bytes, ie .wav header!! any idea why? |
| 12:52.21 | Pon`work | is symmetric RTP supported in Asterisk 1.4.x ? |
| 12:53.34 | uwe | hmm, it seems that when i call a group with ringall strategy, for each phone that rings a .wav file is created which is empty |
| 12:54.04 | *** join/#asterisk toot (n=toot@84.19.255.123) |
| 12:55.39 | *** part/#asterisk kezza491 (n=opera@c211-28-159-50.brasd1.vic.optusnet.com.au) |
| 12:57.58 | *** join/#asterisk Niklas- (i=niklas@moo.dk) |
| 13:00.45 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 13:02.05 | yonahw | is there anythiung similar to hudlite for a non-trixbox install? |
| 13:04.37 | nasls_lsa | some config help ? I am trying to setup an outbound SIP connection. I put a line like: register => user:password@sip_proxy/800 at sip.conf , |
| 13:04.57 | nasls_lsa | what do I have to put in extensions.conf so I can use it ( make my alias for example ) |
| 13:05.42 | nasls_lsa | OUTBOUND=SIP/user:password@sip_proxy ? |
| 13:06.14 | *** join/#asterisk myiagy (n=myiagy@200.175.61.250.static.gvt.net.br) |
| 13:08.18 | angryuser | nasls_lsa: xx,1,Dial(Sip/${number}@yourpeer,[delay],[options]) |
| 13:09.51 | angryuser | nasls_lsa: in sip.conf to register: register => username:password@domain/username |
| 13:10.20 | nasls_lsa | register => username:password@domain/ext_number you mean ? |
| 13:10.34 | angryuser | nasls_lsa: no 2 times username |
| 13:10.41 | angryuser | nasls_lsa: it works for me |
| 13:10.56 | nasls_lsa | and then at extensions.conf ? |
| 13:11.38 | angryuser | nasls_lsa: dont forget to create a [username] in sip.conf a peer |
| 13:11.53 | *** join/#asterisk hellop (n=hellop@udp112969uds.hawaiiantel.net) |
| 13:11.55 | nasls_lsa | aha ! |
| 13:12.09 | *** join/#asterisk gr1ncheux (n=devine@unaffiliated/gr1ncheux) |
| 13:12.12 | angryuser | with options: fromdomain:targerdomain |
| 13:12.34 | angryuser | fromdomain=targerdomain and callerid=username |
| 13:12.38 | nasls_lsa | I thought that was only for clients ( phones ) .. |
| 13:13.11 | angryuser | nasls_lsa: type=peer |
| 13:13.26 | angryuser | ~book |
| 13:13.36 | jbot | i heard book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 13:13.48 | angryuser | nasls_lsa:read more info about sip.conf |
| 13:14.13 | *** join/#asterisk anthonyl (n=anthonyl@r74-193-79-81.pfvlcmta01.grtntx.tl.dh.suddenlink.net) |
| 13:14.27 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 13:14.37 | nasls_lsa | from console as I can see asterisk registers ok with my SIP provider .. |
| 13:19.55 | *** join/#asterisk champster (n=asterisk@AH.tescogroup.com) |
| 13:20.10 | angryuser | xx,1,Dial(Sip/${number}@yourpeer,[delay],[options]) nasls_lsa:dialout then |
| 13:20.27 | nasls_lsa | thanks , I'll try and see .. |
| 13:23.41 | *** join/#asterisk emiquelito (n=evandro@200-155-185-1.static.spo.ifx.net.br) |
| 13:23.47 | *** part/#asterisk emiquelito (n=evandro@200-155-185-1.static.spo.ifx.net.br) |
| 13:24.44 | *** join/#asterisk Thazza (n=mine@229.9.233.220.exetel.com.au) |
| 13:28.19 | *** join/#asterisk MarkWD (n=MarkWD@rrcs-67-78-88-186.sw.biz.rr.com) |
| 13:35.39 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:38.52 | *** join/#asterisk tsurko (n=tsurko@vpn-pppoe-213-240-242-237.megalan.bg) |
| 13:43.45 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 13:47.35 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
| 13:49.16 | *** join/#asterisk _charly_ (i=kroseneg@sunrise.schmidham.net) |
| 13:50.33 | *** join/#asterisk _charly_ (i=kroseneg@sunrise.schmidham.net) |
| 13:51.48 | *** join/#asterisk _charly_ (i=kroseneg@sunrise.schmidham.net) |
| 13:52.57 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 13:53.06 | *** join/#asterisk _charly_ (i=kroseneg@sunrise.schmidham.net) |
| 13:53.18 | [TK]D-Fender | coppice: You about? |
| 13:53.26 | coppice | no |
| 13:53.35 | [TK]D-Fender | coppice: Ok... just checking ;) |
| 13:56.47 | coppice | i'm really really not about |
| 13:56.48 | jeremy_g | is there any script around that can help me import asterisk conf into sql (real time static) |
| 13:56.49 | jeremy_g | the one that wiki refers to i.e. ast2sql.pl is still not downloadable |
| 13:56.52 | jeremy_g | i have mailed the author |
| 13:58.37 | *** join/#asterisk fakhir (i=fakhir@unaffiliated/fakhir) |
| 13:58.46 | coppice | [TK]D-Fender: yes? |
| 13:58.48 | [TK]D-Fender | coppice: Ok, quick question for you : I went to download the latest SpanDSP & app_rxfax/app_txfax to run on 1.2.15 and ran into a lot of make error on * regarding them. I had since heard there may be issues with SpanDSP's & *'s being a function pair looking forward |
| 13:59.10 | [TK]D-Fender | coppice: Looking for some clarity :) |
| 13:59.30 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 13:59.45 | coppice | which version of spandsp are you using? |
| 14:00.40 | [TK]D-Fender | spandsp-0.0.3pre28.tgz |
| 14:00.50 | [TK]D-Fender | Figured the latest.... |
| 14:01.18 | [TK]D-Fender | And I tried 2 different sets of the * apps (one was in a Test folder) |
| 14:02.00 | coppice | you will need the app_rxfax.c from http://www.soft-switch.org/downloads/snapshots/spandsp/test-apps-asterisk-1.2/ to go with that version |
| 14:03.55 | ManxPower | coppice: The most recent SpanDSP and rx_fax solved the last of our known issues with it. The issue is that the page seemed to "wrap around" right to left. |
| 14:04.13 | ManxPower | but only on some faxes |
| 14:06.39 | coppice | i've no idea what that might have been |
| 14:08.01 | coppice | the most recent thing I fixed was a issue of how T.30 is interpreted in doing ECM error recovery. That just chopped pages short with some machines, though. It should tolerate pretty much any interpretation of that issue now |
| 14:09.16 | ManxPower | my users are more happy now. |
| 14:09.27 | ManxPower | They are always upset about SOMETHING. 8-( |
| 14:09.29 | [TK]D-Fender | coppice: Attempting now... |
| 14:10.43 | *** join/#asterisk intralanman (n=lanman@pool-71-253-253-149.nrflva.east.verizon.net) |
| 14:10.48 | coppice | quite a few people seem to have timing issues, but when they use spandsp through iaxmodem things seem more reliable. I have yet to figure out why that should be |
| 14:11.13 | [TK]D-Fender | coppice: Darn... it compiled, but can't find libspandsp.so.0 now on load... |
| 14:11.29 | coppice | isn't life tragic? |
| 14:12.51 | [TK]D-Fender | coppice: I feel somewhat Shakespearian.... |
| 14:13.20 | [TK]D-Fender | Found the file... now to finda way to cope with its location... |
| 14:13.41 | [TK]D-Fender | maybe just copy it :) |
| 14:14.47 | [TK]D-Fender | Hrm... except it doesn't say where it expects to FIND it... |
| 14:15.48 | ManxPower | [TK]D-Fender: /etc/ld.so.conf put the directory in there and run ldconfig -v |
| 14:16.20 | ManxPower | Usually /usr/local/lib |
| 14:18.15 | [TK]D-Fender | coppice & ManxPower : SMASHING.... thanks a bunch! |
| 14:19.37 | [TK]D-Fender | I've so much to learn abou even the basics of Linux programming, libraries, etc... |
| 14:20.21 | *** join/#asterisk ESCulapio__ (n=The@200.88.44.66) |
| 14:23.04 | *** join/#asterisk mrchicken (n=asdasd@200.71.58.39) |
| 14:23.09 | mrchicken | Hello :) |
| 14:24.29 | mrchicken | I just upgraded my MySQL libraries and now when I try to load the mysql cdr addon I get Mar 6 09:28:19 WARNING[31213]: loader.c:325 __load_resource: libmysqlclient.so.14: cannot open shared object file: No such file or directory |
| 14:24.59 | mrchicken | however I do a find / -name libmysqlclient.so and I find it in /usr/lib |
| 14:25.17 | mrchicken | I edited ld.`so.conf |
| 14:25.22 | mrchicken | and then did a ldconfig |
| 14:25.29 | mrchicken | to include /usr/lib |
| 14:25.39 | mrchicken | but no luck :( |
| 14:26.16 | *** join/#asterisk skirmisha (i=507e4918@gateway/web/cgi-irc/ircatwork.com/x-38c2b4d0a6d38ba3) |
| 14:26.21 | skirmisha | hello |
| 14:26.31 | skirmisha | anyone with rxfax and 1.4 version |
| 14:29.02 | *** join/#asterisk af_ (n=getsmart@ip-202-133.sn2.eutelia.it) |
| 14:29.25 | *** join/#asterisk anonymouz666 (n=anonymou@re366.compuland.com.br) |
| 14:30.40 | anonymouz666 | when we see 0? on GotoIF the condition is true or false? |
| 14:30.43 | skirmisha | i am getting strange thing |
| 14:30.57 | skirmisha | whenever i call rxfax it does not pickup anything |
| 14:31.11 | skirmisha | can it be from version of spandsp |
| 14:32.05 | skirmisha | ? |
| 14:33.05 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 14:33.05 | *** mode/#asterisk [+o mog] by ChanServ |
| 14:38.44 | skirmisha | hello |
| 14:38.46 | skirmisha | where are u |
| 14:39.50 | puzzled | coppice: can you please tell me which is the latest/best chan_unicall for 1.2? The one in snapshots/unicall/asterisk-1.2.x-20060205/ or the one in unicall/unicall-0.0.3pre9/asterisk-1.2.x/ |
| 14:41.01 | coppice | unicall/unicall-0.0.3pre9/asterisk-1.2.x/ is well tested. snapshots/unicall/asterisk-1.2.x-20060205 is an experimental one with the interfaces changes, but I never had much feedback about it. most people use 0.0.3pre9 |
| 14:41.27 | coppice | work on Unicall is finally resuming :-) |
| 14:41.39 | skirmisha | what is that for |
| 14:42.41 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@ppp-70-255-106-4.dsl.hstntx.swbell.net) |
| 14:42.51 | puzzled | coppice. thanks. will use the 0.0.3pre9 version then. will it include "dynamic spans"? |
| 14:43.13 | puzzled | skirmisha: funky protocol used in South America and Asia (iirc) |
| 14:43.29 | coppice | what is a dynamic span? |
| 14:44.02 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 14:44.38 | puzzled | coppice:no idea. red-fone mentioned it is missing and they are going through the code to add it |
| 14:45.02 | puzzled | coppice: if you want to have a chat with them about this I can email you my contact |
| 14:45.53 | coppice | that code is pretty much out of date now. If people want to add stuff to my code without talking to me, why should I care |
| 14:46.29 | puzzled | coppice: you have got a point there. shall I refer my contact to you? |
| 14:46.42 | coppice | no |
| 14:46.46 | puzzled | ok |
| 14:47.16 | skirmisha | so anyone who can give me help with fax |
| 14:47.39 | puzzled | skirmisha: there is a tutorial on asteriskguru.com |
| 14:49.04 | skirmisha | well the problem is when rxfax is called nothing happened |
| 14:49.25 | skirmisha | app_rxfax.so Trivial FAX Receive Application 0 |
| 14:49.25 | skirmisha | app_txfax.so Trivial FAX Transmit Application 0 |
| 14:49.34 | skirmisha | this is when i do show modules |
| 14:49.54 | skirmisha | and this is when i called it |
| 14:50.11 | JT | still waiting for red-fone to come out with hardware EC |
| 14:50.32 | skirmisha | Executing [s@macro-faxreceive:3] RxFAX(SIP/............"/var/spool/asterisk/fax/1173195477.0.tif") in new stack |
| 14:50.54 | skirmisha | but asterisk does not open the line and fax is hanging |
| 14:51.07 | skirmisha | means no answer just like on hook |
| 14:51.20 | puzzled | coppice: red-fone told me that they have a telco setup to test R2 should you be interested |
| 14:51.35 | skirmisha | any idea why this is happening |
| 14:51.51 | coppice | there are hundreds of people in south america testing unicall every day :-) |
| 14:51.59 | puzzled | JT: that sounds like a reasonable requirement :) |
| 14:52.13 | JT | puzzled: supposedly they're "working on it" |
| 14:52.54 | puzzled | coppice: hehe point taken |
| 14:55.24 | *** join/#asterisk IPmonger (n=ipmonger@c-68-84-208-206.hsd1.pa.comcast.net) |
| 14:58.03 | *** part/#asterisk kmilitzer (n=km@office-gw.westend.com) |
| 14:58.46 | [TK]D-Fender | skirmisha: Have you considered putting "Answer" in front of that? And then theres the thought of "WTF are you doing trying to fax over SIP?" |
| 14:59.07 | puzzled | [TK]D-Fender: adventurous comes to mind :) |
| 14:59.36 | [TK]D-Fender | puzzled: More like "anuerism" |
| 15:00.42 | puzzled | for those 2 faxes a year |
| 15:00.53 | mutilator | electronic fax/fax to email man |
| 15:01.14 | [TK]D-Fender | chan_smokesignal.so FTW! |
| 15:01.15 | *** join/#asterisk Cyon (n=cyon@216.179.31.170) |
| 15:01.16 | skirmisha | yes i am doing fax over sip |
| 15:01.27 | *** join/#asterisk sergee (n=opera@195.94.224.197) |
| 15:01.55 | puzzled | skirmisha: afaik that is unreliable unless you have rock solid T.38 support in Asterisk |
| 15:02.08 | *** join/#asterisk mercestes (n=merceste@rrcs-71-41-157-70.sw.biz.rr.com) |
| 15:02.19 | Cyon | Doesn't 1.4.1 have rock-solid T.38 support? |
| 15:02.28 | Cyon | (T.38 passthrough) |
| 15:02.48 | puzzled | how about endpoint? |
| 15:04.32 | skirmisha | well i am doing it over g711 |
| 15:04.38 | skirmisha | but still get stuck |
| 15:04.43 | RoyK | puzzled: no endpoint support in asterisk.... |
| 15:04.49 | RoyK | t38ep |
| 15:05.00 | sergee | can anybody help me with AGI? i'm trying to get DTMF into perl script with "WAIT FOR DIGIT 5000", but it always return '1' instantly |
| 15:05.00 | puzzled | skirmisha: doesn't matter. a bit of jitter and some lost packets will kill the fax |
| 15:05.15 | sergee | maybe something wrong with dtmfmode or something? |
| 15:05.16 | puzzled | RoyK: I know |
| 15:05.24 | skirmisha | puzzled: it does not create the file for fax |
| 15:05.37 | skirmisha | so i don;t know where the problem is |
| 15:05.44 | sergee | i tryed that a year ago and everything worked fine, but now i can't get dtmf into my script... |
| 15:05.49 | puzzled | skirmisha: maybe you don't have the right permission in the directory where it wants to write the fax |
| 15:06.00 | skirmisha | nope it is not that |
| 15:06.16 | skirmisha | the fax machine does not start to send the page |
| 15:06.22 | dual-man | hi guys, who is the voice for asterisk sounds |
| 15:06.25 | skirmisha | which means it does not get proper signal |
| 15:06.29 | Cyon | puzzled: Sure, but fixed jitter buffers can account for the fitter, and proper QOS across a private network will deal with packet loss. |
| 15:06.39 | Cyon | And LLQ will really settle jitter as well. |
| 15:06.48 | skirmisha | how can i debug this |
| 15:07.06 | FuriousGeorge | dual-man: her name is allison |
| 15:07.13 | dual-man | can i get custom sounds |
| 15:07.13 | puzzled | dual-man: Allison. check http://thevoice.digium.com/ |
| 15:07.15 | coppice | skirmisha: if the file can't be opened, the fax machine won't start to send the fax |
| 15:07.19 | FuriousGeorge | dual-man: i hear she does private work too |
| 15:07.27 | skirmisha | hmm |
| 15:07.29 | puzzled | dual-man: yes you can buy voice prompts from her |
| 15:07.35 | dual-man | ok |
| 15:08.10 | *** join/#asterisk Ahrimanes (n=ma@x1-6-00-0a-e4-2e-90-43.k707.webspeed.dk) |
| 15:08.17 | puzzled | Cyon: theoretically that sounds nice but in reality FoIP just seems a bitch |
| 15:08.18 | JT | Cyon: or you could just be smart and use t.38... |
| 15:08.52 | Cyon | JT: Ummm, that's what I'm talking about? |
| 15:09.02 | hellop | sergee I'm trying to work with AGI perl... wanna collaborate? |
| 15:09.10 | JT | Cyon: sounded like you were talking about fixing fax over g.711 |
| 15:09.17 | JT | over ip |
| 15:09.19 | skirmisha | still the same |
| 15:09.39 | Cyon | JT: Nah, just responding to puzzled; but t.38 still can't handle tons of packet loss, like a voice conversation. |
| 15:09.56 | sergee | hellop: me too, however, i have a problems with $AGI->wait_for_digit(5000) |
| 15:10.05 | puzzled | Cyon: yup which is what makes it difficult |
| 15:10.05 | Cyon | JT: No matter what you are doing, when dealing with data you really want minimal to no packet loss and minimal jitter. |
| 15:10.14 | Pon`work | is symmetric RTP supported in Asterisk 1.4.x ? |
| 15:10.30 | JT | Cyon: doesn't matter if you do something like email the fax |
| 15:11.11 | *** join/#asterisk heh_v_water (n=burd@71-210-51-58.hlna.qwest.net) |
| 15:11.44 | skirmisha | i can not understand why asterisk did not create file |
| 15:11.56 | FuriousGeorge | fax machine makers need to agree on some FoIP standard, and make their devices with a lan/wan interface |
| 15:12.10 | Qwell | FuriousGeorge: you mean like T.38? |
| 15:12.17 | FuriousGeorge | i assume sip/sdp/and some sort of digital standard |
| 15:12.44 | FuriousGeorge | Qwell: no, im talking about taking all analog out of the loop |
| 15:12.55 | coppice | the standards have existed for several years. the fax makers don't use them |
| 15:14.17 | *** part/#asterisk sergee (n=opera@195.94.224.197) |
| 15:14.49 | angryuser | is there a way to disable a buzz sound which is supposed to notify Sip user that he has a second call? |
| 15:15.22 | FuriousGeorge | angryuser: im pretty sure that is controlled by the sip UA, so depends what phone u are using |
| 15:15.36 | FuriousGeorge | you can limit how many calls go to that phone in asterisk |
| 15:15.51 | angryuser | FuriousGeorge: i am using snom's and buzz soint is on internal calls |
| 15:15.56 | angryuser | *sound |
| 15:16.27 | FuriousGeorge | angryuser: i happen to use snoms myself, and i happen to be in the firmware at this moment, and im pretty sure the option is in there somewhere, lemme check |
| 15:17.21 | angryuser | FuriousGeorge: gm, i have this sound even on distant fxs client(dlink) |
| 15:17.26 | puzzled | coppice: besides when it's ready, would "when" be a matter of weeks or is months more appropriate? |
| 15:17.31 | skirmisha | damn |
| 15:17.39 | skirmisha | is there anyone using rxfax |
| 15:17.56 | *** join/#asterisk _VoicePulse (n=contact@unaffiliated/voicepulse) |
| 15:18.00 | puzzled | skirmisha: nope |
| 15:18.27 | FuriousGeorge | angryuser: are we talking about the same thing? im talking about the tone (not like a buzz) the phone plays when a second call comes in that you can answer if you choose |
| 15:18.47 | coppice | puzzled: when is what ready? |
| 15:19.03 | puzzled | coppice: your shiny new Unicall release |
| 15:19.04 | angryuser | FuriousGeorge: yes we are talking about the same thing |
| 15:19.04 | FuriousGeorge | that is a setting under advanced where you can set it to on, visual only, off |
| 15:19.06 | *** join/#asterisk quidpro (n=quid@CPE00131078ba5d-CM000f9f7eff1e.cpe.net.cable.rogers.com) |
| 15:19.28 | *** join/#asterisk jorguna (n=Sergo@217.147.230.35) |
| 15:19.36 | jorguna | sup ppl |
| 15:19.38 | jorguna | :) |
| 15:19.38 | quidpro | Hmm, out of curiousity what is the symptom of the remote crash bug? Will * totally die or merely hang? |
| 15:19.41 | jorguna | have some problem |
| 15:19.44 | jorguna | wanted to get help |
| 15:19.48 | coppice | puzzled: there should be some stuff appearing in a couple of weeks, and it should roll along from there |
| 15:19.49 | file | /clear |
| 15:20.02 | puzzled | coppice: great, thanks |
| 15:20.07 | jorguna | I've go Cisco IP phone 7906 |
| 15:20.12 | angryuser | FuriousGeorge: ok i will try it no thanks |
| 15:20.20 | jorguna | want to connect it to my my Asterisk |
| 15:20.25 | jorguna | can any1 explain me how? |
| 15:20.33 | FuriousGeorge | angryuser: its called "call waiting indication" and another possible option is "ringer" which i assume would just make the phoen ring as per usual |
| 15:20.37 | angryuser | FuriousGeorge: i mean thank you; |
| 15:20.41 | FuriousGeorge | ;) |
| 15:20.44 | puzzled | jorguna: did you check voip-info.org? |
| 15:20.53 | jorguna | yep |
| 15:20.57 | jorguna | look |
| 15:21.06 | jorguna | my steps: |
| 15:21.09 | *** join/#asterisk BSDTech (n=RNeese@ppp-69-239-113-101.dsl.irvnca.pacbell.net) |
| 15:21.11 | jorguna | 1. made config on aster |
| 15:21.19 | jorguna | 2. copied files to tftp |
| 15:21.25 | jorguna | 3. uploaded files |
| 15:21.29 | jorguna | result |
| 15:21.33 | jorguna | not working |
| 15:21.33 | jorguna | :D |
| 15:21.48 | puzzled | jorguna: sorry can't help you. I don't use a 7906 |
| 15:21.55 | jorguna | :S |
| 15:21.58 | *** join/#asterisk _Vile (n=vile@bc182112.bendcable.com) |
| 15:22.18 | *** join/#asterisk nasls_lsa (n=chatzill@athedsl-219219.otenet.gr) |
| 15:23.54 | angryuser | FuriousGeorge:ringer produce this sound, tryin visual only |
| 15:25.44 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 15:25.45 | *** mode/#asterisk [+o anthm] by ChanServ |
| 15:25.55 | *** join/#asterisk quidpro (n=quid@CPE00131078ba5d-CM000f9f7eff1e.cpe.net.cable.rogers.com) |
| 15:25.58 | quidpro | Argh. |
| 15:26.11 | *** join/#asterisk anthony] (n=anthony@175.21.188.72.cfl.res.rr.com) |
| 15:26.22 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
| 15:28.12 | *** join/#asterisk marv[work] (n=timr@24.214.206.254) |
| 15:28.49 | reber | what is extension "s" ? |
| 15:29.20 | [TK]D-Fender | reber: Go read : http://www.voip-info.org/wiki/view/Asterisk+standard+extensions |
| 15:30.36 | reber | [TK]D-Fender, thanks |
| 15:30.40 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 15:30.44 | coppice | skirmisha: I've had reports of success with recent versions of spandsp, the rxfax in http://www.soft-switch.org/downloads/snapshots/spandsp/test-apps-asterisk-1.4/ and * 1.4. I've never tried it, though |
| 15:30.56 | Chris-NB | hi |
| 15:31.00 | Chris-NB | anyone using dusnet? |
| 15:31.20 | reber | [TK]D-Fender, btw, yesterday 484 error wasn't a problem on the softphone "early dial", but a problem on asterisk extensions. |
| 15:31.48 | *** join/#asterisk Ch0Hag (n=mking@87.127.170.250) |
| 15:31.51 | *** join/#asterisk dasenjo (n=dasenjo@190.24.176.236) |
| 15:32.03 | quidpro | Hmmm, I still wonder why such a big conspiracy of silence on this crash bug. I thought this was OSS. :) |
| 15:32.31 | *** join/#asterisk JerJer (n=PhatJ@pdpc/supporter/bronze/jerjer) |
| 15:34.56 | hellop | shh |
| 15:36.03 | quidpro | Can't find any information on symptoms and what versions are affected by it. It's not like I am trying to find out what the malicious code to send is. |
| 15:37.05 | Corydon-w | quidpro: no conspiracy. We're just allowing everybody to upgrade before the announcement goes out |
| 15:37.36 | *** join/#asterisk techie (n=techie@c-67-181-184-170.hsd1.ca.comcast.net) |
| 15:37.54 | Corydon-w | quidpro: it's the responsible thing to do |
| 15:38.54 | quidpro | Corydon: Understandable... but I seem to have problems with 1.2.15 locking up, not so much in 1.2.14... now don't know if this is because of the bug or if it is something different... so that is why I am hesitant to move to 1.2.16. |
| 15:39.02 | ping2921 | anyone in here has worked with asterisk-java? |
| 15:39.34 | Corydon-w | quidpro: if you're having issues, file a bug, please. We can't fix issues if you don't report them. |
| 15:40.20 | quidpro | Corydon: Would love to... but the lockup in 1.2.15 just froze... nothing in debug or anything. |
| 15:40.37 | JerJer | is this valid/sane exten => _XXX,1,n(hangup),Hangup ? |
| 15:40.42 | JerJer | that's a label right? |
| 15:40.43 | quidpro | So hard to track a bug when there is no papertrail. |
| 15:40.51 | Corydon-w | quidpro: attach with gdb and get a 'thread apply all bt full' |
| 15:41.06 | Corydon-w | JerJer: you have an extra ,1 in there |
| 15:41.11 | JerJer | er |
| 15:41.12 | *** join/#asterisk nasls_lsa (n=chatzill@athedsl-219219.otenet.gr) |
| 15:41.12 | JerJer | ahh |
| 15:41.20 | JerJer | exten => _XXX,n(hangup),Hangup ? |
| 15:41.25 | Corydon-w | Yep |
| 15:41.28 | quidpro | Well I am running 1.2.14 until I find out more what's going on. |
| 15:41.49 | JerJer | then one can do Goto(hangup) from the same context |
| 15:41.54 | Corydon-w | quidpro: what's your IP address again? |
| 15:41.54 | JerJer | or whatever |
| 15:42.01 | Corydon-w | JerJer: correct |
| 15:42.16 | *** join/#asterisk svenna_ (n=svenna@p548D0417.dip0.t-ipconnect.de) |
| 15:43.47 | dual-man | i'm using asterisk mainly for fxo and fxs lines, i bought a voip phone just now to try it out, for some reason i can hear the zap extension clearly but the other end hears only parts of my converstaion, what could cause this? |
| 15:44.05 | dual-man | it keeps cutting in and out |
| 15:44.11 | *** join/#asterisk ppyy (i=ppyy@58.216.25.126) |
| 15:44.13 | dual-man | the voip phone sending |
| 15:45.23 | [TK]D-Fender | JerJer: n = bad, labels = bad, hard numbering = GOOD ! |
| 15:45.32 | dual-man | what is the codec that needs the least bandwidth? |
| 15:46.02 | [TK]D-Fender | dual-man: G.723.1 I believe, but * doesn't really support it |
| 15:46.08 | [TK]D-Fender | dual-man: Runner up = G.729 |
| 15:46.56 | *** join/#asterisk darylvoip (n=darylvoi@c-71-225-231-66.hsd1.pa.comcast.net) |
| 15:46.57 | mercestes | D-Fender is "old skool." He doesn't need any of this fluffy new fangled stuff to make life easier, like the "n" priority |
| 15:47.13 | [TK]D-Fender | mercestes: Doesn't change my life any.... I tend to do it right the FIRST time ;) |
| 15:47.19 | mercestes | lol |
| 15:47.54 | darylvoip | Hey all....quick question. Does anyone know how I can find out how many g729 channels I have in use through the manager interface? I'm trying to track license usage and alert when we're getting close. |
| 15:47.59 | boch | how do i get a random integer for a var in the dialplan ? |
| 15:48.08 | skirmisha | ok i think rxfax does not detect faxes |
| 15:48.36 | [TK]D-Fender | skirmisha: It is not meant to "detect", it is meant to immediately send out the answer tones... |
| 15:48.44 | [TK]D-Fender | skirmisha: and "jsut do it". |
| 15:48.53 | Corydon-w | darylvoip: use the Action: Command to send a Command: show 729 |
| 15:49.00 | [TK]D-Fender | skirmisha: If you want detection, look at the "fax" standard extension. |
| 15:49.07 | Corydon-w | err, show g729 |
| 15:49.15 | jeremy_g | does anyone have a copy of this script http://www.krisk.org/asterisk/ast2sql.pl (taken from http://www.voip-info.org/wiki/view/Asterisk+RealTime+Static) |
| 15:50.03 | nasls_lsa | I was editing / testing extensions.conf and sip.conf , and sudenly asterisk became to slow when it reloads configs .. any ideas why ? |
| 15:50.04 | skirmisha | [TK]D-Fender: should i install nvfax |
| 15:50.11 | darylvoip | Croydon.....that works through the manager interface? I didn't see it documented anywhere. |
| 15:51.04 | Corydon-w | darylvoip: 'show manager commands' shows a Command action |
| 15:51.32 | darylvoip | OK...thanks. Been a while since I've done much in Asterisk. |
| 15:51.44 | Corydon-w | darylvoip: and 'show manager command Command' shows the information about it |
| 15:52.07 | *** join/#asterisk murdmath (n=vircuser@c-24-10-190-87.hsd1.ut.comcast.net) |
| 15:52.41 | *** join/#asterisk badcfe (n=cso@LNeuilly-152-22-86-193.w193-251.abo.wanadoo.fr) |
| 15:52.46 | *** join/#asterisk jpmiles (n=h@200.58.147.131) |
| 15:53.11 | jpmiles | is asterisk good? |
| 15:53.20 | darylvoip | OK...not seeing show g729. Works from the console, of course. |
| 15:53.29 | mercestes | jpmiles: no. It sucks. |
| 15:53.30 | nasls_lsa | jpmiles with one word : YES ! ! ! ! ! |
| 15:53.35 | mercestes | jpmiles: use nortel or mitel. It's much better. |
| 15:53.38 | darylvoip | Oh well. Its not a huge deal. But it it was already there I'd track it. |
| 15:53.46 | Corydon-w | darylvoip: does the manager user you're connecting as have 'command' authority in manager.conf ? |
| 15:53.52 | mercestes | jpmiles: Save yourself the pain, trouble and loss of employement. Go Mitel. seriously. It's worth the money. |
| 15:53.54 | Sweeper | arg |
| 15:53.56 | mercestes | jpmiles: don't look back....just go. |
| 15:54.09 | darylvoip | Corydon..I can make it that way |
| 15:54.11 | mercestes | jpmiles: Good luck. |
| 15:54.14 | Sweeper | what lame options should I be using to encode on-hold music? |
| 15:54.21 | darylvoip | Are you saying I can run arbitrary console commands if its set that way? |
| 15:54.29 | Corydon-w | darylvoip: yes |
| 15:54.41 | darylvoip | Excellent. THanks. Off to voip-info for me. I'm sure its doced there. |
| 15:54.48 | darylvoip | THanks for the help. Very good pointer. |
| 15:55.32 | mercestes | lol. My work here is done. |
| 15:55.32 | murdmath | mercestes: I think you made a friend. |
| 15:55.38 | mercestes | murdmath: I think so too. :) |
| 15:55.41 | skirmisha | this is what i do |
| 15:55.42 | skirmisha | http://scottstuff.net/blog/articles/2004/03/28/faxing-with-asterisk |
| 15:55.47 | skirmisha | but it does not work |
| 15:56.00 | mercestes | skirmisha: So what you mean is, "This is what I *want* to do." |
| 15:56.23 | *** join/#asterisk wunderkin (i=kev@ip72-208-3-221.ph.ph.cox.net) |
| 15:56.32 | mercestes | skirmisha: Might I suggest hylafax + iaxmodem? |
| 15:57.20 | skirmisha | mercestes: i have only SIP |
| 15:57.24 | Sweeper | facking a |
| 15:57.32 | skirmisha | fax is coming over sip |
| 15:57.42 | *** join/#asterisk vgster (n=vgster@81.96.139.59) |
| 15:57.47 | mercestes | skirmisha: Iaxmodem is local only, and you would be hard pressed to install * with no iax support. |
| 15:57.53 | mercestes | skirmisha: Doesn't matter. |
| 15:58.37 | mercestes | skirmisha: Fax can come from telepathy for all * cares, bring it in g711u on sip and out g711u on IAX and viola, no transcoding. |
| 15:58.56 | mercestes | skirmisha: Then you dump it to your local software modem, convert it out to pdf, and then mail the pdf out. |
| 15:59.46 | skirmisha | mercestes: i think we are talking for 2 diff things |
| 15:59.54 | skirmisha | i want to use virtual fax |
| 16:00.07 | *** join/#asterisk booray (n=booray@adsl-75-4-8-7.dsl.irvnca.sbcglobal.net) |
| 16:00.08 | skirmisha | means calling a number routed to asterisk |
| 16:00.23 | mercestes | yes |
| 16:00.23 | skirmisha | and asterisk just accept it and convert it to pdf |
| 16:00.26 | mercestes | yes |
| 16:00.35 | mercestes | and then ....email it out, maybe? |
| 16:01.03 | mercestes | that would be exactly what I meant to say. |
| 16:01.04 | mercestes | http://svn.netdomination.org/gentoo-voip/wiki/GentooAsteriskInstall |
| 16:01.08 | mercestes | great howto |
| 16:01.14 | quidpro | Anybody know what would cause these entries and cause * to go brain dead (responds to cli commands, doesn't process calls).... http://pastebin.ca/383726 |
| 16:01.17 | booray | is there a detailed changelog available for 1.4.1? |
| 16:01.57 | [TK]D-Fender | skirmisha: Did you even look at the URL for that link. its 3 years old! Do you really think you can follow those instructions with todays versions? |
| 16:01.57 | skirmisha | let me see |
| 16:02.10 | Corydon-w | quidpro: it's a simple timing race. Those are usually safe to ignore |
| 16:02.19 | mercestes | D-Fender: I still use it. |
| 16:03.11 | Corydon-w | quidpro: usually that means that a client responded just after the expire for the invite occurred |
| 16:03.12 | skirmisha | mercestes: what do u use |
| 16:03.16 | mercestes | and in my experience the rtxfax stuff is near impossible to even get installed so....yea. |
| 16:03.19 | skirmisha | IAX or rxfax |
| 16:03.26 | [TK]D-Fender | skirmisha: Try the "fax" standard extensionand make sure that T.38 is not getting in the way of your fax reception. |
| 16:03.38 | mercestes | skirmisha: I swore off rxfax after about the 87th time it failed to compile and went with hylafax and iaxmodem. |
| 16:04.06 | skirmisha | mercestes: it is stable |
| 16:04.08 | mercestes | and I didn't try to do that "fax tone detection" crap that asterisk tries to implement. |
| 16:04.25 | [TK]D-Fender | mercestes: I've had pretty good success rate with SpanDSP & compiling has only caused minor questions solved in a days time max. |
| 16:04.34 | mercestes | define stable. |
| 16:04.41 | mercestes | stable as in, works reliably, then yes. |
| 16:04.48 | mercestes | stable as in, never fails. no. |
| 16:04.58 | quidpro | Corydon: Hmm, yeah it doesn't sound like anything... but the fact that * stopped dead after it... strange. Let me go back and see if there was anything previously. It is funy I notice that sometimes when I do SIP debugs in *, it will freeze up occasionally a few hours later. |
| 16:05.05 | mercestes | You'll still have that one guy on the other side of the planet that will swear up and down that they can fax anybody in the world other than you. |
| 16:05.19 | mercestes | but I'v ehad those ppl on standard fax machines on analog lines so... |
| 16:05.50 | skirmisha | i understand |
| 16:05.54 | quidpro | Corydon: Nope nothing else of note. |
| 16:06.20 | mercestes | I don't think D-Fender uses gentoo tho, so that's likely why he had SpanDSP success and I had largely failure. |
| 16:06.35 | *** join/#asterisk Iamnacho (i=Iamnacho@ip68-103-153-140.ks.ok.cox.net) |
| 16:06.55 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 16:06.58 | [TK]D-Fender | mercestes: I've installed it on Slackware, and CentOS without much trouble. |
| 16:07.01 | coppice | skirmisha: are you trying to fax by SIP or from a PSTN connection? |
| 16:07.50 | [TK]D-Fender | coppice: Oh, and an official big "Thank You" for all that you do for us... |
| 16:08.07 | skirmisha | i am trying pstn fax to sip |
| 16:08.41 | coppice | what does that mean? do you have a PSTN line into *, or a SIP call? |
| 16:08.43 | murdmath | Before I call digium about this I figured I'd ask here. When using a T1 and making an outbout call, has anyone experienced a situation where you won't hear the first word the answering party says? |
| 16:08.55 | skirmisha | yes i have ISDN to sip router |
| 16:09.16 | skirmisha | so fax is on pbx -> isdn router -> asterisk |
| 16:09.24 | coppice | rxfax does not work with VoIP calls. |
| 16:09.35 | skirmisha | huh |
| 16:09.57 | skirmisha | that's why it does not work |
| 16:10.01 | coppice | it will never work reliably, so I make no attempt to make it work at all |
| 16:10.03 | skirmisha | so what is working on sip |
| 16:10.30 | skirmisha | mercestes: is your solution working on sip |
| 16:10.50 | *** join/#asterisk x86 (n=x86@p3m/member/x86) |
| 16:10.51 | rajiv | anyone know if the security bug which was fixed in 1.2.16 and 1.4.1 is present in 1.0.12 ? |
| 16:11.00 | mercestes | murdmath: Yes. I usually just do a "wait(1)" for IVRs |
| 16:11.10 | [TK]D-Fender | skirmisha: Lets make this simple : Fax over VoIP = FAILURE. FOREGET IT. |
| 16:11.35 | mercestes | skirmisha: it worked fairly reliably over SIP on a *good* SIP connection. |
| 16:11.50 | mercestes | skirmisha: And why I suggest it because it was the only solution that worked reasonably well. |
| 16:12.08 | mercestes | but they're right. Fax over VoIP = failure. your *going* to run into problems. |
| 16:12.12 | murdmath | mercestes: If I was to call you and you picked up the call and immediattly said Hello I would hear that. |
| 16:12.15 | skirmisha | mercestes: how do u get faxes, directly from isdn or over sip? |
| 16:12.26 | murdmath | mercestes: Wouldn't hear that. |
| 16:13.16 | mercestes | murdmath: Yea, . Old version of *? |
| 16:13.28 | murdmath | mercestes: 1.2.12.1 |
| 16:13.42 | [TK]D-Fender | skirmisha: SIP = VoIP. You really need to think on that more... |
| 16:13.42 | mercestes | skirmisha: over PRI if I can help it but the times I had to use SIP I used hylafax + iaxmodem. |
| 16:14.08 | murdmath | mercestes: Not to old. |
| 16:14.16 | mercestes | skirmisha: let me put it this way. With some black magic and super-leet scripting the company I was workign for was able to get faxes off of a ship in the Atlantic over satelite. |
| 16:14.55 | mercestes | no, I don't have that script but basically i tchecked for a connection before attempting to send. |
| 16:15.12 | quidpro | Maybe it was INMARSAT voice. :) |
| 16:15.18 | mercestes | so, the answer is "no!" but if you absolutely must tread down this path of fax over SIP, I suggest using hylafax + iaxmodem. |
| 16:15.34 | cpm | fax over sip = evil |
| 16:15.42 | mercestes | quidpro: marisat. Thanks, I was trying to remembre the name of it. |
| 16:15.45 | dual-man | what is the difference btn g711a and g729? |
| 16:15.53 | Qwell[] | dual-man: about 18 |
| 16:16.02 | dual-man | which is better |
| 16:16.39 | coppice | G.729 is better because it make lot of people money through patent royalties |
| 16:17.28 | dual-man | do only ip ohones use these codecs or also zap channels? |
| 16:17.40 | *** join/#asterisk ChkDigit (n=mrw@static24-89-65-166.regina.accesscomm.ca) |
| 16:17.41 | dual-man | excuse my ignorance |
| 16:18.17 | coppice | phone lines use G.711 |
| 16:18.29 | *** join/#asterisk thinwires (n=thinwire@24-49-196-96.kntnny.adelphia.net) |
| 16:18.35 | anonymouz666 | -- Executing GotoIf("SIP/10.0.0.1-b4d96300", "1?s-CHANUNAVAIL|10") in new stack |
| 16:18.51 | skirmisha | i'd better forget about it |
| 16:18.55 | anonymouz666 | I don't know where is this damn line on dialplan |
| 16:18.56 | skirmisha | too much hasle |
| 16:19.34 | mercestes | skirmisha: Good man. now you are embracing the way. |
| 16:19.36 | *** join/#asterisk ppyy (n=upirc@211.138.205.197) |
| 16:20.38 | *** part/#asterisk ppyy (n=upirc@211.138.205.197) |
| 16:22.19 | skirmisha | thanks a lot guys |
| 16:22.26 | mercestes | np |
| 16:22.55 | *** join/#asterisk ppyy (n=upirc@211.138.205.197) |
| 16:23.27 | *** join/#asterisk saftsack (n=oliver@pD9E07C86.dip.t-dialin.net) |
| 16:23.53 | *** join/#asterisk progmonkey (n=asdfkle@c-71-195-248-169.hsd1.ma.comcast.net) |
| 16:24.11 | ManxPower | anonymouz666: That is the RESULT of the evaluated line. is would be something with a Goto and ?s-${DIALSTATUS},10) near the end of th line |
| 16:24.26 | progmonkey | Can you Call a MYSQL procedure in Asterisk? |
| 16:24.46 | JerJer | progmonkey: how does one call a mysql procedure without asterisk? |
| 16:25.22 | progmonkey | JerJer: Call <procedure name>(<variables>) |
| 16:25.43 | JerJer | like via the mysql client ? |
| 16:26.03 | progmonkey | yes |
| 16:26.22 | progmonkey | it is basically the same as a select statement |
| 16:26.35 | progmonkey | except your sql code is in the database not in asterisk |
| 16:26.35 | *** join/#asterisk ppyy (n=upirc@211.138.205.197) |
| 16:26.35 | JerJer | Corydon has a res_odbc (i think) |
| 16:26.54 | Corydon-w | JerJer: func_odbc |
| 16:26.58 | JerJer | func - there we go |
| 16:27.22 | progmonkey | where can I look at it? |
| 16:27.25 | *** join/#asterisk jarg (n=jarg@200.56.225.61) |
| 16:27.27 | anonymouz666 | ManxPower: there is nothing with 10 on the dialplan |
| 16:27.35 | JerJer | perhaps that might be time for a func_mysql |
| 16:27.39 | Corydon-w | http://asterisk.org/func_odbc |
| 16:27.51 | anonymouz666 | JerJer what about app_mysql? |
| 16:27.52 | Corydon-w | JerJer: ugh |
| 16:27.58 | JerJer | lol |
| 16:28.36 | ManxPower | anonymouz666: It is in there somewhere. Do you have any #include lines? |
| 16:29.15 | JerJer | Corydon-w: is there any benefit to connecting to mysql via odbc ? |
| 16:29.19 | progmonkey | JerJer: We wrote a patch for mysql procedures in 1.2 but figured they would have included something for 1.4 |
| 16:29.21 | ManxPower | anonymouz666: it very well might be [macro-stdexten] |
| 16:29.22 | JerJer | i really don't know |
| 16:29.35 | Corydon-w | JerJer: none, AFAICT |
| 16:29.51 | Corydon-w | Well, the configuration is simpler, but that's about it |
| 16:30.48 | JerJer | perhaps maybe it doesn't segfault ?? |
| 16:31.00 | JerJer | mysql C lib was totally not forgiving |
| 16:31.23 | JerJer | was/is |
| 16:31.31 | Corydon-w | JerJer: oh, I meant the mysql config is simpler |
| 16:31.34 | anonymouz666 | ManxPower: there is no such macro on the very very confuse AMP dialplan |
| 16:31.42 | JerJer | yep |
| 16:31.59 | Corydon-w | The odbc config is a little more complex, but that's a one-time thing |
| 16:32.39 | Corydon-w | and the API is documented from more than one vendor, which is good in case you need clarification |
| 16:32.59 | Corydon-w | Two companies saying the same thing with different words |
| 16:33.17 | anonymouz666 | there is a damn script called dialparties.agi |
| 16:33.19 | anonymouz666 | argh |
| 16:34.12 | mutilator | #include just sticks whatever is in said text file into the config right where ya put the statement right? |
| 16:34.37 | mutilator | i could do... |
| 16:34.40 | mutilator | [default] |
| 16:34.42 | mutilator | exten => 911,1,Ringing() |
| 16:34.44 | Corydon-w | mutilator: correct |
| 16:34.45 | mutilator | [voicemail] |
| 16:34.49 | mutilator | #include <filename.conf> |
| 16:34.55 | mutilator | [incomingcall] |
| 16:35.00 | mutilator | exten => 411,1,Ringing() |
| 16:35.03 | Corydon-w | mutilator: wrong syntax, #include "filename.conf" |
| 16:35.16 | mutilator | Corydon-w: yea.. but it does what i'm thinking right? |
| 16:35.18 | *** join/#asterisk ppyy (n=upirc@211.138.205.197) |
| 16:35.22 | Corydon-w | mutilator: yes |
| 16:35.22 | mutilator | the general idea |
| 16:35.23 | mutilator | ok |
| 16:35.25 | mutilator | cool thx |
| 16:35.32 | ManxPower | anonymouz666: NOW you understand why we hate AMP so much. |
| 16:36.19 | anonymouz666 | yeah it sucks. |
| 16:36.26 | BSDTech | amp is old and crap |
| 16:36.34 | BSDTech | freepbx is much better |
| 16:36.34 | [TK]D-Fender | ~amp |
| 16:36.42 | jbot | methinks amp is NOT supported here! People using it should join #freepbx (FreePBX is the new name of AMP) |
| 16:36.42 | [TK]D-Fender | ~freepbx |
| 16:36.46 | jbot | well, freepbx is unable to be supported here. It's a complex piece of dialplan and GUI, and can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 16:36.46 | [TK]D-Fender | ~trixbox |
| 16:36.48 | jbot | trixbox is, like, unable to be supported here. Try joining #freepbx and asking there, or use the trixbox forums at http://www.trixbox.org/modules/newbb/ |
| 16:36.49 | BSDTech | and the asterisk-gui leaves alot to be desired |
| 16:36.53 | [TK]D-Fender | They're CrapTASTIC! |
| 16:37.46 | BSDTech | the asterisk-gui is craptastic. its majorly lacking and when it breaks ti takes the whole gui with it |
| 16:38.34 | BSDTech | btw 1.4.1 works on Freebsd and have had no issues |
| 16:38.37 | Pon`work | are real-time includes supported yet? |
| 16:38.53 | BSDTech | zaptel-bsd 1.4 works and libpri 1.4 works |
| 16:39.07 | BSDTech | I should have addons ported today |
| 16:39.35 | JerJer | Corydon-w: i think your func_odbc example is the first time I have seen REPLACE INTO used |
| 16:39.47 | Corydon-w | Heh |
| 16:39.55 | Qwell[] | <3 obscure syntax |
| 16:40.14 | Corydon-w | JerJer: well, I can't do a conditional INSERT or UPDATE and still have a simple example |
| 16:40.51 | Corydon-w | I usually just do an UPDATE and do the conditional INSERT/UPDATE elsewhere |
| 16:41.03 | Corydon-w | i.e. on the web interface |
| 16:41.32 | JerJer | is there more advanced func_odbc examples out there? |
| 16:41.59 | Qwell[] | JerJer: I could dig up the one I wrote, which made him expand the buffer size :P |
| 16:42.00 | Corydon-w | Well, there's a few other examples in the sample config in trunk |
| 16:42.20 | JerJer | cool - i'll have to suck down trunk |
| 16:42.30 | Corydon-w | JerJer: or in svncommunity |
| 16:42.34 | Corydon-w | http://svncommunity.digium.com/view/func_odbc/1.2/ |
| 16:42.51 | Corydon-w | http://svncommunity.digium.com/view/func_odbc/1.2/func_odbc.conf.sample?rev=26&view=markup |
| 16:49.10 | *** join/#asterisk step_quasar (n=step_qua@OL148-144.fibertel.com.ar) |
| 16:52.14 | *** join/#asterisk jm|work (n=jm@sentry.flags.co.uk) |
| 16:53.09 | anonymouz666 | ManxPower: -- Goto (macro-exten-vm,s-CHANUNAVAIL,10) -- there is no such extension/priority on this macro, I will add right now and hope for the best |
| 16:53.49 | anonymouz666 | its amazing there is no such priority on the whole dialplan (10) |
| 16:54.00 | anonymouz666 | someone is setting this |
| 16:54.34 | *** join/#asterisk dual-man (n=dwayne@64-42-247-120.mb.skyweb.ca) |
| 16:54.57 | *** join/#asterisk Bazy (n=bazy@89.137.178.124) |
| 16:55.08 | dual-man | was the callerid on distinctive ring problem ever fixed? I can't seem to get asterisk to pass callerid on distinctive ring |
| 16:56.21 | *** join/#asterisk [[blah]asfd (n=ckwall@63.149.122.93) |
| 16:56.51 | *** part/#asterisk [[blah]asfd (n=ckwall@63.149.122.93) |
| 16:57.54 | *** join/#asterisk infernix (i=nix@unaffiliated/infernix) |
| 17:00.25 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 17:00.57 | *** join/#asterisk russellb (i=russellb@asterisk/developer-and-stable-maintainer/drumkilla) |
| 17:00.57 | *** mode/#asterisk [+o russellb] by ChanServ |
| 17:02.12 | *** join/#asterisk qdk (n=qdk@80.243.125.204) |
| 17:02.39 | *** join/#asterisk RoyK (n=roy@ti211310a080-6495.bb.online.no) |
| 17:04.59 | *** join/#asterisk CunningPike (n=CunningP@dhcp-10-234.district.north-van.bc.ca) |
| 17:07.40 | *** join/#asterisk infernix (i=nix@unaffiliated/infernix) |
| 17:07.45 | *** join/#asterisk squitta (n=squarti@dslb-084-058-021-014.pools.arcor-ip.net) |
| 17:07.56 | squitta | Hi, how can i connect to my local asterisk server, and connect the local asterisk to a SIP Provider Service? |
| 17:08.03 | squitta | What is needed for that? |
| 17:08.12 | squitta | I have an asterisk server installed, with all needed modules and stuff like that |
| 17:11.35 | *** join/#asterisk tuan_modulis (n=chatzill@hvmoduli.enter-net.com) |
| 17:12.34 | _charly_ | squitta: i guess you're german because of your hostname, so do you know the german asterisk book? |
| 17:12.43 | tuan_modulis | here's a question: what's "conditional" call forwarding (as opposed to unconditional forwarding) |
| 17:12.59 | tuan_modulis | what type of conditions? |
| 17:14.28 | _charly_ | squitta: you can find it on http://www.das-asterisk-buch.de/ |
| 17:14.46 | tuan_modulis | hmm, well according to tips and tricks, there's no answer call forwarding and busy call forwarding... |
| 17:14.54 | tuan_modulis | at voipinfo |
| 17:17.42 | squitta | Yeah, i just read it allready. |
| 17:17.57 | FuriousGeorge | do any of the major telcos allow you to "bring your own device" with their voip services? |
| 17:18.11 | FuriousGeorge | or do they all wanna be like vonage and sell you a locked ata |
| 17:18.20 | squitta | But at the moment I am thru with my installation, but cant remember how to configure a softphone, for use with my server |
| 17:18.40 | squitta | i might have something like 501@localhost if thats true, but i dont know :)) |
| 17:20.35 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@ppp-70-255-106-4.dsl.hstntx.swbell.net) |
| 17:24.02 | darviria | could anyone tell me if a system like this is possible....I'd like a way of being able to e-mail my asterisk server and have it send an sms for me, the reason being I currently use skypeout to send sms but it's not really convenient, I need to have a skype client and it has to be a windows machine, the linux version doesn't do sms yet, accessing my e-mail is a lot quicker and more convenient even if I have to do it from webmail |
| 17:24.38 | *** join/#asterisk duco (n=duco@host-85-27-49-12.brutele.be) |
| 17:24.45 | duco | Hello all |
| 17:25.20 | darviria | at home my only windows install is a partition on a laptop, it's not really convenient to boot that up just to send an sms |
| 17:27.19 | *** join/#asterisk oej (n=olle@apollo.webway.se) |
| 17:27.42 | duco | I am running icecast2 who can stream ogg and mp3 files. What is the command line to add in musiconh.conf to permit asterisk read the stream from the icecast? |
| 17:27.56 | Juggie | ~wiki |
| 17:28.15 | duco | with mplayer I run: mplayer http://myserver:8000/mount_point and it works fine. |
| 17:28.33 | Juggie | hmmm |
| 17:28.34 | Juggie | ~centos |
| 17:28.37 | jbot | centos is a rebuild of the Red Hat Enterprise Linux RPMs by the community. Check it out at http://www.centos.org/projects/centos |
| 17:28.42 | Juggie | ~doc |
| 17:28.43 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk |
| 17:28.44 | *** join/#asterisk PupenoR (n=pupeno@200.123.183.91) |
| 17:28.47 | Juggie | there we go |
| 17:28.49 | Juggie | go to the wiki |
| 17:29.36 | duco | Nobody is using stream audio with asterisk? |
| 17:35.04 | BSDTech | lol |
| 17:37.55 | *** join/#asterisk darylvoip (n=darylvoi@c-71-225-231-66.hsd1.pa.comcast.net) |
| 17:40.50 | *** join/#asterisk ajohnson_laptop (n=ajohnson@ariel.gpinc.com) |
| 17:41.21 | *** join/#asterisk CrashHD (n=crashhd@c-67-166-155-233.hsd1.ca.comcast.net) |
| 17:41.45 | ajohnson_laptop | Where can I find some info on what exactly the http interface in Asterisk 1.4 does? |
| 17:43.44 | russellb | there isn't any documentation on it, that i know of ... |
| 17:44.00 | russellb | well, i think there may be some with the AADK (asterisk appliance dev kit) |
| 17:44.01 | ajohnson_laptop | heh, ok. That's what I thought |
| 17:44.03 | russellb | but in any case |
| 17:44.13 | russellb | put simply ... it allows you to use the manager interface over HTTP |
| 17:44.22 | ajohnson_laptop | ah ha |
| 17:44.39 | russellb | so ... think ajax apps |
| 17:44.46 | russellb | that's what the asterisk gui is |
| 17:45.08 | ajohnson_laptop | I think somebody talked about it then |
| 17:45.14 | russellb | mark did probably |
| 17:45.19 | russellb | screenshots are on asterisknow.org i think |
| 17:47.42 | *** join/#asterisk Dr-Linux (n=Nothing@202.125.139.198) |
| 17:47.55 | Dr-Linux | how can i set PIN on dymanic conference room? |
| 17:50.13 | *** join/#asterisk jbot (i=ibot@pdpc/supporter/active/TimRiker/bot/apt) |
| 17:50.13 | *** topic/#asterisk is Asterisk: The Open Source PBX -=- Asterisk 1.4.1 (Mar. 2, 2007), Asterisk 1.2.16 (Mar. 2, 2007), Zaptel 1.2.15 (Mar. 2, 2007) -=- Other fun channels: #asterisk-gui, #asterisknow, #asterisk-commits -=- Join #freepbx for freepbx/trixbox support. -=- |
| 17:51.05 | BSDTech | you have to add the gui its not built in |
| 17:51.09 | BSDTech | its a addon |
| 17:51.22 | russellb | ... |
| 17:51.34 | russellb | whoa re you responding to? |
| 17:51.39 | BSDTech | the asterisk gui |
| 17:51.51 | BSDTech | <ajohnson_laptop> Where can I find some info on what exactly the http interface in Asterisk 1.4 does? |
| 17:51.53 | BSDTech | sorry |
| 17:52.06 | BSDTech | you can look at asterisknow and look at screen shots |
| 17:52.07 | russellb | ah |
| 17:52.17 | ajohnson_laptop | BSDTech : I was actually just curious what it did |
| 17:52.20 | ajohnson_laptop | I've used the GUI |
| 17:53.26 | BSDTech | its a 1/16th functional gui for adding extensions and meemtme rooms and other things . its a basic gui for configuring . its in dev for use on the new asterisk pbx hardware |
| 17:53.38 | step_quasar | as I execute scripts of AGI in AMI ? |
| 17:53.41 | *** join/#asterisk _-zar_- (n=zar@202.182.27.22) |
| 17:53.42 | BSDTech | I use freepbx it has more power |
| 17:53.51 | _-zar_- | .j/join #unbuntu |
| 17:54.05 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 17:54.08 | _-zar_- | lol |
| 17:59.40 | wunderkin | ./join #openpbx |
| 17:59.41 | wunderkin | oops |
| 17:59.42 | wunderkin | :D |
| 18:00.33 | *** part/#asterisk _-zar_- (n=zar@202.182.27.22) |
| 18:03.33 | *** join/#asterisk friedrich| (n=friedric@e177245199.adsl.alicedsl.de) |
| 18:03.57 | step_quasar | <PROTECTED> |
| 18:03.57 | step_quasar | somebody that speaks Spanish? |
| 18:04.45 | *** join/#asterisk rkeels (n=chatzill@99.eedinc.com) |
| 18:05.47 | rkeels | is anyone here who professes to be experts on aplications? I have a couple of questions |
| 18:06.12 | russellb | depends on what you mean by expert |
| 18:06.16 | rkeels | What is the aplication used to log an agent out of the queues |
| 18:06.18 | russellb | expert on how to use them or write them? |
| 18:06.24 | russellb | ah ... |
| 18:06.51 | russellb | either RemoveQueueMember or ... CallbackLogin with some option to say log out, not in |
| 18:06.56 | russellb | i think. |
| 18:07.02 | Corydon-w | AddQueueMember/RemoveQueueMember |
| 18:07.28 | rkeels | I haven't been able to fine any resource that lists all available aplications with *1.4.0 |
| 18:07.37 | russellb | how about ...... show applications |
| 18:07.44 | ajohnson_laptop | dun dun dunnnnn |
| 18:07.44 | russellb | at the asterisk CLI |
| 18:07.48 | FuriousGeorge | can anyone recommend a voip provider w/ POP in nyc? |
| 18:07.52 | Corydon-w | "core show applications" |
| 18:07.59 | russellb | Corydon-w: d'oh |
| 18:08.38 | Corydon-w | russellb: it'll be better, once we wean everybody off 1.2 |
| 18:09.27 | Corydon-w | Then we can proceed with another round of command renames in 1.6... ;-) |
| 18:09.31 | russellb | hehehe |
| 18:09.33 | Strom_C | catsex show dogballs |
| 18:09.53 | russellb | commands are no longer in English ... only hex |
| 18:10.03 | russellb | "core show applications" just becomes FF81B |
| 18:10.07 | Strom_C | deadbeefcafe 101010101 |
| 18:10.35 | step_quasar | as execute agi scripts in cli ? |
| 18:12.40 | *** join/#asterisk bkw_ (i=brian@adsl-70-142-32-235.dsl.tul2ok.sbcglobal.net) |
| 18:13.16 | *** join/#asterisk toombaloomba (n=phph@do.you.like.my.frippers.com) |
| 18:13.44 | JacksLivr | exten => s,n(todmz),Lookupdmz() |
| 18:13.50 | JacksLivr | is not looking up the db |
| 18:14.02 | [TK]D-Fender | russellb: The least you could do is make sure that your hex codes are in in 8-bit increments :) |
| 18:14.14 | russellb | no way |
| 18:14.26 | Qwell[] | russellb: I don't mean to alarm you... |
| 18:14.31 | Qwell[] | russellb: but there is somebody in your office |
| 18:14.34 | JacksLivr | how do i lookup a db that i crceate? |
| 18:14.34 | CrashHD | hah |
| 18:14.39 | russellb | Qwell[]: lol |
| 18:14.42 | Strom_C | and he's EATING YOUR LUNCH |
| 18:14.48 | russellb | Qwell[]: it's my girlfriend, Julie. you can come say hi if you want :) |
| 18:14.50 | Qwell[] | Strom_C: s/h/sh/ :D |
| 18:14.53 | [TK]D-Fender | JacksLivr: "core show function DB" |
| 18:14.58 | Strom_C | hehe |
| 18:15.07 | rkeels | CallbackLogin isn't shown in the show applications command |
| 18:15.13 | Qwell[] | russellb: I will in a bit :) |
| 18:15.22 | rkeels | does that mean that it is depricated in Ast 1.4 |
| 18:15.42 | JacksLivr | asterisk*CLI> core show function DB |
| 18:15.42 | JacksLivr | No such command 'core' (type 'help' for help) |
| 18:15.49 | [TK]D-Fender | rkeels: Yes, check upgrade.txt, and read the changelogs |
| 18:15.59 | rkeels | How do I find out the syntax and options used for the different aplications through the cli |
| 18:16.08 | [TK]D-Fender | JacksLivr: if you're on 1.2 - "show function DB" |
| 18:16.23 | JacksLivr | yeah, i figured that out right before you typed it |
| 18:16.32 | [TK]D-Fender | rkeels: "show show application [applicationname]" |
| 18:16.48 | [TK]D-Fender | rkeels: "core show application [applicationname]" |
| 18:17.05 | russellb | show show? |
| 18:17.32 | Strom_C | that's from the era when Ric Ocasek was contributing code. |
| 18:17.37 | [TK]D-Fender | russellb: yes yes |
| 18:18.23 | russellb | lol .. |
| 18:18.24 | [TK]D-Fender | Yay.... my new 22" LCD has arrived :D |
| 18:18.38 | russellb | ooh, 22 ... i'm jealous |
| 18:18.41 | Strom_C | ooooOOOOOooooo |
| 18:18.50 | [TK]D-Fender | russellb: Very economical these days... |
| 18:18.55 | russellb | but I do have 3 20" widescreen displays on my desk :-D |
| 18:18.56 | Qwell[] | [TK]D-Fender: is it WS? |
| 18:19.08 | [TK]D-Fender | Qwell : Naturally. 1680*1050 |
| 18:19.16 | Qwell[] | pfft |
| 18:19.26 | [TK]D-Fender | Dell E228WFP |
| 18:19.46 | Qwell[] | rev 8? jeebus |
| 18:20.17 | [TK]D-Fender | Qwell : Well a 22" 1680*1050 monitor is about $330 as opposed to a 24" 1920*1200's cost of about $800 |
| 18:20.23 | Qwell[] | (22 inch, rev 8, widescreen, flatpanel - in case that wasn't bloody obvious) |
| 18:20.40 | *** join/#asterisk EmleyMoor (n=phil@topdeck.tinsleyviaduct.com) |
| 18:20.55 | JacksLivr | i do not wanna put or get a value, I want to use it like the lookupblacklist, except with a different family |
| 18:21.16 | file | [TK]D-Fender: I still <3 my Acers |
| 18:21.26 | EmleyMoor | JacksLivr: What is it you want to do? |
| 18:21.50 | *** join/#asterisk nortex (n=breeves@snapper.titanspecialties.com) |
| 18:22.01 | JacksLivr | send to context based on callerid, without have a ton of individual statements in the dialplan |
| 18:22.27 | EmleyMoor | OK - on what basis based on caller ID? |
| 18:22.38 | [TK]D-Fender | file: Yeah, they were a great deal but Dell beat them out on function, and included shipping |
| 18:22.41 | BSDTech | gasterisk = asterisk after a night out at the pub |
| 18:23.03 | EmleyMoor | Just "these caller IDs to one phone, these to another" sort of thing, or like black/white/redlists? |
| 18:23.42 | *** join/#asterisk phillipk (n=pkey@216.248.143.77) |
| 18:24.38 | JacksLivr | EmleyMoor: like when friends call in, i send them to one context, when I dial in from my cell phone, I go to another context, etc. |
| 18:24.59 | EmleyMoor | OK, well, the database is quite suitable for that... |
| 18:25.26 | JacksLivr | i thought so, but im when i lookupfamilyname it fails |
| 18:26.10 | EmleyMoor | You could have GotoIf(${DB_EXISTS(familyname/${CALLERID(num)})} kind of statements |
| 18:26.30 | EmleyMoor | I use those to process my colourlists |
| 18:26.51 | JacksLivr | ok, lemme try that, thanks |
| 18:27.32 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@ppp-70-255-106-4.dsl.hstntx.swbell.net) |
| 18:27.40 | *** join/#asterisk oej (n=olle@apollo.webway.se) |
| 18:27.57 | EmleyMoor | And, if it helps, ${DB_RESULT} contains the value associated with the given key if found |
| 18:28.31 | EmleyMoor | (I simply use 1 for my colourlists, but keep numbers in there for the lastcall related stuff) |
| 18:29.05 | *** join/#asterisk fakhir (i=fakhir@unaffiliated/fakhir) |
| 18:29.20 | JacksLivr | but i wanna not have to do one line for every number |
| 18:29.37 | EmleyMoor | JacksLivr: You won't have to |
| 18:29.51 | EmleyMoor | You do one line for each type of number, and put the numbers in the database |
| 18:30.43 | EmleyMoor | Or you could have a family in there of a given name, with the numbers as keys, and the context names as values... |
| 18:31.06 | *** join/#asterisk pigpen (n=mark@fw.seamans.cc) |
| 18:31.36 | EmleyMoor | You could then do GotoIf(${DB_EXISTS(family/${CALLERID(num)})}?${DB_RESULT},${EXTEN},1) perhaps |
| 18:32.12 | [TK]D-Fender | EmleyMoor: Gettting warmer... |
| 18:32.12 | pigpen | Does anyone know why "polycom-check-cfg" is not present in Asterisk 1.4.1 ? |
| 18:32.29 | pigpen | ie: sip notify polycom-check-cfg <exten> |
| 18:33.01 | pigpen | it was there in 1.4.0 |
| 18:33.45 | EmleyMoor | [TK]D-Fender: Well, that's how I'd do it... T no doubt MTOWTDI |
| 18:34.21 | EmleyMoor | Oh, my other user actually used a softphone instead of his Zap phone last night |
| 18:35.33 | EmleyMoor | Has anyone else noticed that your own voice comes back quite loud with ekiga, or is that just me? |
| 18:36.25 | cpm | pipe down, you're too loud! |
| 18:37.32 | [TK]D-Fender | EmleyMoor: I just had to look at how that gets interpreted... yeah, its clean & dry. Not used to IF's w/o $[] |
| 18:37.57 | [TK]D-Fender | EmleyMoor: Forgot that function returns a perfect response. |
| 18:46.28 | *** part/#asterisk toerkeium (i=oo@201.216.206.221) |
| 18:49.48 | gerphimum | im looking at creating an asterisk box for my small business and i was wondering what options there were in the line of bluetooth headsets - can they operate as their own channel or will they need to pair with some other device to operate |
| 18:53.31 | Strom_C | you're not seriously considering using bluetooth headsets in lieu of telephones, are you? |
| 18:56.48 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 18:59.14 | *** join/#asterisk kFuQ (n=somedude@67.185.123.34) |
| 18:59.32 | [TK]D-Fender | gerphimum: Generally not worth it.... |
| 18:59.48 | [TK]D-Fender | gerphimum: All implementations I've seen were kludgy crap. |
| 19:00.10 | [TK]D-Fender | gerphimum: and still pinned you to your desk. Only difference is that there is no wire directly to your head. |
| 19:03.58 | gerphimum | are you familiar with any desk phones that have bluetooth capability ? |
| 19:04.08 | [TK]D-Fender | gerphimum: Nope... |
| 19:04.25 | gerphimum | odd. that seems like something that would be popular |
| 19:04.34 | [TK]D-Fender | gerphimum: I've worked with wireless headset lifters, etc, and they all really sucked. My guys hated them. |
| 19:04.48 | gerphimum | yeah, i had those at my old office |
| 19:05.59 | gerphimum | they were always breaking - the mechanisms for lifting the headset would lock up, plus the setup on the desk is all around clunky and annoying |
| 19:06.23 | gerphimum | would a bluetooth setup work on a softphone, i wonder.. |
| 19:06.33 | Sweeper | [TK]D-Fender: done much with call queues? I'm having a context problem with them... |
| 19:07.17 | darylvoip | gerphimum: I use one with SJPhone on a MacBook with no problem |
| 19:07.27 | Sweeper | my default context has provisions for voicemail after a set amount of time |
| 19:07.54 | Sweeper | obviously, I don't want this to occur when a queue is calling an agent |
| 19:08.10 | gerphimum | darylvoip >> what are you able to do with the button on the bluetooth headset |
| 19:08.31 | gerphimum | darylvoip >> can you assign it to different things ? |
| 19:08.42 | Sweeper | so I made a new context for the queue, with a goto, and the agent extensions |
| 19:08.43 | darylvoip | never bothered to try. I just redirect audio in/out to it |
| 19:08.52 | gerphimum | k |
| 19:09.05 | Sweeper | but the queue still contacts the agents through the defult context... |
| 19:10.39 | *** join/#asterisk kFuQ (n=somedude@c-67-185-123-34.hsd1.wa.comcast.net) |
| 19:13.39 | *** join/#asterisk kraptv (n=ryan@magic.skylab.org) |
| 19:13.51 | kraptv | I've got a comedian mail question that I can't seem to find the answer to. |
| 19:14.06 | *** join/#asterisk sahafeez (n=sahafeez@67.109.14.227.ptr.us.xo.net) |
| 19:14.11 | EmleyMoor | kraptv: Just ask |
| 19:14.49 | kraptv | I'm running Asterisk 1.2.9.1 and want to know how to change the default greeting for a mailbox by editing a configuration file. |
| 19:15.09 | kraptv | it seems like the interactive greeting from within the phone is not letting her choose the main or temporary greeting. |
| 19:15.50 | Qwell[] | [TK]D-Fender: ! |
| 19:15.53 | Qwell[] | where you at? |
| 19:16.27 | kraptv | voicemail.conf doesn't really indicate menu selection, so, where does it end up getting stored? |
| 19:16.28 | Sweeper | he's dead D: |
| 19:16.38 | wunderkin | the temporary greeting overrides the others |
| 19:16.41 | Qwell[] | probably died sitting there waiting for a polycom to boot ;) |
| 19:16.58 | Sweeper | hahaha, fucking right |
| 19:17.29 | ManxPower | kraptv: you cannot specify the default greetings via a config file. |
| 19:17.33 | bkruse | Qwell[]: a sound station :X |
| 19:17.40 | Sweeper | Qwell[]: any ideas on my problem? :v |
| 19:17.55 | ManxPower | there are many ways you can force it, however. The easiest is to just copy an audio file to whereever the temp greeting is stored |
| 19:21.28 | *** join/#asterisk thekidrio (n=thekidri@66.107.42.13) |
| 19:21.37 | kraptv | that's too bad. many voicemail systems have the ability to select greetings alternately, etc... |
| 19:23.41 | Qwell[] | seriously... seriously? 10 minutes before a polycom even asks for it's config files? |
| 19:24.06 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-88-154-8-247.red.bezeqint.net) |
| 19:24.20 | kraptv | that's unreasonable. are you running on a dedicated vlan or network for your polycoms? |
| 19:24.29 | kraptv | are you running an ip501 or what? |
| 19:24.40 | *** join/#asterisk snowy_owl (i=0@200.218.196.2) |
| 19:24.43 | Qwell[] | 600, and the only thing on this lan is this phone |
| 19:25.01 | snowy_owl | Hi all. I've a question about reinvite |
| 19:25.01 | kraptv | it takes quite a while to upgrade the phones to the latest BootROM and SIP application, but 10 minutes seems way too long. |
| 19:25.02 | bkruse | Qwell[]: EW! |
| 19:25.05 | BSDTech | polycom rocks |
| 19:25.13 | Qwell[] | I want my money back! |
| 19:25.17 | Qwell[] | :D |
| 19:25.17 | kraptv | is it able to upload the log to the tftp server to tell you what it's doing? |
| 19:25.21 | Qwell[] | I think I hosed something |
| 19:25.24 | kraptv | there's the four button salute to force a status upload.... |
| 19:25.26 | Qwell[] | kraptv: It's using http |
| 19:25.41 | kraptv | oh. Hrm. any reason why you wouldn't want to to tftp? |
| 19:25.43 | snowy_owl | why when i set canreinvite=yes the asterisk sends a INVITE when I shut down my phone? |
| 19:25.59 | Qwell[] | because http > tftp |
| 19:26.26 | kraptv | That answer seems puzzling to me. Pardon my ignorance. |
| 19:26.36 | Sweeper | ooookay |
| 19:26.36 | thekidrio | haha |
| 19:26.58 | Sweeper | how to queues decide what context to use to call agents? |
| 19:27.02 | Sweeper | *do |
| 19:27.09 | *** part/#asterisk step_quasar (n=step_qua@OL148-144.fibertel.com.ar) |
| 19:27.43 | snowy_owl | when asterisk should sends just a BYE, it sends a reinvite. I dont know why the reason. The reinvite change the header 'c' to his own IP and, after send it, it sends a BYE |
| 19:27.49 | snowy_owl | is it normal? |
| 19:27.56 | wunderkin | sometimes it took 10 minutes before it asked for the next file for me too... |
| 19:28.37 | Qwell[] | I think my dhcp server is giving out some phone address... |
| 19:28.59 | *** join/#asterisk reber (n=reber@cl-158.dub-01.ie.sixxs.net) |
| 19:30.53 | *** join/#asterisk errr (n=errr@fedora/errr) |
| 19:30.57 | *** join/#asterisk bkruse_home (i=kruz@nat/digium/x-c4d78144a9d784ab) |
| 19:31.22 | snowy_owl | the version 1.2.* sends the reinvite after the OK's invite. But in 1.4.0 this reinvite is sent before the BYE, after the conversation. |
| 19:31.49 | snowy_owl | Am I being so stupid? |
| 19:32.48 | Qwell[] | man, where is this phone getting 192.168.254.1 as a server IP from? |
| 19:33.01 | wunderkin | 192.168.2.290? |
| 19:33.10 | kraptv | do any of you know of the replacement for comedian mail that was supposed to be for asterisk 1.4?? |
| 19:33.25 | wunderkin | krapper, i haven't heard of such a thing |
| 19:33.27 | wunderkin | heh |
| 19:34.32 | bkruse | Qwell[]: lame..... |
| 19:34.38 | bkruse | Qwell[]: what network dude? |
| 19:34.43 | Qwell[] | bkruse: c'mere :D |
| 19:34.46 | bkruse | kk |
| 19:35.23 | [TK]D-Fender | Qwell[]: y0... |
| 19:35.35 | [TK]D-Fender | Qwell[]: : Any my Polycom's DON'T reboot. :) |
| 19:35.49 | *** join/#asterisk Arth_72 (n=Arturo@88-149-232-233.f5.ngi.it) |
| 19:35.51 | Arth_72 | hi |
| 19:35.57 | *** join/#asterisk Defraz (n=t0tal@fw.centrisys.com) |
| 19:36.14 | kraptv | yeah, there was the IMAP comedian mail storage enhancement, but other than that, I heard nothing... |
| 19:36.40 | Arth_72 | can someone help me with asterisk and brifstuff compiling ? |
| 19:37.50 | *** join/#asterisk cekc (n=cekc@66-17-9-220.biz.bkfd.arrival.net) |
| 19:38.10 | cekc | what's that thing called that lets you hop around on your T1 lines listening to what is going on |
| 19:38.35 | ManxPower | cekc: a T-BERD. They are $4,000 - $20,000 |
| 19:38.50 | cekc | ok what's the asterisk software version of it called then |
| 19:39.27 | cekc | i remember one time on an asterisk system dialing a random extension and i could put in the trunk numbers and it'd tune into the call audio |
| 19:39.40 | cekc | maybe it was trixbox |
| 19:40.36 | *** join/#asterisk J4k3- (i=J4k3@dhcp-12-197-128-58.intrastar.net) |
| 19:40.53 | mercestes | zap barge I think. |
| 19:41.09 | cekc | yeahhhhh |
| 19:41.37 | kraptv | thanks for the help, everyone! I'll try to stop by later, perhaps offering some polycom help! |
| 19:41.39 | *** part/#asterisk kraptv (n=ryan@magic.skylab.org) |
| 19:41.41 | cekc | that one works just from the console i think |
| 19:42.01 | *** join/#asterisk oej (n=olle@apollo.webway.se) |
| 19:42.02 | cekc | no wait i guess it's a dialable program |
| 19:42.21 | *** join/#asterisk alexcf (i=alexcf@lexi.me.uk) |
| 19:42.39 | mercestes | It's an asterisk application callable from extensions.conf |
| 19:42.59 | alexcf | hi... how can i block anonymous calls coming into an extension? i've tried anonymous/1234,Hangup() |
| 19:43.07 | mercestes | show application ZapBarge |
| 19:44.32 | cekc | chanspy is another good one |
| 19:48.39 | *** join/#asterisk rvhi2 (n=as@66.175.65.82) |
| 19:49.40 | rvhi2 | hi |
| 19:49.58 | mercestes | then why did you ask? :P |
| 19:50.24 | rvhi2 | i did 'apt-get upgrade' on debian sarge, it replaced the kernel |
| 19:50.30 | rvhi2 | i recompile zaptel |
| 19:50.39 | rvhi2 | when do "modprobe wct4xxp" |
| 19:50.41 | rvhi2 | got error |
| 19:50.49 | rvhi2 | Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters. |
| 19:51.07 | mercestes | chanspy is for SIP. |
| 19:51.39 | rvhi2 | -- /lib/modules/2.4.27-3-686-smp/misc/wct4xxp.o: init_module: No such device |
| 19:52.08 | *** join/#asterisk bitbandit (n=polx@74.211.30.160) |
| 19:52.10 | rvhi2 | any suggestion? |
| 19:54.36 | ManxPower | rvhi2: do a "make clean" before the "make install" |
| 19:54.51 | rvhi2 | i did that |
| 19:54.52 | *** join/#asterisk toot (n=toot@84.19.255.123) |
| 19:55.18 | JunK-Y | chamspy is for any driver. |
| 19:55.30 | mercestes | JunK-Y: True. |
| 19:55.32 | ManxPower | rvhi2: looks like the system does not see the card. Is it listed in "lspci" |
| 19:58.03 | Sweeper | well, if anyone was WONDERING, the way to specify what context the queue calls agents from is done on login |
| 19:58.17 | rvhi2 | lspci saw it, same as another production server |
| 19:58.53 | *** join/#asterisk florz (n=florz@2001:1a50:503c:0:0:0:0:1) |
| 20:02.44 | *** join/#asterisk tmccrary (n=tmccrary@68-77-164-10.ded.ameritech.net) |
| 20:03.15 | Arth_72 | can anyone help me to compile bristuff and asterisk 1.4.1 plz ? i'm getting an error on zaptel compiling .... |
| 20:03.26 | tmccrary | Is there a way with Dial cmd (or anything) to force one way only audio on a call? I have a paging system that can do voice and page, but when a certain number calls, I only want to do paging (and not let them talk through it) |
| 20:05.09 | mercestes | tmccrary: If they are calling into the PBx, audio is outbound only until you call an "answer()". if they are dialing out, show command "page" should give you what you need for outbound audio only. |
| 20:05.30 | mercestes | tmccrary: If you want inbound audio only, I suggest screwing up your NAT settings or improperly configure a firewall. |
| 20:05.39 | ManxPower | Arth_72: do you have the version of BRIstuff specific for 1.4? |
| 20:06.30 | Arth_72 | don't know exactly .. i dl the pre-y |
| 20:06.49 | Arth_72 | sorry bristuff-0.3.0-pre-1y |
| 20:07.06 | ManxPower | Arth_72: you should confirm. |
| 20:07.18 | *** join/#asterisk oej (n=olle@apollo.webway.se) |
| 20:07.35 | *** join/#asterisk Telamon (i=telamon@blk-137-96-217.eastlink.ca) |
| 20:07.41 | Arth_72 | i have that version .. is it ok for 1.4.1 ? or where do i find some doc to know it ? |
| 20:07.43 | *** part/#asterisk Telamon (i=telamon@blk-137-96-217.eastlink.ca) |
| 20:08.20 | ManxPower | Arth_72: I don't know if that version if for 1.4 or not. You should find out from the BRIstuff web site |
| 20:08.37 | Arth_72 | ok, i search again there then ... |
| 20:09.43 | snowy_owl | I just want a word: this reinvite (just a milisecons before the bye message) is normal? Yes or no? |
| 20:09.54 | Arth_72 | uhm think it's not for 1.4.1 ... and there is no dl there for the 1.4.1 |
| 20:09.56 | *** join/#asterisk deb_user (n=none@70-59-111-238.albq.qwest.net) |
| 20:09.58 | Arth_72 | sorry if i asked :P |
| 20:10.09 | deb_user | [TK]D-Fender: are you there? |
| 20:10.15 | [TK]D-Fender | deb_user: yup |
| 20:11.22 | deb_user | [TK]D-Fender: i did as you suggested and disabled the usb controller in the bios |
| 20:11.36 | deb_user | cat /proc/interrupts output is here: http://pastebin.ca/384053 |
| 20:11.51 | deb_user | but...still having feedback and static on the line |
| 20:11.53 | [TK]D-Fender | deb_user: Much improved. |
| 20:12.14 | tmccrary | yeah, Page doesn't really work well |
| 20:12.17 | deb_user | it seems to be only one fxs/fxo pair that's having the problem |
| 20:12.23 | [TK]D-Fender | deb_user: Keeping in mind you also have 2 of those cards chewing up interrupts on your BUS |
| 20:12.58 | [TK]D-Fender | deb_user: Try exchanging the suspected module and see if the problem follows the port. Could jsut be a dead module |
| 20:13.26 | *** join/#asterisk Stridernzl (n=neville@222-152-248-128.jetstream.xtra.co.nz) |
| 20:13.34 | deb_user | [TK]D-Fender: you mean switching it over to a different POTS line? |
| 20:13.42 | Arth_72 | just a stupid question (i'm new on asterisk ... ) do u know if there will be a bristuff version for 1.4.1 shortly ? |
| 20:14.14 | [TK]D-Fender | deb_user: Well yeah, try switching the line first. Then change the module on the card for another and see if the problem follows. that will help pin down if its the line or the module. |
| 20:14.15 | ManxPower | Arth_72: try asking on the BRIstuff mailing list. |
| 20:14.22 | *** join/#asterisk greegoro (n=greegoro@eme90.internetdsl.tpnet.pl) |
| 20:14.24 | Arth_72 | ok, ty ... |
| 20:14.39 | Arth_72 | maybe better if i get the 1.2.15 meanwhile ... |
| 20:14.52 | deb_user | [TK]D-Fender: ok, well I don't have any spare modules lying around, but I'll switch the POTS connections and see what happens |
| 20:15.01 | Sweeper | Arth_72: better off on 1.2 anyways :v |
| 20:15.08 | Arth_72 | y ? |
| 20:15.13 | deb_user | i'll see if it follows the module or the POTS connection... |
| 20:15.18 | deb_user | that should help isolate the prob |
| 20:15.20 | deb_user | thanks |
| 20:15.24 | [TK]D-Fender | deb_user: you have multiple FXO, no? |
| 20:15.31 | deb_user | fender: 3, yeah |
| 20:15.38 | ManxPower | [TK]D-Fender: looks like BLF is still not fixed in 1.4.1 |
| 20:15.39 | [TK]D-Fender | deb_user: swap amongst those 3! |
| 20:15.40 | deb_user | but all of them are in production |
| 20:15.49 | Sweeper | Arth_72: it's production stuff, 1.4 is still devel |
| 20:15.49 | greegoro | Hi, does anybody can help me. I'm starting small VOIP business in my city but I don't know what to you to terminate GSM calls ? |
| 20:16.06 | [TK]D-Fender | deb_user: If its bad on one, it'll be just as bad on the other with no change in impact, but that will confirm what we should BLAME. |
| 20:16.08 | deb_user | [TK]D-Fender: do you mean actually pull the module in the tdm card, and switch it? |
| 20:16.09 | [TK]D-Fender | deb_user: DO IT. |
| 20:16.15 | [TK]D-Fender | deb_user: Yeah. |
| 20:16.20 | greegoro | use VOIP provider ? or GSM |
| 20:16.27 | [TK]D-Fender | deb_user: Swap 2 red modules and see if the problem follows it |
| 20:16.31 | deb_user | groan |
| 20:16.31 | Sweeper | greegoro: voip provider |
| 20:16.42 | Sweeper | gsm in this context is a voice codec |
| 20:16.43 | Arth_72 | ok, ty and sorry for the newbie questions :) |
| 20:16.56 | deb_user | pulling my tdm cards is like running an obstacle course |
| 20:17.02 | tuan_modulis | is there a way to detect if a device is "in use" from the dialplan? |
| 20:17.02 | greegoro | no no :) I'm talking about celluar phones |
| 20:17.17 | Sweeper | greegoro: eh, answer is the same... |
| 20:17.22 | deb_user | but ok...I guess I don't have a choice |
| 20:17.29 | *** join/#asterisk Mavvie (n=edwin@ppp21-177.lns2.syd7.internode.on.net) |
| 20:17.33 | *** join/#asterisk backblue (n=moo@87-196-44-132.net.novis.pt) |
| 20:17.36 | ManxPower | I think the answer is |
| 20:17.57 | ManxPower | I think the answer is "If you have to ask these questions then you are not qualified to start your own VoIP company" |
| 20:18.20 | mutilator | heh |
| 20:18.20 | ManxPower | "and maybe you should be doing more reading and less talking" |
| 20:18.25 | EmleyMoor | tuan_modulis: What kind of device? |
| 20:18.30 | mutilator | pwnt |
| 20:18.50 | tuan_modulis | EmleyMoor: a sip device for instance |
| 20:19.15 | greegoro | OK thank's. I'm looking for good VOIP biling and admin/user management panel |
| 20:19.23 | *** join/#asterisk darylvoip (n=darylvoi@c-71-225-231-66.hsd1.pa.comcast.net) |
| 20:19.37 | [TK]D-Fender | tuan_modulis: "show application chanisavail". if 1.4 add "core " in front |
| 20:19.44 | EmleyMoor | tuan_modulis: You could always check if its ChanIsAvail |
| 20:20.03 | tuan_modulis | thanks |
| 20:20.36 | darylvoip | Does anyone know if the dial options g, C, or S(x) will prevent re-invites. I'm thinking the S might be a problem. |
| 20:22.20 | deb_user | [TK]D-Fender: why not just swap the RJ11 cords coming from the POTS line to the TDM, then change the dialplan to reflect the swap? |
| 20:22.51 | [TK]D-Fender | deb_user: Thats 1 test, depends if the port # is important to you. |
| 20:23.13 | [TK]D-Fender | deb_user: Swap lines first, but it can also be the card on THAT port, and not jsut the module on it. |
| 20:23.28 | [TK]D-Fender | deb_user: There are 3 things in play giving you a max of 8 tests |
| 20:24.41 | *** join/#asterisk IMG-SD (n=IMG-SD@tserv6.imperialgroup.ca) |
| 20:25.30 | *** join/#asterisk Exhar (i=Ask@84-105-192-215.cable.quicknet.nl) |
| 20:26.44 | ajohnson_laptop | rtcp |
| 20:26.55 | ajohnson_laptop | wrong window :-) |
| 20:27.00 | *** join/#asterisk alephant (n=cmd@onion.alephant.net) |
| 20:27.03 | alephant | Hi all! |
| 20:27.07 | yonahw | does anyone know of an alternative to hudlite with similar functionality? |
| 20:27.21 | ajohnson_laptop | and not FOP? |
| 20:27.27 | ajohnson_laptop | (flash operator panel) |
| 20:27.30 | alephant | I have a branch office that I want to connect to the main office by putting an * server at the branch |
| 20:27.35 | rkeels | Does anyone have a recomendation of best practice for replacing the Perl() app |
| 20:27.46 | yonahw | i find FOP to be a little too for an operator |
| 20:27.47 | alephant | Can I pastebin my proposed config additions for somebody to validate/criticize? |
| 20:27.57 | yonahw | i am looking for something smaller that just shows the users i wish to see |
| 20:28.07 | ajohnson_laptop | Other than FOP and Hudlite, I haven't found anything else that really works |
| 20:28.12 | yonahw | ty |
| 20:29.40 | alephant | http://pastebin.ca/384091 |
| 20:29.50 | *** join/#asterisk brea (n=rollergr@c-71-195-248-169.hsd1.ma.comcast.net) |
| 20:30.04 | alephant | In addition to the above pastebin'd config, where would I configure the extensions? |
| 20:30.16 | alephant | the ~2xx extensions at one side and the ~3xx at the other? |
| 20:30.43 | *** join/#asterisk nextime (n=nextime@unaffiliated/nextime) |
| 20:31.46 | nextime | which preemption model is better for asterisk? i usually don't use preempt enabled kernel for servers, but maybe for * it can be better to do preemption? |
| 20:31.55 | *** join/#asterisk thinwires (n=thinwire@24-49-196-96.kntnny.adelphia.net) |
| 20:32.31 | alephant | Anybody? Help with an IAX2 peering config as pastebinned above? |
| 20:32.39 | *** join/#asterisk punkgode (n=Punkgode@rev-200-40-119-222.netgate.com.uy) |
| 20:32.46 | *** part/#asterisk ManxPower (n=manxpowe@71-8-56-64.dhcp.leds.al.charter.com) |
| 20:33.11 | lude | hmm |
| 20:33.16 | lude | this phone refuses to register |
| 20:33.19 | alephant | nextime: I don't pretend to authority on the matter, but it would seem to me that the timer frequency would be a better tweakable... |
| 20:33.21 | lude | username and password match |
| 20:33.33 | lude | but asterisk keeps returning 401 unauthorized |
| 20:33.33 | punkgode | hello, can anyone tell me if there is a way to handle cmds return codes in the dialplan? |
| 20:34.03 | punkgode | I'm using AEL (version 1) |
| 20:34.18 | *** join/#asterisk Ebola (n=Ebola@host86-143-156-147.range86-143.btcentralplus.com) |
| 20:34.19 | *** part/#asterisk backblue (n=moo@87-196-44-132.net.novis.pt) |
| 20:34.59 | nextime | alephant : mumble, i think that using a "non default" timer frequency can do some weird things if you use zaptel devices that need excatly 1000 interrupts for second... no? |
| 20:35.05 | *** join/#asterisk Dane1 (n=DaneM@adsl-75-25-36-35.dsl.chi2ca.sbcglobal.net) |
| 20:35.08 | alephant | nextime: ooh |
| 20:35.17 | alephant | nextime: yeah... good call |
| 20:35.56 | cekc | omg hylafax is better than sex why didn't i try this before |
| 20:36.00 | alephant | "excatly" --> an oldskool substitute for sed? |
| 20:36.08 | alephant | sorry |
| 20:36.22 | alephant | cekc: what exactly are you doing with hylafax!? |
| 20:36.34 | Dane1 | Hello. I'm having a bit of a strange problem with Asterisk and outgoing calls (from in from zap/1 office phone, out from zap/3 phone line). It hangs at "Attempting native bridge of Zap/1-1 and Zap/3-1" and never seems to dial. Any suggestions? |
| 20:36.34 | nextime | alephant : maybe only too many thc in my blood ( about "exactly" _) |
| 20:36.36 | alephant | "bend over and squeal like a fax machine" |
| 20:36.42 | cekc | i don't know yet, but i have an iaxmodem extension setup and sent it a fax and it emailed it to me |
| 20:37.12 | cekc | alephant: but soon I'll have it almost completely replacing winfax |
| 20:37.29 | alephant | cekc: Hm. I'm still liking sex better. |
| 20:38.00 | nextime | so, anyone have idea about preemption or not preemption? |
| 20:38.14 | cekc | you see, winfax is the girl I would take out to dinner and buy nice things for, but hylafax is the dirty slut who gives me exactly what I wanted |
| 20:38.18 | tmccrary | So, there is no way to force one way audio with Asterisk currently? |
| 20:38.20 | alephant | So... any takers on a quick eyeball of my pastebinned attempt at making two * boxen behave like one logical PBX via IAX2? |
| 20:38.38 | nextime | ( i'm working on a "default kernel for generic asterisk servers" ) |
| 20:39.56 | darylvoip | Does "-- Packet2Packet bridging SIP/093691855977-b7baddd8 and SIP/teleglobe-09e44d98" mean that I'm doing a native bridge on that call and shouldn't be routing media? |
| 20:40.19 | darylvoip | ....becuse I'm still showing RTP traffic, and it sure seems like I shouldn't be. |
| 20:41.01 | alephant | cekc: ooooohhh... you're using an "analogy". |
| 20:41.17 | lude | argh wtf |
| 20:41.24 | alephant | cekc: I thought you'd hooked up your serial port to weird places and... nevermind. |
| 20:41.24 | lude | it just rejects this phone |
| 20:41.25 | lude | for now reason |
| 20:41.30 | nextime | reading on asteriskguru seem to be preemption for low-latency desktop the right way. |
| 20:41.38 | cekc | you see using asterisk is a lot like the vietnam war... |
| 20:42.08 | alephant | Who are you, Walter SIPczyck? |
| 20:42.33 | cekc | hehehe |
| 20:42.48 | lude | http://pastebin.ca/384105 |
| 20:42.55 | lude | anyone see anything out of the ordinary there? |
| 20:43.07 | Dane1 | anybody have an idea on the outgoing call problem? |
| 20:43.08 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
| 20:43.27 | *** part/#asterisk BSDTech (n=RNeese@ppp-69-239-113-101.dsl.irvnca.pacbell.net) |
| 20:43.55 | *** join/#asterisk progmonkey (n=asdfkle@c-71-195-248-169.hsd1.ma.comcast.net) |
| 20:44.19 | [TK]D-Fender | Dane1: It'd be nice if you pastebin'd the entire CLI output of the failed call attempt. That message alone tells us nothing. |
| 20:44.31 | progmonkey | Is there a way to use multiple Read() in a row and still read dtmf correctly? |
| 20:44.48 | Dane1 | OK. Where should I paste it? (I'm pretty new on IRC) |
| 20:45.11 | bkruse | progmonkey: you can do get data from agi and keep it down to the limit of digits |
| 20:45.25 | [TK]D-Fender | ~pb |
| 20:45.29 | jbot | well, pb is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 20:45.45 | Dane1 | thanks |
| 20:46.17 | progmonkey | bkruse: I would like to keep it in asterisk afraid too much agi might bog down the system |
| 20:46.38 | bkruse | progmonkey: then you should be able to do a read for dtmf and specific max digits i thought? |
| 20:48.05 | progmonkey | bkruse: I have recordings that are broken down into multiple recordings |
| 20:48.24 | progmonkey | numbers for example |
| 20:48.35 | Dane1 | http://pastebin.ca/384111 |
| 20:48.52 | progmonkey | basically I would like to do a read for 3 recordings at once instead of just 1 |
| 20:49.18 | [TK]D-Fender | Dane1: Your dial is no good |
| 20:49.25 | [TK]D-Fender | -- Executing Dial("Zap/1-1", "Zap/3/#{EXTEN:1}") |
| 20:49.37 | [TK]D-Fender | that should be "$" not "#" |
| 20:50.01 | [TK]D-Fender | Dane1: Which is why it didn't parse out to be the number dialed less the first digit as expected |
| 20:50.01 | Dane1 | aah. Thanks! I'll give that a try. |
| 20:50.16 | bkruse | progmonkey: that doesnt make much sense to me, explain your situation plz :] |
| 20:50.19 | *** part/#asterisk deb_user (n=none@70-59-111-238.albq.qwest.net) |
| 20:50.25 | Dane1 | nodnod |
| 20:50.25 | bkruse | [TK]D-Fender: some people just cant use variables correctly :P |
| 20:50.52 | [TK]D-Fender | bkruse: Some people have a way with words.... other people... euhhh err.... thingy ;) |
| 20:51.03 | bkruse | hahaha, good point :] |
| 20:51.05 | progmonkey | bkruse: Read("you pressed") Read("${varnum}) Read("thankyou") |
| 20:51.34 | *** join/#asterisk sob0l (n=sobol@213.218.117.125) |
| 20:51.39 | *** part/#asterisk nextime (n=nextime@unaffiliated/nextime) |
| 20:51.51 | bkruse | progmonkey: why would you put read("you pressed")? |
| 20:51.55 | JunK-Y | [TK]D-Fender: like me with ur best friend tracy. |
| 20:51.56 | progmonkey | that is a poor example, but basically if you use dtmf on the first recordign it wont play the others |
| 20:52.00 | JunK-Y | ;) |
| 20:52.01 | bkruse | thats reading whatever they enter into the variable you pressed |
| 20:52.11 | bkruse | progmonkey: that makes NO sense |
| 20:52.14 | progmonkey | haha |
| 20:52.18 | progmonkey | k, let me try again |
| 20:52.26 | [TK]D-Fender | JunK-Y: Lol, far from. She is completely neurotic. I just know OF her :) |
| 20:52.50 | JunK-Y | i know you wanted her for all ur life! |
| 20:52.51 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 20:53.01 | JunK-Y | i will call her 2nite to tell her the truth ;) |
| 20:53.28 | bkruse | progmonkey: your going to want exten = background or playback to play back audio |
| 20:53.29 | [TK]D-Fender | JunK-Y: You think *I* want here... and YOU'VE got her number! |
| 20:53.34 | [TK]D-Fender | JunK-Y: *BUSTED* ! |
| 20:53.36 | [TK]D-Fender | :D |
| 20:53.54 | [TK]D-Fender | JunK-Y: Bebe don't hurt me! ;) |
| 20:53.58 | JunK-Y | true true. |
| 20:54.01 | JunK-Y | mouhaha aha |
| 20:54.07 | *** join/#asterisk darken_darken (n=marco@26.188.77.83.cust.bluewin.ch) |
| 20:54.24 | JunK-Y | can we change the topic, since i dont want to be mad. |
| 20:54.25 | JunK-Y | heheh |
| 20:54.36 | bkruse | you only use read to read in dtmf values into a variable |
| 20:54.39 | [TK]D-Fender | JunK-Y: Que ferais une gang de trois comme vous? ;) |
| 20:54.44 | progmonkey | bkruse: Read("incoming_call_from.wav") Read("Bob.wav") Read("press_one_to_accept_or_two_to_hangup.wav") |
| 20:54.45 | bkruse | read is NOT for playing back audio progmonkey |
| 20:54.52 | JunK-Y | [TK]D-Fender: en francais? |
| 20:54.53 | JunK-Y | :) |
| 20:54.56 | bkruse | progmonkey: i see what you mean, change read to playback |
| 20:54.57 | progmonkey | does that make more sense? |
| 20:54.59 | bkruse | yep |
| 20:55.13 | progmonkey | what if they press 1 at the beginning of first playback? |
| 20:55.32 | [TK]D-Fender | progmonkey: You don't use "read" for playback! |
| 20:55.48 | alephant | If you're a public speaker? |
| 20:55.56 | *** join/#asterisk test34 (n=test34@unaffiliated/test34) |
| 20:56.03 | alephant | Orate("incoming_call-from.wav")... |
| 20:56.06 | [TK]D-Fender | progmonkey: Sounds like you want a basic IVR background-ing a bunch of seperate recordings. |
| 20:56.29 | bkruse | progmonkey: then ok |
| 20:56.30 | [TK]D-Fender | progmonkey: And that'd be "Background", not "Read" |
| 20:56.39 | Dane1 | OK...I've made the correction. Now it gives me a "ringing" tone, but it doesn't actually ring my cell phone (the number I dialed). I have already entered the "9" before dialing. |
| 20:56.41 | Arth_72 | just a question ... the patches in the bristuff packages apply to zaptel 1.2.10 .. do i have to apply them anyway to zaptel-1.2.15 ? |
| 20:56.44 | progmonkey | [TK]D-Fender: and I can still have dtmf for all of it? |
| 20:56.45 | bkruse | progmonkey: do this, use background, read is to read in dtmf values into a variable |
| 20:56.49 | bkruse | progmonkey: yes |
| 20:57.08 | [TK]D-Fender | alephant: Or perhaps "Spew", or "Ramble", perhaps "one of the trio of "Nag", "Bitch" or "Whine"? |
| 20:57.10 | progmonkey | bkruse: ok, I will check out background |
| 20:57.14 | bkruse | kk ;] |
| 20:57.18 | progmonkey | thx |
| 20:57.23 | bkruse | np |
| 20:57.45 | alephant | Well, Perl has carp()... |
| 20:58.24 | alephant | I will note that one of the primary advantages of Perl is its being written by a linguist makes it dang funny... |
| 20:58.57 | alephant | Gotta run |
| 20:59.08 | alephant | Thanks for not helping with what I came in here for :-D |
| 20:59.16 | bkruse | +b alephant |
| 20:59.20 | bkruse | :] |
| 20:59.20 | *** join/#asterisk dasenjo (i=dasenjo@gateway/tor/x-720214e32b819c61) |
| 20:59.26 | alephant | ? +b? |
| 20:59.30 | bkruse | +b = ban |
| 20:59.35 | alephant | lol |
| 20:59.38 | [TK]D-Fender | alephant: I'm a cunning linguist myself... quite popular with the ladies :) |
| 21:00.21 | alephant | Nah, you're just a cunny linguist. |
| 21:00.29 | alephant | ANYway |
| 21:00.56 | [TK]D-Fender | Darn.. he changed his exit msg :) |
| 21:01.22 | dasenjo | Hi! I'm obtaining the following warning: Unable to handle indication 3 for 'SIP/XXX' |
| 21:01.44 | dasenjo | and when this occurs I have no ring for the calling pary |
| 21:01.54 | *** join/#asterisk _Vile (n=vile@bc182112.bendcable.com) |
| 21:01.54 | progmonkey | bkruse: Have you used Background with AEL? |
| 21:02.27 | bkruse | actually no |
| 21:02.29 | [TK]D-Fender | progmonkey: There is absolutiely nothing special with AEL, ok? It just gets parsed back to standard logic in the core. |
| 21:02.31 | bkruse | ael is very useful, howver |
| 21:02.36 | bkruse | yep |
| 21:02.53 | [TK]D-Fender | AEL stole time that could have been put to better use elsewhere in the core.... |
| 21:03.01 | progmonkey | well I love AEL |
| 21:03.01 | codefreeze | progmonkey: I have. Having a problem? |
| 21:03.20 | progmonkey | anyway, my extensions are not all named 1 or 2 |
| 21:03.29 | progmonkey | so background wont jump where I want |
| 21:03.44 | dasenjo | I'm passing the r option to the Dial command ... if I do not pass it, the warning disappears, but I need it in some circustamces |
| 21:04.28 | progmonkey | bkruse: I will just make an app for it, thx anyways |
| 21:04.44 | [TK]D-Fender | progmonkey: You should be able to use Background just fine in AEL... |
| 21:04.59 | [TK]D-Fender | progmonkey: like I said, it all gets parsed back to the base anyways |
| 21:05.12 | bkruse | progmonkey: haha, np :] |
| 21:05.54 | progmonkey | [TK]D-Fender: I may be wrong but Background reads a dtmf and then transfers you to an extension of that dtmf number. Since my extensions are not numbers how would it work for me? |
| 21:06.10 | [TK]D-Fender | No that is NOT what background does |
| 21:06.22 | bkruse | [TK]D-Fender: not at all, ahaha |
| 21:06.41 | [TK]D-Fender | go read the instructions on it and read up on * standard extensions, and basic IVR implementation. |
| 21:07.09 | progmonkey | This application will play the given list of files while waiting for an extension to be dialed by the calling channel. |
| 21:07.27 | progmonkey | That's exactly what I am reading |
| 21:07.37 | [TK]D-Fender | What you require can and should be done all nice & neat in your dialpla, without multiple reads or any of that. |
| 21:07.40 | *** join/#asterisk supjigatr (n=syslod@152.53.16.10) |
| 21:07.49 | progmonkey | ugh |
| 21:07.50 | progmonkey | hehe |
| 21:07.52 | progmonkey | thanks anyway |
| 21:07.54 | [TK]D-Fender | progmonkey: Reading from where? |
| 21:08.06 | progmonkey | http://www.voip-info.org/wiki-Asterisk+cmd+Background |
| 21:08.07 | supjigatr | Anyone see polycoms just randomly reboot when used with asterisk? |
| 21:08.30 | Qwell[] | supjigatr: yeah, there was some weird firmware bug or something, I think |
| 21:08.39 | Qwell[] | I'd try upgrading |
| 21:08.41 | supjigatr | Any idea how to fix it? |
| 21:08.43 | *** join/#asterisk rgsteele (n=chatzill@nat-pool.agora-net.com) |
| 21:09.18 | [TK]D-Fender | progmonkey: Actually, so far that isn't really off. Whats the issue? You want to play a buch of recordings, and be able to interrupt them with DTMF to go do something else... |
| 21:09.32 | supjigatr | I think I have the lastest version but not upgraded past the bootrom that theres no way back from. |
| 21:09.35 | [TK]D-Fender | Qwell[]: You were looking for me earlier? |
| 21:09.46 | Qwell[] | [TK]D-Fender: no, I was just gonna blame you |
| 21:10.14 | [TK]D-Fender | Qwell[]: What for this time? All this cataloging is taking its toll.... |
| 21:10.24 | rgsteele | Hey guys. I'm trying to figure out why my Asterisk PBX keeps dropping calls. Is there any way to increase the log level so that I can glean more from the logs after a call drop? |
| 21:10.25 | *** join/#asterisk anthonyl (n=anthonyl@r74-193-79-81.pfvlcmta01.grtntx.tl.dh.suddenlink.net) |
| 21:10.30 | Qwell[] | dunno, it was being stupid with ignoring the server IP from dhcp |
| 21:10.39 | Qwell[] | turns out I didn't have option-66 support enabled |
| 21:10.54 | rgsteele | I need something outside of firing asterisk up in debugging mode - something that just writes gobs of info during normal operation |
| 21:10.59 | [TK]D-Fender | Qwell[]: Yeah... WTF... I should have done that for you! I am SOOO bad! |
| 21:11.05 | Qwell[] | [TK]D-Fender: exactly |
| 21:11.13 | Qwell[] | bkruse agrees, I'm sure |
| 21:11.17 | Juggie | rgsteele, its called full logging, check logger.conf |
| 21:11.19 | bkruse | :D |
| 21:11.26 | Qwell[] | [TK]D-Fender: here's the weird part though...maybe you can shed some light on it |
| 21:11.27 | rgsteele | Juggie: Thanks, will do! |
| 21:11.30 | *** join/#asterisk sbingner (n=sam@pdpc/supporter/sustaining/sbingner) |
| 21:11.35 | bkruse | Qwell[]: now ya know for future reference, and i def think that is some cool stuff your doing :D |
| 21:11.38 | Qwell[] | if I plugged it into a different LAN, it would get serverip just fine |
| 21:11.42 | codefreeze | progmonkey: It plays the sound(s) until dtmf is detected, then stops; it basically plays an interruptible sound, just as [TK]D-Fender said. You might use this near the end of the s exten. Then, any dtmf will stop the playback, and be used to select which exten to go to. |
| 21:12.02 | Qwell[] | [TK]D-Fender: Is it supposed to get "Server IP" from dhcp? |
| 21:12.05 | bkruse | codefreeze: yep |
| 21:12.12 | bkruse | [TK]D-Fender: yes right? |
| 21:12.19 | Juggie | rgsteele, becareful though, it can get quite big. |
| 21:12.30 | bkruse | just like normal dhcp, it hands you the gateway, or in udhcpd(i thought) the "router" |
| 21:12.47 | [TK]D-Fender | Qwell[]: it can. You can tell it in provisioning to use it or not. opt 66 is for the boot server but for that to work, you should have 0.0.0.0 in your BootROM. |
| 21:13.17 | rgsteele | Juggie: I will. I don't see much in logger.conf... maybe voip-info.org will have better documentation. |
| 21:13.30 | Qwell[] | well, in the setup thingie, it had server address set to 192.168.254.1, and when I got dhcp from this device, it kept it at that - but when I plugged it into our lan, it got the right address |
| 21:13.35 | Qwell[] | *before* provisioning |
| 21:13.48 | bkruse | yep |
| 21:13.51 | *** join/#asterisk heh_v_water (n=heh_v_wa@71-210-51-58.hlna.qwest.net) |
| 21:13.55 | bkruse | pre-tftp loading, just at the dhcp end |
| 21:14.08 | rgsteele | Juggie: Looks like "verbose" |
| 21:14.42 | wunderkin | is there a digiumite developer i can speak with? i don't want to air my dirty laundry about digium publically |
| 21:14.51 | Qwell[] | wunderkin: sure, msg me |
| 21:15.00 | Juggie | rgsteele, did you enable full => notice,warning,error,debug,verbose |
| 21:15.44 | rgsteele | No. Right now I've just got: |
| 21:15.50 | rgsteele | console => notice,warning,error |
| 21:15.52 | rgsteele | messages => notice,warning,error |
| 21:15.57 | rgsteele | That's it |
| 21:15.59 | Juggie | didnt you ask me to log to a file? |
| 21:16.28 | rgsteele | I was going to append verbose to the list of log levels associated with messags. |
| 21:16.42 | *** join/#asterisk Assid (n=assid@202.88.132.238) |
| 21:16.42 | rkeels | has anyone had trouble with SIP NOTIFY with Ast 1.4 |
| 21:16.54 | Assid | heya |
| 21:17.03 | bkruse | Assid: whats up! |
| 21:17.11 | rkeels | I am using Polycom 430's |
| 21:17.21 | Assid | bkruse: getting my ass kicked.. |
| 21:17.29 | Assid | office is pissed off cause voicepulse having major issues.. |
| 21:17.30 | bkruse | Assid: ;[ |
| 21:17.33 | bkruse | lame. |
| 21:17.40 | bkruse | all their customers? |
| 21:17.41 | Assid | and past 2 weeks is bad.. and they need to atleast get calls out |
| 21:17.42 | bkruse | or just you guys!? |
| 21:17.56 | bkruse | i agree |
| 21:18.04 | Assid | well.. we 2 weeks.. and todayt they have a service message sent out |
| 21:22.07 | *** part/#asterisk supjigatr (n=syslod@152.53.16.10) |
| 21:26.34 | [TK]D-Fender | I'm off, BBIAB |
| 21:27.13 | Arth_72 | c u and thanks for the help |
| 21:27.16 | *** part/#asterisk Arth_72 (n=Arturo@88-149-232-233.f5.ngi.it) |
| 21:29.57 | *** join/#asterisk eltech (i=G00Ds@ool-457c93b6.dyn.optonline.net) |
| 21:30.49 | *** join/#asterisk sharp (n=sharp@2001:470:1f01:ffff:0:0:0:1c23) |
| 21:31.41 | Assid | err |
| 21:31.49 | Assid | anyone know any decent good providers |
| 21:32.01 | Assid | im freaking tired of these night by people coming in |
| 21:33.12 | *** join/#asterisk Hmmhesays (n=Neg@24-119-151-57.cpe.cableone.net) |
| 21:36.52 | *** join/#asterisk Assid (n=assid@59.183.14.221) |
| 21:36.54 | Assid | hi |
| 21:37.35 | Assid | okay seriously tho.. anyoneknow a good provider for pstn termination ? |
| 21:37.44 | Assid | not the fly by night operators |
| 21:39.03 | *** join/#asterisk IronHelix (n=irc@ool-45785cfe.dyn.optonline.net) |
| 21:39.15 | IronHelix | good morning all |
| 21:39.41 | tuan_modulis | how exactly does SetGroup(name) or Set(GROUP()=name) work exactly? Every time a caller adds this to the stack, it increments the group counter? |
| 21:41.46 | *** join/#asterisk KuJaX (n=one@customtrading.dsl.xmission.com) |
| 21:41.48 | *** part/#asterisk queztor (i=questor@office.endoria.net) |
| 21:42.19 | KuJaX | I have Trixbox and I just hired a new employee. For training, I want him to listen in on my phone call conversations. What is the best way of doing this? |
| 21:42.45 | Hmmhesays | yeah I just ordered a mid atx tower for 32 bucks |
| 21:42.45 | Hmmhesays | woo |
| 21:42.54 | tuan_modulis | how about recording the conversations? |
| 21:43.00 | JunK-Y | KuJaX: train him :) |
| 21:43.06 | tuan_modulis | unless u want to bridge the calles |
| 21:43.08 | tuan_modulis | calls |
| 21:43.09 | JunK-Y | seriously, use chanspy. |
| 21:43.18 | tuan_modulis | or that T_T |
| 21:43.48 | IronHelix | yeah i think chanspy would be the best way... either that or record calls (subject to local laws) and let him browse thru them |
| 21:44.00 | tuan_modulis | that's an asterisk feature or 3rd party app? |
| 21:44.12 | *** join/#asterisk bbbcoder (n=chatzill@mail.hurdmanivr.com) |
| 21:44.17 | cekc | anyone know if a hylafax irc channel exists and if so on which network? |
| 21:44.27 | IronHelix | chanspy comes with asterisk... not sure what trix calls it |
| 21:44.40 | tuan_modulis | oo, learn new things everyday |
| 21:44.43 | tuan_modulis | it's an actual command |
| 21:45.03 | IronHelix | yeah |
| 21:45.16 | bbbcoder | Where are the standard feature code extensions, such as *69, defined? I thought it was iana.org, but I can't find them and am not sure what these are called in order to google it. |
| 21:45.26 | Assid | okay seriously.. does anyone know of a good provider for incoming/outgoing? |
| 21:45.42 | IronHelix | connect.voicepulse.com? |
| 21:46.08 | Assid | IronHelix: is thaty for me? |
| 21:46.19 | IronHelix | yuppers |
| 21:46.23 | Assid | besides voicepulse |
| 21:46.32 | Sweeper | what could be causing my t1 port to go into red alarm every 5-10 minutes, and come back up automagically? |
| 21:46.47 | Assid | erven if outgoing isfrom a different provider |
| 21:46.48 | IronHelix | http://en.wikipedia.org/wiki/Vertical_service_code ? |
| 21:46.53 | IronHelix | hmmm |
| 21:47.00 | IronHelix | you could try voxee or junction networks |
| 21:47.05 | IronHelix | i dont have much experience w/ them tho |
| 21:47.14 | KuJaX | JunK-Y , chanspy froze our entire system up last time I tried it. |
| 21:47.49 | *** join/#asterisk Dr-Linux|home (n=nahin@DSL-202-59-73-131.nexlinx.net.pk) |
| 21:47.52 | bkruse | Sweeper: can you still receive calls then? |
| 21:48.02 | Dr-Linux|home | how can i call socket from the dialplan? |
| 21:48.08 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 21:48.27 | Hmmhesays | system call? |
| 21:48.30 | bbbcoder | *70 usually disables call waiting. What is this called and where are all the common ones defined? |
| 21:48.45 | Hmmhesays | google |
| 21:49.43 | Assid | there was one which specially ran asterisk or was mentioned somewhere that they are some kinda link up with the main project or something like that |
| 21:49.43 | Assid | i cant remember the name |
| 21:49.43 | Dr-Linux|home | Hmmhesays: backend database api, on socket communication |
| 21:50.09 | IronHelix | bbbcoder see wikipedia link above |
| 21:51.02 | IronHelix | assid maybe you mean asterlink? wasnt officially associated with Digium but a handful of devs worked on it as i recall |
| 21:51.02 | Assid | aah yes |
| 21:51.05 | Assid | im guessing they got superior call qualituy |
| 21:51.09 | JunK-Y | KuJaX: have you reported the bug? |
| 21:51.25 | Strom_C | bbbcoder: |
| 21:51.26 | Strom_C | ~vsc |
| 21:51.28 | jbot | well, vsc is Vertical Service Codes - these codes are generally reserved for specific uses, and it's a bad idea to conflict with the official assignments. A list of assigned VSCs for North America is at http://nanpa.com/number_resource_info/vsc_assignments.html |
| 21:52.22 | IronHelix | bbl |
| 21:52.23 | IronHelix\AFK | Thanks |
| 21:52.45 | bbbcoder | That's it. nanpa.com is what I'm looking for. The wikipedia article works too, of course, but maybe it's less official... |
| 21:52.52 | *** join/#asterisk heh_v_water (n=heh_v_wa@71-210-51-58.hlna.qwest.net) |
| 21:52.55 | bbbcoder | Thanks all. |
| 21:53.54 | Assid | is asterlink prepaid? |
| 21:54.53 | Dr-Linux|home | any clue on my question? |
| 21:54.54 | *** join/#asterisk infernix (i=nix@unaffiliated/infernix) |
| 21:56.06 | *** join/#asterisk ars247 (n=no@64-142-43-180.dsl.static.sonic.net) |
| 21:58.55 | Sweeper | bkruse: well, the red alarm only lasts for an instant, but all the calls get dropped |
| 21:59.21 | *** part/#asterisk techie (n=techie@c-67-181-184-170.hsd1.ca.comcast.net) |
| 22:00.19 | *** join/#asterisk techie (n=gus@voipops.net) |
| 22:02.55 | *** join/#asterisk logan|w (i=nothing@station441.dallasix.net) |
| 22:06.14 | *** join/#asterisk [TK]D-Fender (n=chatzill@64.235.216.2) |
| 22:10.16 | bkruse | dang |
| 22:11.06 | [TK]D-Fender | Yay, 22" of Dell goodness :) |
| 22:12.29 | Strom_C | are you basking in the glow of its expansive resolutiony goodness? :) |
| 22:12.54 | [TK]D-Fender | indeed |
| 22:13.47 | [TK]D-Fender | And just plain BIG. I sold my 19" at full retail for $240 and got this 22" for $330. Nice trade up, and I as I'm now on DVI for primary I can use it for setting up a secondar system easy as well on VGA |
| 22:15.25 | *** join/#asterisk delmar (n=delmar@ip-58-28-158-154.ubs-dsl.xnet.co.nz) |
| 22:15.45 | Assid | err anyone know any free carrier for tollfree termination ? |
| 22:15.55 | Assid | trxtel seems down |
| 22:16.43 | *** join/#asterisk gr1ncheux (n=devine@unaffiliated/gr1ncheux) |
| 22:17.12 | *** join/#asterisk popow (n=happytig@2607ds3-fb.0.fullrate.dk) |
| 22:17.32 | popow | Hello ppl |
| 22:18.26 | delmar | hi everyone. Here is a curly one.... I have just setup a basic firewall on a box, and since doing so, calls on Asterisk are dropping LOTS and quite randomly. at first I thought it was the cron job that reloads the script every so often, but while on a call I ran the script a bunch of times and couldnt knock myself off. I had a 15min long call work fine... and thought nothing more of it, until this morning when it started again. |
| 22:18.26 | delmar | It doesn't seem to be the provider. I have reversed the changes just now, so I am waiting to see if it does still happen but in the mean time, would anyone here have any ideas.. netfilter breaking Asterisk? |
| 22:18.38 | popow | We have a setup with an isdn 30 line going into a patton remote access server box, and from the patton to a pc with a tcp/gateway running |
| 22:18.46 | *** join/#asterisk dseeb_ (n=dcb@CPE-58-169-130-113.vic.bigpond.net.au) |
| 22:19.01 | popow | So is there anyway an asterisk server could replace the remote server? |
| 22:19.36 | delmar | popow, isdn 30 line.. E1 ? |
| 22:19.54 | popow | isdn30 = E1 yes |
| 22:20.13 | delmar | popow, then yes. using a digium T1/E1 card in a PC. |
| 22:20.31 | popow | We have that card already |
| 22:20.39 | delmar | popow, there u go. |
| 22:20.41 | popow | And the line in is fittted |
| 22:21.06 | delmar | popow, so u are all set. |
| 22:21.12 | popow | but how to make the traffic go out on the ethenet port of the asterisk server |
| 22:21.33 | delmar | popow, to where |
| 22:21.48 | popow | Right now we have E1 out from asterisk to patton to tcp/gateway |
| 22:22.28 | delmar | popow, so instead you want it to go from the E1 on Asterisk, to where? another Asterisk box? to hard phones? what? |
| 22:22.42 | popow | Asterisk E1 -> patton -> out of patton remote access server via eth0 -> machine with tcp/gateway |
| 22:22.52 | popow | So instead i want |
| 22:23.39 | popow | E1 -> asterisk -> eth0 out of asterisk server direct to -> tcp/gateway machine |
| 22:24.12 | *** join/#asterisk tuan_modulis (n=chatzill@hvmoduli.enter-net.com) |
| 22:24.15 | delmar | popow, so give the E1 box an ethernet connection |
| 22:24.22 | rkeels | Can anyone tell me how to get a list of standard global variables ie what is the variable for getting a callerid into a function |
| 22:24.38 | delmar | rkeels, the wiki ? |
| 22:24.58 | [TK]D-Fender | ~book |
| 22:25.00 | jbot | methinks book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 22:25.00 | [TK]D-Fender | ~docs |
| 22:25.02 | jbot | i guess docs is Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com |
| 22:25.02 | [TK]D-Fender | ~wikis |
| 22:25.04 | jbot | methinks wikis is http://www.voip-info.org |
| 22:25.10 | popow | yes if i put ethernet betwen asterisk and machine with tcp/gateway |
| 22:25.16 | *** join/#asterisk ToyMan (n=Stuart@user-12lcqvl.cable.mindspring.com) |
| 22:25.40 | [TK]D-Fender | rkeels: lookup "asterisk variables", "asterisk standard extensions", and "asterisk functions" ont he WIKI. |
| 22:25.51 | popow | then the all the magic the patton remote server does has to be done in asterisk server |
| 22:26.06 | rkeels | thx I will try that |
| 22:26.39 | popow | So I am not asking how to do it. Only I ask can asterisk do this? |
| 22:27.07 | [TK]D-Fender | popow: yes |
| 22:27.09 | popow | I.e can I put the patton to the dustbin :-) |
| 22:28.16 | rkeels | SWEET, could it be any easier? |
| 22:28.48 | tuan_modulis | where is digium based anyway? |
| 22:28.53 | popow | Nice :-) thx delmar & TKD-fender |
| 22:29.51 | [TK]D-Fender | tuan_modulis: Sol 3, northern polar hemisphere |
| 22:29.55 | popow | Any links to getting E1 lines routed to eth0 port? |
| 22:30.20 | [TK]D-Fender | popow: E1 != ethernet. |
| 22:30.38 | tuan_modulis | heh |
| 22:30.47 | [TK]D-Fender | popow: Perhaps you should be more specific as to WHAT you will be speaking on the ethernet side. |
| 22:30.49 | *** join/#asterisk nosbig (n=nosbig@rrcs-70-62-223-6.central.biz.rr.com) |
| 22:31.03 | popow | Oki i will explain |
| 22:31.11 | nosbig | Anyone here using bandwidth.com SIP trunks? plural? |
| 22:31.27 | popow | I have E1 line comming into asterisk on a songoma card |
| 22:32.06 | nosbig | I have a working dialplan for outbound long-distance calling. My users enter 9, followed by the full phone number, including area code (10-digit dialling due to overlay). |
| 22:33.02 | popow | And from another port on the same card E1 going out to a patton remote server gw, going out of patton is eth0 ethernet connecting to a windows box running a small program/gateway that forwards the tcp/ traffic to |
| 22:33.09 | popow | a compagny in france |
| 22:33.33 | nosbig | That works perfectly. Outbound local calling is another matter entirely. I have duplicated the long-distance stanza, removing the 9 and tacking it onto the +1 for E.164 compatibility. |
| 22:33.46 | nosbig | But I get the following error on an attempt to call: |
| 22:33.58 | popow | So I wanted to skip the remote access server patton and connect asterisk directly to the windows box (eth0 ethernet) |
| 22:34.19 | *** join/#asterisk Dr-Linux|home (n=nahin@DSL-202-59-73-131.nexlinx.net.pk) |
| 22:34.30 | nosbig | Forbidden - wrong password on authentication for INVITE to '""XXXX XXX" <sip:+19995551212@4.79.212.236>;tag=as6e2e9fbe |
| 22:34.46 | [TK]D-Fender | popow: You are not being clear what PROTOCOL is being sent to the Windows box, and for what purpose. |
| 22:34.52 | nosbig | But there are no usernames and passwords in use anywhere, per BW.com's instructions... |
| 22:35.24 | nosbig | What does this error point to? Any why can I call long-distance (or local numbers dialled as long-distance) easily. |
| 22:36.40 | popow | tcp traffic from a pos terminal sent over pstn net to the E1 phonenumber on asteriskserver -> out to patton remote server -> to local windows machine- > which sends the tcp/traffic to a server in france for billing |
| 22:38.24 | [TK]D-Fender | popow: A dial-up type connection from the POS? |
| 22:38.28 | popow | So yes |
| 22:38.31 | popow | yes |
| 22:38.33 | popow | <PROTECTED> |
| 22:38.52 | popow | so the patton is one big modem :-) |
| 22:39.01 | [TK]D-Fender | popow: What kind of connection exactly is thes POS doing? |
| 22:39.45 | popow | it send tcp traffic over pstn. It is a solution for selling electronic pins for sale on a pos terminal |
| 22:40.46 | popow | So the local winows machine we have act as gateway for this traffic. We have 2 gateways 1 for adsl and one for pstn |
| 22:43.24 | [TK]D-Fender | popow: You are still not ver SPECIFIC. Analog FXS device? ISDN? Usually "TCP over PSTN" implies taht along the way the call is going through an analog modem. |
| 22:43.47 | popow | it is pstn pos -> analog line |
| 22:44.07 | popow | SO yes to analog modem |
| 22:45.34 | popow | So the pos being PSTN analog modem based POS dial up to a number which we have an ISDN30 (E1 line) on which is an astersisk server with songoma card |
| 22:45.42 | [TK]D-Fender | popow: At that point you'll want an analog card on your * server, and you're set. |
| 22:47.05 | popow | So analog card today is there in the form of and patton remote server with E1 interface connect to windows box. So I wanted to skip the patton box. |
| 22:47.33 | popow | Using songoma E1 in and *server eth0 out |
| 22:47.35 | [TK]D-Fender | popow: You don't need 10 extra hops to plug a stupid FXS devine into your * server. |
| 22:47.41 | [TK]D-Fender | device* |
| 22:48.42 | popow | the custermer is many external |
| 22:49.02 | popow | that dials to our isdn30 number |
| 22:49.42 | popow | the device is not onside |
| 22:50.22 | popow | Old setup was like this E1 line in to patton and vupti into windows box |
| 22:50.28 | popow | no *server |
| 22:50.49 | popow | So now we have *server and wanted to skip the patton |
| 22:50.53 | [TK]D-Fender | popow: Might jsut be able to use an ATA on ULAW if you're lucky |
| 22:51.04 | [TK]D-Fender | ALAW actually. |
| 22:52.44 | popow | Well ATA box aint the setup .. I cant seem to explain the setup properly.. Ill fix a drawing another day and ask again |
| 22:52.49 | popow | thx for the try :-) |
| 22:54.07 | *** join/#asterisk mtaht4 (n=m@c-71-198-23-124.hsd1.ca.comcast.net) |
| 22:55.32 | [TK]D-Fender | popow: Yes, a drawing would be nice. You seem to describing items at the far ends, but being very vague about how they come together. Draw a complete picture and host it somewhere. |
| 22:56.54 | *** part/#asterisk dasenjo (i=dasenjo@gateway/tor/x-720214e32b819c61) |
| 22:57.31 | popow | thx fender i will make a drawing |
| 22:58.36 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 22:59.35 | nosbig | OK. I figured out why my dialplans were not allowing me out in certain circumstances. For some reason, bandwidth.com is refusing my connection when I try to set caller id (even to the same number as our "fromuser" parameter). |
| 23:03.10 | popow | Anyone in here heard of portaswitch ? |
| 23:03.32 | Aurs | are there any good alternatives to callgroup/pickupgroup? that isn't limited to 64 groups? |
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| 23:20.45 | marv[work] | so....why does ast_select take fds....instead of say hot chicks |
| 23:21.16 | ManxPower | That is a mystery even to the developers |
| 23:21.17 | *** join/#asterisk topping (n=topping@204.152.96.238) |
| 23:22.37 | tuan_modulis | wat's an asterisk regular expression to extract the number from say Local/605@adsadsasd.... [0-9]+ doesnt seem to work |
| 23:23.35 | bkruse | marv[work]: im writing a patch now |
| 23:27.27 | *** join/#asterisk orkid (n=orkid@bas1-barrie18-1242377764.dsl.bell.ca) |
| 23:28.43 | marv[work] | bkruse: thanks. Do you think you can backport it to 1.2 since it's technically a bug fix? Or is it too big of a structural and/or policy change? |
| 23:29.32 | tuan_modulis | ack im so stupid, i should've RTFM |
| 23:31.48 | tuan_modulis | i love how in voipinfo the documentation is so obscure though |
| 23:31.57 | tuan_modulis | and i quote: However, if ${cid} contains foo123 then ${cid2} will be empty. Supposedly you could use =~ instead of : and the string matching would work anywhere, but I don't think that works correctly (at least, I could not get =~ to work). |
| 23:32.19 | bkruse | marv[work]: well, i would want to port it from 1.2 up to 1.4 then to trunk, as its suppose to be i believe |
| 23:32.21 | tuan_modulis | for regular expressions |
| 23:32.32 | bkruse | its going to be a hard fix, not to mention bsd support |
| 23:32.32 | bkruse | :P |
| 23:33.39 | *** join/#asterisk puzzled (n=patrick@535427C9.cable.casema.nl) |
| 23:35.27 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
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| 23:53.57 | tuan_modulis | for dead channels, what's the best way to get rid of them short of doing a cron job at certain time intervals for restarting the service? |
| 23:54.05 | *** join/#asterisk freestyle (i=freestyl@2002:c951:8358:0:0:0:c951:8358) |
| 23:54.21 | tuan_modulis | well, not best... im looking for alternatives |
| 23:54.37 | tuan_modulis | something that'll allow me to not hire a monkey |
| 23:54.39 | nosbig | tuan_modulis: Are you talking about calling channels? |
| 23:54.54 | tuan_modulis | nosbig: yes |
| 23:54.59 | nosbig | tuan_modulis: My active channel list clears out automatically... |
| 23:55.31 | tuan_modulis | nosbig: but dont u get dead channels from time to time? |
| 23:55.39 | tuan_modulis | like, they're just not used |
| 23:55.53 | tuan_modulis | and they disappear when u do a restart |
| 23:57.20 | nosbig | tuan_modulis: No, I don't get dead channels. I use Zap (for another 18 hours or so) and SIP. Occassional IAX. ALl are torn down within a few seconds of inactivity. |
| 23:57.38 | *** join/#asterisk mtgh (n=chatzill@dsl093-001-038.det1.dsl.speakeasy.net) |
| 23:57.47 | tuan_modulis | nosbig: ic, thx for that info |
| 23:58.32 | nosbig | np. What channels are you trying to use? What version of *? |
| 23:59.03 | ars247 | nosbig? you know anything regarding one way audio with 1.4? |
| 23:59.08 | tuan_modulis | well, apparently, there's something fishy going on, I'm using sip channels with 1.2.16 |
| 23:59.31 | tuan_modulis | but actually, this is a new problem, the channels are still there even with service restart |
| 23:59.41 | nosbig | Not had any experience with 1.4 ATM. |
