| 00:00.51 | mog | ManxPower, it uses dhcp |
| 00:00.53 | mog | not bootp |
| 00:01.01 | mog | it might be able to do bootp too |
| 00:01.04 | mog | but it defaults to dhcp |
| 00:01.59 | ManxPower | mog: has this changed? |
| 00:02.10 | *** part/#asterisk Fieldy (n=toon@gentoo/contributor/Fieldy) |
| 00:02.36 | mog | it has always been that way |
| 00:02.44 | mog | at least since i got here 3 years ago |
| 00:03.21 | ManxPower | werid. I was sure the one IAXy we got required bootp. Sorry. |
| 00:03.58 | ManxPower | was any of the other information I gave incorrect? |
| 00:04.33 | robin_sz | thanks to all those who put up with my whining and offered help over the weekends fun with bristuff |
| 00:04.41 | robin_sz | the system works nicely now |
| 00:04.58 | robin_sz | just one teeny, weeny thing ... |
| 00:05.10 | robin_sz | the zaphfc incoming numbers ... |
| 00:05.11 | mog | it just dhcp/static, ulaw/alaw/adpcm and does not support dns, has fall over support, as well as heartbeat, supports cid, cw, cidcw, and threeway |
| 00:05.13 | mog | thats it |
| 00:05.25 | robin_sz | the callerid is missing the leading 0 |
| 00:05.40 | robin_sz | any clooes? |
| 00:06.39 | ManxPower | robin_sz: there may be some prefix options |
| 00:06.51 | robin_sz | ummm |
| 00:07.23 | ManxPower | at least there is for PRI |
| 00:09.18 | *** part/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net) |
| 00:10.18 | robin_sz | stripmsd? |
| 00:10.23 | *** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.ct.comcast.net) |
| 00:10.27 | ZaVoid | morning |
| 00:12.14 | ManxPower | robin_sz: no, that would remove |
| 00:13.06 | ManxPower | ; sample 1 for Germany |
| 00:13.07 | ManxPower | ;internationalprefix = 00 |
| 00:13.07 | ManxPower | ;nationalprefix = 0 |
| 00:13.07 | ManxPower | ;localprefix = 0711 |
| 00:13.19 | robin_sz | anyway ... its causing some grief :( |
| 00:13.30 | ManxPower | there |
| 00:13.34 | robin_sz | hmmm |
| 00:13.54 | robin_sz | how does it know if its local, national or international? |
| 00:14.13 | robin_sz | to add th eprefixes? |
| 00:15.15 | ManxPower | I suspect the info comes in on an ISDN message |
| 00:15.19 | robin_sz | hmmm |
| 00:15.33 | ManxPower | I do not know if BRIstuff supports this or not. |
| 00:15.59 | robin_sz | I know when i had pridilplan=local it failed ot make calls, so i set it to unkown instead, that worked |
| 00:16.44 | ManxPower | yes, that IS why the comments say that option is almost never used. |
| 00:17.09 | robin_sz | just wondering if that will mess up the outgoing numbers again |
| 00:17.29 | ZaVoid | nature of address issue? |
| 00:19.16 | *** join/#asterisk ltdwk- (n=z@203-173-10-9.perm.iinet.net.au) |
| 00:19.47 | *** join/#asterisk bmg505 (n=leon@196.209.180.71) |
| 00:20.13 | *** join/#asterisk SuperID (n=gary@c-65-96-225-97.hsd1.ma.comcast.net) |
| 00:21.12 | ManxPower | robin_sz: it should not |
| 00:34.44 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 00:34.44 | blitzrage | evening all |
| 00:35.26 | blitzrage | if I call a Macro(), and I Set() a variable, which I want available outside of the Macro() (specifically in the 'h' extension, but just assume I want access to it outside of the Macro()) can anyone think of how I might do that? |
| 00:35.37 | blitzrage | Yes, I have tried using Set(__MY_VAR=foo) |
| 00:36.42 | blitzrage | wow, there really isn't anyone around tonight |
| 00:36.53 | *** join/#asterisk LoneShadow (i=duh@nat/cisco/x-0df14fcfdb6294b0) |
| 00:37.05 | *** part/#asterisk LoneShadow (i=duh@nat/cisco/x-0df14fcfdb6294b0) |
| 00:39.07 | xpot | has anyone here ever used OpenSER as a load balancer for Asterisk? |
| 00:41.05 | ZaVoid | yeah |
| 00:41.08 | ZaVoid | we have |
| 00:41.15 | ZaVoid | simple dispatcher list |
| 00:41.46 | xpot | I tried the instructions at voip-info and I still get errors.. do you think you can help me out? |
| 00:43.25 | xpot | I performed the info posted at: <http://openser.org/dokuwiki/doku.php/asterisk:load-balancing-and-ha> and I get errors |
| 00:43.37 | xpot | although I am not sure where: if ( method=="INVITE" ) ... is supposed to be placed in the cfg? |
| 00:45.43 | ZaVoid | i'm really not that very good with |
| 00:45.46 | ZaVoid | i barely got ours to work :( |
| 00:46.03 | *** join/#asterisk dj-fu (n=aj@unaffiliated/dj-fu) |
| 00:46.09 | xpot | ok, thanks anyways |
| 00:46.58 | blitzrage | there is an #openser channel, funny enough |
| 00:47.18 | Corydon76-home | That's funny? You need to get out more. |
| 00:47.26 | xpot | blitzrage: I am already in there... it seems everyone is sleeping or dead in there too |
| 00:47.31 | blitzrage | Corydon76-home: it's true... my humor is very dry |
| 00:47.43 | blitzrage | guess some people DO get out :) |
| 00:47.51 | xpot | -=0) hehe |
| 00:48.05 | Corydon76-home | waaaaay out |
| 00:48.13 | Corydon76-home | (even for Toronto) |
| 00:48.15 | blitzrage | :) |
| 00:48.40 | Corydon76-home | We need a piece on SLA |
| 00:49.15 | blitzrage | yes... |
| 00:49.16 | blitzrage | ideally |
| 00:49.22 | blitzrage | but we need to get the rest done first :( |
| 00:49.30 | blitzrage | Corydon76-home: did you see my question above? |
| 00:50.16 | *** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 00:50.17 | Corydon76-home | No |
| 00:50.33 | Corydon76-home | I've been trolling CL for you-know-what |
| 00:50.33 | blitzrage | ooo... I wonder if I set the variable before I call the Macro, then reset it within the Macro() if the variable will be available outside of it.... |
| 00:50.59 | Corydon76-home | You should be able to set a variable within Macro just fine |
| 00:51.14 | blitzrage | I can |
| 00:51.15 | Corydon76-home | as long as it's not ARG1...ARGn |
| 00:51.20 | blitzrage | but then it doesn't seem to be available in 'h' |
| 00:51.30 | Corydon76-home | Eh? |
| 00:51.30 | blitzrage | the other thing I wonder if it matters which leg of the call hung up |
| 00:51.51 | Corydon76-home | should be in 'h' just fine |
| 00:51.54 | blitzrage | you'd think so |
| 00:51.59 | blitzrage | here, let me show you what I'm doing |
| 00:52.01 | Corydon76-home | You don't mean the M option in Dial, do you? |
| 00:52.04 | blitzrage | yes |
| 00:52.05 | blitzrage | I do |
| 00:52.15 | Corydon76-home | Well, that's why it's not available |
| 00:52.24 | blitzrage | it doesn't hang around in memory? |
| 00:52.35 | Corydon76-home | the M option calls the Macro on the CALLED channel, not the originating channel |
| 00:52.41 | Corydon76-home | So it's a different namespace |
| 00:53.51 | blitzrage | right -- which is why I was thinking it depends who hangs up? |
| 00:53.51 | Corydon76-home | You're not understanding |
| 00:53.51 | blitzrage | or does that not apply here |
| 00:53.51 | Corydon76-home | h is for the CALLING channel |
| 00:53.52 | blitzrage | aha |
| 00:53.52 | blitzrage | ok, that makes sense then |
| 00:53.52 | blitzrage | hrmmm |
| 00:53.52 | blitzrage | so I guess the only way to get around that is to set a global variable... |
| 00:54.37 | Corydon76-home | unless you can call ImportVar before the bridged channel goes away... |
| 00:54.38 | blitzrage | where would I call that? |
| 00:54.38 | Corydon76-home | but I don't know how you can do that |
| 00:54.39 | blitzrage | ya... |
| 00:54.43 | blitzrage | I don't see where it would get called |
| 00:55.07 | blitzrage | unless I pass the channel name to the M(), then call ImportVar() inside that? |
| 00:55.18 | Corydon76-home | Possible, sure |
| 00:55.31 | Corydon76-home | but that pulls the variable into the outgoing channel |
| 00:55.42 | Corydon76-home | it doesn't pull variables back into the incoming channel |
| 00:55.43 | blitzrage | nope, that won't work |
| 00:55.51 | blitzrage | now that I've read the description |
| 00:56.19 | blitzrage | only way I can see it then is to use a global |
| 00:56.24 | blitzrage | I'll just have to set it to be unique |
| 00:56.31 | blitzrage | then reset it back to 0 in 'h' |
| 00:57.58 | Corydon76-home | You could potentially use a database... |
| 00:58.11 | Corydon76-home | pass in the calling channel name, then use that as a database key |
| 00:58.44 | blitzrage | crap... I DO have to set it in the DB or else the other Asterisk boxes won't know about it |
| 00:58.49 | blitzrage | shit... I don't like that at all |
| 00:59.17 | Corydon76-home | then set both channel name and hostname in the database, plus an id, triple key |
| 01:00.06 | blitzrage | actually, I'm already setting the value I need to in the DB .... I can check that, so ok, thats alright |
| 01:00.17 | blitzrage | yep... I think this global var might do everything I need |
| 01:00.36 | Corydon76-home | How are you dealing with possible conflicts? |
| 01:00.54 | blitzrage | which conflicts? The globla var will contain a unique identifier |
| 01:01.03 | Corydon76-home | Ah... |
| 01:01.07 | blitzrage | ya |
| 01:01.11 | blitzrage | that's about all I can think of :) |
| 01:01.24 | Corydon76-home | Um, tiny problem |
| 01:01.29 | blitzrage | I'll have a bunch of global vars kicking around.... but at least they'll be cleared if Asterisk crashes |
| 01:01.49 | Corydon76-home | There is no way in the dialplan to clear global variables, so your global namespace is going to eat memory |
| 01:01.57 | blitzrage | ya... this I realized... |
| 01:02.08 | blitzrage | I don't think I have much of a choice though |
| 01:02.25 | Corydon76-home | You can clear database rows, in two places though |
| 01:02.27 | blitzrage | and the amount of variables I'm gonna be using... I'm hoping won't use toooo much space |
| 01:02.45 | Corydon76-home | a) when you originate an incoming channel, clear them, and in the 'h' extension, clear them |
| 01:03.32 | Corydon76-home | At least with a DB, you have ultimate control over storage |
| 01:04.39 | Corydon76-home | I have, however, done something like this before, as a hack for a customer... |
| 01:04.56 | blitzrage | actually... since the only time this is gonna be checked, is when 'h' is executed, even if Asterisk crashes, it won't matter, because it won't be used to block the call from hitting the agent... |
| 01:05.04 | blitzrage | it'll be the data in the DB that does that... |
| 01:05.06 | Corydon76-home | I stuck the last SIP response code in a global variable, unique to each peer... |
| 01:05.18 | Corydon76-home | SIP_${PEER}_STATUS |
| 01:05.33 | blitzrage | yah, that was the same idea I was basically going with in re: to the global var... |
| 01:06.06 | Corydon76-home | Oh, if that's the case, then you don't have to worry too badly about filling up namespace |
| 01:06.09 | *** join/#asterisk ez` (n=ez@c66.110.149-45.clta.globetrotter.net) |
| 01:06.47 | Corydon76-home | However, things might suck if someone parks a call, then makes a second call... then you have two concurrent calls to the same SIP peer and a possible conflict |
| 01:07.14 | blitzrage | this will only be on termination, and never origination |
| 01:07.22 | Corydon76-home | Ah... |
| 01:07.36 | blitzrage | basically, I'm using this to track whether someone is on a call or not, and block calls from the Queue going to them |
| 01:08.17 | blitzrage | will also use a GROUP_COUNT() to check for then they originate a call as to their status |
| 01:08.44 | *** join/#asterisk thoughtpolice (n=austin@c75-111-139-133.plaicmtc01.tx.dh.suddenlink.net) |
| 01:08.47 | blitzrage | actually... probably just set the same column in the DB will be sufficient |
| 01:09.29 | blitzrage | the only thing is I'm going to have stale values if Asterisk ever crashes |
| 01:09.53 | blitzrage | that field won't be cleared, and when it comes back up, Asterisk will still think the phone is unavailable |
| 01:10.56 | blitzrage | although... if I don't keep it in the database, and keep it in a global var, then I can use DUNDi to check the variables that way... so when Asterisk crashes... I won't end up with anything stale... |
| 01:11.35 | blitzrage | ya... I think that might be the best way of doing it |
| 01:12.10 | *** join/#asterisk Strom_M (n=strom@netblock-66-159-243-60.dslextreme.com) |
| 01:13.18 | CCFL_Man2 | who here has asterisk working nicely with an adit 600? |
| 01:19.54 | *** join/#asterisk raptorrat (n=rathomps@cpe-66-25-25-138.houston.res.rr.com) |
| 01:20.55 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 01:20.55 | *** mode/#asterisk [+o anthm] by ChanServ |
| 01:22.54 | *** join/#asterisk ghento (n=ghento@CPE0014bfaf7d7a-CM00111ae4684c.cpe.net.cable.rogers.com) |
| 01:23.38 | *** part/#asterisk [[blah]asfd (n=ckwall@c-71-195-196-198.hsd1.ut.comcast.net) |
| 01:23.39 | *** join/#asterisk Cabal (n=Cabal@unaffiliated/cabal) |
| 01:25.42 | ghento | hi everyone. just wondering, has anyone had trouble with skypeout -> SIP? Asterisk isn't recognizing skype's dtmf inputs |
| 01:27.09 | *** join/#asterisk [[blah]asfd (n=ckwall@c-71-195-196-198.hsd1.ut.comcast.net) |
| 01:27.25 | [[blah]asfd | besides the support, are there any advantages to the asterisk business addition? |
| 01:35.31 | *** part/#asterisk raptorrat (n=rathomps@cpe-66-25-25-138.houston.res.rr.com) |
| 01:37.57 | *** join/#asterisk raptorrat (n=rathomps@cpe-66-25-25-138.houston.res.rr.com) |
| 01:43.31 | Corydon76-home | [[blah]asfd: No, the support is pretty much why you buy ABE |
| 01:43.34 | SuperID | I want to try out my first asterisk box on a home lan. Besides soft phones, what external services do I need to get started? Usage is all US domestic, virtually all local calls, mix of inbound/outbound. Recommendations appreciated |
| 01:43.51 | *** join/#asterisk khronos (n=khronos@c-76-110-134-230.hsd1.fl.comcast.net) |
| 01:43.52 | Corydon76-home | ABE is for people who buy RHEL for the enterprise level support |
| 01:47.06 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 01:55.50 | *** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.ct.comcast.net) |
| 01:56.09 | ZaVoid | whats the diff between exten => s,1 and n,1 |
| 01:57.32 | JunK-Y | s is a special extension, n means next, so it follows any extension,1 |
| 01:57.44 | JunK-Y | n,1 is impossible btw, it must be n > 1 |
| 01:58.06 | Strom_M | n is a valid extension name |
| 01:58.22 | ZaVoid | yeah i got tons like s,n |
| 01:58.27 | ZaVoid | but some like s,1 s,2 s,3 |
| 01:58.38 | JunK-Y | ya, true, n is for the priority! |
| 01:58.38 | Strom_M | ZaVoid: look up "asterisk special extensions" |
| 01:58.43 | ZaVoid | thanks strom |
| 01:58.53 | ZaVoid | n is for prio? |
| 01:59.01 | JunK-Y | Strom_M: kick me back to my school books! |
| 01:59.25 | ZaVoid | http://www.voip-info.org/wiki/index.php?page=Asterisk+config+extensions.conf |
| 01:59.26 | JunK-Y | ZaVoid: instead of saying, foo,2, foo,3, ya can say foo,n, foo,n |
| 01:59.28 | ZaVoid | ahh sweet thanks man |
| 01:59.45 | ZaVoid | ok so the fact it goes 1,2,3,n,n,n,n, is just lazy on whoever made it orginally basically? |
| 02:00.22 | JunK-Y | yes, so if ya need to insert something at priority 11, that will be really easier if using lot of priorities n. |
| 02:00.30 | ZaVoid | right |
| 02:00.35 | ZaVoid | i see now |
| 02:01.08 | ZaVoid | but it reads the exten's in a macro in order form top down right? |
| 02:01.27 | ZaVoid | so the 1,2,3 wouldn't matter as much as order in the .conf file? |
| 02:02.28 | *** join/#asterisk shadou (n=aj@unaffiliated/dj-fu) |
| 02:03.58 | *** part/#asterisk SuperID (n=gary@c-65-96-225-97.hsd1.ma.comcast.net) |
| 02:06.29 | *** join/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net) |
| 02:06.29 | *** mode/#asterisk [+o mog] by ChanServ |
| 02:07.59 | blitzrage | the order does matter |
| 02:08.17 | blitzrage | well... not in the extensions.conf I guess it doesn't -- but you can't skip priorities |
| 02:08.33 | blitzrage | although why you wouldn't order them in the appropriate order boggles my mind :) |
| 02:08.48 | ZaVoid | right makes sense |
| 02:09.36 | *** part/#asterisk [[blah]asfd (n=ckwall@c-71-195-196-198.hsd1.ut.comcast.net) |
| 02:10.49 | Qwell | blitzrage: n priority |
| 02:11.19 | blitzrage | right... order with 'n' would... of course... matter :) |
| 02:11.28 | blitzrage | I meant numbered priorities I suppose :) |
| 02:14.15 | ZaVoid | lol |
| 02:19.04 | *** join/#asterisk sysreq (n=sysreq@modemcable171.134-81-70.mc.videotron.ca) |
| 02:20.18 | *** join/#asterisk dj-fu (n=aj@unaffiliated/dj-fu) |
| 02:22.37 | *** join/#asterisk Fieldy (i=Xzm5rO7z@gentoo/contributor/Fieldy) |
| 02:26.22 | *** join/#asterisk xtr (i=94752345@64.46.20.91) |
| 02:41.17 | *** join/#asterisk ohadz (n=ohad@cpe-69-203-27-50.nyc.res.rr.com) |
| 02:43.50 | *** join/#asterisk RyanW (n=cableguy@ge0-0-15-lns0.207alg.qx21.net) |
| 02:45.13 | RyanW | http://pastebin.ca/478313 would the caller in this instance be billed, did asterisk technicall answer the call? |
| 02:46.09 | SwK | RyanW that call should not have been answered |
| 02:46.35 | Strom_M | RyanW: no answer on that call |
| 02:46.41 | SwK | but you have to becareful and check the apps... some apps like playback, zapateller etc auto answer the calls |
| 02:46.45 | Qwell | BUT, your question is ambiguous |
| 02:46.51 | Qwell | *should* the caller be billed? No. |
| 02:46.57 | Qwell | *would* the caller be billed? That's up to you. |
| 02:47.15 | Strom_M | if you want to be sure, though, make sure you send PROCEEDING at the start of the call, and flag any app that auto-answers with "noanswer" |
| 02:47.16 | SwK | yeah what qwell said (or depends on your carrier heh) |
| 02:48.37 | RyanW | is that done using the progress application/ |
| 02:48.59 | *** join/#asterisk BSD_Tech (n=bsdtech@adsl-69-230-166-20.dsl.irvnca.pacbell.net) |
| 02:49.29 | Strom_M | yes |
| 02:49.36 | RyanW | Thanks guys, |
| 02:50.19 | russellb | meepmeep |
| 02:54.58 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 02:56.16 | *** part/#asterisk Rusty1 (n=Rusty1@cpe-72-226-96-74.nycap.res.rr.com) |
| 02:57.29 | *** join/#asterisk ManxPower (n=manxpowe@83.sub-70-216-112.myvzw.com) |
| 02:57.44 | ghento | can you no longer use your own variables inside the System() function? like, System(...${RECORDED_FILE}..) works, but System(..${FOO}...) is being replaced with nothing (even though it is Set()) |
| 02:57.50 | ghento | ^ in 1.4 |
| 02:58.12 | russellb | no, that should still be working fine |
| 02:58.30 | ManxPower | ghento: put a Noop(FOO is ${FOO}) in the dialplan and see what the console shows |
| 02:58.47 | ghento | ok |
| 03:00.54 | ghento | ah, stupid me. I had spaces within my Set() ..Set(FOO = ...) ..removing them and it's working. thanks guys! |
| 03:01.14 | Strom_M | instant catsex |
| 03:01.40 | russellb | owned by whitespace |
| 03:01.43 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 03:02.00 | *** join/#asterisk threat (i=threat@60-240-43-214.static.tpgi.com.au) |
| 03:02.10 | ghento | haha |
| 03:02.20 | *** join/#asterisk flashnet (n=flashnet@211.223.75.49) |
| 03:02.36 | ghento | the whites are always ruining my work |
| 03:12.44 | SwK | you know that not being abot to handle whitespace in set annoys me |
| 03:12.44 | ManxPower | ghento: Noop for debug is the single most powerful tool you have for debugging dialplan issues |
| 03:12.44 | ManxPower | well stop putting in extra whitespace |
| 03:12.45 | ghento | ManxPower: thanks, gonna be using it a lot from now on. |
| 03:12.46 | *** join/#asterisk Dimik (n=Dimik@unaffiliated/dimik) |
| 03:13.03 | SwK | set(foo = bar) can be easier to read then set(foo=bar) at times |
| 03:13.03 | SwK | (liek when I been drunking |
| 03:13.03 | *** join/#asterisk remmo (n=junk@smack.isp.net.au) |
| 03:15.17 | *** join/#asterisk jbot (i=ibot@pdpc/supporter/active/TimRiker/bot/apt) |
| 03:15.17 | *** topic/#asterisk is Asterisk: The Open Source PBX -=- Asterisk 1.4.4 (April 27, 2007) Asterisk 1.2.18 (April 24, 2007), Zaptel 1.2.17.1, 1.4.2.1 (April 25, 2007) -=- Other fun channels: #asterisk-gui, #asterisknow, #asterisk-commits -=- Join #freepbx for freepbx/#trixbox for trixbox support. |
| 03:19.16 | *** join/#asterisk MrChicken (n=Dorphals@pcsp163-73.supercabletv.net.co) |
| 03:19.17 | MrChicken | Hello |
| 03:19.58 | MrChicken | I'm looking for the beta firmware of the grandstream phone (GXP2000) |
| 03:20.05 | MrChicken | can anybody help me? |
| 03:24.29 | *** join/#asterisk Mavvie (n=edwin@ppp161-18.lns4.syd6.internode.on.net) |
| 03:24.43 | RyanW | http://www.voip-info.org/wiki/view/GXP-2000 |
| 03:25.10 | *** join/#asterisk Mavvie (n=edwin@ppp161-18.lns4.syd6.internode.on.net) |
| 03:25.50 | *** join/#asterisk stkn__ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 03:27.19 | ManxPower | SwK: If you want to see some horrible macros that used to be mostly compat with asterisk 1.x see http://www.fnords.org/~eric/macros.inc |
| 03:27.47 | SwK | thats ok i'll pass |
| 03:27.52 | SwK | i have my own horrible macros |
| 03:30.39 | *** join/#asterisk Beirdo (n=gjhurlbu@unaffiliated/beirdo) |
| 03:30.40 | MrChicken | RyanW --> yeah. but I cant find anything there |
| 03:30.46 | MrChicken | I mean, the links are dead |
| 03:32.53 | ohadz | hi, i am trying to setup asterisk-gui -- when i login i get a blank page where i should have -- Welcome to Asterisk GUI setup wizard |
| 03:33.40 | *** part/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net) |
| 03:35.44 | MrChicken | ohadz --> Did you add the webenabled=yes line to the manager.conf? |
| 03:36.38 | ohadz | MrChicken, yes |
| 03:37.07 | MrChicken | permissions OK? |
| 03:37.46 | ohadz | MrChicken, [manager] secret=manager read = system,call,log,verbose,command,agent,user,config write = system,call,log,verbose,command,agent,user,config |
| 03:38.22 | wolferine | what manager command do I use to track phone calls, both incoming and outgoing? |
| 03:38.24 | MrChicken | no i meant on the files :P |
| 03:38.46 | ohadz | MrChicken, what should it be? |
| 03:38.49 | ohadz | 777? |
| 03:38.50 | ohadz | :) |
| 03:39.15 | MrChicken | perhaps :p |
| 03:39.32 | MrChicken | Although lately I thnk asterisk whines for 777d files |
| 03:39.43 | MrChicken | I think u should check the logs |
| 03:39.51 | MrChicken | good luck finding them :) |
| 03:40.50 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 03:40.56 | ohadz | MrChicken, i get the page "Welcome to Asterisk GUI setup wizard" but nothing in it -- no option to click "next" |
| 03:41.29 | MrChicken | ah I had the exact same prob |
| 03:41.37 | MrChicken | it was all in the manager.conv |
| 03:41.39 | MrChicken | conf |
| 03:41.40 | ohadz | MrChicken, what did you do? |
| 03:41.59 | MrChicken | the webenabled=yes |
| 03:42.05 | MrChicken | and there is another line |
| 03:42.22 | MrChicken | but I cant remember what the damned line was :$ |
| 03:42.28 | ohadz | ... |
| 03:42.42 | MrChicken | maybe in #asterisknow u'll get lucky? |
| 03:42.44 | MrChicken | If ya dont |
| 03:42.49 | MrChicken | add me |
| 03:42.54 | MrChicken | david@megatherion.com |
| 03:42.56 | MrChicken | thats my msn |
| 03:43.04 | MrChicken | talk to me tomorrow at 10am |
| 03:43.07 | MrChicken | 12 hrs |
| 03:43.13 | MrChicken | and I'll give ya my config |
| 03:43.23 | ohadz | cool. thanks |
| 03:44.42 | MrChicken | now if you would be willing to sacrifice your neihbor (only if its a she and pretty) in some kind of ritual to me... I'd be thankful |
| 03:45.10 | MrChicken | ;) |
| 03:45.54 | MrChicken | Anywho.. I'm outta here |
| 03:46.05 | MrChicken | nobody gave me the GXP2000 firmware |
| 03:47.13 | *** join/#asterisk fbffff (n=fbffff@r74-193-79-81.pfvlcmta01.grtntx.tl.dh.suddenlink.net) |
| 03:47.19 | ManxPower | MrChicken: The wiki url did not have a link to it |
| 03:47.54 | MrChicken | ManxPower ... It Did, but grandstream removed it from their site |
| 03:48.05 | MrChicken | as a matter of fact the only thing in GS beta site is the lang pack |
| 03:51.31 | MrChicken | so.... |
| 03:51.45 | MrChicken | am I out of luck? |
| 03:52.29 | *** join/#asterisk bmg505 (n=leon@196.209.183.106) |
| 03:52.42 | *** part/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 03:53.37 | MrChicken | guess so |
| 04:03.02 | *** join/#asterisk Sargun (n=xbmodder@atarack/staff/sargun) |
| 04:03.11 | Sargun | Anyone here got Fax -> PSTN working in asterisk? |
| 04:04.19 | *** join/#asterisk salviadud (n=pelos@189.160.61.226) |
| 04:04.42 | Sargun | Anyone got Fax working in asterisk at all? |
| 04:04.54 | dj-fu | gave up |
| 04:04.55 | dj-fu | lol |
| 04:05.55 | Sargun | Oh. |
| 04:05.59 | Sargun | That's happy news to start with |
| 04:06.13 | Sargun | Time to test my ATA |
| 04:07.12 | *** join/#asterisk ExR90 (n=exr9001@cpe-76-166-105-25.socal.res.rr.com) |
| 04:07.33 | ExR90 | I have googled for some time on this without luck. Are there any known issues with asterisk-1.4.4 and festival 1.95? I get whenever I call Festival inside the dialplan: WARNING[31901]: app_festival.c:393 festival_exec: festival_client: gethostbyname failed. I saw a reference to it in an old bug but it was closed many versions ago being fixed. My festival.conf has all options commented out. I have tried it with them uncommented as well |
| 04:08.24 | *** join/#asterisk Beirdo (n=gjhurlbu@unaffiliated/beirdo) |
| 04:11.12 | andrew` | how do I get my sip debug output routed to a file as well as the console? |
| 04:12.39 | errr | Sargun: I do, I used IAXModem and hylafax for our fax solution |
| 04:13.35 | andrew` | nvm got it |
| 04:14.05 | Sargun | errr, With an ATA, how was the fax hooked to asterisk? |
| 04:15.16 | errr | Sargun: IAXModem is like an IAX2 extension that is dialed, an incoming call comes in its passed to the IAX2 extension for the IAXModem and then hylafax takes over |
| 04:16.21 | errr | its all software based.. there is no hardware involved at all |
| 04:18.19 | andrew` | apparently not..i have a debug log file but it won't show sip debug info |
| 04:18.28 | andrew` | added in all the same entries as console has |
| 04:19.30 | *** join/#asterisk Crescendo_ (n=martinda@adsl-072-151-080-148.sip.rmo.bellsouth.net) |
| 04:19.40 | andrew` | oops i'd missed verbose |
| 04:19.41 | andrew` | there we go |
| 04:19.42 | andrew` | :) |
| 04:25.53 | *** join/#asterisk wunderkin (n=wunderki@ip68-108-204-139.ph.ph.cox.net) |
| 04:28.13 | wolferine | what manager command do I use to track phone calls, both incoming and outgoing? |
| 04:28.55 | SwK | wolferine: you dont use a manager command to track it, you take the output of the manager interface and corelate that to useable information |
| 04:31.24 | wolferine | oh |
| 04:31.29 | wolferine | so its a bit complex |
| 04:31.35 | wolferine | thanks SwK |
| 04:32.03 | SwK | yeah |
| 04:33.05 | SwK | but i'm sure if you look around you might find something that does what you want or you can also help out one of the guys that hang out here with a little bounty money for waht you arte looking for |
| 04:33.29 | ExR90 | bounty money? |
| 04:33.43 | wolferine | um i would rather have the knowledge, then to pay someone else |
| 04:33.49 | ExR90 | grin |
| 04:35.35 | SwK | hah |
| 04:36.32 | *** join/#asterisk AJaymn (n=mypocket@66-188-80-40.dhcp.mdsn.wi.charter.com) |
| 04:36.45 | SwK | wolferine, then your best bet is set it up and start watching what it does |
| 04:36.51 | AJaymn | anyone been able to use sip_notify.conf to get a PAP rebooted? |
| 04:36.58 | *** join/#asterisk ctooley (n=ctooley@209.33.108.198) |
| 04:37.18 | CCFL_Man2 | anyone here use or used a cisco 7912? |
| 04:37.34 | CCFL_Man2 | i'd like to know which firmware supports ringtones |
| 04:37.42 | wolferine | SwK, |
| 04:37.46 | wolferine | yeah |
| 04:39.40 | CCFL_Man2 | i want a half life geiger counter ringtone |
| 04:40.53 | *** join/#asterisk antlers (n=antlers@ip70-173-89-173.lv.lv.cox.net) |
| 04:44.46 | wunderkin | damn i don't see any openser 1.2.0 ipkg for openwrt.. i have to build one? :D |
| 04:50.44 | *** join/#asterisk senski (n=samllewe@60.234.20.178) |
| 05:01.15 | *** part/#asterisk senski (n=samllewe@60.234.20.178) |
| 05:02.48 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 05:12.33 | ExR90 | Is there a variable that can tell me who (based on peer/friend/user definition) who is "on the channel". I want to identify internal people for logic in my dialplan for security. I dont think callerid would be good. |
| 05:12.54 | ExR90 | the channel variable returns what I need, but also some additional information that changes |
| 05:20.47 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 05:22.05 | RyanW | checkout the setvar=FOO=BAR in sip.conf |
| 05:22.19 | ExR90 | danke |
| 05:31.26 | *** join/#asterisk mocker (n=mocker@198.247.173.227) |
| 05:32.46 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 05:33.38 | blitzrage | anyone here ever setup a static Agent via realtime using the Local channel? I can get SIP./username working, but Local/username@context/n seems to pass back invalid |
| 05:33.58 | blitzrage | for some reason I thought I'd figured this out... but of course I didn't write down any notes, heh |
| 05:34.32 | *** join/#asterisk wierdo (n=noname@digsys34-217.pip.digsys.bg) |
| 05:35.07 | blitzrage | oh wierd.. it just doesn't liek the /n |
| 05:39.19 | *** join/#asterisk mbranca (n=matteo@81.208.92.210) |
| 05:41.49 | *** join/#asterisk dlynes_ (n=dlynes@d207-216-161-56.bchsia.telus.net) |
| 05:44.37 | *** join/#asterisk RyanW (n=cableguy@ge0-0-15-lns0.207alg.qx21.net) |
| 05:44.55 | RyanW | whats the simplest way to introduce a 2 week rotation into my dial plan ? |
| 05:45.22 | RyanW | can i have 26 include lines specifying a date range ? |
| 05:48.11 | *** join/#asterisk ohadz (n=ohad@cpe-69-203-27-50.nyc.res.rr.com) |
| 05:48.38 | ohadz | what does this mean -- Really destroying SIP dialog '75a3d0e9-24e972ed@10.1.7.167' Method: ACK |
| 05:49.30 | ExR90 | I saw it when I had my debug and verbose high |
| 05:51.40 | ExR90 | If I have in a particular persons configuration in their entry in the SIP.CONF file a set=VAR=VALUE, how do I reference it inside the dialplan when a channel is up associated with that SIP context? I tried NoOp(${VAR}) but it always says empty |
| 05:52.04 | ExR90 | do I need to have set=__VAR=VALUE in the SIP.CONF for that context? |
| 05:53.29 | RyanW | set-var=MYVAR=MYVALUE |
| 05:53.37 | RyanW | then NoOp(${MYVAR}) |
| 05:54.05 | ExR90 | ah |
| 05:57.32 | mocker | Guh. |
| 05:57.52 | mocker | Compiling spandsp/fax support is still crappy. |
| 05:57.52 | mocker | ;) |
| 05:58.21 | mocker | Anyone used AsterFax? |
| 05:58.28 | *** join/#asterisk hermuli (n=Eladamri@a88-112-255-26.elisa-laajakaista.fi) |
| 06:08.22 | ExR90 | RyanW: Do you know if its possible for SIP endpoints/devices/phones can pass their own values? IE: I am assigning each SIP context a variable called zCOS. This will be used later in dialplan logic to do class of service. This wouldnt work if a phone or endpoint could just pass their own value for zCOS and override what I have defined. |
| 06:08.44 | ExR90 | ehm: values=variables |
| 06:11.03 | CCFL_Man2 | anyone use cisco 7912 phones? |
| 06:17.02 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
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| 06:46.17 | grinsbalu | CCFL_Man2: yeah, i'm doin with chan_sccp |
| 06:46.21 | grinsbalu | and they work well |
| 06:46.28 | grinsbalu | easy integration |
| 06:46.43 | grinsbalu | oh |
| 06:47.17 | grinsbalu | i read 7902 |
| 06:47.41 | grinsbalu | but looks the same |
| 06:47.51 | grinsbalu | with a little display |
| 06:48.26 | grinsbalu | 7902 isn't updateable to sip... so 7912 couldn't be as well... |
| 06:49.49 | *** join/#asterisk CBU[^_^]M`` (n=love@210.213.145.222) |
| 06:57.23 | *** join/#asterisk af_ (n=getsmart@81-174-8-57.f5.ngi.it) |
| 07:03.27 | ExR90 | My brain is fried. Can someone check my logic here. Basically I want to compare these two variables. If they match, move on. If not, jump to somewhere else. |
| 07:03.30 | ExR90 | exten -> s,n,GotoIf(["${zExten}" = "${GreetingManager1}"]?:NoPermission) |
| 07:03.54 | ExR90 | Its always coming up true and doing nothing even when I know the two are different |
| 07:04.19 | Strom_M | Gotoif($["${ |
| 07:04.27 | blitzrage | exten => s,n,GotoIf($[${zExten} = ${GreetingManager1}]?:NoPermission) |
| 07:04.47 | ExR90 | So no quotes, but a leading $ before the enclosing ['s? |
| 07:05.03 | blitzrage | quotes are superfluous, but they won't hurt anything |
| 07:05.07 | blitzrage | its the $[ you're missing |
| 07:05.17 | ExR90 | What is $['s function? |
| 07:05.24 | ExR90 | err $[ xxxx ] |
| 07:05.26 | blitzrage | $[...] means expression |
| 07:05.40 | blitzrage | comparisons, math, etc.... |
| 07:05.50 | blitzrage | $[${FOO} + ${BAR}] |
| 07:05.55 | ExR90 | Is there any case where you would NOT use that then? |
| 07:06.02 | blitzrage | $[${FOO} & ${BAR}] |
| 07:06.03 | blitzrage | etc |
| 07:06.06 | blitzrage | no |
| 07:06.09 | ExR90 | ok |
| 07:06.12 | ExR90 | Lemme plug it in. |
| 07:06.21 | blitzrage | everytime you do an expression, matching, etc..., always encase it in $[...] |
| 07:06.34 | blitzrage | While($[${X} < 10]) |
| 07:06.36 | blitzrage | for example |
| 07:06.41 | ExR90 | nod |
| 07:06.51 | blitzrage | Set(X=$[${X} + 1]) |
| 07:06.53 | blitzrage | another example |
| 07:06.58 | blitzrage | you probably get it now |
| 07:07.21 | blitzrage | and with that... its 3:07am, and I gotta get up and work again by 10am |
| 07:10.35 | *** join/#asterisk Iamnacho (i=Iamnacho@ip68-103-153-140.ks.ok.cox.net) |
| 07:10.35 | ExR90 | Still trouble, so I simplified to this and it still comes back true: |
| 07:10.35 | ExR90 | exten -> s,n,GotoIf($[201 = 201]?:NoPermission) |
| 07:10.40 | ExR90 | of course it would |
| 07:10.40 | ExR90 | slap |
| 07:11.30 | ExR90 | well changing it to this still gets me true: |
| 07:11.30 | ExR90 | exten -> s,n,GotoIf($[201 = 205]?:NoPermission) |
| 07:13.32 | ExR90 | Well, im done. off to sleep. Fresh mind will help tomorrow. |
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| 08:06.11 | *** join/#asterisk abasic (n=alen@78-0-79-21.adsl.net.t-com.hr) |
| 08:06.20 | abasic | hello |
| 08:06.49 | abasic | is anyone tryed cdr_yada |
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| 09:03.02 | swyrus | Good morning all, (at least where I am) |
| 09:03.25 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 09:03.47 | swyrus | I have having problems making or recieving calls from my sip provider |
| 09:04.24 | swyrus | asterisk keeps saying Failed to authenticate on INVITE to................ |
| 09:04.38 | swyrus | any ideas please |
| 09:09.56 | *** join/#asterisk `pariah (n=josh@unaffiliated/pariah) |
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| 09:28.24 | abasic | cdr_odbc oracle problems |
| 09:30.52 | abasic | need help |
| 09:30.57 | abasic | anyone |
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| 10:43.52 | abasic | hello |
| 10:44.13 | abasic | i need a help with cdr_odbc with oracle |
| 10:45.02 | badcfe | i cant find how i reload the h323.conf in * |
| 10:46.17 | badcfe | on the cli theres no such thing as an h323 reload |
| 10:47.05 | *** join/#asterisk Zefk (n=Zefk@wsc-fo.b.astral.ro) |
| 10:48.23 | kaldemar | badcfe: you can reload single modules with the reload command. for example "reload chan_h323.so". asterisk re-reads the configuration file then. |
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| 11:11.42 | badcfe | kaldemar: thank you. it confirms what i was 1 percent certain about |
| 11:12.02 | *** join/#asterisk bawb2 (n=bawb2@ip50210.estcmp.ku.edu) |
| 11:15.43 | Zefk | Hi. Is it possible to compile and/or encrypt the dialplan before deliver the solution to the final customer? Thx. |
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| 11:30.10 | *** part/#asterisk ctooley (n=ctooley@209.33.108.198) |
| 11:35.13 | JunK-Y | Zefk: i had the suggestion by Leif Madsen, I will take a look at this. |
| 11:39.47 | *** join/#asterisk octoban (n=miha@219.83.106.136) |
| 11:40.19 | octoban | sorry wrong button |
| 11:41.02 | *** join/#asterisk d4rkst4r75 (n=d4rkst4r@ip-41-112.sn1.eutelia.it) |
| 11:41.20 | d4rkst4r75 | hi to all |
| 11:41.29 | d4rkst4r75 | is there some sangoma a10x user? |
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| 11:51.33 | CBU[^_^]M`` | hello... can i use SPA 3102 on asteriks? |
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| 12:08.21 | LeddyHM | anyone have examples of multiple outbound providers? |
| 12:08.40 | LeddyHM | I have a working single provider setup that I'm trying to add a *XX to dial second provider |
| 12:09.40 | *** join/#asterisk Fieldy (i=SBKhAPTK@gentoo/contributor/Fieldy) |
| 12:10.56 | CBU[^_^]M`` | hello... can i use SPA 3102 on asteriks? |
| 12:12.58 | LeddyHM | I've got the *code registered but can't figure out why it's not passing digits |
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| 12:26.41 | *** join/#asterisk TJBraza (n=tj@201-34-69-62.paemt704.dsl.brasiltelecom.net.br) |
| 12:26.50 | TJBraza | Hello |
| 12:26.58 | TJBraza | can anyone help me with a DeadAGI issue? |
| 12:28.29 | TJBraza | i'm using cdr_mysql, and executing a DeadAGI script on the hangup extension, but the cdr isn't written out yet at this point |
| 12:28.40 | TJBraza | and i need access to the cdr after the call is hungup |
| 12:29.03 | mosty | TJBraza: does mysql have triggers? |
| 12:29.33 | TJBraza | no.. actually.. nevermind.. i just found this http://bugs.digium.com/view.php?id=6193 |
| 12:29.47 | TJBraza | after about 5 hours of mocking around |
| 12:32.06 | *** join/#asterisk threat (i=threat@60-240-43-214.static.tpgi.com.au) |
| 12:32.32 | *** join/#asterisk dhakatel (n=root@58.65.224.5) |
| 12:33.04 | dhakatel | is there any soft phone in linux suppot g729. |
| 12:33.06 | dhakatel | ? |
| 12:33.34 | dhakatel | or is Xten is abaliable for linux? |
| 12:37.33 | mosty | dhakatel: not that i know of, mostly because g729 is patented |
| 12:37.43 | mosty | besides asterisk |
| 12:38.28 | dhakatel | I know |
| 12:39.07 | *** join/#asterisk s0lid (n=jlq@210.213.243.34) |
| 12:39.51 | dhakatel | does g729 codes abaliable in ITU work in linux? |
| 12:39.58 | tzanger | has anyone noticed that in 1.4 the main voicemail menu doesn't seem to be interruptible? i.e the first VoiceMailMain announcements "you have x new and y old, press 1 to hear new, 2 to change folders, etc." ? |
| 12:40.17 | mosty | dhakatel: i don't understand your question |
| 12:40.55 | dhakatel | I can download source code of g729 from ITU. |
| 12:41.24 | dhakatel | but I don't know if it work in linux properly |
| 12:42.18 | *** join/#asterisk TJBraza (n=tj@201-34-69-62.paemt704.dsl.brasiltelecom.net.br) |
| 12:42.37 | mosty | dhakatel: you can buy g729 codecs from digium |
| 12:43.18 | dhakatel | good idea |
| 12:43.41 | mosty | dhakatel: if this is for business, just do that |
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| 12:57.58 | *** join/#asterisk Lithium_Ion (n=lithium@d57-71-44.home.cgocable.net) |
| 12:59.51 | Lithium_Ion | Hi all, I'm having trouble getting incoming calls on my trixbox installation. I trying to determine if it's my provider. I noticed the sip peer isn't registering, could this be part of the problem? |
| 13:00.14 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 13:00.34 | LeddyHM | ANy idea why one of our phones is displaying this every few seconds in the cli: Incoming call: Got SIP response 500 "Internal Server Error" |
| 13:00.45 | mosty | if you're not registered to a server, you cant receive calls from them unless they have you setup as a static ip |
| 13:02.07 | Lithium_Ion | I figured but I've been going back and forth with my voip provider (zingotel) and they had me change the password a couple times but it's still not registering |
| 13:03.15 | mosty | you should probably ask in #trixbox how to get logs |
| 13:03.31 | LeddyHM | #asterisk != #trixbox |
| 13:03.34 | Lithium_Ion | thx |
| 13:04.23 | LeddyHM | sweet, got my dual provider setup |
| 13:04.31 | LeddyHM | I feel like smart and stuff |
| 13:05.19 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 13:09.01 | Lithium_Ion | What can i do to resolve jitter issues? |
| 13:09.24 | tzanger | Lithium_Ion: better network. :-) |
| 13:10.04 | Lithium_Ion | lol yeah that always helps |
| 13:12.33 | *** join/#asterisk skyphyr (n=alanj@135.196.58.222) |
| 13:12.42 | *** join/#asterisk Ifaistos (n=stelios@ipa226.211.tellas.gr) |
| 13:13.02 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:13.39 | Lithium_Ion | Anyone know of a good provider that supports asterisk? |
| 13:13.47 | *** join/#asterisk af_ (n=getsmart@81-174-8-57.f5.ngi.it) |
| 13:14.09 | skyphyr | hi all - anyone have experience with iax2 hardphones? only found 3 options and only one had a review (which wasn't good) |
| 13:15.17 | mosty | skyphyr: the general experience is that none of them are any good |
| 13:15.58 | skyphyr | that's a shame - so running sip is the best option for now? |
| 13:16.45 | skyphyr | this is only for a small office - only need 4-6 of them |
| 13:16.45 | skyphyr | whether that makes any of the drawbacks of them negligible |
| 13:17.09 | mosty | sip is fine on a lan |
| 13:17.19 | mosty | sip only has problems crossing nat |
| 13:17.48 | skyphyr | ahh cool - so seeing I'll be running an asterisk server I just set that up to hook up to a voip provider with iax and run sip on the network internally? |
| 13:18.24 | skyphyr | any particular sip phones provide a good balance of quality and price? |
| 13:18.30 | mosty | sip from the phones to the asterisk box, then sip or iax to your upstream provider (whichever they support) |
| 13:18.30 | LeddyHM | Polycom 501 |
| 13:18.41 | mosty | skyphyr: everybody loves polycom phones |
| 13:18.51 | skyphyr | thanks mosty and LeddyHM :-) |
| 13:18.56 | Lithium_Ion | Could my registration problem be because it has to traverse a nat? |
| 13:18.59 | skyphyr | you've been enourmously helpful |
| 13:19.03 | *** join/#asterisk Putzz (n=me@CPE00155824933c-CM00111ae07ac2.cpe.net.cable.rogers.com) |
| 13:20.24 | skyphyr | I notice the 501 is a £55 hit on top of the 301 - is there a big difference in the two? |
| 13:21.16 | mosty | Lithium_Ion: unlikely |
| 13:21.46 | Lithium_Ion | Yeah I figured. Because I can call out fine |
| 13:22.53 | mosty | Lithium_Ion: you need to find out where trixbox keeps it logs, and turn on debug and verbose logging, then force a re-register attempt and see what is happening |
| 13:23.40 | LeddyHM | huge difference |
| 13:23.44 | LeddyHM | 501 > 301 |
| 13:25.09 | Lithium_Ion | how do I force a reregister? |
| 13:25.51 | mosty | iax2 reload. or sip reload |
| 13:28.31 | skyphyr | thanks again LeddyHM |
| 13:28.46 | Lithium_Ion | Says SIP/2.0 423 Interval Too Brief |
| 13:30.39 | LeddyHM | www.vendor.com |
| 13:30.48 | LeddyHM | support@vendor.com :) |
| 13:31.13 | Lithium_Ion | I wish, they take forever to respond |
| 13:31.37 | LeddyHM | new provider? |
| 13:31.44 | *** join/#asterisk oej (n=olle@213.115.215.55) |
| 13:31.47 | LeddyHM | ours sucks |
| 13:31.56 | LeddyHM | can't even get DTMF working reliably |
| 13:32.00 | Lithium_Ion | Know any good ones? :P |
| 13:32.11 | Lithium_Ion | Guess that's a no |
| 13:32.15 | *** join/#asterisk jer (n=jtregunn@unaffiliated/jer) |
| 13:32.22 | LeddyHM | voip-info.org has a good list |
| 13:32.26 | LeddyHM | based on location |
| 13:33.38 | *** join/#asterisk toot (n=toot@84.19.255.123) |
| 13:34.08 | toot | hmm. silly question but TE207P + PCIExpress? that work ( i didn't think it did but client has bought server anyway :() |
| 13:34.25 | Lithium_Ion | thanks |
| 13:35.33 | Lithium_Ion | leddy: Thanks |
| 13:36.16 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 13:36.16 | *** mode/#asterisk [+o anthm] by ChanServ |
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| 13:40.50 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 13:40.52 | d4rkst4r75 | hi to all |
| 13:41.07 | d4rkst4r75 | someone with sangoma a10x boards online? |
| 13:41.21 | *** join/#asterisk nasls_lsa (n=chatzill@87.203.223.194) |
| 13:42.03 | *** join/#asterisk anonymouz666 (n=anonymou@re366.compuland.com.br) |
| 13:42.51 | *** join/#asterisk Inkubot (n=fgutierr@200.75.4.10) |
| 13:43.18 | Inkubot | hi |
| 13:43.55 | *** join/#asterisk MrWup (i=root@80-46-38-124.static.dsl.as9105.com) |
| 13:44.21 | Inkubot | i have a problem with libss7. I can't load the module... when i'm trying to load the module show me this: Module 'chan_ss7.so' does not provide a description. |
| 13:44.28 | MrWup | if i use an external app to connect to the asterisk manager interface, is it possible to tell asterisk to set up a call between say number x and number y |
| 13:44.29 | Inkubot | i have a te110p card |
| 13:44.37 | *** join/#asterisk oej (n=olle@213.115.215.55) |
| 13:44.41 | MrWup | so that the external app can have like a list of phone numbers in a directory |
| 13:44.45 | MrWup | and u can right click a number |
| 13:45.05 | MrWup | and your phone will ring, and when you pick up your phone, you're connected to the other person u right clicked? |
| 13:46.34 | [TK]D-Fender | MrWup: Yes : "Originate" |
| 13:46.42 | MrWup | oooh |
| 13:47.06 | Inkubot | http://pastebin.ca/478961 here is my zaptel.conf and zapata.conf |
| 13:47.26 | MrWup | thing is |
| 13:47.30 | MrWup | originate asks for a channel |
| 13:47.36 | MrWup | how do i know which channel? |
| 13:47.42 | MrWup | cause i have SIP phones with 9 channels |
| 13:47.51 | MrWup | cant it just automatically pick the next free channel? |
| 13:47.54 | MrWup | like if i did a dial? |
| 13:48.53 | MrWup | ohhhhhh |
| 13:48.55 | MrWup | nevermind i get it |
| 13:49.04 | *** join/#asterisk SuPrSluG (n=SuPrSluG@pool-71-241-171-123.buff.east.verizon.net) |
| 13:49.04 | MrWup | i originate between say a zap channel and an extension in the dialplan |
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| 14:00.13 | *** join/#asterisk manopulus (n=manopulu@213.197.181.154) |
| 14:01.42 | manopulus | hello, if I do regexten in sip.conf, how I can know, what is peer number or name, when I dialing. i.e. [111222] regexten=222333, incoming call come to 222333 and I have to lookup and dial to 111222. I want to manage DID number, assigned to sip account this way. |
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| 14:10.03 | *** mode/#asterisk [+o mog] by ChanServ |
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| 14:16.45 | tzanger | meh; unlimitel says that their gateway giving ringback before TDM has given ringback is "normal" |
| 14:16.58 | tzanger | "ringbeep beep beep beep" is not normal iMO |
| 14:18.48 | *** join/#asterisk Here_And_There (n=Here_And@74.92.105.41) |
| 14:18.52 | *** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust826.leed.cable.ntl.com) |
| 14:21.56 | *** join/#asterisk bluelinq (n=bluelinq@dsl-7-36.cofs.net) |
| 14:22.25 | bluelinq | Hi, I have an ivr that is not timemint out, as soon as the announcement is done it just hangs up |
| 14:24.09 | Corydon76-home | Then you're not running WaitExten after the announcement |
| 14:24.16 | bluelinq | how do I check that? |
| 14:25.11 | Corydon76-home | extensions.conf |
| 14:25.45 | Corydon76-home | Sounds like a 1.0 dialplan. |
| 14:26.11 | Corydon76-home | We no longer implicitly wait for an extension. You have to explicitly specify that now |
| 14:26.23 | Corydon76-home | (if that's what you want) |
| 14:26.25 | anonymouz666 | if it wasn't for chan_unicall I'd jump to 1.4 forever |
| 14:27.14 | bluelinq | Corydon I have nothing with the word waitexten in any of the extension files |
| 14:27.29 | Corydon76-home | bluelinq: sounds like you found your problem |
| 14:28.06 | Corydon76-home | Alternatively, you can set autofallthrough=no in [general] |
| 14:28.57 | bluelinq | no general either... |
| 14:30.53 | *** join/#asterisk SwK (n=SwK@65.192.110.34) |
| 14:31.07 | *** join/#asterisk joshaidan (n=brianj@thunderbay-voip-4.vianet.ca) |
| 14:31.18 | tzanger | hmm |
| 14:31.29 | tzanger | I was certain that Asterisk supported SIP early audio |
| 14:31.37 | tzanger | can someone more knowledgable than I comment on that? |
| 14:31.48 | tzanger | from Unlimitel: "My customers who are using early audio SIP trunking doesn't seem to have this |
| 14:31.51 | tzanger | problem. Can you check if your IP-PBX can do early audio streaming?" |
| 14:32.26 | Lithium_Ion | When I register with my provider it says interval too brief and says min-expires:600 but I have defaultexpirey=600 in my sip.conf. Any ideas? |
| 14:32.53 | *** join/#asterisk snook3r (n=ariel@bzq-219-46-202.isdn.bezeqint.net) |
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| 14:37.11 | Lithium_Ion | The the registration is sending out 120 why? |
| 14:37.43 | Lithium_Ion | Expires: 120 |
| 14:39.35 | *** join/#asterisk syzygyBSD (n=chatzill@66.226.228.204.cpe.speedyquick.net) |
| 14:40.20 | *** join/#asterisk NoCarrier (n=NoCarrie@unaffiliated/badpacket) |
| 14:42.05 | [TK]D-Fender | Corydon-w: WaitExten.... the best app we NEVER NEEDED |
| 14:42.41 | syzygyBSD | [TK]D-Fender: are you saying you have never used it? |
| 14:42.52 | bluelinq | Corydon, where do you put the autofallthrough? |
| 14:42.58 | [TK]D-Fender | syzygyBSD: I did once, but never needed it. |
| 14:43.05 | [TK]D-Fender | bluelinq: [general] |
| 14:43.14 | bluelinq | in which file? |
| 14:43.20 | [TK]D-Fender | bluelinq: extensions.conf |
| 14:43.27 | syzygyBSD | thats like saying you used your penis once but never NEEDED it |
| 14:43.40 | syzygyBSD | I'm still glad it is there |
| 14:43.53 | bluelinq | there is no [general] |
| 14:44.01 | *** join/#asterisk rmayorga (i=rmyorg@unaffiliated/rmayorga) |
| 14:44.17 | [TK]D-Fender | syzygyBSD: WaitExten does precious little that standard "s" + Autofallthrough (also only came into existance BECAUSE of WaitExten) didn't already, nor still does. |
| 14:44.25 | [TK]D-Fender | bluelinq: Go make one then! |
| 14:44.34 | bluelinq | ok ok.. be that way... |
| 14:46.14 | bluelinq | seems to be working... |
| 14:46.37 | *** join/#asterisk jm|work (n=jm@sentry.flags.co.uk) |
| 14:47.26 | bluelinq | ok thanks, working good. Wonder why it wasn't there |
| 14:47.41 | [TK]D-Fender | bluelinq: Because the dialplan is 100% YOUR job. |
| 14:48.15 | [TK]D-Fender | bluelinq: Capabilities are standard, implementation is not. |
| 14:49.07 | *** join/#asterisk tcseke (n=chatzill@22-36.adsl.etel.hu) |
| 14:49.39 | syzygyBSD | :) |
| 14:49.50 | bluelinq | agree, but I have never seen that in any of the documentation for trixbox, freepbx or asterisk... |
| 14:50.10 | [TK]D-Fender | bluelinq: FreePBX?! Don't make me laugh. |
| 14:50.32 | bluelinq | agree again, but is better than anything else out there |
| 14:50.38 | *** join/#asterisk sigmounte (n=sigmount@www.sighq.net) |
| 14:50.47 | [TK]D-Fender | bluelinq: You aren't expected to ever TOUCH the configs. It s CANNED system that turns a PC in a telephony TOASTER |
| 14:50.55 | sigmounte | hello ! anyone with a grandstream gxp 2000 here ? |
| 14:50.59 | [TK]D-Fender | bluelinq: Better is a matter of perspective. |
| 14:51.20 | syzygyBSD | poor grandstream... |
| 14:51.40 | bluelinq | yeah, but for the ave person it can do most of what is needed. I hate their login in the ivr, but I guess is a matter of getting used to |
| 14:51.41 | sigmounte | sysko, poor why ? |
| 14:52.27 | [TK]D-Fender | ~gs |
| 14:52.29 | jbot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
| 14:52.31 | [TK]D-Fender | ^^^^^ |
| 14:52.53 | sigmounte | yes ok , but too late for me , so what's the best choice ? |
| 14:53.00 | [TK]D-Fender | bluelinq: Average people don't NEED a PBX. |
| 14:53.00 | syzygyBSD | lol, who added that to the bot |
| 14:53.03 | bluelinq | cheap phones, cheap quality.. like FreePBX...lol |
| 14:53.06 | [TK]D-Fender | :D |
| 14:53.21 | bluelinq | ok, a bit higher that average then.. |
| 14:53.43 | [TK]D-Fender | bluelinq: Sure, but don't get any dreams of support of flexibility |
| 14:55.01 | bluelinq | nahh... flexibility is my middle name.. just need to ask nice people like you that really know and make us look like a bunch of idiots, thats all |
| 14:55.29 | [TK]D-Fender | heh |
| 14:55.40 | [TK]D-Fender | nah, I wasn't condescending at all back there! |
| 14:56.11 | [TK]D-Fender | bluelinq: If I really rip into you, the whole world will know it ;) |
| 14:56.18 | bluelinq | not taken that way... I meant it really in the good way... |
| 14:56.52 | bluelinq | anyway, thanks that did the trick |
| 14:57.20 | [TK]D-Fender | bluelinq: np... well on your way to being free of GUI's |
| 14:57.54 | [TK]D-Fender | bluelinq: If you don't need to make changes all the time and give that power to idiots you'll appreciate the control more... |
| 14:57.56 | *** join/#asterisk hfb (n=hfb@pool-72-67-156-130.lsanca.dsl-w.verizon.net) |
| 14:58.26 | bluelinq | the gui does help a bit, not polished and not 100% true, but does some good |
| 15:00.00 | [TK]D-Fender | bluelinq: Getting you up and running sure, but its a LOT to undo. |
| 15:00.11 | [TK]D-Fender | You'll effectively be having to start frmo scratch |
| 15:02.23 | *** join/#asterisk Dr-Linux (n=Nothing@202.125.139.198) |
| 15:02.35 | Dr-Linux | anybody is using web-meetme or Cepstral? |
| 15:02.45 | *** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net) |
| 15:04.03 | sigmounte | Anyone with a grandstream gxp 2000 here who can help me ? |
| 15:04.05 | shido6 | firmware? |
| 15:04.18 | [TK]D-Fender | sigmounte: .... |
| 15:04.19 | [TK]D-Fender | ~ask |
| 15:04.21 | jbot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there, just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 15:04.44 | bluelinq | TK as long as there was a how-to in doing so yes. I have read some documentation and I have never seen that, but this is also my first real install |
| 15:05.39 | sigmounte | [TK]D-Fender, my grand stream always send a busy/congested paquet to my asterisk after 60 seconds , and i'm unable to find where to change this value |
| 15:06.11 | *** join/#asterisk _VoiceMeUp_COM (n=_VoiceMe@145-27.mc.cite.net) |
| 15:06.37 | _VoiceMeUp_COM | anyoen have this prob ?asterisk seems to drop a call when it |
| 15:06.37 | _VoiceMeUp_COM | tries to preregister (user a) again and does not maintain the previous registration. |
| 15:06.47 | bluelinq | TK what elso normally goes into the general section? |
| 15:07.04 | _VoiceMeUp_COM | EX: user -> us -> out if user re-register while on call after 15 min.. then call drops |
| 15:08.36 | shido6 | wow |
| 15:09.01 | shido6 | im starting to not like my 942 |
| 15:09.21 | shido6 | it randomly decides... I'm going to reboot when you press my buttotns. |
| 15:13.41 | *** join/#asterisk nomadsoul (n=nomadsou@unaffiliated/nomadsoul) |
| 15:15.17 | coppice | the 942 looks very fake |
| 15:17.14 | mosty | shido6: faulty unit? |
| 15:17.14 | [TK]D-Fender | ~phones |
| 15:17.17 | jbot | i guess phones is http://bani.anime.net/phones/. SIP Hardphones in order of quality/suggestability : Polycom (any), Aastra 480i, Cisco 7940(+), Linksys SPA-9XX, Snom |
| 15:17.17 | shido6 | maybe... |
| 15:17.17 | *** join/#asterisk NoCarrier (n=NoCarrie@unaffiliated/badpacket) |
| 15:17.17 | shido6 | i use the phone everyday for hours at a time |
| 15:17.18 | *** join/#asterisk Inkubot (n=fgutierr@200.75.4.10) |
| 15:17.18 | shido6 | it reboots when i push buttons maybe once every 8 days |
| 15:17.51 | shido6 | wish i handt sold the 7960s |
| 15:18.00 | shido6 | hadn't |
| 15:18.01 | _VoiceMeUp_COM | 7960's are loya |
| 15:18.02 | _VoiceMeUp_COM | l |
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| 15:32.03 | Runlvl | It was OK before you touched it |
| 15:32.06 | Runlvl | sounds like a Windows problem, try calling Microsoft support |
| 15:32.21 | shido6 | :) |
| 15:32.21 | nomadsoul | Runlvl: lol |
| 15:32.25 | Runlvl | jaja |
| 15:33.03 | neverblue | my Dad did that once |
| 15:33.06 | neverblue | oh, I laughed |
| 15:33.31 | neverblue | his cracked Nero told him there was an error, and for some reason he decided to call M$ for support |
| 15:33.59 | Runlvl | neverblue: jajaj |
| 15:35.19 | [TK]D-Fender | Polycom > All |
| 15:35.26 | [TK]D-Fender | Nero.... feh |
| 15:35.32 | [TK]D-Fender | CDBurnerXP Pro |
| 15:35.39 | [TK]D-Fender | Freeware/OSS! |
| 15:35.48 | Runlvl | K3b. |
| 15:35.49 | dlynes_laptop | bah growisofs > all |
| 15:35.54 | *** join/#asterisk NoCarrier (n=NoCarrie@unaffiliated/badpacket) |
| 15:35.56 | [TK]D-Fender | The only paid item I validate for people is the OS itself |
| 15:36.03 | *** part/#asterisk NoCarrier (n=NoCarrie@unaffiliated/badpacket) |
| 15:36.05 | *** join/#asterisk ploieel (n=ploieel@Fb2c8.f.ppp-pool.de) |
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| 15:42.40 | *** join/#asterisk jart (n=user@ool-43509aa5.dyn.optonline.net) |
| 15:42.54 | jart | is anyone having an issue with call-limit limiting when there are no calls? |
| 15:43.15 | *** join/#asterisk darkmug (n=Dennis@143.106.167.234) |
| 15:43.31 | Strom_M | that question sounds unnervingly like 'how much wood could a woodchuck chuck if a woodchuck could chuck wood?' |
| 15:45.18 | jart | asterisk is rejecting calls to some of my sip phones due to call limit when the phones are free |
| 15:46.49 | *** join/#asterisk stimpie (n=michiel@ip565faf27.direct-adsl.nl) |
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| 15:55.02 | irule | anybody have an example on VMAuthenticate? thanks |
| 15:55.24 | [TK]D-Fender | irule: "show application vmauthenticate" |
| 15:55.44 | [TK]D-Fender | irule: the syntax is clearly written, and how/how it will return |
| 15:56.08 | irule | should I type that in asterisk CLI or something? |
| 15:57.12 | irule | cool thanks |
| 15:59.24 | *** part/#asterisk jart (n=user@ool-43509aa5.dyn.optonline.net) |
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| 16:00.43 | redax | hi |
| 16:01.44 | redax | is it possible to implement a special kind of Dial() where specified targets should be ringed using ALERT_INFO=some while otheres with ALERT_INFO=bar |
| 16:01.45 | redax | ? |
| 16:02.04 | Strom_M | yeah, use the local channel |
| 16:02.19 | redax | hm? |
| 16:02.26 | redax | Dial(Local/someext) ? |
| 16:02.38 | Strom_M | Dial(Local/exten@context) |
| 16:02.41 | ghento | hi everyone. using the Record() function , if i want to save to a .wav file, is it saved as 8khz, 8-bit or 16-bit? |
| 16:02.48 | Strom_M | 16 bit |
| 16:02.52 | Strom_M | it'll sound better |
| 16:02.54 | redax | ah! :) |
| 16:03.32 | redax | Strom_M: thanks, great! |
| 16:03.56 | ghento | Strom_M: where do I specify this? |
| 16:04.06 | *** join/#asterisk tsurko (n=tsurko@77.70.24.142) |
| 16:04.08 | Strom_M | ghento: oh wait, I misread your question |
| 16:04.21 | Strom_M | if you're recording it yourself, save it as 16 bit |
| 16:04.25 | ghento | ok |
| 16:04.41 | Strom_M | if you record() to wav, I think it automatically saves as 16 bit |
| 16:04.52 | bkruse | Strom_M: i believe so |
| 16:04.52 | ghento | great, thanks |
| 16:04.56 | bkruse | format_wav tells me so :D |
| 16:05.03 | Strom_M | woot |
| 16:05.09 | Strom_M | i win, then. |
| 16:05.41 | bkruse | yep, i think so |
| 16:06.50 | *** join/#asterisk deeperror (n=deeperro@mail.banctel.com) |
| 16:06.56 | *** join/#asterisk coolbeans (n=null@adsl-074-247-038-249.sip.bhm.bellsouth.net) |
| 16:07.30 | deeperror | I'm having an issue with no ring back after upgrading from 1.2.8 - .18 any clues? |
| 16:07.41 | coolbeans | Hi all. I understand that IP calls are initiated with SIP and then the bulk of the voice data is sent via RTP. When managing firewall QoS, is giving SIP high priority good enough or does something have to be done with RTP as well? |
| 16:08.01 | deeperror | RTP is the most important |
| 16:08.23 | deeperror | what are you using for qos? |
| 16:09.04 | coolbeans | A Fortigate. |
| 16:09.24 | coolbeans | I'm just wondering if SIP initiated connections via RTP will follow the traffic rules. |
| 16:09.36 | deeperror | not 100% sure |
| 16:10.39 | Strom_M | RTP has to be given the priority as well; to the router they appear separate |
| 16:10.55 | Strom_M | they're on separate port numbers, have separate sequence numbering, etc etc |
| 16:11.52 | coolbeans | Right, but the fortigate has a built-in service called "SIP" but no built-in service called RTP. Hrm.. guess I need to call them to make sure.. |
| 16:12.14 | Strom_M | maaaaaaaaaaaybe it's smart enough to read the SIP headers |
| 16:12.22 | deeperror | would routing by IP be a more effective way of handling qos? all traffic on x.x.x.1 is voice and takes high pri over x.x.x.2 |
| 16:12.35 | coolbeans | Ahh, just found this: Currently, FortiOS v3.00 has a session-helper in the kernel that parses SIP messages encapsulated in UDP and |
| 16:12.35 | coolbeans | opens the necessary pinholes in the firewall and performs NAT, where required, to allow RTP to flow through the FortiGate. |
| 16:12.51 | coolbeans | Now I wonder if the shaping rules apply... |
| 16:19.53 | *** join/#asterisk ircrly (i=astrutt@punk.valuetel.net) |
| 16:22.30 | *** join/#asterisk bhrobinson (n=brobinso@northtx1-static.telwestonline.com) |
| 16:22.49 | bhrobinson | hi... has anyone tried the artdio phones? |
| 16:26.43 | *** join/#asterisk FuriousGeorge (n=brian@ool-43536ea8.dyn.optonline.net) |
| 16:27.09 | FuriousGeorge | hey all |
| 16:27.21 | *** join/#asterisk DocHolliday (i=RgRabbit@gateway/gpg-tor/key-0x0E4F6D6C) |
| 16:27.38 | DocHolliday | docelmo? |
| 16:28.08 | FuriousGeorge | i was just wondering, how crucial (no pun intended) to stability is ecc memory? ive never heard anyone recommend it specifically, in the context of an * server |
| 16:28.30 | FuriousGeorge | is it just a given? |
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| 16:38.29 | [TK]D-Fender | bhrobinson: looks like cheap junk... write them off like GS |
| 16:38.50 | [TK]D-Fender | FuriousGeorge: Pretty much |
| 16:39.05 | bhrobinson | lol... I was just trying to get a rise out of anyone :) |
| 16:39.33 | coolbeans | Ok, the new Polycom HD 550 is cool. |
| 16:39.39 | coolbeans | We just got a bunch of them in. |
| 16:39.50 | Grandstream4Ever | OMG GRANDSTREAM !!! |
| 16:41.01 | *** join/#asterisk anonymouz666 (n=anonymou@re366.compuland.com.br) |
| 16:41.40 | BSD_Tech | Call Grandstream to get the latest firm ware |
| 16:41.41 | [TK]D-Fender | coolbeans: 550 is a waste..... good phone, but jsut not worth the $ |
| 16:41.47 | BSD_Tech | and they work great |
| 16:41.58 | BSD_Tech | 550? |
| 16:42.04 | JT | *for a grandstream |
| 16:42.09 | BSD_Tech | what 550 ? |
| 16:42.12 | JT | polycom |
| 16:42.19 | BSD_Tech | you mean the 650 ? |
| 16:42.21 | JT | you know, the brand we always recommend ;) |
| 16:42.24 | JT | polycom |
| 16:42.25 | JT | no |
| 16:42.26 | JT | 550 |
| 16:43.49 | BSD_Tech | POlycom |
| 16:43.59 | JT | ? |
| 16:44.01 | BSD_Tech | POLYCOM the only brand I live for |
| 16:44.07 | JT | heh |
| 16:45.01 | BSD_Tech | but I just dod a 25 user setup with grandstream gxp2000 and got the latest firmware from GS and they work great |
| 16:45.03 | [TK]D-Fender | IP 550 costs the same as the 601 without exapansion or 6 line-keys. I'm not paying that for a friggen backlight... |
| 16:45.10 | BSD_Tech | its what the client wanted |
| 16:45.24 | BSD_Tech | yeah |
| 16:45.31 | [TK]D-Fender | Clients need to be told what they want before being stuck with what they won't |
| 16:46.41 | *** join/#asterisk mbranca (n=matteo@81.208.92.210) [NETSPLIT VICTIM] |
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| 16:49.28 | ZaVoid | sup all |
| 16:49.53 | ZaVoid | anyone know any one looking for Tier 1 VoIP NOC work? We are looking for 1-2 people who know asterisk/sip and voip |
| 16:51.30 | MrWup | how do i play dialtone to someone when they enter my context? |
| 16:51.36 | MrWup | for example, they dial 9 for an outside line, then i want to play them some dialtone |
| 16:52.06 | [TK]D-Fender | MrWup: "show application disa" |
| 16:52.14 | [TK]D-Fender | MrWup: And prepare for CDR Hell |
| 16:52.34 | tzanger | you can also use ReadExten I think |
| 16:52.36 | MrWup | i do all my own databasing =] |
| 16:52.44 | tzanger | CDRs are hell any way you look at them |
| 16:53.01 | MrWup | ive found it quite easy actually... by not using the asterisk cdr thing |
| 16:53.17 | MrWup | i use my own mysql database which is updated with times, transfers etc based on part dial plan part manager interface |
| 16:53.25 | MrWup | thanks fender btw |
| 16:54.22 | MrWup | oh no |
| 16:54.25 | MrWup | i dont want disa etc |
| 16:54.30 | MrWup | i just want to play the dialtone sound |
| 16:54.38 | MrWup | i do all my own processing via php you see |
| 16:54.40 | BSD_Tech | disa is for the gods |
| 16:54.48 | *** join/#asterisk `pariah (n=josh@unaffiliated/pariah) |
| 16:54.51 | BSD_Tech | loly end users should not have it |
| 16:55.43 | *** join/#asterisk mbranca (n=matteo@81.208.92.210) [NETSPLIT VICTIM] |
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| 16:57.34 | MrWup | ahh |
| 16:57.37 | MrWup | playtones(dial) |
| 17:03.08 | *** join/#asterisk gammah (n=gammah@70.253.197.130) |
| 17:06.21 | Lithium_Ion | How can I tell if asterisk is getting hit by incoming calls? |
| 17:06.36 | _mm_ | check your log files? |
| 17:07.21 | Lithium_Ion | yeah i checked it but Im not exactly sure what I should be seeing |
| 17:07.39 | *** join/#asterisk fysa (i=fysa@653414hfc175.tampabay.res.rr.com) |
| 17:08.33 | _mm_ | well, it depends on your config, but you should see some execute statements as it goes your the config |
| 17:08.40 | _mm_ | is it a busy server? |
| 17:08.53 | Lithium_Ion | no |
| 17:09.17 | _mm_ | can you call it and watch the logs so you know its an incoming call so you can then know what you are looking for? |
| 17:09.19 | Lithium_Ion | i'm just having a bitch of a time convincing my provider the problem is on their end not mine |
| 17:09.29 | _mm_ | ah |
| 17:10.21 | Lithium_Ion | i tried that and nothing happens in the log when I call in |
| 17:10.42 | _mm_ | do you have an extension setup to simulate a call coming from outside? |
| 17:11.22 | Lithium_Ion | yes |
| 17:11.22 | fysa | We have a call center that we're looking to eventually migrate to an Asterisk-based system, but it's PSTN currently and using a big ugly Access database to hold leads. What are the better call center systems out there with Asterisk support? Any GPL? |
| 17:11.49 | _mm_ | lith: so when you dial that internal extension nothing happens in the logs? does the system answer? |
| 17:12.14 | *** join/#asterisk Fieldy (i=BM94y19g@gentoo/contributor/Fieldy) |
| 17:13.02 | _VoiceMeUp_COM | i think we ahve the lowest plan avail now ;) |
| 17:13.28 | *** part/#asterisk deeperror (n=deeperro@mail.banctel.com) |
| 17:14.19 | Lithium_Ion | when i dial 7777 everything works fine |
| 17:14.36 | _mm_ | do you have PRIs? |
| 17:16.29 | Lithium_Ion | it's a simple setup to one voip provider with 4 internal extentions, outgoing works but incoming doesnt |
| 17:16.30 | Lithium_Ion | no pris |
| 17:16.30 | Lithium_Ion | i think |
| 17:16.30 | _mm_ | oh its a voip provider |
| 17:16.30 | *** join/#asterisk toerkeium (n=oo@201.216.206.221) |
| 17:16.30 | *** join/#asterisk Cresl1n (i=matt@nat/digium/x-3215053e3d73246b) |
| 17:16.30 | *** mode/#asterisk [+o Cresl1n] by ChanServ |
| 17:16.30 | Lithium_Ion | ya |
| 17:16.49 | _mm_ | and presumably you're registered properly since outgoing calls work |
| 17:16.54 | Lithium_Ion | yes |
| 17:17.01 | Lithium_Ion | everything is registered |
| 17:17.22 | *** join/#asterisk mbranca (n=matteo@host-210-mi.linuxserver.it) [NETSPLIT VICTIM] |
| 17:18.29 | _mm_ | turn on debugging ? |
| 17:20.26 | Lithium_Ion | Yeah nothing is coming in, thanks for your help. |
| 17:20.37 | *** join/#asterisk grantm (n=grantm@kolob.wingateservices.com) |
| 17:21.37 | _mm_ | sorry i wasn't much more help though |
| 17:21.37 | _mm_ | heh |
| 17:22.01 | Lithium_Ion | I'm ripping out my hair with these guys |
| 17:22.44 | _mm_ | whos the provider |
| 17:22.50 | Lithium_Ion | zingotel |
| 17:24.31 | *** join/#asterisk rmayorga (i=rmyorg@unaffiliated/rmayorga) |
| 17:25.21 | *** join/#asterisk EmleyMoor (i=phil@topdeck.tinsleyviaduct.com) |
| 17:25.37 | *** join/#asterisk ringhals (i=fwuser@firewall.drgutah.com) |
| 17:25.52 | *** join/#asterisk sivana (n=sivana@gromit.mixdown.ca) |
| 17:26.05 | [TK]D-Fender | yup... thats a 1st tier looking name... |
| 17:26.11 | Lithium_Ion | zingotel's support is garbage |
| 17:26.19 | EmleyMoor | What IP phones are there on the market that log in to three accounts at the same time? |
| 17:27.39 | EmleyMoor | Failing that, what IP phones are there that can be made to ring distinctively depending on the extension in my dialplan through which they are being called? |
| 17:27.57 | Corydon76-home | Well, Polycom will log in to multiple accounts |
| 17:27.58 | HarryR | EmleyMoor, I'm using an old Snom 200 which can do both (something like 8 accounts and 10 different ringtones) |
| 17:28.32 | EmleyMoor | It seems to always come down to Polycom or Snom |
| 17:28.49 | HarryR | yup :) |
| 17:28.51 | MrWup | i use aastra 9133i |
| 17:28.55 | MrWup | good phone |
| 17:29.14 | *** join/#asterisk SwK (n=SwK@65.192.110.34) |
| 17:29.16 | EmleyMoor | MrWup: Can that log in to three accounts at once? |
| 17:29.28 | MrWup | erm |
| 17:29.52 | _VoiceMeUp_COM | yes |
| 17:29.53 | _VoiceMeUp_COM | lol |
| 17:30.13 | MrWup | yeah |
| 17:30.23 | MrWup | just checked on my admin panel =] |
| 17:30.29 | EmleyMoor | Excellent... it seems that many will :-) |
| 17:30.33 | MrWup | can log into 9 accounts |
| 17:31.06 | neverblue | I want to pull out info (about incoming and outgoing calls from my phone only), using php and the manager, what manger function would I use to view the data? |
| 17:31.39 | MrWup | status |
| 17:31.50 | *** join/#asterisk Skaag (n=skaag@l85-130-153-179.broadband.actcom.net.il) |
| 17:32.15 | neverblue | thanks MrWup |
| 17:33.09 | *** join/#asterisk Skaag (n=skaag@dip5-fw.corp.ukl.corp.ukl.yahoo.com) |
| 17:33.44 | MrWup | just used it myself |
| 17:33.49 | MrWup | if you log into manager and call status |
| 17:33.55 | MrWup | it will send you an event every time something happens |
| 17:34.02 | MrWup | u just gotta wait |
| 17:34.41 | MrWup | if u log in, call status, and log out at the same time... it will send you a list of current connections/ringing/etc (like a snapshot of the whole thing) and then disconnect |
| 17:34.46 | MrWup | just been using it myself |
| 17:34.48 | MrWup | very useful |
| 17:35.43 | bkruse | MrWup: events: off |
| 17:35.43 | bkruse | :D |
| 17:37.02 | syzygyBSD | wait.. reboot for security reasons? |
| 17:37.15 | syzygyBSD | not running linux then... |
| 17:37.24 | EmleyMoor | Security issue discovered in the kernel |
| 17:37.44 | EmleyMoor | Running Linux |
| 17:37.44 | syzygyBSD | hmm, ok |
| 17:38.07 | syzygyBSD | that sucks, have to recompile everything then? |
| 17:38.24 | EmleyMoor | syzygyBSD: Hopefully not - using stock kernels at the moment |
| 17:38.24 | MrWup | bkruse, oh =] |
| 17:39.10 | bkruse | :D |
| 17:39.17 | EmleyMoor | Might have to recompile zaptel - we'll see |
| 17:39.39 | MrWup | i hacked channels.c to get zombie channel variables |
| 17:40.23 | syzygyBSD | EmleyMoor: probably all modules, libpri too |
| 17:40.38 | EmleyMoor | syzygyBSD: I don't use libpri |
| 17:41.03 | syzygyBSD | well that makes it easier |
| 17:42.40 | EmleyMoor | If I didn't have a rotary phone, zaptel would be marginally less of a pain |
| 17:43.33 | syzygyBSD | lol... wow, why don't you get a sip phone? |
| 17:43.48 | EmleyMoor | I'm going to get a SIP (or IAX) phone for the lounge... but I happen to like the rotary phone I have in the bedroom |
| 17:44.43 | syzygyBSD | does it have speed dial? |
| 17:45.21 | EmleyMoor | No - though I am probably going to set up a central speed dial soon |
| 17:45.46 | *** join/#asterisk ghento (n=ghento@CPE0014bfaf7d7a-CM00111ae4684c.cpe.net.cable.rogers.com) |
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| 17:54.11 | *** join/#asterisk rbd (n=rbd@adsl-074-229-183-112.sip.rmo.bellsouth.net) |
| 17:55.04 | *** join/#asterisk FlyboySR22 (n=rsears@hq.fw.americanis.net) |
| 17:55.13 | rbd | I have a custom asterisk app (.c source file) that I link against asterisk 1.2.... if I link it against one rev of asterisk 1.2 (say 1.2.14), is it safe to use that resulting .so binary in another rev (1.2.12 for instance)? |
| 17:55.49 | *** join/#asterisk RobH (n=RobH@rrcs-24-73-86-239.se.biz.rr.com) |
| 17:56.34 | RobH | Would anyone be able to assist with this? Music on hold is not working, I checked everything listed in this thread: http://forums.digium.com/viewtopic.php?t=4873&highlight=stopped+music+hold |
| 17:56.49 | RobH | Music on hold gives the following when placing someone on hold: |
| 17:56.58 | RobH | <PROTECTED> |
| 17:56.58 | RobH | May 9 13:55:26 NOTICE[21324]: res_musiconhold.c:533 monmp3thread: Request to schedule in the past?!?! |
| 17:56.58 | RobH | <PROTECTED> |
| 17:57.21 | *** join/#asterisk bzik (n=ohad@19-231-13-72.static.cosmoweb.net) |
| 17:57.31 | Ifaistos | rbd: Well it should work, provided that the headers/api your app is using have not changed between the versions |
| 17:58.45 | rbd | Ifaistos: yeah, it's a pretty basic app. I would think between revs they wouldn't change the function prototypes anyhow (it would be more of a major/minor release thing) |
| 17:59.06 | rbd | or change some semantics or data types |
| 18:01.31 | [TK]D-Fender | RobH: Looks like you're set up to use MPG123 for MoH. |
| 18:02.08 | [TK]D-Fender | RobH: Make sure you have the right version, that you have MP3 files in the proper folder with the proper permissions, and that your MP3's are non-VBR w/o ID3 |
| 18:02.22 | *** join/#asterisk oej (n=olle@apollo.webway.se) |
| 18:02.50 | RobH | Ok, I kind of got that, will have to do more searching but with those in frame. Any suggestions or links for setting this up as you say? |
| 18:03.20 | MrWup | gah |
| 18:03.32 | RobH | But its more than I had, Thank you =] |
| 18:03.33 | MrWup | im dialling out on a zap channel and the person actually answers and we speak |
| 18:03.41 | MrWup | but asterisk says noone answered and times out |
| 18:03.52 | [TK]D-Fender | RobH: I have just asked you to verify what you are using. Both the MoH files you intend to use, and the version o fthe tool that will play them |
| 18:03.57 | MrWup | (the same channel works fine for incoming calls and asterisk detects when the zap channel hangs up) |
| 18:04.04 | MrWup | anyone know what might be wrong? |
| 18:04.18 | MrWup | ive done all the obvious, like fxotune my digium card, set all uk tone groups ztcfg etc |
| 18:04.34 | MrWup | exten => 123,1,Dial(zap/1/9ww${EXTEN},20) |
| 18:04.35 | RobH | Yep, checking now. |
| 18:04.42 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@cpe-24-162-48-94.houston.res.rr.com) |
| 18:04.54 | MrWup | i dial the talking clock, and it answers, but just dies cause * says noanswer =[ |
| 18:05.38 | RobH | bleh, got it to work, stupid typo, ignore last request |
| 18:05.42 | RobH | its playing now |
| 18:05.53 | RobH | Although, while playing it does output May 9 14:05:10 NOTICE[21324]: res_musiconhold.c:533 monmp3thread: Request to schedule in the past?!?! |
| 18:06.05 | RobH | Is that something to be concerned about? |
| 18:06.12 | Dr-Linux | anybody is using web-meetme or Cepstral? |
| 18:06.33 | Dr-Linux | does Cepstral demo version work with asterisk? |
| 18:06.56 | *** join/#asterisk darkmug (n=Dennis@143.106.167.234) |
| 18:07.09 | Corydon76-home | MrWup: are you probing wctdm with the kernel option opermode=uk ? |
| 18:08.04 | coolbeans | Hey guys, with Polycom phones, I can't get my GMTOffset setting to 'stick'... I have to manually set it via the web interface... Anyone else have this issue? |
| 18:08.09 | Corydon76-home | Cepstral works fine in demo mode, if you don't mind the prefix message on everything |
| 18:08.47 | [TK]D-Fender | RobH: usually nothing to worry about |
| 18:09.02 | Corydon76-home | "This voice is unlicensed. Please visit ... to purchase a license. <your message>" |
| 18:09.10 | [TK]D-Fender | coolbeans: You need to set it to override DHCP. |
| 18:09.23 | [TK]D-Fender | coolbeans: and you should NEVER touch the web interface. EVER |
| 18:09.27 | RobH | [TK]D-Fender: I appreciate the assit, thank you! |
| 18:09.52 | [TK]D-Fender | coolbeans: What are you putting into gmtoffset in provisioning? |
| 18:09.57 | coolbeans | [TK]D-Fender: Thanks. I'll go hunt down that setting. ;) |
| 18:10.02 | [TK]D-Fender | RobH: np |
| 18:10.10 | Dr-Linux | Corydon76-home: does Cepstral work? |
| 18:10.17 | *** join/#asterisk Here_And_There (n=Here_And@74.92.105.41) |
| 18:10.19 | Corydon76-home | Dr-Linux: yes, works fine |
| 18:10.43 | Corydon76-home | It's a little digital sometimes, but it works |
| 18:11.46 | Dr-Linux | Corydon76-home: i've installed everything fine, and i'm donig this in extensions.conf "exten => 3999,1,Cepstral(Hello world)" |
| 18:12.02 | Corydon76-home | Did you compile the application? |
| 18:12.22 | Dr-Linux | Corydon-w: but when i dial it, it execute and my asterisk gets crashed with message "Segmantation fault (core)" |
| 18:12.45 | Dr-Linux | Corydon76-home: yes all fine, |
| 18:13.14 | Corydon76-home | Might also want to try Answering the channel first |
| 18:13.16 | Dr-Linux | ivr1*CLI> show application Cepstral |
| 18:13.16 | Dr-Linux | ivr1*CLI> |
| 18:13.16 | Dr-Linux | <PROTECTED> |
| 18:13.41 | Dr-Linux | hhm.. |
| 18:14.09 | Dr-Linux | Corydon-w: also i can't see "show cesptral .." commands? |
| 18:14.25 | Dr-Linux | as i hve no voices yet, ofcouse i want to buy but first i want to check |
| 18:14.42 | Corydon76-home | Uh, have no voices yet? |
| 18:14.48 | Dr-Linux | but this is dangour as it crashes my asterisk |
| 18:15.03 | Dr-Linux | Corydon-w: yes, but i wanna test it first .. the demo version |
| 18:15.27 | Corydon76-home | Well, if it crashed, you might want to get a backtrace |
| 18:15.38 | Dr-Linux | Corydon-w: i installed David-8k ..... swift/engine/etc stuff |
| 18:17.16 | Dr-Linux | Corydon-w: my question is that, "does it should voices in the CLI if we are using demo" |
| 18:17.46 | Dr-Linux | bcoz i can't see commands show cepstral |
| 18:18.33 | Corydon76-home | Where'd you find the source code? |
| 18:18.33 | coolbeans | [TK]D-Fender: Found the DHCP Override setting, no probme. I have a local configs for each timezone, a phone1.cfg, and the master sip.cfg. Which one takes priority? I have my custome SNTP settings in my timezone.cfg (i.e., central, easter, etc) but there is also a setting in sip.cfg for timezones. Thanks for any help. |
| 18:19.00 | Dr-Linux | Corydon76-home: i download david one from here: http://www.cepstral.com/cgi-bin/downloads?type=1143746987 |
| 18:19.12 | Dr-Linux | i ran it and all stuff done |
| 18:19.16 | Corydon76-home | Dr-Linux: I mean app_cepstral |
| 18:19.37 | [TK]D-Fender | coolbeans: What the heck is "timezone.cfg"? and what value are you using for the offset? |
| 18:19.41 | Dr-Linux | Corydon-w: that i downloaded from wiki for 1.2 |
| 18:19.48 | Dr-Linux | and i compiled .. all went fine |
| 18:19.53 | Dr-Linux | no error |
| 18:20.02 | *** join/#asterisk astraeis (n=sbma44@dsl092-174-002.wdc2.dsl.speakeasy.net) |
| 18:20.36 | Dr-Linux | Corydon76-home: from here: http://www.voip-info.org/wiki/index.php?page=App_Cepstral |
| 18:20.44 | Dr-Linux | the first one |
| 18:21.06 | coolbeans | Offset is central "-6", I have seperate files for our eastern and central timezone settings included in my <mac>.cfg, so my question was which file takes precedence since I have SNTP settings in both my <timezone>.cfg files and the master sip.cfg which everywhere I read says don't change. |
| 18:21.18 | _VoiceMeUp_COM | hheh got a hacker signing up lol |
| 18:21.35 | astraeis | I've got a slightly odd situation: I have an AGI script that runs fine on one asterisk installation, but not on another. I've confirmed that the script is being called in the dial plan -- it's just that it returns immediately (ret val 0) without doing anything. I've trimmed it down to a perl script that just waits a few seconds then exits, and confirmed that it works when called from the path that the asterisk log reports. But |
| 18:21.36 | Dr-Linux | Corydon76-home: also i loaded app_cepstral.so module sucessfully |
| 18:21.53 | Corydon76-home | coolbeans: uh, -21600, you mean? |
| 18:21.54 | astraeis | I should add that the logs don't report any errors -- it looks like the script executed properly. |
| 18:22.09 | coolbeans | It's in seconds? |
| 18:22.19 | coolbeans | In the web interface, it's in hours? |
| 18:22.35 | Qwell[] | <[TK]D-Fender> coolbeans: and you should NEVER touch the web interface. EVER |
| 18:22.41 | coolbeans | lol, got it. |
| 18:23.06 | [TK]D-Fender | coolbeans: I didn't ask what you thought you KNEW, I asked what you DID. |
| 18:23.24 | Corydon76-home | Dr-Linux: well, that's what wrong. That implementation doesn't have CLI commands |
| 18:23.32 | MrWup | Corydon76-home, yeah i am |
| 18:23.42 | Corydon76-home | Dr-Linux: this one does: http://www.automated.it/asterisk/app_cepstral_swift.tar.gz |
| 18:23.45 | coolbeans | I have "-6" in the .cfg file, should it be in hours or seconds? |
| 18:23.47 | MrWup | Corydon76-home, got opermode=uk, got all modes set to uk, confirmed with dmesg |
| 18:23.51 | coolbeans | (and thanks for your help) |
| 18:23.55 | MrWup | hangup detection fine, callerid fine |
| 18:23.55 | *** join/#asterisk NightTiger (n=derekm@ottawa-hs-64-26-171-99.s-ip.magma.ca) |
| 18:24.00 | MrWup | but when i dial, it doesnt answer |
| 18:24.07 | MrWup | it does autofallthrough status NOANSWER |
| 18:24.14 | [TK]D-Fender | coolbeans: its is SECONDS. thats the problem. |
| 18:24.21 | coolbeans | Got it. Thanks, TK. |
| 18:24.21 | Dr-Linux | Corydon-w: yes i know that does, but that want me to recompile asterisk source |
| 18:24.31 | [TK]D-Fender | coolbeans: So congratulations... you are offset by 6 whole seconds! |
| 18:24.46 | coolbeans | lol |
| 18:24.57 | Dr-Linux | Corydon76-home: i don't need commands, but if all fine, it should work |
| 18:25.33 | Dr-Linux | Corydon76-home: maybe i'm missing something, but i can't fine much on the web |
| 18:25.57 | MrWup | gah well |
| 18:26.00 | Dr-Linux | s/fine/find |
| 18:26.07 | MrWup | spose id better go home. there's no solving stupid zap channels |
| 18:26.07 | Corydon76-home | Dr-Linux: well, try emailling the author of the module, then |
| 18:26.42 | NightTiger | Where would I find pointers to using asterisk over a satellite hookup? |
| 18:27.38 | Corydon76-home | From a priest |
| 18:27.46 | Corydon76-home | You're going to have to pray |
| 18:27.58 | Dr-Linux | Corydon76-home: i already emailed him |
| 18:28.12 | Corydon76-home | Dr-Linux: what'd he say? |
| 18:28.33 | Dr-Linux | never reply |
| 18:28.56 | Corydon76-home | When did you email him? |
| 18:29.04 | [TK]D-Fender | coolbeans: you should be setting for -21600 for CST |
| 18:29.10 | drako | how I can make voicemail.conf to send a message to 2 diferents mailboxes, it keep sending one and paging one. |
| 18:29.58 | Corydon76-home | drako: Voicemail(mbox1@default&mbox2@other) |
| 18:30.03 | Dr-Linux | Corydon76-home: actually when i installed swift engine, it ask me to test it, i issued the command for "Hello word" i gave me nothing, so it means it worked as i'm on linux remotely |
| 18:30.33 | NightTiger | Uh... I'm being serious. Reception is *just fine*, but transmit is choppy on the recieveing end, so I'm guessing I need to tell the Asterisk admin to adjust some sort of buffer, or something, somewhere. But, I have no idea what I'm looking for |
| 18:30.44 | *** join/#asterisk Aquavette (n=noone@74-137-181-254.dhcp.insightbb.com) |
| 18:30.57 | Corydon76-home | NightTiger: jitter buffer |
| 18:31.09 | Aquavette | Is this the correct channel to ask questions? |
| 18:31.12 | *** join/#asterisk LoneShadow (i=duh@nat/cisco/x-6db5653a3e8ff8a6) |
| 18:31.19 | Corydon76-home | ~ask |
| 18:31.29 | jbot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there, just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 18:31.39 | NightTiger | Corydon76-home: Thanks. I'll start there... |
| 18:31.55 | Aquavette | I have a Asteriks Box outside a NAT, and I have several SIP clients behind NAT |
| 18:31.58 | Aquavette | do I need a stun server? |
| 18:32.03 | Corydon76-home | NightTiger: the main problem is the lalf second latency you get with geosynch |
| 18:32.11 | Corydon76-home | err, half second |
| 18:32.42 | Corydon76-home | Aquavette: no, you just need to set nat=yes for each sip client in Asterisk |
| 18:33.07 | *** join/#asterisk tausq (i=randolph@64.81.244.94) |
| 18:33.17 | Aquavette | what about port forwards throught the clients NATS? Do i need to specify specific RTP Ports or let Asteriks do it automatically |
| 18:33.17 | Aquavette | ? |
| 18:33.19 | Corydon76-home | Aquavette: and set registration low, like every 60 seconds |
| 18:33.34 | tausq | hi, has anybody tried to directly connect a fxs port to a fxo port on a tdm400 card? |
| 18:33.36 | NightTiger | Corydon76-home: That's not so much an issue, actually. They just have to be able to actually _hear_ me. |
| 18:34.15 | Corydon76-home | NightTiger: yeah... and the half second latency doesn't help one bit |
| 18:34.49 | NightTiger | You were never a CBer, were you? ;-) |
| 18:35.05 | Corydon76-home | No, just ham |
| 18:35.15 | drako | Corydon76-home, but in voicemail.conf? |
| 18:35.21 | Corydon76-home | drako: no |
| 18:35.28 | robin_sz | still have a problem with zaphfc/bristuffed truncating the first digit of incoming numbers ... I dont quite understand the pridialplan stuff |
| 18:35.41 | drako | it can be done only in extension.conf? |
| 18:35.45 | [TK]D-Fender | Aquavette: Leave registration normal in most cases, and set qualify=yes for their entries in addition to "nat=yes" |
| 18:35.49 | Corydon76-home | drako: you want two emails, set it up in your local mailer |
| 18:35.59 | drako | got it |
| 18:36.13 | drako | just wanted to know if it was possible from voicemail.conf |
| 18:36.17 | Aquavette | [TK]D-Fender: So that should solve my issue |
| 18:36.18 | Corydon76-home | No |
| 18:38.28 | Dr-Linux | Corydon76-home: does asterisk 1.4 Cepstral application? |
| 18:39.08 | Corydon76-home | What? |
| 18:39.18 | Dr-Linux | sorry for bad english :P |
| 18:39.39 | *** join/#asterisk adorah (n=Michael@87.69.57.73.cable.012.net.il) |
| 18:39.43 | Dr-Linux | Corydon76-home: i mean , is Cepstral application available in asterisk 1.4 by default? |
| 18:39.48 | Corydon76-home | No |
| 18:41.06 | *** join/#asterisk DigitalKNK (n=DigitalK@adsl-69-232-146-57.dsl.irvnca.pacbell.net) |
| 18:41.14 | bkw__ | Dr-Linux, go port the line list http://www.freeswitch.org/node/50 to 1.4? |
| 18:41.36 | bkw__ | s/line/one/ |
| 18:42.42 | Dr-Linux | bkw__: port one list? |
| 18:42.49 | *** join/#asterisk keulin (n=cray@AMontpellier-152-1-17-37.w81-251.abo.wanadoo.fr) |
| 18:43.17 | redax | anybody using Linksys SPA9xx? |
| 18:43.33 | pifiu | yeah |
| 18:43.35 | pifiu | good phones |
| 18:43.40 | redax | seems like User uploadable ringtones is not selectable using ALERT_INFO= |
| 18:43.52 | redax | does anybody knows any trick? |
| 18:44.19 | redax | ALERT_INFO= Office;Simple-[1-5] ; Classic-[1-5] |
| 18:44.29 | [TK]D-Fender | Aquavette: Well I don't know what other factors may be involved, but that is all you should have to do. |
| 18:44.30 | redax | they all selectable. |
| 18:44.39 | [TK]D-Fender | Aquavette: You should NOT be forwarding ports on the remote side |
| 18:47.04 | bkw__ | Call-Info |
| 18:47.07 | Dr-Linux | bkw__: there are some more cool stuff, will see them later, but first this Capstral thing |
| 18:47.07 | bkw__ | check that out |
| 18:47.12 | *** part/#asterisk tausq (i=randolph@64.81.244.94) |
| 18:47.32 | Dr-Linux | bkw__: me? |
| 18:48.26 | Aquavette | [TK]D-Fender: Problems solved, thanks |
| 18:49.13 | [TK]D-Fender | Aquavette: ywc |
| 18:49.48 | redax | bkw__: SipAddHeader Call-Info ? |
| 18:54.14 | *** join/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net) |
| 18:54.14 | *** mode/#asterisk [+o mog] by ChanServ |
| 18:54.39 | *** part/#asterisk NightTiger (n=derekm@ottawa-hs-64-26-171-99.s-ip.magma.ca) |
| 19:00.58 | *** join/#asterisk w9sh (n=w9sh@adsl-068-209-117-205.sip.asm.bellsouth.net) |
| 19:03.43 | *** join/#asterisk jetlagmk2 (n=jetlag@pool-70-18-182-200.pskn.east.verizon.net) |
| 19:06.10 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 19:22.04 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 19:22.06 | *** join/#asterisk ftexcom (n=ftexcom@14.Red-80-26-4.staticIP.rima-tde.net) |
| 19:22.15 | ftexcom | hui |
| 19:22.24 | ftexcom | is there anyway to block outbound calls from an extension? |
| 19:23.02 | pipwerk | yes, put it in an context where it cant reach the outbound extension |
| 19:24.00 | *** join/#asterisk angom (n=angom@red-corp-201.143.54.251.telnor.net) |
| 19:25.57 | robin_sz | so, how to add a leading zero to the incoming cid in chan_zap or zapat.conf |
| 19:27.01 | *** join/#asterisk Gpl_Source (n=The_natu@unaffiliated/gplsource) |
| 19:28.24 | redax | Set(CALLERID(number)=0${CALLERID(number}) ? |
| 19:29.09 | sevard | Does anyone know if "asterisk dialplan scripting" has a more official title? |
| 19:29.40 | *** join/#asterisk JuanCri (n=juancri@65.111.169.15) |
| 19:30.38 | pipwerk | sevard: AEL2 you mean? |
| 19:31.52 | sevard | Asterisk Extension Language, nifty. |
| 19:33.02 | pipwerk | ael2 is pretty okay |
| 19:36.36 | sevard | so extensions.ael is a bit different then extensions.conf, is there a naming distinction? |
| 19:36.53 | sevard | than* |
| 19:44.24 | redax | ahh. this is insane. seems like only the Simple-* ; Class-* and Office rings are selectable in SPA-9xx |
| 19:45.44 | redax | I'd like to have `silent' ring with spa9xx ; where only the display shows the caller, + the top red light flashes |
| 19:46.42 | robin_sz | redax, OK, thanks ... I'll try it |
| 19:48.06 | sevard | redax: you can rip out the speaker |
| 19:48.30 | *** join/#asterisk ghento (n=ghento@CPE001310e1d7b7-CM0014e8b5f018.cpe.net.cable.rogers.com) |
| 19:49.38 | redax | sevard: :-) |
| 19:50.15 | sevard | 8-) |
| 19:50.34 | redax | the other method to implement this feature, to set the default ring in the spa9xx to "No Ring" and set ALERT_INFO=Classic-1 whenever I need ring |
| 19:50.52 | redax | the strange thing is I can't choose ALERT_INFO=No Ring |
| 19:50.59 | redax | neither the uploaded user rings |
| 19:51.28 | robin_sz | redax, mmm Set(CALLERID(number)=0${CALLERID(number}) still sees its all wrong in the CDR logs and some other places |
| 19:52.53 | redax | this way the user looses the feature to choose his favorite ringtone |
| 19:53.40 | robin_sz | LOOSES? sigh |
| 19:53.52 | robin_sz | redax, you are american, right? |
| 19:55.58 | redax | robin_sz: no |
| 19:59.23 | robin_sz | oh well, worth a guess :) |
| 19:59.26 | redax | just a tired european |
| 19:59.26 | redax | robin_sz: try to set / fix the CDR data by SetCDR |
| 19:59.26 | robin_sz | its normally USAians who get loose/lose mixed up |
| 19:59.28 | *** join/#asterisk Cresl1n (i=matt@nat/digium/x-cf320b337db44918) |
| 19:59.28 | *** mode/#asterisk [+o Cresl1n] by ChanServ |
| 19:59.28 | robin_sz | redax, out of curiosity, which bit of europe then ? |
| 19:59.29 | *** join/#asterisk Brijn (n=bas@204.244.176.116.net-conex.com) |
| 19:59.29 | Brijn | Good day all |
| 19:59.29 | [TK]D-Fender | robin_sz: time to start hacking chan_zap.so |
| 19:59.29 | redax | Set(CDR(clid)=...) the syntax, (I hope) |
| 19:59.29 | redax | robin_sz: hungary |
| 19:59.30 | robin_sz | [TK]D-Fender, probably |
| 19:59.30 | [TK]D-Fender | redax: Yeah, I'm hungry too... good thing I'm outta here in 35 mins :) |
| 19:59.30 | robin_sz | redax, ah, kewl, im half hungarian myself :) |
| 19:59.30 | redax | the world is so small :) |
| 19:59.30 | robin_sz | remind me to visit the place some day |
| 19:59.31 | anonymouz666 | ahhh |
| 19:59.31 | anonymouz666 | writing udev rule sucks balls |
| 19:59.31 | Brijn | I want to hook up a wireless phone (DECT not Wifi) to an ATA.. Is there a "best" ATA for use with *? |
| 19:59.31 | robin_sz | so may to choose from |
| 19:59.31 | robin_sz | Linksys, sipura |
| 19:59.31 | redax | linksys pap2's good |
| 19:59.31 | Brijn | redax: OK, that is the one I was looking at |
| 19:59.32 | *** part/#asterisk HockeyInJune (n=HockeyIn@pool-68-161-188-234.nycmny.east.verizon.net) |
| 19:59.32 | robin_sz | cisco ata186 .. aka Sipurar somehting is reliable |
| 19:59.44 | Brijn | robin_sz: used the cisco ones in a previous live, the webinterface was horrible then |
| 19:59.50 | redax | I like linksys voip stuffs, even if I would kill them right now |
| 20:00.09 | robin_sz | Brijn, yeah, but its set and forget, you only ever use it at set up and then ... never again |
| 20:00.17 | Brijn | How are the Linksys iPhones.. Different solution to the same problem |
| 20:00.27 | robin_sz | i have the linksys SPA2002 ... |
| 20:00.32 | redax | pap2 configuration is easy via XML :) |
| 20:00.39 | robin_sz | perfect ONCE you have it set up ... |
| 20:00.41 | *** join/#asterisk sysreq (n=sysreq@132.208.95.78) |
| 20:00.48 | *** join/#asterisk rajiv (n=rajiv@gentoo/developer/rajiv) |
| 20:00.51 | *** join/#asterisk LeBowlingAlley (n=derek@71.16.158.170) |
| 20:01.05 | Brijn | redax: Do you mean it's provisioned from your server? |
| 20:01.08 | robin_sz | it is hard to convince it your * server is on the LAN not WAN side |
| 20:01.09 | LeBowlingAlley | what does "ExtensionState: 8" mean? |
| 20:01.24 | LeBowlingAlley | in the CLI |
| 20:01.34 | Brijn | robin_sz: :-) |
| 20:01.35 | redax | BrianR___: yep. or http://your.pap2/admin/resync?http://your.server/config.xml |
| 20:01.43 | redax | with lynx |
| 20:01.53 | redax | s/BrianR___/brijn/ |
| 20:03.07 | Brijn | redax: That's an interesting trick.. haven't seen that before, but makes a lot of sense |
| 20:03.40 | Brijn | Nobody experience with the IPhone's?? They are expensive, but it would be nice if they work well |
| 20:04.33 | redax | for i in `cat pap2-devices.txt | awk '{print $2}'' ; do lynx -dump http://$i/admin/resync |
| 20:04.41 | redax | for i in `cat pap2-devices.txt | awk '{print $2}'' ; do lynx -dump http://$i/admin/resync |
| 20:04.44 | redax | grr |
| 20:05.12 | redax | for i in `cat pap2-devices.txt | awk '{print $2}'' ; do lynx -dump "http://$i/admin/resync?http://192.168.1.100/cfg/$i.xml" ; done |
| 20:05.16 | redax | real mass deployment |
| 20:06.28 | Brijn | redax: What would an admin be with scripts and pipes :) |
| 20:07.15 | redax | hehh :) |
| 20:07.30 | Brijn | s/with/without/g |
| 20:08.38 | Brijn | The PAP2 and SPA2002 are the same thing? |
| 20:08.54 | redax | almost the same |
| 20:09.33 | redax | if I remember, the webinterface has some difference |
| 20:09.39 | Brijn | Cisco = linksys = sipura now? |
| 20:09.53 | Brijn | Different product lines, but all owned by Cisco |
| 20:10.48 | [TK]D-Fender | SPA-2002 supports T.38 & acts as a router as well. PAP2 = cheap ass poor choice |
| 20:12.34 | redax | t.38 ? cool |
| 20:12.55 | redax | but what is `acts' ? |
| 20:12.56 | brad_mssw | really? thought only the 2100 supported T.38 ... not the 200X series |
| 20:14.06 | *** join/#asterisk Dimik (n=Dimik@unaffiliated/dimik) |
| 20:14.13 | Brijn | [TK]D-Fender: I only need one analog port. Best choice for basic ATA is SPA-2002, and stay away from PAP2? |
| 20:14.22 | brad_mssw | http://www.sipura.com/Documents/faq/Section_1.html #3 says only 2100 series supports T.38 |
| 20:16.51 | *** part/#asterisk BSD_Tech (n=bsdtech@adsl-69-230-166-20.dsl.irvnca.pacbell.net) |
| 20:16.55 | [TK]D-Fender | Brijn: yup |
| 20:17.26 | Brijn | [TK]D-Fender: OK, do you have any experience with the Linksys Wifi phones. Good, Ok, Horrible? |
| 20:17.47 | ido | so i'd like to set up a nice fancy shmancy asterisk box for my apartment next year...i'd like it to interface with our landline. what do you folks recommend for that? |
| 20:17.54 | ido | should i get an all-in-one sort of dealie? |
| 20:19.46 | [TK]D-Fender | ~wifisip |
| 20:19.49 | jbot | Wi-Fi SIP phones suck. All of them. HARD. Some only slightly less than others... |
| 20:19.56 | Brijn | hahahaha |
| 20:20.07 | *** join/#asterisk Waverly360 (n=irc@209.12.249.243) |
| 20:20.08 | ido | the linksys wip-300 seems nice :/ |
| 20:20.12 | Brijn | That's clear enough |
| 20:20.15 | ido | wip3something |
| 20:20.51 | robin_sz | [TK]D-Fender, chan_zap is huge and twisted, I fear a simple hack to add al eadign zero is beyond me |
| 20:21.23 | *** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 20:21.40 | *** join/#asterisk theacolyte (n=theacoly@unaffiliated/theacolyte) |
| 20:21.42 | [TK]D-Fender | robin_sz: then your hack should fit right in! |
| 20:21.47 | *** join/#asterisk Trionnis (i=lordkuri@c-67-162-33-95.hsd1.il.comcast.net) |
| 20:22.07 | *** join/#asterisk bkruse (i=bkruse@nat/digium/x-11c6c3481b4f5164) |
| 20:22.13 | theacolyte | I've spent entirely too much time looking on asterisk.org trying to find an HCL to see what I can use for hardware... can anyone point me in the right direction please? |
| 20:22.40 | Qwell[] | theacolyte: any hardware made in the last 15 years should suffice. |
| 20:22.42 | [TK]D-Fender | I have a WIP-300. Looks nice. Connects to networks ok. no NAT support (SHIT), no Conferences (SHIT), no TRANSFER (SHIT!!!!!) |
| 20:22.49 | Trionnis | does anyone know of an example or existing application that will let me do call routing based on the number of calls that have been sent out a specific channel, and then reroute them once it hits a certain number? Just don't want to reinvent the wheel if I don't have to :) |
| 20:23.09 | [TK]D-Fender | Trionnis: No, there is no app for this... only dialplan. Get coding. |
| 20:23.15 | theacolyte | So... I could use a v92 modem laying on my desk? |
| 20:23.42 | [TK]D-Fender | theacolyte: Digium, Sangoma, and Rhino boards for TDM/digital. |
| 20:23.47 | theacolyte | Want to start with a single line install to test it out... |
| 20:23.55 | theacolyte | [TK]D-Fender.:| I'm just doing analog for now |
| 20:23.58 | [TK]D-Fender | theacolyte: No, odds are you random modem hanging around in completely useless |
| 20:24.06 | coolbeans | Wow. The Polycom 550 sounds freakin' great. |
| 20:24.15 | Trionnis | well, here's a quick description, perhaps you know a better way to do what I'm trying to do |
| 20:24.20 | theacolyte | So... what's a good, cheap analog card... or if you can point me in the right direction I could look myself |
| 20:24.38 | theacolyte | Investigating replacing our Altigen with Asterisk |
| 20:24.51 | *** join/#asterisk pariah (n=j0sh@unaffiliated/pariah) |
| 20:25.06 | Trionnis | I have a pots line with at&t (blah) that allows 30 outgoing local calls, then starts charging per call. What I want to do is send all local calls out that channel until it hits 30, then send them out voipjet or something similar until the "month" resets. |
| 20:25.20 | robin_sz | [TK]D-Fender, solved it ... nationalprefix=0 |
| 20:25.44 | robin_sz | I think ... |
| 20:25.46 | Trionnis | was thinking that tying it to mysql/pgsql/etc might be a good way to do it, unless you know of a better way :) |
| 20:25.57 | [TK]D-Fender | Trionnis: I presume you mean 30 free LD calls..... not jsut general... |
| 20:26.04 | Trionnis | local exchange |
| 20:26.16 | redax | a simple agi script can do that routing without any sql |
| 20:26.18 | Trionnis | I already know how to do the lookup for that, it's the call counting part that's throwing me |
| 20:26.28 | Trionnis | the sql is for the exchanges |
| 20:26.36 | [TK]D-Fender | Trionnis: And yes, you are definately looking at some dialplan logic to handle this. a dozen or few lines. |
| 20:26.42 | Trionnis | since they tend to change slightly on at&t's whim |
| 20:26.49 | Trionnis | well yes, I know that :) |
| 20:27.02 | Trionnis | what I'm specifically wanting to know is the counting part |
| 20:27.05 | robin_sz | Trionnis, callcounting in SQL? |
| 20:27.35 | [TK]D-Fender | Trionnis: use AstDB to store the counter and the current month. Test one, increment the other, route accordingly |
| 20:27.38 | Trionnis | I can use sql for storing the count if it's convienent |
| 20:27.51 | Trionnis | hm, that might work |
| 20:27.54 | astraeis | anybody have any idea why an AGI script might exit immediately without errors instead of executing? |
| 20:27.57 | [TK]D-Fender | AstDB is much more convenient. No need for SQL |
| 20:27.59 | Trionnis | don't know if it goes off the calendar month tho |
| 20:27.59 | robin_sz | update line set callount = callcount + 1 where lineid =012330300303 |
| 20:28.08 | robin_sz | or use astdb, like he said |
| 20:28.11 | astraeis | (it runs fine when called from the command line) |
| 20:28.22 | Trionnis | I'll have to ask 12 different people from at&t to hopefully get 2 or 3 usable answers |
| 20:28.25 | Trionnis | :) |
| 20:28.27 | [TK]D-Fender | ok, heading home... back in a bit |
| 20:29.12 | robin_sz | update line set callount = callcount + 1 where lineid =012330300303 and monthid=5 |
| 20:29.28 | Trionnis | the hardest part I'm having with this is that I'm pretty sure the counting will go off of my billing date, not the calendar month |
| 20:29.33 | robin_sz | or better still, dont do that, grab it out of ther CDR log |
| 20:29.38 | Trionnis | so I'll have to find a way to reset the count externally |
| 20:29.47 | Trionnis | yeah, I was thinking about using the cdr for it |
| 20:30.09 | robin_sz | just write a sql statment to grok the number of calls in the current charging period mad eon that line |
| 20:30.28 | Trionnis | I suppose I could stick a small bash script on a cron to reset the count on the billing date |
| 20:30.28 | robin_sz | nah |
| 20:31.05 | robin_sz | select count(*) from calls where line = 1 and datstamp >= $someDate |
| 20:31.12 | Trionnis | well, yes |
| 20:31.17 | Trionnis | but how to get $somedate |
| 20:31.21 | Trionnis | that's the catch :) |
| 20:31.37 | Brijn | Voicemail question: I don't get a mail.. Set logging real high, looked at the logfiles.. Doesn't show anywhere if it tries to send the mail and fails.. |
| 20:31.46 | robin_sz | make up a list of predicted rollover dates in a table |
| 20:31.50 | Trionnis | I'd have to redo the script each month, and I really want this to be automated :) |
| 20:31.55 | Brijn | Trionnis: Use the mysql date function - the period you want to look at |
| 20:32.07 | Trionnis | doh, forgot about that, haha |
| 20:32.10 | Trionnis | that could work |
| 20:32.25 | robin_sz | make the list at least 10 years long, slect the date that fall immediatelyt before the current date |
| 20:32.25 | *** join/#asterisk xtr (i=01928375@216.19.191.191.novuscom.net) |
| 20:32.51 | Trionnis | hm, now the hard part |
| 20:32.59 | robin_sz | being bothered? |
| 20:33.02 | Trionnis | how to make it so it only logs a call if the call actually conencts |
| 20:33.50 | Trionnis | that's where I'd probably be better to grok the cdr instead |
| 20:33.50 | Brijn | Trionnis: duration probably > 0 ? |
| 20:33.51 | Trionnis | that might work too |
| 20:33.51 | Brijn | Why redo something if there is already table with all the detail you need |
| 20:33.51 | robin_sz | and when you have done all this you will save ~ $3 a month |
| 20:34.35 | Trionnis | I'm a cheap bastard, what can I say? |
| 20:34.35 | robin_sz | Trionnis, what is the cost of 30 calls on voipgate or whatver? |
| 20:34.37 | Trionnis | :) |
| 20:34.40 | robin_sz | $2? |
| 20:34.46 | Trionnis | plus, it's a learning experience :) |
| 20:35.09 | robin_sz | one of the things I learnt a long time ago was not to waste time for $2 |
| 20:35.22 | Trionnis | learning != wasting time, imho |
| 20:35.28 | Trionnis | but to each their own :) |
| 20:35.56 | robin_sz | just stick it permanelty on voipgate or whatever, save those 30 calls for when it innevitably is down or can;t connect, or sounds like a recording of soup |
| 20:36.47 | robin_sz | route any calls with a leading 9 to the pstn |
| 20:36.59 | redax | wohooo |
| 20:37.03 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@cpe-24-162-48-94.houston.res.rr.com) |
| 20:38.08 | redax | great. it works. in a different way of course, you don't need to upload the ringtone, you just need to set the Ringtone1-9 like this: `n=myring;w=tftp://tftp-server/silence.wav' |
| 20:38.17 | redax | and ALERT_INFO should be myring |
| 20:38.23 | Trionnis | I could do that, but where's the fun in that? lol |
| 20:38.46 | redax | Trionnis: last time I did trunk routing with Random() :=) |
| 20:38.51 | Trionnis | haha |
| 20:38.58 | redax | better than nothing |
| 20:39.03 | Trionnis | true |
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| 20:39.29 | redax | bot trunk has 500minutes free call :) |
| 20:39.31 | redax | both |
| 20:39.43 | Trionnis | ya |
| 20:39.50 | Trionnis | I've got 3 trunks on this box |
| 20:39.54 | Trionnis | 2 voip, 1 pots |
| 20:40.05 | Trionnis | I don't think I'll have a problem getting a voip call out |
| 20:40.06 | Trionnis | : |
| 20:40.09 | Trionnis | :) |
| 20:40.44 | *** join/#asterisk adorah (n=Michael@87.69.57.73.cable.012.net.il) |
| 20:42.33 | Brijn | Voicemail question: I don't get a mail.. Set logging real high, looked at the logfiles.. Doesn't show anywhere if it tries to send the mail and fails.. |
| 20:44.28 | *** join/#asterisk shinao1 (n=shinao1@dial-pool1.lagos.starcomms.net) |
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| 20:46.41 | redax | when Dial(Local/ext@context) executed, and context miss `ext' extension, which one will be looked for? |
| 20:47.22 | *** join/#asterisk darkmug (n=Dennis@143.106.167.234) |
| 20:48.05 | Waverly360 | Has anyone here successfully built an rpm out of the latest wanpipe driver for the sangoma cards? |
| 20:52.22 | adorah | <Brijn>u have to set first a sendmail client |
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| 20:57.32 | Brijn | Sendmail client was set |
| 21:00.16 | *** part/#asterisk LeBowlingAlley (n=derek@71.16.158.170) |
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| 21:12.59 | neverblue | Master.cvs, the first entry (in a line) is what value? |
| 21:13.24 | Dantenix | hi all, I have an Axis 210 camera with audio support, that is: using the axis software I can listen and talk from/to the camera's mic and speaker. Shold be possible to hook the audio stream from the camera with asterisk, in the way I'll can listen and talk from a regular asterisk's extension? |
| 21:15.53 | *** join/#asterisk mattchis (n=mattchis@216.54.143.246) |
| 21:17.36 | mattchis | I have just upgraded from 1.4.0 to 1.4.4 and now it will not play any sound files. Does anyone know what might be the cause for this? |
| 21:19.59 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 21:22.58 | *** join/#asterisk jhiver (i=jhiver@165-242.206-83.static-ip.oleane.fr) |
| 21:23.01 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 21:23.44 | jhiver | hi all |
| 21:23.59 | *** join/#asterisk kevinfries (n=kfries@71.216.94.233) |
| 21:24.15 | jhiver | it's been a long time since i last came in this channel, how's the asterisk world going? =) |
| 21:25.02 | kevinfries | Can anyone point me at a resource to help me write an IVR only application with Asterisk... I.e. no phone extensions |
| 21:26.29 | *** join/#asterisk fbffff (n=fbffff@r74-193-79-81.pfvlcmta01.grtntx.tl.dh.suddenlink.net) |
| 21:26.33 | *** join/#asterisk FastFeet (n=FastFeet@CPE0013109fd25b-CM000f9fa60d7a.cpe.net.cable.rogers.com) |
| 21:27.52 | kevinfries | Can anyone point me at a resource to help me write an IVR only application with Asterisk... i.e. no phone extensions |
| 21:28.44 | [TK]D-Fender | kevinfries, ... |
| 21:28.45 | [TK]D-Fender | ~docs |
| 21:28.47 | jbot | [docs] 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 |
| 21:28.47 | [TK]D-Fender | ~wikis |
| 21:28.49 | jbot | i guess wikis is http://www.voip-info.org |
| 21:28.53 | [TK]D-Fender | ~book |
| 21:28.56 | jbot | book is probably a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 21:29.21 | [TK]D-Fender | kevinfries, And don't repeat questions around here every 2 minutes you don't get answered. |
| 21:29.31 | *** join/#asterisk gbilly (n=gbilly@S01060004e2caa4a2.gv.shawcable.net) |
| 21:29.40 | [TK]D-Fender | kevinfries, broken records tend to get slapped around in here :) |
| 21:30.00 | [TK]D-Fender | jhiver, Much the same as usual |
| 21:30.02 | jhiver | i guess that answers my question about the asterisk world: same as usual =) |
| 21:30.11 | [TK]D-Fender | jhiver, ;) |
| 21:30.13 | jhiver | typical =) |
| 21:30.16 | kevinfries | I wasn't even sure this channel had anyone listening because I saw no chatter, so I just repeated the question |
| 21:30.53 | [TK]D-Fender | kevinfries, General etiquette says you should wait at LEAST 10 minutes before repeating the first time, 20+ for the next. |
| 21:31.12 | robin_sz | [TK]D-Fender, you're repating yourself now ... |
| 21:31.36 | [TK]D-Fender | robin_sz, Yeah, I said the exact same thing to some other newb abour a month ago! |
| 21:31.57 | robin_sz | and do they listen? |
| 21:32.05 | gbilly | Hi, about how much would I expect to spend on a business landline that can forward to mutiple cell phones? Do I need 2 landlines? |
| 21:32.33 | [TK]D-Fender | robin_sz, variably..... |
| 21:37.55 | *** join/#asterisk diclophis-work (n=jbardin@65.203.37.58) |
| 21:37.57 | diclophis-work | hello all |
| 21:38.03 | diclophis-work | for asterisk 1.2 how do i set the callerid? |
| 21:38.13 | diclophis-work | is it Set(CALLERID=1231231234) ? |
| 21:38.54 | [TK]D-Fender | Set(CALLERID(num)=1231231234) |
| 21:39.06 | diclophis-work | i see |
| 21:39.07 | diclophis-work | close |
| 21:40.02 | Black-Kakugane-1 | anyone know where I can find more information on the dialplan syntax "exten => s/1235551212,1,<action>" where 1235551212 is the caller's number? |
| 21:40.58 | Corydon76-home | callerid rarely includes the 1 prefix |
| 21:41.25 | Black-Kakugane-1 | so it does get set from the caller id info? |
| 21:41.40 | Corydon76-home | Correct |
| 21:41.51 | Black-Kakugane-1 | and i was just using 123 as a (bad) example areacode |
| 21:42.25 | Corydon76-home | 6152345678 is what I usually use for an example |
| 21:42.54 | Black-Kakugane-1 | is there anything else I can match on? I can't find any info on anything other then just plain 's' |
| 21:43.13 | Corydon76-home | What type of line do you have? |
| 21:43.27 | Black-Kakugane-1 | mainly sip and iax trunks |
| 21:43.47 | [TK]D-Fender | Black-Kakugane-1, "s" is the exten. This is usually what calls are reveived against for inbound analog Zap channels |
| 21:43.48 | Corydon76-home | Then it's unlikely to be s they're sending as the initial extension |
| 21:44.02 | Corydon76-home | It's more likely to be your DID that they're sending |
| 21:44.18 | robin_sz | Corydon76-home, 61 .. australia? |
| 21:44.36 | robin_sz | or christmas island? |
| 21:44.43 | diclophis-work | what are the CALLERID(rdnis) and CALLERID(dnid) used for? |
| 21:44.45 | diclophis-work | and ani/ |
| 21:45.05 | Black-Kakugane-1 | yeah, i'm working off an old *@home dialplan that does "_X.,n,Goto(ext-did,s,1)" |
| 21:45.25 | Black-Kakugane-1 | I just couldn't find any documentation on the /<num> syntax |
| 21:45.38 | Corydon76-home | ~trixbox |
| 21:45.46 | jbot | Trixbox is a full linux distro that includes , FreePBX, and other 3rd party add-ons. It is these things on top of which make it seriously painful to support and hence you will find little help here for it. Try asking in #trixbox , or their forums & WIKI at http://www.trixbox.org |
| 21:46.01 | Black-Kakugane-1 | yeah, the web interface has been killed |
| 21:46.19 | Black-Kakugane-1 | i'm just using it's dailplans |
| 21:46.40 | Corydon76-home | Ugh, you should probably start from scratch |
| 21:46.46 | *** join/#asterisk Fieldy (i=rVkIRCw7@gentoo/contributor/Fieldy) |
| 21:47.35 | *** join/#asterisk JerJer (n=PhatJ@pdpc/supporter/bronze/jerjer) |
| 21:47.43 | Black-Kakugane-1 | i probaby will eventually, but that's why i was looking for documentation for that syntax |
| 21:47.46 | JerJer | ok what replaced DBGet in 1.4 ? |
| 21:47.59 | [TK]D-Fender | JerJer, "show function DB" |
| 21:48.05 | JerJer | grr |
| 21:48.06 | robin_sz | monkeys? |
| 21:49.02 | JerJer | its great to see that function DB is not documented in doc/ |
| 21:49.42 | JerJer | [TK]D-Fender: and its CORE show function DB |
| 21:50.31 | [TK]D-Fender | JerJer, Mine gives you a WARNING, so its still good till 1.6 ;) |
| 21:50.57 | JerJer | no warning on svn trunk code |
| 21:51.02 | JerJer | so pre-1.6 |
| 21:51.38 | Black-Kakugane-1 | so, is there documentation for "exten => (exten)/(caller id),1,(action)" anywhere? |
| 21:51.59 | JerJer | Black-Kakugane-1: somewhere |
| 21:52.05 | Black-Kakugane-1 | ... |
| 21:52.06 | [TK]D-Fender | Black-Kakugane-1, you've pretty much broken it down right there. |
| 21:52.08 | [TK]D-Fender | ~book |
| 21:52.09 | jbot | from memory, book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 21:52.10 | JerJer | but what else do you need |
| 21:52.11 | [TK]D-Fender | ^^^^ |
| 21:52.37 | Black-Kakugane-1 | read it, didn't see it mentioned |
| 21:52.47 | FastFeet | A new revision of this book is soon to be released... I belive in July ? |
| 21:53.08 | JerJer | search for anti ex-girlfriend logic :) |
| 21:53.19 | [TK]D-Fender | Black-Kakugane-1, its a normal exten with a CID requirement to match. what more is there to say? |
| 21:53.23 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 21:54.14 | *** join/#asterisk fakhir (i=fakhir@unaffiliated/fakhir) |
| 21:55.24 | Black-Kakugane-1 | i was just looking for docs saying what else i could match on (CID name? other info?) and what it would use with no CID info present (Unavailiable? <empty>?) |
| 21:56.30 | [TK]D-Fender | Black-Kakugane-1, For anything other than a number match : "show application gotoif" |
| 21:57.20 | [TK]D-Fender | Black-Kakugane-1, the old CID match syntax is a nearly worthless implementation which could have been done in straight dialplan. |
| 22:00.12 | Black-Kakugane-1 | ok |
| 22:02.01 | Black-Kakugane-1 | next question: i made a simple Ansyer(), MP3Player(), Hangup() plan. it works fine over a IAX trunk, but when that same provider is set to SIP, all i get is ~2 sec of garbled audio and then silence |
| 22:02.17 | FastFeet | Could someone help me out here: I get: # |
| 22:02.17 | FastFeet | WARNING[4400]: app_dial.c:1081 dial_exec_full: Unable to create channel of type 'IAX2' (cause 3 - No route to destination) |
| 22:02.17 | FastFeet | # |
| 22:02.17 | FastFeet | == Everyone is busy/congested at this time (1:0/0/1) |
| 22:02.17 | FastFeet | # |
| 22:02.19 | FastFeet | -- Timeout on SIP/user3-08554b40 |
| 22:02.23 | FastFeet | more info here: http://pastebin.ca/479782 |
| 22:03.00 | JerJer | ok - what replaced Read in 1.4 ? |
| 22:04.39 | [TK]D-Fender | JerJer, "core show application read" :) |
| 22:04.43 | *** join/#asterisk aydiosmio (i=aydiosmi@judecca.aculei.net) |
| 22:04.59 | [TK]D-Fender | JerJer, And fear not.... we can't tell the difference ;) |
| 22:04.59 | aydiosmio | anyone know the default password for the web login for the GS budgetones? |
| 22:05.06 | FastFeet | I don't have this problem at home, but at work I keep getting this: Unable to create channel of type IAX2 (cause -- No route to destination) when I place calls from across the Net to a PSTN line. |
| 22:05.07 | JerJer | *CLI> core show application Read |
| 22:05.07 | JerJer | Your application(s) is (are) not registered |
| 22:05.09 | FastFeet | http://pastebin.ca/479782 |
| 22:05.26 | [TK]D-Fender | JerJer, Your install is broken then |
| 22:05.45 | JerJer | its very much stock svn from a few days ago |
| 22:06.24 | *** join/#asterisk `Sean (i=Un1x@74.114.247.106) |
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| 22:08.34 | aydiosmio | 123, figures. |
| 22:10.17 | *** join/#asterisk Here_And_There (n=Here_And@pool-68-238-252-162.phlapa.fios.verizon.net) |
| 22:10.46 | FastFeet | Could this ( http://pastebin.ca/479782 ) be an issue due to a firewall perhaps? |
| 22:11.23 | *** join/#asterisk cspot (i=cspot@ip68-1-63-100.pn.at.cox.net) |
| 22:14.12 | Black-Kakugane-1 | FastFeet: you try without the 'username@' ? |
| 22:14.54 | FastFeet | Thanks: No I have not, I was under the impression I needed the username. |
| 22:15.29 | FastFeet | I did try username:password@ though.... no luck |
| 22:15.32 | Black-Kakugane-1 | i don't think you need it there |
| 22:15.36 | *** join/#asterisk jm|home (n=jm|home@zen.jamiem.com) |
| 22:15.39 | FastFeet | This all works fine here at home, but not at work |
| 22:15.54 | Black-Kakugane-1 | i think it's turning the part after the @ into a host |
| 22:16.31 | FastFeet | well, that does make sense. |
| 22:17.10 | FastFeet | so you suggest: |
| 22:17.18 | FastFeet | exten => _9.,1,Dial(IAX2/${EXTEN:1}) ; voip out |
| 22:17.49 | Black-Kakugane-1 | exten => _9.,1,Dial(IAX2/iaxout/${EXTEN:1}) |
| 22:17.52 | *** join/#asterisk chino (n=daquino@c-68-36-237-152.hsd1.nj.comcast.net) |
| 22:17.57 | FastFeet | ahhh |
| 22:18.01 | FastFeet | Thanks.. |
| 22:18.07 | FastFeet | I shall try this |
| 22:18.21 | chino | is there any free pots->sip gateways so i could call my computer from my phone |
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| 22:22.10 | *** part/#asterisk dasenjo (n=dasenjo@190.24.177.31) |
| 22:25.50 | *** join/#asterisk gammah (n=gammah@70.253.197.130) |
| 22:26.02 | FastFeet | Black-Kakugane-1: Thanks seems to have helped. :-) |
| 22:27.00 | *** join/#asterisk kiscokid (n=ron@208.106.33.66) |
| 22:27.35 | *** join/#asterisk kuku5 (n=kuku5@c-71-201-219-72.hsd1.il.comcast.net) |
| 22:28.29 | kuku5 | Is there a reason why in 1.2.18 on a cisco 7940 when a call comes in it shows the caller id number in the caller id name line (shows the number twice), but when you later forward that call, the name appears ? |
| 22:33.58 | *** join/#asterisk derrick_cmr (n=kamgas@dyn-83-152-251-245.ppp.tiscali.fr) |
| 22:34.22 | derrick_cmr | Bonjour |
| 22:34.57 | derrick_cmr | Je cherche l'aide pour compiler zaptel sur Debian 4 |
| 22:35.15 | derrick_cmr | j'ai un probleme avec make linux26 |
| 22:36.45 | *** join/#asterisk viler (i=1000@134.42.128.66.PPPoECali.dynamic.telesat.net.co) |
| 22:39.50 | *** join/#asterisk xtr (i=94752345@216.19.191.191) |
| 22:49.50 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 22:54.00 | *** join/#asterisk findlay (n=justin@67.137.24.115) |
| 22:54.23 | findlay | is there a special number you call to access voicemail? |
| 22:55.56 | Qwell[] | findlay: whatever you set up to be your voicemail number |
| 23:01.31 | *** join/#asterisk alexrrr (n=alex@200.49.165.118) |
| 23:01.43 | alexrrr | lame, an empty chat room |
| 23:02.08 | Qwell[] | empty? |
| 23:02.35 | alexrrr | hi there... im just starting with Xchat... used to work with mIRC before |
| 23:02.44 | alexrrr | Oh |
| 23:02.47 | Qwell[] | even if it were empty... |
| 23:02.48 | alexrrr | i see everyone now sorry |
| 23:02.58 | Qwell[] | wouldn't that mean...you were...talking to...yourself? |
| 23:03.10 | alexrrr | hehehe, yes... practicing my english i guess. |
| 23:03.30 | alexrrr | im the kind of guy who thinks of himself in third person. |
| 23:03.46 | findlay | Qwell[]: so, you just dial that number (the mailbox number)? |
| 23:08.49 | findlay | man this configuration is ugly |
| 23:13.32 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 23:14.14 | *** join/#asterisk Putzz (n=me@CPE00155824933c-CM00111ae07ac2.cpe.net.cable.rogers.com) |
| 23:14.43 | *** join/#asterisk dotSlashW (i=hidden-u@mail.intouchhealth.com) |
| 23:14.46 | dotSlashW | <PROTECTED> |
| 23:14.55 | dotSlashW | some people here recommended axvoice |
| 23:15.10 | dotSlashW | <PROTECTED> |
| 23:17.51 | findlay | dotSlashW: my experience is limited, but I've been good so far with arrivaltel.com |
| 23:19.39 | fbffff | lame question, anyone know of a decent voip provider in the uk? |
| 23:20.08 | fbffff | just for sip trunking |
| 23:22.34 | *** join/#asterisk dj-fu (n=aj@unaffiliated/dj-fu) |
| 23:23.50 | *** join/#asterisk ManxPower (n=manxpowe@201.sub-70-216-204.myvzw.com) |
| 23:25.03 | fbffff | nevermind i found a company that will work |
| 23:27.52 | *** join/#asterisk Cyber-Dogg (i=Cyber-Do@24-178-240-97.dhcp.stls.mo.charter.com) |
| 23:28.48 | mocker | Hmm, hopefully copying app_rxfax and txfax from one server to another won't make my system puke. |
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| 23:37.29 | LeddyHM | how do I "echo" stuff to the console to show values of variable to make sure they exist? |
| 23:37.44 | Juggie | Noop(${MYVAR}) |
| 23:37.59 | LeddyHM | whoot |
| 23:38.01 | LeddyHM | thanks |
| 23:39.50 | *** join/#asterisk xtr-II (i=94752345@216.19.191.191.novuscom.net) |
| 23:42.40 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177583012.dsl.bell.ca) |
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| 23:47.13 | LeddyHM | hrm |
| 23:47.23 | LeddyHM | can't seem to get the extension |
| 23:47.40 | LeddyHM | $EXTEN == the number dialed |
| 23:47.59 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 23:51.45 | mocker | LeddyHM: ${EXTEN} |
| 23:51.46 | mocker | ? |
| 23:53.35 | LeddyHM | yes |
| 23:54.53 | ManxPower | ${EXTEN} is the currently executing extension in extensions.conf. Usually that is the dialed number, but with macros and gotos it might not be |
| 23:55.28 | LeddyHM | I have a *XXX macro to dial out |
| 23:55.29 | *** join/#asterisk Putzz (n=me@CPE00155824933c-CM00111ae07ac2.cpe.net.cable.rogers.com) |
| 23:55.48 | Putzz | is it possible to turn up the volume on voip calls |
| 23:55.58 | ManxPower | Putzz: no. |
| 23:56.11 | Putzz | didnt think so thanks I had a little bet going ;-) |
| 23:56.18 | ManxPower | Putzz: that is the job of the PSTN/VOIP device and/or the actuall VoIP phone |
| 23:56.40 | Putzz | ok cool |
| 23:56.48 | ManxPower | Remember you don't usually WANT the audio to even go thru asterisk if you don't have to |
| 23:57.25 | ManxPower | adjusting audio levels even in the analog and t-1 world can suck |
| 23:57.37 | FastFeet | I don't have this problem at home, but at work I keep getting this: Unable to create channel of type IAX2 (cause -- No route to destination) when I place calls from across the Net to a PSTN line. |
| 23:57.40 | FastFeet | http://pastebin.ca/479782 |
| 23:58.27 | Putzz | FastFeet: cause 3 - No route to destination |
| 23:58.35 | Putzz | check your trunk settings |
| 23:58.36 | FastFeet | yes |
| 23:58.41 | ManxPower | FastFeet: you have a [iaxout] section of iax.conf? |
| 23:58.56 | FastFeet | yes: http://pastebin.ca/479782 |
| 23:59.17 | FastFeet | Putzz: could you expand on what you said? |
| 23:59.53 | ManxPower | you can ping iax.callwithus.com ? |