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| 00:06.37 | Inez | :( |
| 00:06.43 | *** join/#asterisk [Outcast] (n=bill@222-154-60-227.jetstream.xtra.co.nz) |
| 00:07.33 | Marshall16 | [Global] Rebooting. |
| 00:09.20 | CrazyTux | Anyone here know anything about CPL? |
| 00:13.42 | *** join/#asterisk Dovid (n=Dovid@barak.cellcom.co.il) |
| 00:14.07 | Dovid | anyone know if there will be an astricon in asia anytime soon or when the next one will be in the US ? |
| 00:14.30 | Dovid | on thier site they dont seem to have any listings of future dates |
| 00:15.03 | *** join/#asterisk bkw__ (n=ASSERT_K@216.138.69.138) |
| 00:15.27 | bkw__ | Let me tell you how chidlish digium and Mark Spencer is. I walk into a restaurant with them all here at Astricon wearing my sangoma shirt and he asked me to leave. |
| 00:15.56 | Dovid | u serious ? |
| 00:15.56 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 00:16.03 | bkw__ | yes |
| 00:16.08 | Adam12 | Wow, I didn't think Mark Spencer owned a restaurant! :) |
| 00:16.12 | Dovid | btw how old is mark ? he looks like a kid |
| 00:16.17 | bkw__ | 28 |
| 00:16.36 | Dovid | damn |
| 00:16.40 | Dovid | filthy rich boy |
| 00:16.46 | CtRiX | bkw_, trolling again :-) |
| 00:16.46 | Inez | I must try with 1.4 |
| 00:16.47 | Dovid | i knes he wasnt over 30 |
| 00:17.12 | *** mode/#asterisk [+b *!n=*@216.138.69.138] by Corydon-w |
| 00:17.18 | *** kick/#asterisk [bkw__!n=tilghman@pdpc/supporter/sustaining/Corydon76-home] by Corydon-w (troll) |
| 00:17.32 | Dovid | he asked u to leave cause u were wearin the competitors shirt ? u know u didnt have to leave. i woulda rubbed it in his face. |
| 00:17.35 | Dovid | lol |
| 00:17.49 | Dovid | Corydon: can I PM u ? |
| 00:17.52 | CtRiX | another demo of how the truth may hurt |
| 00:17.57 | mishehu | Corydon-w: uhm, flex that muscle |
| 00:17.58 | Corydon-w | Nope |
| 00:17.59 | [TK]D-Fender | Dovid : Careful... "The Man" is listening ;) |
| 00:18.15 | Dovid | i know - i want to ask a question but not here |
| 00:18.33 | mishehu | Dovid: I've had sat there and told him how great I think the competitor's product is |
| 00:18.33 | Dovid | to be fare to him i want to ask in PM |
| 00:18.40 | mishehu | that's rude as hell what he did. |
| 00:18.52 | Dovid | who ? |
| 00:19.02 | mishehu | Dovid: about bkw and the digium crew |
| 00:19.11 | CtRiX | another demo of how sangoma is stealing someone's business wirh better products |
| 00:19.16 | Dovid | what bkw did or mark ? |
| 00:19.38 | mishehu | Dovid: commenting about what I would have done if I was bkw_. |
| 00:19.46 | *** mode/#asterisk [+b *!n=*@aretha.navynet.il] by Corydon-w |
| 00:19.51 | *** kick/#asterisk [CtRiX!n=tilghman@pdpc/supporter/sustaining/Corydon76-home] by Corydon-w (trolling) |
| 00:20.08 | [TK]D-Fender | *sigh* |
| 00:20.17 | *** part/#asterisk Adam12 (n=adam@d150-182-137.home.cgocable.net) |
| 00:21.01 | Dovid | lol. i still cant get over the humor that the coders at digium have - first time i logge outa the console - i got thans for all the fish :) |
| 00:21.06 | Dovid | Adam12 reminded me |
| 00:21.27 | *** join/#asterisk CtRiX (n=CtRiX@aretha.navynet.it) |
| 00:21.42 | *** join/#asterisk sqldoug (n=dougkres@mail.ideafit.com) |
| 00:21.53 | Druken | when you drink that much coke... you need to release the humour somewhere.... |
| 00:22.00 | Dovid | lol |
| 00:22.03 | *** mode/#asterisk [-b *!n=*@aretha.navynet.il] by Corydon-w |
| 00:22.06 | *** mode/#asterisk [+b *!n=*@aretha.navynet.it] by Corydon-w |
| 00:22.09 | mishehu | ooooh |
| 00:22.12 | [TK]D-Fender | Druken : Don't start what I'm fully willing to finish ;) |
| 00:22.40 | Druken | [TK]D-Fender: that almost sounds like a threat :) |
| 00:22.46 | Dovid | what is -b and +B ? i am a newbie to irc. |
| 00:22.48 | CrazyTux | Can someone here explain what exactly CPL is? |
| 00:22.57 | Dovid | CPL = ? |
| 00:23.00 | [TK]D-Fender | Druken : I'm clearly not trying hard enough :D |
| 00:23.04 | CrazyTux | Call Processing Language, but what exactly is it used for/with? |
| 00:23.28 | [TK]D-Fender | Dovid : Channel banning. |
| 00:23.29 | *** join/#asterisk ariel_ (n=Ariel@dsl-20-177.cofs.net) |
| 00:23.36 | Druken | [TK]D-Fender: apparently not... :P |
| 00:23.41 | Dovid | which is ? |
| 00:23.48 | *** join/#asterisk [Outcast] (n=bill@222-154-63-14.jetstream.xtra.co.nz) |
| 00:24.14 | [TK]D-Fender | Dovid : If you're banned, you can't join the channel. basically getting the door slammed in your face. |
| 00:24.15 | Dovid | oh i get it. he baeen the entire dns of the troll |
| 00:24.21 | Dovid | i know that |
| 00:24.27 | Dovid | i am allways a newbie at something |
| 00:24.30 | Dovid | Grrrrrrrrrrrrrrrrrrrrrrrrrrrrrr |
| 00:24.45 | sqldoug | Hello, All. I'm having a problem with DTMF recognition, and was wondering if anyone here could help. |
| 00:25.09 | [TK]D-Fender | Dovid : Banning IP / netmask ranges helps ensure if he's on a dynamic IP that he probably won't be able to sign in anywhere on that provider |
| 00:25.20 | *** join/#asterisk juanjoc (n=juanjoc@248-32-235-201.fibertel.com.ar) |
| 00:25.26 | *** join/#asterisk Marshall16 (n=Marshall@cpe-76-181-166-122.columbus.res.rr.com) |
| 00:25.31 | Dovid | but dosent that block anyone on from that ISP ? |
| 00:25.32 | JT | wtf, bkw banned?? |
| 00:25.53 | Dovid | its called the troll treatment :) works well |
| 00:25.56 | JT | a fake? |
| 00:26.04 | JT | ssh Dovid |
| 00:26.43 | Dovid | huh ? |
| 00:26.43 | [TK]D-Fender | JT : (connection refused) |
| 00:26.56 | JT | [TK]D-Fender: i'm sorry? |
| 00:27.03 | Dovid | sqldoug: what seems to be the problem ? |
| 00:27.03 | JT | ah |
| 00:27.05 | JT | :P |
| 00:27.09 | aptura | Why doesnt mark do the smart thing and lure away some of of samgomas best engineers. |
| 00:27.23 | mishehu | JT: he was banned for saying something true I guess. |
| 00:27.26 | [TK]D-Fender | aptura : Stoke the fire why don't you..... |
| 00:27.29 | JT | so was that a fake bkw? |
| 00:27.33 | mishehu | I don't have scrollback right now. |
| 00:27.38 | [TK]D-Fender | ALL OF YOU. sheesh..... |
| 00:27.46 | JT | lol no scrollback |
| 00:28.20 | Corydon-w | No, he was banned for trolling. |
| 00:28.38 | Corydon-w | He tried trolling Mark in the restaurant and Mark asked him to leave. That's all. |
| 00:28.39 | wulfy814 | how do I troubleshoot svn trunk seg faulting? |
| 00:28.47 | JT | my goodness, talk about kneejerk reaction |
| 00:28.49 | JT | what did he do there? |
| 00:29.04 | Druken | trolling? did i miss something? |
| 00:29.08 | Dovid | lol. there is allways another side to the storry |
| 00:29.12 | Dovid | he said he just walked in |
| 00:29.20 | Dovid | seems he said something to mark too |
| 00:29.27 | Corydon-w | JT: bkw has a long history in the community, and he left the community in a very public and acrimonious way |
| 00:29.31 | sqldoug | I'm running Asterisk 1.2.9 with a TDM411P(?) w/a T1 line. Some incoming calls have problem with 1,2,3 tone recognition. |
| 00:29.45 | JT | Corydon-w: so what happened at the restaurant? |
| 00:29.53 | [TK]D-Fender | "Understanding is a three edged sword" |
| 00:30.08 | Dovid | sqldoug: if it was over voip i could help. ask TK he is better at this |
| 00:30.19 | Corydon-w | JT: he made it very clear that he no longer wanted to be associated with the Asterisk community, and then was surprised when the rest of the community turned their backs on him. |
| 00:30.29 | *** join/#asterisk hmmhesays (n=hmmhesay@24-117-135-28.cpe.cableone.net) |
| 00:30.59 | Dovid | Corydon: This was tonight or in the past ? |
| 00:31.05 | Corydon-w | Dovid: in the past |
| 00:31.08 | JT | Corydon-w: yes i understand, but what happened at the restaurant? :) |
| 00:31.11 | Dovid | ah ok |
| 00:31.18 | carrar | A three edge sword would be a rectagle looking sword 3 sharp sides? :) |
| 00:31.25 | Dovid | JT: He dont wana talk about it. Geeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeez |
| 00:31.30 | mishehu | Corydon-w: that's looking at things thru rosy goggles. I'm pretty sure that the story is much more complicated than that, and that the "asterisk community" is likely to be just as at fault. |
| 00:31.31 | JT | Dovid: shush |
| 00:31.36 | Corydon-w | JT: bkw walked in, trying to associate with the Asterisk party, and Mark asked him to leave. |
| 00:31.37 | *** join/#asterisk Mad_guy (n=austin@ip68-1-213-212.dl.dl.cox.net) |
| 00:31.40 | JT | Dovid: if you don't want to contribute, then don't |
| 00:31.46 | JT | Corydon-w: okay |
| 00:32.03 | aptura | how many asterisk users were there with mark |
| 00:32.27 | Corydon-w | aptura: Mark always invites a large crowd |
| 00:32.33 | Dovid | Corydon: do u know when the next astricon in the US will be ? I am stuck outa the country now - i wana meed my idle Mark |
| 00:32.46 | JT | Corydon-w: just seems that everyone holds a bit too much of a grudge |
| 00:32.55 | JT | on all sides |
| 00:33.00 | Corydon-w | Dovid: I'm informed that there will be another Astricon next year, in Los Angeles |
| 00:33.50 | CrazyTux | Corydon-w, what kind of large scale do you think asterisk can handle? compared to something like openser? |
| 00:33.57 | aptura | Corydon-w he invites on a informat invite |
| 00:34.28 | Dovid | The one thing I can say is that mark never bashed Sangoma openly - i know a while back that a digium employee made a comment to some one prasing sangoma on the biz list and that was the last time i saw that digium email on the list - and mark came on the list to apologize - it was the first time i ever saw him on the list (and i have been on for 2 + years now) |
| 00:34.33 | Corydon-w | JT: it's not really a grudge. Like I said, there's history. |
| 00:34.57 | mishehu | Corydon-w: history on *both* sides |
| 00:34.57 | Dovid | Corydon: do u know any dates and aprox. location ? i wana book the hotel now so i know i will be there :) |
| 00:35.11 | Corydon-w | Dovid: sorry, check the Astricon website |
| 00:35.35 | [TK]D-Fender | Grudge = insurmountable history. |
| 00:35.44 | Dovid | Corydon: there is nothing on there - other than the current one - they never seem to post about them till a few weeks b4. i wana book now so i commit myself so i goto go. |
| 00:35.48 | mishehu | [TK]D-Fender: Grudge == a really bad movie ;-) |
| 00:36.24 | Corydon-w | [TK]D-Fender: if bkw turned over a new leaf tomorrow, I think we could work with him again fairly quickly. Unfortunately, that doesn't seem likely. |
| 00:36.46 | [TK]D-Fender | mishehu : And now.. the sequel ;) |
| 00:36.52 | Dovid | lol |
| 00:37.06 | Dovid | Coming soon to theates - A troll in IRC...... |
| 00:37.11 | mishehu | [TK]D-Fender: I'm waiting for the MPAA to remind me WHY I should go to movie theatres anymore. ;-) |
| 00:37.17 | mishehu | Dovid: Return of the Troll |
| 00:37.21 | mishehu | heh |
| 00:37.21 | Dovid | hehe |
| 00:37.35 | [TK]D-Fender | Corydon-w : I never said which side(s) were unrelenting. I hope only to provide insight and a point of reflection and hope everyone works everything out. |
| 00:37.43 | Druken | mishehu: cause then your girl can give you head in the back row..... |
| 00:37.48 | Druken | why else would one go ? |
| 00:37.57 | Dovid | hey - lets not get personal here |
| 00:38.00 | Dovid | lol - |
| 00:38.06 | mishehu | Druken: man, why pay $20+ for head when I can get it from her for free ? |
| 00:38.20 | Druken | hehe |
| 00:38.24 | CrazyTux | mishehu, who said anything about paying? |
| 00:38.24 | Dovid | ever see who goes to the horror movies ? lil kids that wana get it on when mommy and daddy arent there |
| 00:38.30 | Druken | yeah, bit watch a much bigger tv.... |
| 00:38.33 | mishehu | CrazyTux: then you are a pirate! |
| 00:38.41 | Dovid | $20 for movie dumb dumb - not the ladies of the evening |
| 00:38.47 | CrazyTux | ohhh |
| 00:38.49 | wulfy814 | ok so I'm confused |
| 00:38.49 | CrazyTux | lol, oh ok |
| 00:38.50 | CrazyTux | :) |
| 00:39.07 | wulfy814 | I have compiled asterisk svn trunk, and I have no zap commands |
| 00:39.19 | wulfy814 | actually I did zaptel, libpri, and then asterisk |
| 00:39.29 | Corydon-w | CrazyTux: in answer to your earlier question, SER and Asterisk do different things, so it's unfair to try to compare the number of calls each can process |
| 00:39.37 | wulfy814 | everything seems ok , wanpipe is started, ran ztcfg -vvv prior to running asterisk |
| 00:39.40 | Corydon-w | SER is a proxy; Asterisk is a gateway |
| 00:39.57 | CrazyTux | Corydon-w, whats so different about them? They both are for processing VoIP calls, correct? |
| 00:40.00 | Corydon-w | CrazyTux: in fact, there are many people who choose to deploy both together |
| 00:40.14 | mishehu | I thought * was a PBX |
| 00:40.19 | Dovid | is openSER used for clustering ? |
| 00:40.22 | mishehu | more specifically than a gateway |
| 00:40.23 | CrazyTux | Corydon-w, any more insight on deploying both together? |
| 00:40.41 | Corydon-w | CrazyTux: a proxy relays agents. A gateway may do translations of codecs, protocols, etc. |
| 00:41.02 | Corydon-w | CrazyTux: SER is typically used in front of Asterisk for load balancing |
| 00:41.20 | aptura | what does vonage use |
| 00:41.28 | Inez | Corydon-w I am still playing with it, I installed 1.4 asterisk and still doesn't work. |
| 00:41.52 | CrazyTux | Corydon-w, so openser is a proxy? |
| 00:41.57 | Corydon-w | CrazyTux: correct |
| 00:42.23 | CrazyTux | Corydon-w, so I would setup a cluster with openser, and send off calls to my asterisk systems to implement load balancing/ |
| 00:42.37 | Dovid | correct |
| 00:42.42 | Corydon-w | CrazyTux: that's certainly one possible setup |
| 00:42.53 | CrazyTux | Corydon-w, can I private message you? |
| 00:43.10 | Corydon-w | No. I've been trying to leave and go home for at least half an hour |
| 00:43.31 | CrazyTux | Corydon-w, It would be quick, not exactly a help question... |
| 00:46.17 | saftsack | are there other softPBXs than asterisk and openpbx which are free? |
| 00:46.36 | [TK]D-Fender | saftsack : SipX |
| 00:47.01 | sqldoug | [TK]D-Fender: Do you have advice for DTMF recognition problems with a T1 into a TE412P? |
| 00:48.03 | saftsack | [TK]D-Fender, this is a softswitch ;) |
| 00:48.20 | *** join/#asterisk LakeSolon (n=blake@64-83-227-227.dhcp.stcd.mn.charter.com) |
| 00:49.22 | [TK]D-Fender | saftsack : SipX is a PBX, not just a soft-switch |
| 00:49.32 | [TK]D-Fender | sqldoug : What have you tried to date? |
| 00:49.52 | JT | there's also freeswitch, but that's not really a pbx |
| 00:50.12 | [TK]D-Fender | JT : not yet anyways. |
| 00:50.13 | saftsack | [TK]D-Fender, i use sipx and i have the same feeling as you regarding to its pbx features but officially its a softswitch ;) |
| 00:52.56 | *** join/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 00:54.59 | jtexter3 | sqldoug: I changed my 412P to use software for DTMF, and that's fixed my issues |
| 00:56.08 | *** join/#asterisk toerkeium (i=oo@201.216.206.221) |
| 00:57.33 | *** join/#asterisk rbd (n=rbd@adsl-074-229-183-112.sip.rmo.bellsouth.net) |
| 00:58.01 | *** join/#asterisk marksters_daddy (i=bill@gateway/tor/x-066650f56f85983b) |
| 00:58.13 | rbd | can anyone recommend any sip providers with colocation facilities (or colocation companies with VoIP/SIP services)? |
| 00:58.30 | *** join/#asterisk benjk (n=benjamin@f8a01-0357.din.or.jp) |
| 00:58.31 | rbd | multiple POPs around the US would be desired. I know AT&T has a SIP program but it's in beta still |
| 00:58.48 | marksters_daddy | markster is a punk ass bitch and digium blow chunks |
| 00:59.09 | *** part/#asterisk riddlebox (n=james@24-171-10-102.dhcp.stls.mo.charter.com) |
| 00:59.24 | JT | morning benjk |
| 00:59.27 | *** join/#asterisk riddlebox (n=james@24-171-10-102.dhcp.stls.mo.charter.com) |
| 01:00.15 | benjk | hi JT, how's it going? |
| 01:00.39 | JT | not bad |
| 01:00.46 | JT | you've been away for a bit i take it |
| 01:00.53 | benjk | did you get your BRI working? |
| 01:01.03 | benjk | I wasn't all too well the last couple of days |
| 01:01.14 | JT | ah sorry to hear |
| 01:01.20 | JT | yes i got my bri working |
| 01:01.24 | benjk | cool |
| 01:01.32 | JT | my original crossover cable was a failure though |
| 01:01.36 | benjk | ah |
| 01:01.42 | JT | i worked out why |
| 01:01.57 | benjk | but the drivers and zaptel.c are working fine? |
| 01:02.10 | JT | TE and NT ports have switched tx/rx ports by default |
| 01:02.21 | benjk | makes sense |
| 01:02.21 | JT | so a normal ethernet cable can patch them |
| 01:02.28 | JT | but not the second port on an OctoBRI |
| 01:02.33 | marksters_daddy | zaptel.c only works if you use sangoma |
| 01:02.49 | JT | as they pairing is incorrect and causes crosstalk interference and HDLC CRC errors |
| 01:02.53 | benjk | Sangoma do not have any BRI cards |
| 01:03.08 | saftsack | octobri from junghanns? |
| 01:03.13 | JT | so i have a revised document on how to create a patch cable for OctoBRI |
| 01:03.23 | *** join/#asterisk sahafeez (n=sahafeez@67.109.14.227.ptr.us.xo.net) |
| 01:03.53 | JT | which allows you to loop ports together |
| 01:04.15 | JT | so loop 4 ISDN2 ports |
| 01:04.28 | sahafeez | from http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions - so if i have #1 there is no way in hell to get it to work with out SER? |
| 01:06.22 | [TK]D-Fender | sahafeez : I've never run into a situation that didn't work short of a terrible router. |
| 01:06.59 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 01:07.33 | benjk | [TK]D-Fender, that doesn't mean anything, unless you can claim you have been to the less developed world with sucky internet infrastructure |
| 01:07.47 | sahafeez | hum. well i have binat setup - so on the firewall there is another ip that is translated in to the asterisk box. all traffic is allowed - if you ask for external-pbx it gets translated to internal-pbx with no rules and i cannot get it work from a sip provider |
| 01:08.17 | JT | benjk: still interested in the pinout? |
| 01:08.46 | *** join/#asterisk [hC] (n=hardcore@70.70.128.99) |
| 01:08.50 | *** join/#asterisk file2 (n=IrcNet@out.clearnet.com) |
| 01:08.50 | *** mode/#asterisk [+o file2] by ChanServ |
| 01:08.56 | file2 | moooooo |
| 01:08.57 | *** join/#asterisk Blackthorn (n=blacktho@w-l4.smyth.net) |
| 01:09.02 | [hC] | Is there a way to make polycom phones show the original caller id when doing an attended transfer? |
| 01:09.05 | ariel_ | boooooo mooooo |
| 01:09.13 | [hC] | or is this only possible if you use asterisk's "transfer" feature |
| 01:09.40 | kronic | what's a method for determining if a queue existed or if adding a caller to a queue was successful, return codes? |
| 01:09.46 | file2 | greetings from dinner! |
| 01:09.53 | Blackthorn | I setup the g729 codec today and earlier this morning set it up with a sipura spa-2000. No problems. I have a Sipura 2100 unit here at home and when ever i set the unit to g729a the * server says "no regoniziable codec" Got any suggestions? |
| 01:09.55 | wulfy814 | I'm trying to build asterisk from source (svn trun) and I'm not ending up with a chan_zap.so? what am I missing |
| 01:09.55 | citats | mmmmm dinner |
| 01:10.19 | file2 | i can see my chicken being cooked in front of me. |
| 01:10.52 | *** join/#asterisk kio (n=kio@ool-4577ae5e.dyn.optonline.net) |
| 01:10.54 | citats | file2: so which came first? the chicken or the egg? |
| 01:10.55 | *** join/#asterisk dahunter3 (n=dahunter@pool-71-110-86-116.lsanca.dsl-w.verizon.net) |
| 01:10.59 | [TK]D-Fender | wulfy814 : Downloaded and compiled zaptel before compiling *? |
| 01:11.22 | *** join/#asterisk arcanine (n=arcanine@203.82.44.179) |
| 01:11.28 | [hC] | Yikes.. Any of you guys using app_page? |
| 01:11.30 | file2 | neither. |
| 01:11.33 | ariel_ | egggggggg |
| 01:11.37 | marksters_daddy | asterisk = seg fault waiting to happen |
| 01:11.38 | [hC] | I just tried it on about 80 phones and asterisk didnt enjoy it. |
| 01:12.16 | benjk | JT, yeah, sure, it will come in handy sooner or later ;) |
| 01:12.18 | wulfy814 | [TK]D-Fender, yeah it compiled clean -- zaptel (svn trunk), libpri (svn trunk), asterisk (wanpipe prior to any of these, I have an A200) |
| 01:12.40 | JT | benjk: want me to copy and paste it to /msg, or email you an .xls or just the text? |
| 01:12.58 | arcanine | is it possible that i can monitor engaged client |
| 01:13.55 | *** join/#asterisk hansin321 (n=Miranda@c-67-190-5-42.hsd1.co.comcast.net) |
| 01:13.55 | wulfy814 | do I have to tell it anything about zaptel |
| 01:13.55 | file2 | bbs. |
| 01:13.55 | benjk | email is better, if you don't mind |
| 01:13.55 | JT | text or xls |
| 01:13.56 | wulfy814 | I had to do "./configure", usually I just use the tarball and don't need to do that |
| 01:13.56 | benjk | either format is fine |
| 01:13.56 | JT | okay |
| 01:14.07 | wulfy814 | I'm hoping to try out the hints with parked calls otherwise I wouldn't be using trunk |
| 01:14.11 | [TK]D-Fender | wulfy814 : You need to do libpri, zaptel, wanpipe, zaptel, * |
| 01:14.14 | arcanine | something like call barging |
| 01:14.43 | arcanine | the diff is i can only listen to the conversation |
| 01:14.50 | wulfy814 | [TK]D-Fender: I'll give it a shot |
| 01:15.05 | wulfy814 | do I need to do anything special to start the process over? |
| 01:15.40 | marksters_daddy | Corydon is a tool!!!!!!!!! |
| 01:15.57 | *** join/#asterisk xachen (i=justin@pdpc/supporter/student/xachen) |
| 01:16.20 | [TK]D-Fender | wulfy814 : I'd suggest trashing the extracted fodlers from your source, and then flushing your modules folders. |
| 01:16.36 | *** join/#asterisk tengulre (n=tengulre@221.11.5.182) |
| 01:16.37 | sahafeez | if i want to take a call via sip from external should i put it in a different context? |
| 01:17.18 | *** join/#asterisk Blanker (n=piovrd@ozvoip.dsl.onthenet.net) |
| 01:17.20 | wulfy814 | [TK]D-Fender: not to be stupid, but I did "svn checkout" which extracted folders should I be trashing? |
| 01:17.56 | wulfy814 | I understand the flushing of the modules folder |
| 01:17.57 | Blanker | in asterisk can agents be used for outbound dialing? |
| 01:18.07 | [TK]D-Fender | wulfy814 : redo the whole pile. Make sure to wipe all compiled modules |
| 01:20.00 | *** join/#asterisk dasenjo (n=dasenjo@208.195.215.211) |
| 01:20.30 | JT | benjk: sent |
| 01:20.38 | benjk | thanks |
| 01:23.41 | SpaceBass | So, anyone have a recommendation for a IAX or SIP provider with a starter plan (like $5/month) that allows SetCallerID? |
| 01:24.02 | kronic | how can you use an application in an if expression? Goto Execif etc...? |
| 01:24.40 | [TK]D-Fender | SpaceBass : Where? |
| 01:24.58 | SpaceBass | [TK]D-Fender, Virginia, USA |
| 01:25.08 | file2 | foodage is done. |
| 01:25.11 | Strom_C | as opposed to Virginia, France ;) |
| 01:25.15 | bsdfreak | lol |
| 01:25.33 | SpaceBass | although they should |
| 01:25.39 | bsdfreak | where in va? |
| 01:25.39 | SpaceBass | b/c its the best state...but I digress |
| 01:26.01 | SpaceBass | I'm in Richmond, need a did in 434 which is Lynchburg area |
| 01:26.04 | bsdfreak | ah |
| 01:26.10 | benjk | where's Virginia, France? |
| 01:26.15 | bsdfreak | haha |
| 01:26.16 | benjk | which departement? |
| 01:26.20 | SpaceBass | and I'm on my way to France in 3 days...but not Virginia, france |
| 01:26.33 | SpaceBass | :) |
| 01:26.48 | SpaceBass | Virginia France is right near West Virginia, Spain |
| 01:27.36 | benjk | with your geography skills, I figure you must be American |
| 01:28.12 | SpaceBass | alors, et vous? |
| 01:28.32 | bsdfreak | haha |
| 01:28.38 | sahafeez | if i have NAT=yes set and sip show peers shows N for nat what am i missing |
| 01:28.53 | SpaceBass | sahafeez, whats the issue? |
| 01:29.49 | sahafeez | i am trying to get SIP to work between my asterisk box - nat'd and an external SIP provider. the NAT is setup as one to one and letting all traffic thru |
| 01:30.26 | *** join/#asterisk lule (i=lule@host62.200-117-164.telecom.net.ar) |
| 01:30.47 | SpaceBass | sahafeez, 1:1 NAT, so no ports to open.....have you done a tcpdump to and looked for the registration? |
| 01:30.47 | *** part/#asterisk lule (i=lule@host62.200-117-164.telecom.net.ar) |
| 01:31.20 | sahafeez | 18:26:58.194615 65.175.129.133.5060 > 67.109.14.228.5060: udp 1132 |
| 01:31.21 | sahafeez | 18:26:58.195197 67.109.14.228.5060 > 65.175.129.133.5060: udp 702 (DF) |
| 01:31.21 | sahafeez | 18:26:58.307261 65.175.129.133.5060 > 67.109.14.228.5060: udp 417 |
| 01:31.21 | sahafeez | 18:27:00.921001 67.109.14.228.5060 > 66.237.65.67.5060: udp 483 (DF) |
| 01:31.21 | sahafeez | 18:27:00.921109 67.109.14.228.5060 > 66.237.65.67.5060: udp 483 (DF) |
| 01:31.31 | sahafeez | i am 228 |
| 01:31.37 | aptura | dont dump that here |
| 01:32.26 | *** join/#asterisk MikeJ (n=mikej@d14-69-8-30.try.wideopenwest.com) |
| 01:32.30 | sahafeez | sip show peers show status ok |
| 01:32.37 | tengulre | hi,all |
| 01:33.17 | SpaceBass | sahafeez, what about TRP? |
| 01:33.21 | SpaceBass | or RTP rather |
| 01:33.30 | sahafeez | never see it |
| 01:33.36 | tengulre | anybody can tell me how to play the /var/lib/asterisk/moh/fpm-sunshine.wav in extensions.conf? |
| 01:34.30 | SpaceBass | sahafeez, 5060 is only for SIP registration. Audio is sent over RTP ports (rtp.conf) |
| 01:34.52 | Blackthorn | rtp? i thought it was udp? |
| 01:34.58 | [TK]D-Fender | tengulre : Playback(/var/lib/asterisk/moh/fpm-sunshine) |
| 01:35.09 | sahafeez | it never makes it that far. dialing the nubmer gives the the info you see above and a busy. i never see anything in asterisk -vvvv |
| 01:35.16 | SpaceBass | rtp packets are sent over UDP |
| 01:35.25 | Blackthorn | oh :P |
| 01:35.57 | SpaceBass | sahafeez, do a sip show registry |
| 01:36.42 | sahafeez | SpaceBass, get nothing |
| 01:36.57 | tengulre | [TK]D-Fender: thanks , I try it now.. :) |
| 01:37.53 | MikeJ | so any word when 1.4 is going to release? |
| 01:37.54 | SpaceBass | sahafeez, do a tcpdump to a file, then reload asterisk ... then open the file in etheral or something and look for the registration |
| 01:38.27 | SpaceBass | sahafeez, regardless, once it registers you'll have to have RTP ports forwarded as well...or at least make sure they are clearing your 1:1 nat |
| 01:38.54 | sahafeez | bi-nat with pass all both ways at this point. |
| 01:39.16 | benjk | sahafeez, build a tunnel |
| 01:39.30 | sahafeez | not an option |
| 01:39.55 | benjk | why not? |
| 01:40.05 | benjk | its far easier |
| 01:40.19 | sahafeez | question - if add externip=bla in sip.conf my internal calls - do not ring - do i need to be in a differnet context |
| 01:40.27 | sahafeez | benjk: sip provier |
| 01:40.30 | SpaceBass | benjk, where in the world are you? |
| 01:40.31 | sahafeez | provider |
| 01:40.37 | sahafeez | both not my box |
| 01:40.44 | sahafeez | or netowkr |
| 01:40.46 | benjk | JP |
| 01:40.49 | SpaceBass | meaning quite literally |
| 01:41.14 | kronic | anyone how could I for example do a While(Queue(queue${INC})) |
| 01:41.17 | SpaceBass | sahafeez, you can also add a localnet |
| 01:41.31 | sahafeez | did. same issue. |
| 01:41.32 | kronic | using the return code as the condition |
| 01:41.52 | SpaceBass | Japan? very cool |
| 01:42.08 | SpaceBass | and on that note, I'm getting back to some important tv watching |
| 01:42.27 | benjk | I would never ever use a provider with a server behind NAT |
| 01:42.30 | tengulre | [TK]-D-Fender: thanks , It is successfull!! |
| 01:42.33 | sahafeez | SpaceBass: real quick - http://rafb.net/paste/results/oaHyiC31.html |
| 01:43.04 | tengulre | Japenese girl!! lol. :-D) |
| 01:44.00 | SpaceBass | sahafeez, looks good |
| 01:44.12 | benjk | how likely is it that there is any Japanese girl called Benjamin? |
| 01:44.23 | sahafeez | yes, and with the external ip in there i cannot dial any internal ext. |
| 01:55.23 | [TK]D-Fender | sahafeez : Add "canreinvite=no", and "nat=yes" to [general] |
| 01:55.32 | Blackthorn | I setup the g729 codec today and earlier this morning set it up with a sipura spa-2000. No problems. I have a Sipura 2100 unit here at home and when ever i set the unit to g729a the * server says "no regoniziable codec" Got any suggestions? |
| 01:56.06 | sahafeez | Fender: hum, not everything is nat tho so do i still add nat=yes to general |
| 01:57.01 | benjk | sahafeez, a provider who has the server behind NAT is not to be trusted |
| 01:57.16 | sahafeez | no, i am nat'd. not him |
| 01:57.27 | [TK]D-Fender | sahafeez : Thats what "localnet" is for |
| 01:57.32 | benjk | earlier you said it was NAT at both ends |
| 01:57.36 | [TK]D-Fender | sahafeez : so it knows when to use it. |
| 01:57.56 | sahafeez | yah. did not work. one sec. try again with your additions |
| 02:01.47 | Blackthorn | Not sure this helps, havn't been following the convo closely but: I was told to always let * handle the nat and not the devices. |
| 02:02.12 | Blackthorn | So the devices are set to nat=no and asterisk in the sip.conf file for each sip device i have nat=yes and keepalive=yes. |
| 02:02.12 | sahafeez | Fender: http://rafb.net/paste/results/sbf2T457.html |
| 02:02.46 | Blackthorn | wait. might not be keepalive=yes |
| 02:02.59 | sahafeez | so if i add the exterip command my internal exten to exten does not work. it dials with no sound and the other side does not ring |
| 02:03.04 | [TK]D-Fender | ocalnet=192.168.22.0/255.255.255.0 |
| 02:03.04 | Blackthorn | something else but basicly means sends the keepalive packets |
| 02:03.14 | sahafeez | same thing set that way |
| 02:03.21 | [TK]D-Fender | sahafeez : "there is no "localmask" its all on 1 line |
| 02:03.35 | sahafeez | just thought i would try it the way you see it as i was in the wiki both ways |
| 02:03.42 | [TK]D-Fender | sahafeez : Also would be a good ide to add "qualify=yes" to [general] as well |
| 02:04.25 | Blackthorn | ahh yes tahts it. qualify=yes :))) |
| 02:05.09 | Blackthorn | in my end units I tell it to register every 5 to 10 minutes. in case the link drops (all my voip units are on wireless) |
| 02:06.27 | sahafeez | Fender: still does not work. |
| 02:06.36 | sahafeez | i lose my internal calls. |
| 02:06.56 | [TK]D-Fender | sahafeez : pastebin your new config, and show me a call. |
| 02:07.04 | sahafeez | one sec.. |
| 02:07.11 | [TK]D-Fender | sahafeez : and the peer config for that internal phone. |
| 02:08.16 | sahafeez | the only call that works now is a call to a external phone - ie, it is not on the local net but connect via vpn. desk to desk stopped working |
| 02:09.05 | [TK]D-Fender | sahafeez : Pastebin MORE please... |
| 02:09.18 | sahafeez | working on it. |
| 02:09.46 | *** join/#asterisk shellshark (n=x86@get.hooked.on.voip.with.shellshark.net) |
| 02:10.48 | file | moo |
| 02:11.30 | sahafeez | Fender: http://rafb.net/paste/results/MA6MPC84.html |
| 02:13.06 | [TK]D-Fender | sahafeez : Look at the default IP of [4211]. Sure as hell doesn't look like a match for what you claimed your localnet looked like now does it? |
| 02:13.49 | sahafeez | fuck. i renumbered and ... ahhah! one sec.. |
| 02:14.00 | [TK]D-Fender | sahafeez : Also note that you even used that same default IP for both [4211] and [4212]. Also bad. Are you awake? |
| 02:14.22 | sahafeez | the ip is the ip of the pbx. |
| 02:14.31 | [TK]D-Fender | sahafeez : You'll ditch the default IP altogether if you know whats good for you... |
| 02:14.41 | sahafeez | it has worked that way and i was told to set it up that way a year ago. |
| 02:15.12 | [TK]D-Fender | sahafeez : localnet=192.168.22.0/255.255.255.0 does NOT match 192.168.119.4 |
| 02:15.30 | sahafeez | wow! channg the 22 to 119 fixed that part of it. damn i feel dumb. |
| 02:15.32 | [TK]D-Fender | sahafeez : i think you'd better take a much closer look at what you're doing. |
| 02:15.43 | sahafeez | you are saying i should dump the default ip? |
| 02:15.45 | [TK]D-Fender | sahafeez : And get RID of the "defaultip" lines. |
| 02:15.52 | sahafeez | k. will do. |
| 02:16.10 | [TK]D-Fender | sahafeez : Yes. let your phones simply register like they should. Never assume ANYTHING. |
| 02:17.03 | xachen | Is JerJer at Astricon? |
| 02:17.18 | sahafeez | that being said, i still do not see anything beyond the 5060 ports on the inbound sip call. just did it again with a tcpdump greped for the external ip of the pbx (.228) |
| 02:17.23 | sahafeez | and thank you btw |
| 02:17.41 | *** join/#asterisk saftsack (n=saftsack@p54A7F0D0.dip.t-dialin.net) |
| 02:22.02 | wulfy814 | [TK]D-Fender: it worked! not sure what I was doing wrong - I nuked it all and redownloaded |
| 02:22.07 | *** join/#asterisk nowwhat (n=ARF@24-148-32-14.nwb-bsr1.chi-nwb.il.cable.rcn.com) |
| 02:22.15 | wulfy814 | then followed your instructions libpri - zaptel - wanpipe - zaptel - * |
| 02:23.15 | nowwhat | Anyone have a second to give me a hint on a problem? My google skills are falling short. |
| 02:23.41 | sahafeez | Fender: fixed everything you corrected. thanks. here is were i am at http://rafb.net/paste/results/upceF124.html |
| 02:24.49 | file | good golly I missed some activity |
| 02:25.33 | aptura | file :) |
| 02:25.43 | aptura | you mean the resteraunt talk? |
| 02:25.53 | file | yeah, catching up on my -users mail |
| 02:26.16 | sahafeez | Fender: never see anything come in in asterisk - get a busy. the firewall - binat is wide open and i have checked it 10 times. asterisk i am not so good at but the firewall i know in my sleep. |
| 02:26.59 | aptura | btw came across a sweet wall mount pbx like atx case only issue is its for eatx motherboard. Its tough though. I bet its a little expensive since the company does millitary contracts. |
| 02:27.26 | [TK]D-Fender | sahafeez : What are you forwarding to *? |
| 02:28.21 | sahafeez | not forwarding - doing binat - one to one translation letting everything thru. the rules are in slut mode right now. |
| 02:28.42 | *** join/#asterisk sbingner (n=thanotos@pdpc/supporter/sustaining/sbingner) |
| 02:29.33 | [TK]D-Fender | sahafeez : What range? |
| 02:30.00 | sahafeez | no range - everything is forwarded - wide open - anything for external.pbx goes to internal.pbx |
| 02:30.04 | [TK]D-Fender | sahafeez : And what happened to your phones? |
| 02:31.41 | sahafeez | sorry do not follow. the other stuff works now. this all started because i needed a number on the east coast. so i got a sip provider. i setup like you see in the paste, i dail the number and see traffic via tcpdump for the 5060 ports then nothing and get a busy on the call. using my cell to dial the 202 number |
| 02:32.00 | nowwhat | anyone know where a wiring diagram is for fxs - RJ45 to RJ11? |
| 02:32.52 | sbingner | ... |
| 02:33.02 | sbingner | middle pair to middle pair |
| 02:33.06 | [TK]D-Fender | sahafeez : So everything works now? |
| 02:34.11 | nowwhat | sbingner, thank you. Silly question I know, just being over carefull. |
| 02:34.18 | sahafeez | sorry no. the internal stuff works now with the exterip cmd in there now you pointed out my typo and i removed the default ip stuff. the inbound sip still does not work. |
| 02:34.35 | sahafeez | Fender: http://rafb.net/paste/results/upceF124.html |
| 02:35.24 | sahafeez | that is the cfg and what asterisk see and the tcpdump. i see the traffic for the 5060 port, then nothing. i get a fast busy on the inbound call from my cell. there is no debug in asterisk |
| 02:36.16 | sahafeez | there are no dropped packets on the firewall - i am watching in real time - and there is no rule between the external.pbx address and the internal.pbx * box cept let it all hang out. |
| 02:37.42 | [TK]D-Fender | sahafeez : I've never seen an inbound setup like yours. Try insecure = very. if that doesn't work, add a catch-all exten to your [incoming] and see if anything lands there. |
| 02:39.39 | sahafeez | Fender: would i not see anything in the debug even if my extensions.conf is screwed up? i will try the very |
| 02:54.46 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 02:55.29 | *** join/#asterisk Buglouse (n=SourceRa@66.97.121.210) |
| 02:56.24 | iCEBrkr | http://www.cyberdyne.org/~icebrkr/ss.png |
| 02:56.27 | iCEBrkr | lets go back in time. |
| 02:56.30 | iCEBrkr | 1997 |
| 02:58.28 | iCEBrkr | back when shit was K-Rad |
| 03:02.02 | *** join/#asterisk Buglouse (n=SourceRa@66.97.121.210) |
| 03:05.06 | *** join/#asterisk Buglouse (n=SourceRa@66.97.121.210) |
| 03:08.24 | tengulre | anybody can tell me a web photo manager? |
| 03:08.48 | MikeJ | google |
| 03:11.47 | *** join/#asterisk nowwhat (n=ARF@24-148-32-14.nwb-bsr1.chi-nwb.il.cable.rcn.com) |
| 03:12.45 | nowwhat | anyone had experance with "ZT_CHANCONFIG failed on channel 2: Invalid argument (22)" From the ztcfg util? |
| 03:20.53 | *** join/#asterisk hansin321 (n=Miranda@c-67-190-5-42.hsd1.co.comcast.net) |
| 03:21.06 | sqldoug | nowwhat: what's your /etc/zaptel.conf look like? It looks like it's getting a value it doesn't like. |
| 03:24.01 | nowwhat | pretty simple streight off digium #2 x100p fxsks=1-2 |
| 03:24.12 | nowwhat | trying not to spam |
| 03:24.18 | nowwhat | fxoks=3-4 |
| 03:24.37 | nowwhat | that is for tdm400p with 2 fxs cards |
| 03:31.35 | sqldoug | nowwhat: How about the uncommented lines up to that point? |
| 03:31.47 | sqldoug | nowwhat: sorry - gotta bail. My ride's leaving. |
| 03:33.19 | *** mode/#asterisk [-bbbb Zeynep-[23f]!*@* *!*@672db68b04f7a2c2.session.tor %Kk_`away!*@* %A-Turin[play]!*@*] by Corydon76-home |
| 03:34.35 | *** mode/#asterisk [-bbb %Lucas_Fernando!*@* %JasonBecker!*@* %ms34!*@*] by Corydon76-home |
| 03:35.47 | MikeJ | wheee |
| 03:37.35 | JT | as in more detailed than TFOT :) |
| 03:37.40 | JT | it seems a bit quirky |
| 03:37.43 | Corydon76-home | JT: you mean other than sample.call ? |
| 03:38.22 | citats | JT: dont think it gets more detailed than pbx_spool.c :) |
| 03:45.27 | *** join/#asterisk cyscapes (n=cyscapes@ip70-176-173-30.ph.ph.cox.net) |
| 03:49.20 | nowwhat | anyone know how Digium is on RMAs of almost discontiuned hardware (x100p)? |
| 03:53.04 | *** join/#asterisk bmg505 (n=leon@c1-54-6.rndf.isadsl.co.za) |
| 03:53.23 | [TK]D-Fender | nowwhat : What do you mean "almost"? They've been discontinued for YEARS |
| 03:53.46 | denon | yeah, that's what I was thinking too.. |
| 03:53.57 | denon | almost antique is maybe closer to the truth |
| 03:54.58 | nowwhat | heh |
| 03:55.18 | nowwhat | wow, has it really been that long. That's kinda sad |
| 03:55.19 | denon | if you actually wanted an x100p though, I suspect it'd be easier and cheaper just to buy another one, than to rma it |
| 03:57.00 | *** join/#asterisk k-man_ (n=jason@unaffiliated/k-man) |
| 03:57.04 | k-man_ | hello |
| 03:58.48 | *** join/#asterisk BigBadHoss (n=hoss@65.4.26.233) |
| 03:59.27 | *** join/#asterisk techie (n=techie@c-67-182-22-174.hsd1.ca.comcast.net) |
| 04:01.05 | kilobit2001 | what does the answer() command do? |
| 04:01.15 | BigBadHoss | answers a call |
| 04:01.26 | kilobit2001 | what if answer() is issued twice? |
| 04:01.31 | BigBadHoss | read 'the book' |
| 04:01.54 | k-man_ | is it possible to have a setup where you dial out on your PC and it transfers your call to your voip handset automaticaly after it dialed out? |
| 04:02.15 | BigBadHoss | yes |
| 04:02.16 | kilobit2001 | BigBadHoss-- You Read the book. |
| 04:02.34 | BigBadHoss | haha |
| 04:02.39 | k-man_ | BigBadHoss, what would that be called and where would i read up on it? |
| 04:02.45 | BigBadHoss | i think the second does nothing |
| 04:02.57 | BigBadHoss | well you can always write a dialplan for it |
| 04:03.13 | k-man_ | dialplan? |
| 04:04.27 | BigBadHoss | im not sure exactly how it would be done, but if you can think it, its possible |
| 04:04.40 | k-man_ | ok |
| 04:04.41 | k-man_ | thanks |
| 04:05.03 | BigBadHoss | lemme get you a linjk |
| 04:05.22 | BigBadHoss | http://www.nufone.net/downloads/asteriskdocs/AsteriskTFOT.zip |
| 04:05.32 | BigBadHoss | actualkly |
| 04:05.33 | BigBadHoss | http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 04:05.38 | JT | ~thebook |
| 04:05.43 | jbot | hmm... thebook is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 04:05.44 | BigBadHoss | read parts about the dialplan |
| 04:06.00 | BigBadHoss | or everything if you have time |
| 04:06.27 | k-man_ | your talking to me right BigBadHoss? |
| 04:06.43 | BigBadHoss | yes |
| 04:06.55 | k-man_ | thanks |
| 04:07.15 | BigBadHoss | youll see what i mean by endless possibiliteis if you read the book |
| 04:08.08 | FuriousGeorge | anyone using voicepulse |
| 04:08.13 | *** join/#asterisk sahafeez (n=sahafeez@ip68-6-223-92.sd.sd.cox.net) |
| 04:08.23 | FuriousGeorge | their suggested config is retarded |
| 04:08.49 | FuriousGeorge | dial/vocepule01/${EXTEN} gives me a no authority found |
| 04:08.50 | FuriousGeorge | of course |
| 04:10.05 | BigBadHoss | heheh |
| 04:10.12 | BigBadHoss | they stole my mony |
| 04:10.16 | *** join/#asterisk topping (n=topping@207.47.6.182.static.nextweb.net) |
| 04:10.24 | FuriousGeorge | BigBadHoss: how so |
| 04:10.38 | BigBadHoss | well, i was signing up |
| 04:10.52 | BigBadHoss | it kept saying cannot authorize card or some crap |
| 04:11.10 | BigBadHoss | so i kept trying different combinations of addresses, phone numbers, etc |
| 04:11.25 | BigBadHoss | they charged me the amount i was tryibng to charge every time |
| 04:11.30 | FuriousGeorge | lol |
| 04:11.35 | BigBadHoss | it took me like 3 months to see that money again |
| 04:11.40 | BigBadHoss | it was like $200 |
| 04:11.42 | *** join/#asterisk scurb (i=scurb@216.138.122.150) |
| 04:11.47 | FuriousGeorge | anyone know if multiparking is scheduled to be introduced in the 1.4 release cycle? |
| 04:11.52 | BigBadHoss | i called them and they just bullshiited about |
| 04:12.00 | FuriousGeorge | BigBadHoss: did you ask for a refund? |
| 04:12.06 | BigBadHoss | yeah |
| 04:12.15 | BigBadHoss | this was a while back though |
| 04:12.19 | JT | err |
| 04:12.24 | JT | it was a credit card |
| 04:12.31 | JT | why didn't you just reverse payment |
| 04:12.47 | BigBadHoss | it was a debit/credit |
| 04:12.51 | BigBadHoss | not as easy |
| 04:13.00 | JT | i see |
| 04:13.16 | FuriousGeorge | i know at my bank they will take your word for it and put the burden of proof on the seller |
| 04:13.23 | FuriousGeorge | if you dispute a charge |
| 04:13.28 | JT | i thought it didn't matter if it was debit, as long as it was from a credit card company, eg visa/mastercard/amex etc |
| 04:13.42 | FuriousGeorge | yeah, the debit/credit deals are also backed by a credit company |
| 04:13.53 | BigBadHoss | well i eventually got the money back, at least |
| 04:14.02 | FuriousGeorge | you still use them? |
| 04:14.02 | BigBadHoss | but i wont fool with them again |
| 04:14.06 | JT | yeah i'm talking about stuff that works through the credit card system |
| 04:14.11 | BigBadHoss | i use voipstreet now |
| 04:14.13 | JT | not eftpos debit cards |
| 04:14.24 | BigBadHoss | along with voipjet |
| 04:14.25 | *** join/#asterisk De_Mon (n=de_mon@fl-69-69-136-104.dyn.embarqhsd.net) |
| 04:14.39 | FuriousGeorge | BigBadHoss: how come you stopped using them in the end? behind this double billing thing |
| 04:15.09 | BigBadHoss | well, they said something was wrong with my address, but my card works everywhere else |
| 04:15.17 | BigBadHoss | so i called them |
| 04:15.25 | BigBadHoss | they still wouldnt do anything for me |
| 04:16.09 | *** join/#asterisk topping (n=topping@207.47.6.182.static.nextweb.net) |
| 04:17.48 | FuriousGeorge | wierd |
| 04:18.12 | FuriousGeorge | im already a little annoyed that i have to contact them bc their configuration instruction is no good |
| 04:18.18 | FuriousGeorge | but thats par for the course, it seems |
| 04:21.08 | BigBadHoss | get voipstreet |
| 04:21.25 | BigBadHoss | they have been great to me so far |
| 04:22.16 | BigBadHoss | not the cheapeast |
| 04:22.28 | BigBadHoss | but i get under 10 minute support response times |
| 04:22.34 | BigBadHoss | from the helpdesk software |
| 04:22.46 | BigBadHoss | they even offered to ssh in and check it out, for free |
| 04:27.57 | *** join/#asterisk atapi (n=virgill4@c-65-34-182-167.hsd1.fl.comcast.net) |
| 04:28.12 | *** part/#asterisk k-man_ (n=jason@unaffiliated/k-man) |
| 04:31.25 | *** join/#asterisk drcode (n=user1@87.69.59.186.cable.012.net.il) |
| 04:31.27 | drcode | hi all |
| 04:34.45 | drcode | hi all |
| 04:35.00 | drcode | can I put sip users into mysql or sqlite? |
| 04:41.48 | *** join/#asterisk techie (n=techie@c-67-182-22-174.hsd1.ca.comcast.net) |
| 04:49.02 | *** join/#asterisk kgpsathish (n=sathish@61.246.251.18) |
| 04:50.35 | *** part/#asterisk kgpsathish (n=sathish@61.246.251.18) |
| 04:53.21 | BigBadHoss | night all |
| 04:59.27 | *** part/#asterisk drcode (n=user1@87.69.59.186.cable.012.net.il) |
| 05:01.00 | *** join/#asterisk atapi (n=virgill4@c-65-34-182-167.hsd1.fl.comcast.net) |
| 05:05.29 | *** join/#asterisk marcus2 (i=marcus@atlantis.outer.org) |
| 05:05.53 | marcus2 | so i have a strange problem with 1.4.0b3 |
| 05:06.06 | marcus2 | AGI seems somewhat broken |
| 05:10.30 | *** join/#asterisk inv_Arp (n=junya@c-71-206-88-100.hsd1.fl.comcast.net) |
| 05:11.59 | tengulre | afternoon here |
| 05:12.22 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 05:12.22 | *** mode/#asterisk [+o mog] by ChanServ |
| 05:12.55 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 05:12.55 | *** mode/#asterisk [+o russellb] by ChanServ |
| 05:13.57 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 05:15.49 | *** part/#asterisk atapi (n=virgill4@c-65-34-182-167.hsd1.fl.comcast.net) |
| 05:21.28 | *** join/#asterisk qw3rty (n=qw3rty@c-71-57-75-55.hsd1.il.comcast.net) |
| 05:21.48 | *** join/#asterisk inv_Arp (i=junya@c-71-206-88-100.hsd1.fl.comcast.net) |
| 05:22.29 | qw3rty | I need to take a number that is dialed and replace it with a different number before it goes out of a trunk. Can I do that in the dial rules of my trunk? |
| 05:25.42 | citats | qw3rty: exten => 5095551234,1,Dial(Zap/g1/5098675309) |
| 05:26.05 | qw3rty | citats, thankx I will give that a try |
| 05:26.52 | citats | qw3rty: plenty of ways to skin the cat, thats easiest way i could think of if you've only got a few numbers to do it on and dont mind loading up your extensions.conf with them |
| 05:27.17 | tengulre | anybody know where have Dialogic card driver for asterisk ? |
| 05:27.23 | qw3rty | actually it's just one number so this will work great |
| 05:27.36 | *** join/#asterisk ShipHead (i=ShipHead@gateway/tor/x-5e0c52d70aec1172) |
| 05:28.19 | citats | tengulre: afaik you have to contact digium sales for that |
| 05:28.40 | tengulre | citats: is it not free? |
| 05:28.52 | citats | tengulre: nope, I believe its $10 per channel |
| 05:29.08 | tengulre | citats: thanks! |
| 05:29.32 | russellb | no dialogic channel driver has been released as far as i know ... |
| 05:29.56 | russellb | in any case, the sales team would know |
| 05:30.06 | tengulre | russellb: thanks |
| 05:30.25 | JT | i've heard it mentioned here that there's a pay driver available, too |
| 05:30.45 | russellb | There has been one in the works for a while, I just don't think it has been released |
| 05:32.45 | qw3rty | citats, thanks again that works perfectly |
| 05:52.23 | *** join/#asterisk DarKnesS_WolF (n=wolf@62.114.197.57) |
| 05:53.56 | *** join/#asterisk ixx (i=foobar@cpe-70-112-73-77.austin.res.rr.com) |
| 05:54.16 | *** join/#asterisk Mattwj2005 (n=Matt@76.17.131.68) |
| 05:54.39 | Mattwj2005 | Good evening everyone :) |
| 05:56.32 | tengulre | good afternoon. |
| 05:56.33 | tengulre | here |
| 05:58.12 | Mattwj2005 | :) |
| 05:58.54 | Mattwj2005 | where are you at tenguire? Europe I am guessing |
| 06:03.19 | *** join/#asterisk Jubei_ (n=jubei@jubei.noc.tuc.gr) |
| 06:03.36 | tengulre | Mattwj2005: Asia! |
| 06:03.46 | Mattwj2005 | oh okay |
| 06:03.58 | Mattwj2005 | I was trying to guess from what I knew about timezones |
| 06:04.00 | Mattwj2005 | :) |
| 06:04.40 | Mattwj2005 | so what are you working on tonight tenguire? |
| 06:05.18 | tengulre | Mattwj2005: It is 14:05pm here. |
| 06:05.25 | Jubei_ | :) |
| 06:05.32 | Mattwj2005 | 1:05 am here! |
| 06:05.44 | Jubei_ | 09.00 am here :) |
| 06:06.04 | Mattwj2005 | wow everyone is from around the world :) |
| 06:06.09 | Jubei_ | indeed :D |
| 06:06.14 | tengulre | yes! |
| 06:06.20 | tengulre | I m chinese. |
| 06:06.25 | Jubei_ | <-- Greece |
| 06:06.33 | tengulre | a beginner with asterisk. |
| 06:06.40 | tengulre | :) |
| 06:06.45 | Mattwj2005 | American....don't hate us because of our bad leaders ;) |
| 06:06.53 | Jubei_ | :) |
| 06:06.57 | *** join/#asterisk humba1 (n=aaaazz@pool-70-20-23-115.bstnma.fios.verizon.net) |
| 06:07.02 | tengulre | <PROTECTED> |
| 06:07.20 | Mattwj2005 | I know some about Asterisk....not an expert though |
| 06:07.27 | humba1 | having trouble with the findme/followme example at http://www.voip-info.org/wiki/view/Asterisk+tips+findme |
| 06:07.30 | tengulre | I have a friend is American too. |
| 06:07.31 | humba1 | with sip, the call gets bridged before the callee can screen it |
| 06:08.23 | tengulre | Mattwj2005, which linux release are you runnig asterisk on that? |
| 06:08.34 | tengulre | s/runnig/running |
| 06:08.34 | Jubei_ | hmbal, any idea why I might be getting "Oct 23 11:57:06 NOTICE[2913]: chan_sip.c:11084 handle_request_register: Registration from '<sip:test@host.domain>' failed for '192.168.1.1' - Not a local SIP domain" |
| 06:08.38 | Mattwj2005 | well right now I am trying to rebuild my server |
| 06:08.48 | Mattwj2005 | I had debian before....now I am trying gentoo :) |
| 06:09.20 | tengulre | Mattwj2005, I running debian . |
| 06:09.46 | Mattwj2005 | debian is really good |
| 06:10.20 | tengulre | Mattwj2005, less people using linux or Unix in my country, but I very very like linux. |
| 06:10.25 | tengulre | yes |
| 06:10.25 | humba1 | Jubei_, sounds like the username isn't defined |
| 06:10.28 | Mattwj2005 | I have used a lot of linux distros....debian and ones based off of that are my favorite....I decided to try a hard one |
| 06:10.38 | humba1 | or bad password |
| 06:10.50 | tengulre | Mattwj2005, :) |
| 06:10.52 | Jubei_ | hmmm |
| 06:11.31 | Mattwj2005 | yeah it is good....Windows is very popular in the US....Linux is only like the 3rd most popular operating system |
| 06:11.46 | tengulre | but my english is low. most of article need translate tool when I read it. :( |
| 06:12.22 | Mattwj2005 | yeah....that has to be difficult |
| 06:13.13 | tengulre | Mattwj2005: how to learning english, lol? |
| 06:13.26 | Mattwj2005 | ummmm |
| 06:13.31 | Mattwj2005 | I am not sure! |
| 06:13.44 | Mattwj2005 | I was born here....so I am not much help |
| 06:14.11 | Jubei_ | humba1, my soft spi client, when I try to set up an account for registration has: user/password and then in a "more options" section has "Authentication Login" and "Realm/Domain". any idea what those are? |
| 06:14.46 | Jubei_ | soft sip* |
| 06:16.07 | tengulre | Mattwj2005, that only a joke. [ WELCOME TO MY COUNTRY IF YOU WANT ] I will working now, or else our boss will kill us. lol |
| 06:16.24 | Mattwj2005 | lol |
| 06:16.29 | Mattwj2005 | I know the feeling! :) |
| 06:16.44 | *** join/#asterisk lorinc (n=ang@caracas-1337.adsl.interware.hu) |
| 06:18.46 | JT | tengulre: are you in china? |
| 06:21.19 | tengulre | JT: yeah! |
| 06:21.43 | JT | didn't know the great firewall of china allowed you to irc :) |
| 06:22.28 | tengulre | JT, :( why say that? |
| 06:23.08 | JT | sorry if it made you uncomfortable with me saying that |
| 06:24.03 | *** join/#asterisk Aurs (n=Aurs@host-81-191-123-189.bluecom.no) |
| 06:24.07 | Mattwj2005 | I know America has a lot of problems.... |
| 06:24.08 | Mattwj2005 | :) |
| 06:24.13 | tengulre | JT: not at all! my country is very great and have long long history |
| 06:25.33 | JT | tengulre: can i ask if you're at least aware of the firewall's presence? |
| 06:26.11 | Mattwj2005 | so what is everyone working on? |
| 06:26.29 | tengulre | JT: of course if I known. |
| 06:26.37 | Jubei_ | Mattwj2005, trying to get my sip client to register with asterisk. so simple, yet giving me such a hard time |
| 06:26.49 | Mattwj2005 | hmmm |
| 06:27.13 | Mattwj2005 | how do you have asterisk configured and what softphone? |
| 06:27.51 | JT | tengulre: ah ok, sometimes i can't be sure, heh, we don't really know in the west what the average chinese citizen does, or does not know |
| 06:28.11 | *** part/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 06:28.38 | shellshark | tengulre: greetings from the USA :) |
| 06:30.36 | Jubei_ | Mattwj2005, eer.. asterisk 1.2.12.1 with just sip and a test greeting waiting to be played, then my softphone is ekiga (linux) http://pastebin.ca/218518 |
| 06:30.42 | tengulre | JT: sorry,I don't know, |
| 06:31.13 | *** join/#asterisk rokis (n=rokera@201.132.105.50) |
| 06:32.24 | Mattwj2005 | what are the fields in ekiga? |
| 06:32.39 | JT | tengulre: just that i saw a documentary on tv the other day saying a lot of chinese university students did not know about tiananmen square, which made me curious |
| 06:33.03 | Jubei_ | Mattwj2005, registrar (asterisk's hostname) user and password :) |
| 06:33.17 | *** join/#asterisk tetsuzan (n=raizen@200.180.124.12) |
| 06:33.18 | tengulre | JT: I don't know too. :( |
| 06:33.41 | tetsuzan | hi all |
| 06:33.57 | Mattwj2005 | hmm |
| 06:34.19 | Mattwj2005 | I don't know what to say....it looks obvious....but I don't know what to saw |
| 06:34.50 | Jubei_ | Mattwj2005, I know it's crazy :/ |
| 06:35.01 | *** part/#asterisk Jubei_ (n=jubei@jubei.noc.tuc.gr) |
| 06:35.05 | *** join/#asterisk Jubei_ (n=jubei@jubei.noc.tuc.gr) |
| 06:35.15 | tengulre | JT: because I have gone there never, I think It is not necessary . |
| 06:35.29 | JT | okay |
| 06:35.39 | Mattwj2005 | account name I am not sure |
| 06:35.41 | *** join/#asterisk alerios (n=alerios@190.24.98.181) |
| 06:35.58 | Mattwj2005 | register I think is the extension |
| 06:36.02 | Jubei_ | account name is just something descriptive, nothing to do with authentication |
| 06:36.03 | Mattwj2005 | username is the username filed |
| 06:36.22 | Mattwj2005 | password is the password field |
| 06:36.23 | Jubei_ | password is the secret field |
| 06:36.52 | Mattwj2005 | exactly |
| 06:37.20 | *** join/#asterisk af_ (n=af@ip-173-17.sn1.eutelia.it) |
| 06:37.24 | Mattwj2005 | authentation field should be the username |
| 06:37.25 | tengulre | JT: I feel our education is failed. most of sutdents don't know what OS mean? especially female. |
| 06:37.39 | JT | what is os? |
| 06:37.43 | JT | operating system? |
| 06:38.04 | tengulre | JT: yes. |
| 06:38.06 | Mattwj2005 | realm/domain should be the domain name or IP address of the Asterisk server |
| 06:38.12 | JT | ah |
| 06:38.17 | JT | don't worry |
| 06:38.27 | Jubei_ | Mattwj2005, that's exactly what everything is :D |
| 06:38.31 | JT | most people not into computers here don't really know what it means |
| 06:38.47 | Mattwj2005 | try setting the registrator as the extension |
| 06:39.14 | Jubei_ | it's the "Not a Local SIP domain" that's troubling me |
| 06:40.09 | Mattwj2005 | try your extension@thedomain |
| 06:40.35 | tengulre | JT: but I m point the computer & science university. |
| 06:40.39 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 06:41.03 | JT | tengulre: can you receive private messages? i tried messaging you |
| 06:41.27 | JT | people at computer & science university don't know what an operating system is? |
| 06:41.28 | Mattwj2005 | Jubel I wish I was more help |
| 06:42.08 | tetsuzan | wo |
| 06:42.11 | *** part/#asterisk techie (n=techie@c-67-182-22-174.hsd1.ca.comcast.net) |
| 06:42.33 | tetsuzan | i have 5 sip lines |
| 06:42.43 | tetsuzan | and i want to round-robin |
| 06:42.49 | tetsuzan | outbound calls |
| 06:42.51 | tetsuzan | for this 5 lines |
| 06:43.00 | Jubei_ | Mattwj2005, thanks anyway :) |
| 06:43.05 | *** join/#asterisk prttp (i=Ftv@45.Red-83-50-35.dynamicIP.rima-tde.net) |
| 06:43.08 | FuriousGeorge | you are allowed one cuncurrent call per sip channel? |
| 06:43.08 | tetsuzan | i know that zapata has this function |
| 06:43.11 | tetsuzan | yes |
| 06:43.12 | Mattwj2005 | no problem I like to help :) |
| 06:43.13 | tetsuzan | one |
| 06:43.14 | FuriousGeorge | tetsuzan: ? |
| 06:43.24 | FuriousGeorge | jump based on dialstatus |
| 06:43.30 | tetsuzan | but, |
| 06:43.37 | *** join/#asterisk eject_ck (n=Miranda@mail.interlink.ck.ua) |
| 06:43.40 | tetsuzan | if i have a low traffic |
| 06:43.51 | tetsuzan | only 3 lines will be on fully use |
| 06:44.02 | tetsuzan | and the lastest lines? |
| 06:44.04 | Mattwj2005 | the good news is I have a faster Internet connection right now |
| 06:44.23 | FuriousGeorge | i dont know what you mean. why cant you jump based on DIALSTATUS |
| 06:44.27 | Mattwj2005 | 6.25 Download and 384 kbps upload |
| 06:44.36 | Mattwj2005 | 6.25 Mbps that is |
| 06:44.41 | tetsuzan | i can, |
| 06:44.43 | tetsuzan | but, |
| 06:44.57 | FuriousGeorge | Mattwj2005: big B or little b |
| 06:45.14 | Mattwj2005 | little b in both cases |
| 06:45.19 | Mattwj2005 | bits :) |
| 06:45.22 | FuriousGeorge | dsl? |
| 06:45.28 | tetsuzan | i want to set it properly |
| 06:45.28 | Mattwj2005 | nope cable |
| 06:45.42 | Mattwj2005 | my upload sucks but download is sweet |
| 06:45.51 | tetsuzan | if i by 5 broadvoice lines, |
| 06:45.51 | FuriousGeorge | tetsuzan: jumping based on dialstatus is the most propper way to do it afaik |
| 06:46.06 | tetsuzan | thanks george, |
| 06:46.16 | tetsuzan | but, if i buy 5 broadvoice lines, |
| 06:46.38 | FuriousGeorge | Mattwj2005: im using optonline. one day six months ago they just doubled their speed |
| 06:46.57 | Mattwj2005 | what are you getting for speeds? |
| 06:46.58 | FuriousGeorge | i guess its bc verizon is coming out with that FIOS |
| 06:47.30 | FuriousGeorge | now i think its something like 16 and 3 mbs |
| 06:47.50 | Mattwj2005 | yeah speeds suck here |
| 06:47.52 | Mattwj2005 | :( |
| 06:47.54 | FuriousGeorge | but practically i get between 12 and 1.5 |
| 06:48.10 | *** join/#asterisk UlbabraB (n=UlbabraB@host241-43-static.72-81-b.business.telecomitalia.it) |
| 06:48.19 | FuriousGeorge | i hear in places in s korea they get gig e to their house |
| 06:48.31 | Mattwj2005 | I would believe it |
| 06:48.47 | Mattwj2005 | Japan gets really fast Internet too |
| 06:48.53 | tengulre | who are come from KOREA? |
| 06:48.58 | FuriousGeorge | tetsuzan: i have no idea what is significant about five lines |
| 06:49.29 | tetsuzan | sorry george my poor english, |
| 06:49.29 | FuriousGeorge | in fact, if you want my advice use a provider that allows a few concurrent calls per channel |
| 06:49.33 | FuriousGeorge | if not unlimited |
| 06:49.47 | tetsuzan | unlimited calls, it?s possible |
| 06:49.53 | FuriousGeorge | tengulre: im not from korea, i heard somepeople over there in cities get fast internet for cheap |
| 06:50.01 | tetsuzan | but, unlimited concurrent calls, i don't know |
| 06:50.06 | tetsuzan | because i live in brazil, |
| 06:50.13 | tetsuzan | and the delay to broadvoice |
| 06:50.21 | tetsuzan | is for about 180ms |
| 06:50.31 | FuriousGeorge | well no, you will have to pay about 1-2 cents a minute per call in the us. how many minutes of calling do you expect to make |
| 06:50.31 | tetsuzan | and for anothers provider are 300, 400ms |
| 06:51.07 | Mattwj2005 | I am getting delay around 50 ms |
| 06:51.38 | FuriousGeorge | i found this provider a few towns over im pinging at 10ms |
| 06:51.47 | tetsuzan | here, i'm on 210ms |
| 06:51.49 | tetsuzan | bv |
| 06:52.07 | Mattwj2005 | nice FuriousGeorge |
| 06:52.09 | tetsuzan | here, voip is very slow |
| 06:52.17 | tetsuzan | you have to buy an international service |
| 06:52.25 | tetsuzan | and the delay.... |
| 06:52.40 | Mattwj2005 | I hear guys in space where actually using voip to communicate with their families...huge delay though |
| 06:53.09 | FuriousGeorge | tetsuzan: try ping connect02.voicepulse.com |
| 06:53.32 | FuriousGeorge | tetsuzan: you are in brazil, you said? i work in newark, nj |
| 06:53.42 | FuriousGeorge | mointo portuguese, forgive my spelling |
| 06:53.47 | tetsuzan | [root@shirran ~]# ping connect02.voicepulse.com |
| 06:53.47 | tetsuzan | PING connect02.voicepulse.com (64.61.93.90): 56 data bytes |
| 06:53.47 | tetsuzan | 64 bytes from 64.61.93.90: icmp_seq=0 ttl=50 time=193.395 ms |
| 06:53.48 | tetsuzan | 64 bytes from 64.61.93.90: icmp_seq=1 ttl=50 time=193.520 ms |
| 06:53.48 | tetsuzan | 64 bytes from 64.61.93.90: icmp_seq=2 ttl=50 time=193.283 ms |
| 06:53.48 | tetsuzan | 64 bytes from 64.61.93.90: icmp_seq=3 ttl=50 time=193.529 ms |
| 06:53.50 | tetsuzan | 64 bytes from 64.61.93.90: icmp_seq=4 ttl=50 time=193.656 ms |
| 06:53.52 | tetsuzan | ^C |
| 06:53.54 | FuriousGeorge | watch it |
| 06:53.54 | tetsuzan | --- connect02.voicepulse.com ping statistics --- |
| 06:53.56 | tetsuzan | 5 packets transmitted, 5 packets received, 0% packet loss |
| 06:53.56 | FuriousGeorge | lol |
| 06:53.58 | tetsuzan | round-trip min/avg/max/stddev = 193.283/193.477/193.656/0.127 ms |
| 06:54.00 | tetsuzan | [root@shirran ~]# |
| 06:54.02 | tetsuzan | sorry |
| 06:54.06 | tetsuzan | sorry |
| 06:54.08 | tetsuzan | :| |
| 06:54.42 | Mattwj2005 | here is the article -> http://fridge.ubuntu.com/node?from=16 about half way down |
| 06:54.43 | FuriousGeorge | try switch-1.asterlink.com |
| 06:54.49 | tengulre | here, VoIP is very slow too, and Telecom company refuse VoIP over their network. |
| 06:54.49 | tetsuzan | let me see.. |
| 06:55.13 | tetsuzan | tengulre but can you use a sip proxy? |
| 06:55.22 | tetsuzan | or, rtp packets are denied? |
| 06:55.43 | tetsuzan | (asterlink) 64 bytes from 66.250.69.13: icmp_seq=0 ttl=53 time=197.559 ms |
| 06:55.45 | tengulre | tetsuzan: RTP packets are denied. |
| 06:55.51 | *** join/#asterisk DarKnesS_WolF (n=wolf@62.114.197.28) |
| 06:56.07 | tengulre | 64 bytes from 66.250.69.13: icmp_seq=2 ttl=52 time=329 ms |
| 06:56.18 | JT | tengulre: can you use iax? |
| 06:56.22 | tetsuzan | tengulre freebsd? |
| 06:56.32 | tetsuzan | or, a sip proxy maybe |
| 06:56.33 | tengulre | JT, they same to that. |
| 06:56.45 | tetsuzan | rtp packet are on 10000-20000 |
| 06:56.48 | tetsuzan | by default |
| 06:57.39 | tengulre | the internet provider analyse the packet, if like VoIP protocol then disconnect it. |
| 06:57.48 | shellshark | which is obscene for most installations, as you'll probably never use more than 4 or 5 ports unless you're doing something crazy ;) |
| 06:57.48 | tengulre | :( |
| 06:58.06 | shellshark | tengulre: check out RTPS |
| 06:58.11 | shellshark | tengulre: RTP+SSL :) |
| 06:58.27 | tetsuzan | you have to patch |
| 06:58.31 | tengulre | shellshark: yes! but I don't know how to in asterisk? |
| 06:58.32 | tetsuzan | isn't it? |
| 06:58.39 | tetsuzan | tengulre a patch |
| 06:58.40 | tengulre | tetsuzan: debian. |
| 06:58.46 | *** join/#asterisk kristalino (n=kristali@cl-157.dub-01.ie.sixxs.net) |
| 06:58.50 | tengulre | :( |
| 06:58.54 | tetsuzan | let me see.... |
| 06:59.08 | tengulre | tetsuzan: I m beginner with asterisk. |
| 06:59.29 | tetsuzan | me too.. |
| 06:59.29 | tetsuzan | :) |
| 06:59.33 | tetsuzan | srtp |
| 06:59.44 | tetsuzan | http://www.google.com.br/search?hl=pt-BR&q=asterisk+srtp&btnG=Pesquisar&meta= |
| 06:59.45 | tengulre | our company have a Server running asterisk (E1 + SIP) |
| 07:00.01 | tetsuzan | no, pure sip |
| 07:00.04 | tetsuzan | :) |
| 07:00.10 | tetsuzan | i used a generic clone of x100p |
| 07:00.15 | tetsuzan | hehe... |
| 07:00.32 | tengulre | tetsuzan: but which softphone can supporte SRTP? |
| 07:01.14 | tetsuzan | http://support.counterpath.net/viewtopic.php?p=29758&sid=4e0ae96faecf73fbbe73c6f4f2e968e1 |
| 07:01.16 | tetsuzan | :) |
| 07:01.28 | tetsuzan | http://en.wikipedia.org/wiki/Secure_Real-time_Transport_Protocol |
| 07:01.31 | Inez | re |
| 07:01.40 | tetsuzan | counterpath popular softphone offers SRTP in the version 1.5 |
| 07:01.46 | Inez | So, Do somebody use L option for Dial command on Local channel? |
| 07:01.47 | tengulre | s/supporte/support. |
| 07:02.54 | tengulre | tetsuzan: thanks. |
| 07:03.00 | tetsuzan | furiousgeorge thanks for your opinion |
| 07:03.06 | tetsuzan | telgulre =) |
| 07:03.44 | tetsuzan | or, vpn |
| 07:03.52 | tetsuzan | rtp over vpn |
| 07:04.09 | tetsuzan | assuming that you have 2 asterisk server at 2 peers |
| 07:04.25 | tengulre | tetsuzan:VPN?? good idea! |
| 07:04.31 | tetsuzan | openvpn |
| 07:04.35 | shellshark | openvpn++ |
| 07:04.41 | tetsuzan | with freebsd i use native ppp |
| 07:04.45 | tetsuzan | for tunneling |
| 07:04.49 | FuriousGeorge | tetsuzan: its not really an opinion, im pretty sure thats how you are supposed to do it. your other option is to priority jump or use chanisavail, both of those ways i believe are deprecated. short of doing something horrid with contexts its your only real option |
| 07:05.01 | shellshark | tetsuzan: interesting... you dont have a TUN/TAP driver in BSD? |
| 07:05.43 | tetsuzan | i use vtun |
| 07:05.44 | tetsuzan | for a long time |
| 07:06.00 | shellshark | over ppp? |
| 07:06.01 | tetsuzan | but native ppp do the work |
| 07:06.05 | tetsuzan | uhum |
| 07:06.06 | tetsuzan | yes |
| 07:06.11 | shellshark | why add the extra overhead of PPP if you dont need it? |
| 07:06.45 | tetsuzan | ppp works well, and i dont have to install nothing to do the job |
| 07:07.00 | tetsuzan | it's a little overhead |
| 07:07.04 | shellshark | ppp adds overhead to the packets though |
| 07:07.06 | *** join/#asterisk benjk (n=benjamin@f8a01-0357.din.or.jp) |
| 07:07.08 | tetsuzan | :) |
| 07:07.29 | shellshark | you've got PPP and LCP overhead, and the other end has to run PPP also |
| 07:07.35 | *** join/#asterisk rkr245 (n=ravi@81.21.33.35) |
| 07:07.57 | tetsuzan | my experience with vtun, was a ugly one.... |
| 07:08.08 | shellshark | openvpn on linux you can use the stock tun/tap driver that comes with linux, and it adds no packet overhead |
| 07:08.15 | shellshark | and requires nothing special on either side |
| 07:08.19 | tetsuzan | a catastrophic one... |
| 07:08.26 | rkr245 | HI all |
| 07:08.29 | *** join/#asterisk DarKnesS_WolF (n=wolf@62.114.187.61) |
| 07:08.29 | tetsuzan | maybe openvpn |
| 07:08.35 | rkr245 | Good Morning |
| 07:08.45 | tetsuzan | rkr245 04:05 |
| 07:08.55 | tetsuzan | :) |
| 07:09.10 | shellshark | tetsuzan: does china block NTP also? :) |
| 07:09.47 | rkr245 | tetsuzan ;-) for me its 10 'clock in morning |
| 07:10.48 | tetsuzan | :) |
| 07:11.54 | Mattwj2005 | 2:11 am :) |
| 07:11.59 | rkr245 | tetsuzan , are you using asterisk B2bua ? |
| 07:13.27 | tetsuzan | i think so |
| 07:13.28 | tetsuzan | :) |
| 07:14.36 | rkr245 | tetsuzan , I found this asterisk b2bua some latest version |
| 07:14.57 | rkr245 | and it is a patch only working with asterisk-1.2 beta |
| 07:15.06 | rkr245 | compiling was o.k |
| 07:15.15 | Inez | no one use L option for dial command? |
| 07:15.21 | *** join/#asterisk thorbear (n=thorbear@212.247.4.149) |
| 07:16.04 | tetsuzan | rkr245 don't you think about SER? |
| 07:16.37 | rkr245 | yes |
| 07:17.00 | rkr245 | I am forwarding the call from SER to asterisk which is listening at port 6060 |
| 07:17.10 | rkr245 | but from there I am dead |
| 07:17.29 | *** join/#asterisk atapi2 (n=virgill4@c-65-34-182-167.hsd1.fl.comcast.net) |
| 07:17.52 | rkr245 | can you please tell a simple scenario how to work from there :-) |
| 07:18.02 | tetsuzan | wow.. |
| 07:18.20 | tetsuzan | 1 minute... |
| 07:18.20 | tetsuzan | :) |
| 07:18.27 | rkr245 | O.K |
| 07:19.48 | tetsuzan | http://www.openser.org/pipermail/users/2005-September/000977.html |
| 07:21.04 | rkr245 | just a min . iam opening the above url |
| 07:21.19 | tetsuzan | http://www.mail-archive.com/b2bua-users@lists.berlios.de/msg00020.html |
| 07:21.21 | tetsuzan | this second |
| 07:21.29 | tetsuzan | Asterisk 1.2.7.1 patched witt B2BUA worked very well with GSM codec |
| 07:21.59 | tetsuzan | sorry rkr245, but i have to sleep |
| 07:22.10 | tetsuzan | good work... |
| 07:22.16 | tetsuzan | bye all |
| 07:22.28 | rkr245 | byeee |
| 07:29.10 | *** join/#asterisk petit-toon (n=petit-to@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr) |
| 07:29.29 | *** join/#asterisk sbingner (n=thanotos@pdpc/supporter/sustaining/sbingner) |
| 07:30.38 | tengulre | my asterisk server is 221.11.5.182, SIP users are 2001~2008 password is same to username, my account is 2001, :) calling me now. |
| 07:31.07 | tengulre | exten => _2XXX,1,Dial(SIP/${EXTEN}, 20,tr) |
| 07:31.10 | tengulre | ;) |
| 07:31.34 | *** join/#asterisk scanna (n=scannach@81-174-16-211.f5.ngi.it) |
| 07:31.52 | *** join/#asterisk Defraz (n=t0tal@24-119-94-19.cpe.cableone.net) |
| 07:32.56 | tengulre | Mattwj2005, hi |
| 07:33.22 | Mattwj2005 | hi tengulre :) |
| 07:33.43 | tengulre | Mattwj2005, do you have free time now. |
| 07:33.58 | Mattwj2005 | about ready to go to bed |
| 07:34.00 | Mattwj2005 | what is up? |
| 07:35.16 | tengulre | nothing! :) good night! |
| 07:35.27 | Mattwj2005 | oh okay have a good night :) |
| 07:35.38 | Mattwj2005 | or should I say day ;) |
| 07:36.06 | *** join/#asterisk jeremy_g (n=jerms@static-213-115-44-90.sme.bredbandsbolaget.se) |
| 07:36.10 | tengulre | ;) |
| 07:36.14 | *** part/#asterisk Mattwj2005 (n=Matt@76.17.131.68) |
| 07:36.35 | tengulre | I want to test my asterisk server. |
| 07:37.44 | jeremy_g | sure :) |
| 07:37.54 | tengulre | jeremy_g: ? |
| 07:38.20 | tengulre | jeremy_g: pls register to my asterisk server with 2002/2002 if you want. |
| 07:38.37 | *** join/#asterisk Strom_C (n=strom@netblock-66-159-243-60.dslextreme.com) |
| 07:39.14 | *** join/#asterisk pif (n=ldm@zenon.apartia.fr) |
| 07:40.59 | tengulre | jeremy_g, hi! |
| 07:41.10 | tengulre | anybody alive? |
| 07:41.36 | jeremy_g | can't do that right now bud! |
| 07:41.46 | jeremy_g | busy in some serious testing myself |
| 07:42.00 | jeremy_g | using my latpop as a test server with the main server :( |
| 07:46.57 | *** join/#asterisk qdk (n=qdk@213.150.62.32) |
| 07:50.49 | Inez | i'm alive |
| 07:55.04 | *** join/#asterisk moon06 (n=michael@cim06-1-82-228-240-97.fbx.proxad.net) |
| 07:55.05 | jeremy_g | Inez:are you german |
| 07:56.28 | *** join/#asterisk Doce (n=Doce@m2d0e36d0.tmodns.net) |
| 07:56.42 | *** join/#asterisk petit-toon (n=petit-to@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr) |
| 07:56.45 | Doce | Wazzup |
| 07:56.48 | Inez | jeremy_g no, poland |
| 08:00.34 | *** join/#asterisk redax (n=redax@r6.hu) |
| 08:00.35 | redax | hi |
| 08:01.06 | redax | is it possible to set any Custom variable in AGI scripts that could be used later on the extensions.conf ? |
| 08:02.52 | *** join/#asterisk sulex (n=sulex@pdpc/supporter/active/sulex) |
| 08:05.53 | *** join/#asterisk sulex (n=sulex@pdpc/supporter/active/sulex) |
| 08:07.08 | tengulre | Inez: my asterisk server is 221.11.5.182, SIP users are 2001~2008 password is same to username, my account is 2001, :) calling me now. exten=>_2XXX,1,Dial(SIP/${EXTEN},20,tr) |
| 08:07.13 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 08:09.11 | moon06 | hey all |
| 08:09.11 | moon06 | I got a problem with ISDN/CAPI and ringtones |
| 08:09.11 | moon06 | when I call within France, I hear the ringtones before somebody answers |
| 08:09.11 | moon06 | when I call to other countries, there's no ringtone ... :/ |
| 08:09.46 | *** join/#asterisk oa (n=oal@62.61.133.90.generic-hostname.arrownet.dk) |
| 08:10.27 | oa | hi, does anyone know what causes asterisk to masquerade a channel? |
| 08:14.44 | tengulre | my asterisk server is 221.11.5.182, SIP users are 2001~2008 password is same to username, my account is 2001, :) calling me now. exten=>_2XXX,1,Dial(SIP/${EXTEN},20,tr) |
| 08:16.07 | jgoo | guys, I am setting variables in agi,yet the voicemail isn't seeing them |
| 08:16.41 | jgoo | channel.setVariable("FTS_SUBJECT", subject); |
| 08:16.46 | *** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust944.leed.cable.ntl.com) |
| 08:16.55 | jgoo | in the voicemail app I look for ${FTS_SUBJECT} but it is blank |
| 08:18.09 | *** join/#asterisk ShipHead (i=ShipHead@gateway/tor/x-f0bd48985b841235) |
| 08:21.09 | *** join/#asterisk Givur (n=mail@p54BCFCA3.dip.t-dialin.net) |
| 08:21.16 | Givur | Good morning |
| 08:21.38 | *** join/#asterisk jtar (n=John@host-84-9-186-77.bulldogdsl.com) |
| 08:21.43 | jeremy_g | I want to send a certain type of sip msg(temporarily unavailable 480) from an asterisk box to another asterisk box. how do i do that? |
| 08:23.07 | jgoo | jeremy_g: you need to first know when you want this to happen |
| 08:24.07 | *** join/#asterisk ghenry (n=ghenry@mail.suretecsystems.com) |
| 08:24.10 | jgoo | there is a whole feature of DND, if you just want those features, look at it, if this is something custom and peculiar, what is your setup? |
| 08:24.31 | jgoo | ping: jeremy_g |
| 08:25.38 | *** join/#asterisk santoshr (i=1063@203.199.110.93) |
| 08:26.52 | jeremy_g | oh sorry |
| 08:27.01 | santoshr | Oct 24 13:55:34 WARNING[6398]: db.c:171 ast_db_put: Unable to put value '59.183.191.23:5060:60:1555380466 :sip:15553804663@59.183.191.23:5060' for key '15553804663' in family 'SIP/Registry' wht is this |
| 08:27.29 | jeremy_g | jgoo:i have programmed * to do a certain thing when it receives a sip msg (480) |
| 08:27.41 | jeremy_g | jgoo:just want to check whethers its working or not :) |
| 08:27.43 | jgoo | jeremy_g: how are you developing that? an agi script? what language? |
| 08:28.39 | jgoo | jeremy_g: you can write a fake sip client to send SIP commands to the box, not much harder than a http client, just different protocol commands (if there is no specific encryption being used I guess) |
| 08:29.02 | jeremy_g | no encryption |
| 08:29.09 | jeremy_g | but i dont have time to write a sip client |
| 08:29.15 | jgoo | maybe someone already wrote SimpleSIPClient.java |
| 08:29.17 | jeremy_g | it ll take more than an hour right? |
| 08:29.35 | jeremy_g | aha :) |
| 08:29.42 | jgoo | it'll take 20-30 minutes to find a decent page that just describes the SIP protocol I guess |
| 08:29.48 | jeremy_g | but cant asterisk be programmed to send a certain sip msg |
| 08:29.50 | *** join/#asterisk DarKnesS_WolF (n=wolf@62.114.187.63) |
| 08:30.09 | santoshr | Oct 24 13:55:34 WARNING[6398]: db.c:171 ast_db_put: Unable to put value |
| 08:30.28 | jgoo | I am not sure. I am trying to do simple things in AGI that aren't supported apparently :( I started reading the .c code of the areas, not badly written. |
| 08:31.51 | jeremy_g | jgoo:mmm |
| 08:32.09 | *** join/#asterisk chat_jokey (n=chat_jok@202-149-32-1.static.exatt.net) |
| 08:32.36 | *** join/#asterisk hack1 (i=1076@203.199.110.93) |
| 08:33.10 | hack1 | WARNING[6398]: db.c:171 ast_db_put: Unable to put value '59.183.18.38:5060:300:15553966398:sip:15553966398@59.183.18.38:5060' for key '15553966398' in family 'SIP/Registry' does anyone know this error |
| 08:33.17 | benjk | if you can read one page in 20 secs, then it will take you 20 hours of reading the "page" that describes SIP |
| 08:33.49 | hack1 | benjk: ?? what |
| 08:34.13 | jgoo | what? |
| 08:34.21 | benjk | the total body of RFCs that describes SIP is about 3600 pages by now |
| 08:34.51 | jgoo | benjk: which is why it would be nice to find a page that describes the basic network handshake, I haven't seen SIP protocol, but I hope it is human readable |
| 08:34.59 | *** join/#asterisk [hC] (n=hardcore@S01060004e21ea953.vc.shawcable.net) |
| 08:35.00 | hack1 | does anyone know this error:- ast_db_put: Unable to put value '59.182.11.204:5060:300:15550077191:sip:15550077191@59.182.11.204:5060' for key '15550077191' in family 'SIP/Registry' |
| 08:35.29 | jgoo | anyway, this is important, I am setting variables on a channel, using agi, and the variables are not getting picked up by voicemail |
| 08:35.38 | benjk | dream on, there's no way you can describe this in a single page |
| 08:35.53 | jgoo | I execute setVariable("FOO", omglol); and then channel.exec("VoiceMail", voicemail); |
| 08:35.55 | tengulre | does the asterisk supported Private Asterisk HTTP Servers? |
| 08:36.36 | jgoo | benjk: web pages can be quite long... I was tlaking about a web page, and I am talking about finding how to auth a client, and send 1 sip command, again, I didn't say it was trivial) |
| 08:36.45 | hack1 | zigman: do you know this ast_db_put: Unable to put value '59.183.18.38:5060:300:15553966398:sip:15553966398@59.183.18.38:5060' for key '15553966398' in family 'SIP/Registry' |
| 08:36.50 | *** join/#asterisk ShipHead (i=ShipHead@gateway/tor/x-a029fb686ab09835) |
| 08:37.12 | benjk | the RFCs are all on the ietf website |
| 08:37.12 | *** join/#asterisk darkskiez (n=mbryars@bb-87-81-62-203.ukonline.co.uk) |
| 08:37.14 | jgoo | So, I am setting a variable on one channel, I execute channel.setVariable("FOO", omglol); and then channel.exec("VoiceMail", voicemail);, inside the voicemail app, ${FOO} is blank |
| 08:40.41 | santoshr | hack1: ..... any clues dude |
| 08:42.21 | tengulre | hi,all! how to using asterisk http server??? |
| 08:47.30 | qdk | tengulre: did you try the dov/ folder? |
| 08:47.48 | benjk | dove folder? |
| 08:47.51 | tengulre | qdk: ?? I don't understand! |
| 08:47.56 | tengulre | dove? |
| 08:47.58 | tengulre | where? |
| 08:48.00 | *** join/#asterisk soylentgreen (n=fgast@193.238.89.34) |
| 08:48.08 | qdk | doc/ folder |
| 08:48.16 | qdk | as in documentation. |
| 08:48.30 | benjk | I thought dove as in hawk |
| 08:48.33 | tengulre | what does that mean /var/lib/asterisk/static-http ?? |
| 08:49.08 | qdk | benjk: :-) |
| 08:49.14 | benjk | :) |
| 08:50.39 | *** join/#asterisk RayJWPi (n=RayJWPi@pD9E82DB1.dip0.t-ipconnect.de) |
| 08:51.03 | qdk | tengulre: it means that someone did a great job providing documentation of that feature, so that a lot can read that again and again giving them almost no reasons to ask questions already answered in that particular doc. |
| 08:51.50 | benjk | qdk, fyi, I have looked at chan_ss7, it's got no M2PA nor any proprietary IP or ethernet transport |
| 08:54.09 | santoshr | Oct 24 13:55:34 WARNING[6398]: db.c:171 ast_db_put: Unable to put value '59.183.191.23:5060:60:1555380466 :sip:15553804663@59.183.191.23:5060' for key '15553804663' in family 'SIP/Registry' |
| 08:54.29 | santoshr | guys need a little help on this.. anyone recognize this.......... |
| 08:55.14 | qdk | benjk: yes, i know that. :-) but the layers seem well written and ISUP almost completely implemented. |
| 08:56.05 | benjk | I only responded to your speculation that it probably does SS7 over ethernet |
| 08:56.13 | *** join/#asterisk Strom_C (n=strom@netblock-66-159-243-60.dslextreme.com) |
| 08:56.22 | qdk | benjk: so it could be a good foundation to start from, i think.. though i havent read nearly as much as you have of the SS7 protocol. |
| 08:56.34 | benjk | I wouldn't call it a good foundation at all |
| 08:56.40 | benjk | its based on zaptel |
| 08:56.54 | benjk | zaptel is anything but a foundation for ss7 |
| 08:57.20 | benjk | their wiki pages even admit that they are having problems with FISUs |
| 08:57.54 | *** join/#asterisk darkskiez (n=mbryars@bb-87-81-62-203.ukonline.co.uk) |
| 08:57.55 | qdk | benjk: well that layer could probably easily be replaced. Im not talking hardware wise, especially when i wanna eliminate that part and go eth. |
| 08:58.13 | benjk | you are talking about the entire MTP |
| 08:58.27 | benjk | since they didn't care to separate the layers |
| 08:58.31 | qdk | benjk: yesyes, we covered that earlier. |
| 08:59.33 | qdk | benjk: chan_ss7 didnt do that? coz i was wondering where mtp2 were, coz the doc says its there, so they must have put it all in mtp.c |
| 08:59.53 | benjk | indeed, they mangled it |
| 09:00.14 | benjk | which isn't much of a surprise |
| 09:00.39 | benjk | I have yet to see an implementation where mtp2 and mtp3 are separated |
| 09:00.44 | qdk | benjk: ok, thats not so good, we can agree on that, but when i hacked a few small things in it, it seem quite decent done. |
| 09:01.17 | qdk | benjk: as in, it would probably not require that much to seperate it, and go from there. |
| 09:01.19 | benjk | it also has dependencies on asterisk stuff |
| 09:02.06 | qdk | benjk: ok. |
| 09:02.39 | benjk | I rather rewrite a new properly layered MTP stack than trying to untie yet another gordian knot |
| 09:03.26 | benjk | you've got dependencies on zaptel, you've got the layers mixed up, you've got dependencies on asterisk |
| 09:03.32 | qdk | benjk: Ok, that might be better... one can always look at the good/bad stuff of the other implementations. |
| 09:05.44 | qdk | benjk: I did figure that no matter how well it will be done, it will not give me failover on the PBX unit, only the switching units, so im still kinda stuck in my chase for live call-failover. |
| 09:05.59 | *** join/#asterisk flot (n=flot@87.251.134.36) |
| 09:06.17 | benjk | yes, what they call clustering isn't exactly what you expect |
| 09:07.57 | flot | hi all! I write patch to * v. 1.4. Path for context transfer. Need post this patch in SVN |
| 09:10.16 | qdk | benjk: wasnt talking about chan_ss7, but any/all ss7... now and in the feature. |
| 09:10.55 | benjk | the model I described to you previously has the capability to do live failover |
| 09:11.04 | qdk | benjk: i know that chan_ss7 cluster can be used to forward the call to another server, which then might have some free channels on its E1 line(s). |
| 09:11.43 | benjk | the chan_ss7 "cluster" only works if the incoming calls come from another ss7 node |
| 09:11.44 | qdk | benjk: yes, routing failover, but thats only half the story. |
| 09:12.19 | benjk | I was talking about failing over a call in progress |
| 09:12.30 | benjk | that's what I had described to you before |
| 09:13.27 | qdk | benjk: yes, i know... but the SS7 knows nothing about the call processing, it can reroute the call to another dest. if the current dies. |
| 09:14.01 | benjk | yes, but it doesn't take care of any calls from any other source |
| 09:14.28 | qdk | it? what it? |
| 09:14.36 | benjk | also they say it only works for incomig, not outgoing calls |
| 09:14.45 | benjk | it = chan_ss7 |
| 09:14.50 | qdk | oh, you are still stuck in chan_ss7. |
| 09:15.18 | qdk | qdk benjk: wasnt talking about chan_ss7, but any/all ss7... now and in the feature. |
| 09:15.31 | benjk | well, you are mistaken then |
| 09:15.52 | qdk | im pretty sure im not... |
| 09:16.12 | benjk | ss7 can very well fail over calls in progress |
| 09:16.51 | qdk | SS7 of any kind dont know shit about eg. a conference, if the unit providing the conference dies... byebye conference no matter how amazing you implement SS7. |
| 09:17.19 | benjk | put a mechanism like I described before on the access node that will do IP address switching if its mate dies and you get what you want |
| 09:17.45 | *** join/#asterisk srbaker (n=srbaker@142.179.107.250) |
| 09:17.45 | benjk | this can be implemented as an SCCP user |
| 09:17.47 | srbaker | hey folks |
| 09:17.53 | srbaker | i signed up for VOIPjet |
| 09:18.03 | srbaker | i'm looking for a hardware based compatible box |
| 09:18.14 | srbaker | i just want to plug a regular POTS phone into a box and have it work |
| 09:18.16 | srbaker | any thoughts? |
| 09:18.23 | srbaker | recommendations for hardware? |
| 09:18.57 | benjk | this VoIPjet thing you mentioned, is it SIP based? |
| 09:19.02 | srbaker | uh |
| 09:19.04 | srbaker | it says IAX |
| 09:19.22 | benjk | in that case, there are two boxes available |
| 09:19.24 | srbaker | they don't mention sip, so i'm assuming not |
| 09:19.25 | benjk | Digium's IAXy |
| 09:19.44 | benjk | and a box (the name of which escapes me now) from a company called ATCOM |
| 09:20.08 | srbaker | the S101I ? |
| 09:20.13 | srbaker | this bitch? |
| 09:20.13 | srbaker | http://www.digium.com/en/products/hardware/s101i.php |
| 09:20.41 | benjk | that's the IAXy |
| 09:20.46 | srbaker | okay. those are the only two? |
| 09:21.02 | *** join/#asterisk Aurs (n=Aurs@host-81-191-123-189.bluecom.no) |
| 09:21.15 | srbaker | i'm currently on Vonage. the sound quality sucks (worse than CB radio). sometimes the phone doesn't bother ringing. voice mail doesn't work. |
| 09:21.19 | srbaker | and so on and so on |
| 09:21.34 | srbaker | i really wanted a vonage-like service though. full turn key, i do nothing, service. |
| 09:21.44 | srbaker | benjk: www.atcom.cn ? |
| 09:22.52 | benjk | http://www.atcom.cn/En_products_AG188.html |
| 09:23.03 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
| 09:23.26 | srbaker | why wan/lan ports? |
| 09:23.37 | srbaker | do i want it before my router? is that the point? |
| 09:23.42 | benjk | http://www.atcom.cn/En_products_AG168V.html |
| 09:24.08 | benjk | you don't have to use it as a router |
| 09:24.09 | srbaker | k |
| 09:24.13 | srbaker | i just want non-ass sound qualityu |
| 09:24.18 | srbaker | there's also a better than 2 second delay on vonage |
| 09:24.29 | srbaker | i want little or no *noticeable* delay, and telephone quality |
| 09:24.34 | benjk | that's not the fault of the adapter, really |
| 09:24.38 | benjk | its the network |
| 09:24.43 | srbaker | no, it's the fault of vonage |
| 09:24.45 | *** join/#asterisk Tili (n=tili@202.133.65.111) |
| 09:24.45 | srbaker | okay, last questin |
| 09:24.58 | srbaker | is it possible to unlock my PAP2 that i got from vonage? or the motorola box? and user them with another provider? |
| 09:25.08 | benjk | I don't know |
| 09:25.16 | srbaker | i lied, not last question. who do you recommend for service? |
| 09:25.25 | jgoo | hrm, in agi, the right way to set variables is now just Set right? |
| 09:25.33 | jgoo | not SetVariable or SetVar ? |
| 09:25.49 | qdk | srbaker: its ok, we just added it to the bill. |
| 09:25.54 | marcus2 | so i'm having this problem with agi on 1.4.0b3... agi scripts get the agi_* variables when they are called, but commands that they send to asterisk seem to get ignored |
| 09:26.21 | benjk | srbaker, if you have a US issued credit card, try Voicepulse |
| 09:26.23 | srbaker | oh. apparently VOIPjet doesn't provide inbound service |
| 09:26.29 | srbaker | any recommendations for that? |
| 09:26.33 | benjk | same |
| 09:26.33 | srbaker | id on't have a us issued credit card. :( |
| 09:26.44 | benjk | Voicepulse do inbound (DID) and outbound |
| 09:26.57 | srbaker | wonder if they can give me local number in Nanaimo, BC though |
| 09:26.59 | marcus2 | i have a bunch of linksys PAP2s that i unlocked from vonage |
| 09:27.08 | marcus2 | it wasnt too difficult, there were lots of docs online at the time |
| 09:27.12 | marcus2 | dunno if you can still do it that way |
| 09:27.20 | srbaker | marcus2: ah |
| 09:27.24 | srbaker | marcus2: can they do iax2? |
| 09:27.25 | benjk | there are hundreds of providers |
| 09:27.31 | marcus2 | nope |
| 09:27.34 | srbaker | okay, i'll look around at canadian providers |
| 09:27.38 | srbaker | and see what i can find |
| 09:27.38 | benjk | its impossible to keep track of them all |
| 09:27.49 | marcus2 | srbaker; i have them coupled with WRT54s that run asterisk |
| 09:27.52 | benjk | VoIP-Info.org should list some |
| 09:28.13 | srbaker | i'm going to get my editor to send me a dead tree copy of the asterisk book too |
| 09:28.16 | benjk | the ideal solution is to use two or three (or even more) providers in parallel |
| 09:28.38 | srbaker | ah |
| 09:28.53 | jeremy_g | um trying to use SIPGetHeader(var=headername[|options]) to make asterisk do sth when it receives a 480 |
| 09:29.02 | jeremy_g | what should be the headername to deal with responses |
| 09:29.15 | benjk | that's one of the nice things about running your own IP-PBX (such as Asterisk, Bayonne, Freeswitch, OpenPBX, Yate) |
| 09:29.21 | jeremy_g | like SIPGetHeader(var=To) |
| 09:29.43 | jeremy_g | like SIPGetHeader(var=???) to get what came in the field SIP/2.0 480 Temp Unava |
| 09:29.51 | jgoo | jeremy_g: code, result, something like that |
| 09:30.20 | jgoo | there is no get all headers? |
| 09:30.23 | benjk | you can register with multiple providers are the same time and when you dial and don't get through on one provider's netowrk, have the pbx simply try the next one in the list until successful |
| 09:31.17 | sbingner | well for outgoing you don't need to register -- you just need it properly configured ;) |
| 09:31.27 | jeremy_g | i wanan grab the response code jgoo |
| 09:31.53 | jeremy_g | if response code = 480, then do this <-- implementing this using SIPGetHeader |
| 09:32.19 | Inez | do anyone use Dial command with L option? |
| 09:32.23 | benjk | depends on the provider, if they require you to register, then you will need to register |
| 09:32.28 | jgoo | http://www.iana.org/assignments/sip-parameters |
| 09:32.33 | jgoo | @ jeremy_g |
| 09:34.03 | srbaker | okay, i'm going to try and get some snooze |
| 09:34.04 | srbaker | ttyl |
| 09:34.24 | jgoo | jeremy_g: try SIPGetHeader(var=SIP) |
| 09:34.37 | jgoo | might contains the code |
| 09:34.55 | *** part/#asterisk santoshr (i=1063@203.199.110.93) |
| 09:39.48 | *** join/#asterisk inspired (n=mikael@85.221.7.59) |
| 09:42.23 | jeremy_g | jgoo: chan_sip.c:13193 sip_getheader: SIP Header SIP not found for channel variable RESPONSE |
| 09:42.35 | jeremy_g | jgoo: got above when i used var as RESPONSE |
| 09:42.56 | jeremy_g | jgoo: 2,SIPGetHeader(RESPONSE=SIP) |
| 09:45.04 | jeremy_g | it works if i ,2,SIPGetHeader(RESPONSE=To) |
| 09:45.07 | *** join/#asterisk fourcheeze (n=rich@office.callmaster.co.uk) |
| 09:45.15 | jeremy_g | buts that the content for To header that i get then |
| 09:49.15 | fourcheeze | I've got a customer who has a requirement for call queueing such that the destination is a PSTN number which we would dial via SIP through a provider |
| 09:49.22 | fourcheeze | can asterisk handle queues like that? |
| 09:52.31 | *** join/#asterisk RayJWPi (n=RayJWPi@pD9E82DB1.dip0.t-ipconnect.de) |
| 09:52.50 | jeremy_g | i guess SIP/2.0 is not a header field at all :) |
| 09:53.06 | fourcheeze | So it would go something like this: |
| 09:53.06 | fourcheeze | A person calls the inbound number and a call comes in to *. * tries to call the destination and if that's busy then the caller is held in a queue |
| 09:53.27 | fourcheeze | what I don't understand is how * becomes aware that the line is no longer busy |
| 09:53.29 | jeremy_g | is it possible to grab the response code of the msg received by * |
| 09:58.15 | *** join/#asterisk tengulre (n=tengulre@221.11.5.182) |
| 09:58.29 | tengulre | hi,all |
| 09:59.58 | *** part/#asterisk hack1 (i=1076@203.199.110.93) |
| 10:00.37 | *** join/#asterisk hack1 (i=1076@203.199.110.93) |
| 10:00.45 | *** join/#asterisk jmls (n=asterisk@host81-159-167-80.range81-159.btcentralplus.com) |
| 10:01.12 | *** join/#asterisk stkn (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 10:01.13 | jgoo | so I have tried the following: |
| 10:01.29 | jgoo | channel.setVariable("__FTSA", "FTSA"); channel.exec("SetVar", "__FTSB=" + escapeAndQuote("FTSB")); channel.exec("Set", "__FTSC=" + escapeAndQuote("FTSC") ); |
| 10:01.30 | marcus2 | is there an easy way to execute an agi script when a channel closes? |
| 10:01.45 | jgoo | I can make my own abstract command and try that |
| 10:02.17 | jmls | jgoo, in the dialplan, print the channel name. In your agi, print the channel name - are they the same ? |
| 10:03.12 | jgoo | ill try |
| 10:09.27 | jgoo | wrong chan :p |
| 10:09.37 | jgoo | just testing this script |
| 10:09.50 | jgoo | I made two set command myself, and print the channel names |
| 10:10.48 | RoyK | does * support any ipv6 yet? |
| 10:10.48 | jgoo | hrm, Set(__Foo="TESTA") kills the chan |
| 10:11.07 | RoyK | jgoo: that's a feature, not a bug! |
| 10:11.37 | jgoo | RoyK: My favourite feature is the Segfault command! it segfaults asterisk! (real :p ) |
| 10:11.50 | flot | I find error in CDR. |
| 10:12.26 | jmls | royk is an evil person |
| 10:12.31 | jgoo | Damn, so it is your fault! |
| 10:12.57 | RoyK | jmls: http://karlsbakk.net/fun/ugly-hint.txt |
| 10:14.18 | jmls | holy sh*t |
| 10:14.23 | jmls | that was evil |
| 10:16.10 | *** join/#asterisk angryuser (n=uk@i03v-213-44-169-43.d4.club-internet.fr) |
| 10:16.26 | jgoo | hahaha but so funnt |
| 10:16.28 | jgoo | *funny |
| 10:17.11 | fourcheeze | truuly evil |
| 10:17.21 | marcus2 | ok this is really annoying |
| 10:17.43 | marcus2 | asterisk is native bridging two iax2 channels even when i'm telling it not to :/ |
| 10:20.04 | *** join/#asterisk tparcina (n=tomo@20-136.dsl.iskon.hr) |
| 10:20.21 | fourcheeze | ok, I'm going to try a different tack here: |
| 10:20.33 | fourcheeze | What's the best way to have queue agents dial in to asterisk |
| 10:20.34 | tparcina | HELP ME PLEASE, WORKING ASTERISK HAS STOP. - IT DOESN't REGISTER ANY PHONE!!! |
| 10:20.52 | fourcheeze | also your caps lock has stuck |
| 10:21.21 | tparcina | YES, but this one is realy important, this asterisk is in production! |
| 10:21.32 | tparcina | I don't know what else to check |
| 10:21.38 | fourcheeze | what does it say onthe console? |
| 10:21.39 | jmls | can you ping the phone from the asterisk box ? |
| 10:21.59 | RoyK | tparcina: if you type in capital letters, it'll magically make asterisk work better |
| 10:22.03 | jmls | :) |
| 10:22.08 | tparcina | sodnely it has disconnect all sip phones and it doesn't receive reigistration from them |
| 10:22.21 | jmls | can you ping the phones from the asterisk box ? |
| 10:22.26 | jmls | (#2) |
| 10:22.33 | tparcina | jmls: wait a sec |
| 10:22.42 | jgoo | tparcina: reboot |
| 10:22.49 | jmls | hey, I've got all day :) |
| 10:22.50 | fourcheeze | no don't reboot |
| 10:22.55 | angryuser | hi everybody:) |
| 10:23.00 | tparcina | jmls: yes, i can ping them |
| 10:23.07 | jmls | network is fine then |
| 10:23.09 | tparcina | jgoo: i have rebooted |
| 10:23.17 | jmls | yikes |
| 10:23.33 | fourcheeze | tparcina: pick a particular phone, debug it's IP number and tell it to reregister |
| 10:23.41 | tparcina | i was talking on the phone, when I hang up, afther that i couldn't make anyphone call. |
| 10:23.48 | jmls | tparcina: you were sure that asterisk was running ? |
| 10:23.54 | jmls | sounds like it crashed |
| 10:24.01 | angryuser | i have a small script which check if Zap/3 is avail, but i am unable to get it to work |
| 10:24.01 | jmls | and didn't restart |
| 10:24.03 | tparcina | i have loged on the asterisk and sow there is no phone registered |
| 10:24.24 | angryuser | can someone helm me please http://pastebin.ca/218759 |
| 10:24.31 | tparcina | fourcheeze: I'll do that, thank you |
| 10:26.22 | fourcheeze | angryuser: what doesn't work? |
| 10:27.12 | angryuser | $AVAILCHAN dont get set to ZAP/3 |
| 10:27.15 | tparcina | fourcheeze: I have started sip debug ip 10.0.0.203 and I have tried to register that phone, but I didn't se anything on CLI. just like asterisk doens't receive register request |
| 10:27.39 | tparcina | jmls: I'm sure that asterisk was running, because I was tolking thrue it on my phone |
| 10:28.13 | jmls | tparcina: you said that you after you hung up you couldn't make any more calls. Perhaps it crashed when you hung up. |
| 10:28.29 | tparcina | jmls: yes, but now asterisk is up |
| 10:28.41 | *** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin) |
| 10:28.46 | jmls | i *know* that now. Just trying to help with all possibilities |
| 10:28.53 | PakiPenguin | hello everyone :) |
| 10:29.07 | jmls | tparcina: is your * server on a fixed IP address ? |
| 10:29.20 | jmls | PakiPenguin: yo the penguin |
| 10:29.32 | PakiPenguin | hey :) jmls |
| 10:30.19 | tparcina | jmls: yes, it has fixed IP |
| 10:30.22 | jmls | (also very glad that d is not next to n) |
| 10:30.35 | PakiPenguin | lol |
| 10:30.46 | angryuser | do i need to declare $AVAILCHAN variable in general to get ChanIsAvail() to work? |
| 10:30.49 | tparcina | for some reason asterisk doens't get sip requests from phones |
| 10:30.55 | jmls | angryuser: no |
| 10:31.34 | jmls | angryuser: what does ${AVAILSTATUS} tell you ? |
| 10:32.12 | angryuser | jmls: when i do ChanIsAvail(Zap/3) and then Noop(${AVAILCHAN}) it doens contain chain |
| 10:32.30 | angryuser | jmls: il chesk |
| 10:32.34 | angryuser | jmls: il check |
| 10:33.24 | angryuser | availstatus is = 0 |
| 10:33.58 | jgoo | ok, I print out the channel in the voicemail, and I get **UNKNOWN** |
| 10:33.59 | jgoo | helpful |
| 10:34.07 | jgoo | @jmls |
| 10:34.37 | tparcina | ANYBODY PLEASE, asterisk doens't receive SIP register request messages |
| 10:34.58 | angryuser | tparcina, firewall? |
| 10:35.16 | PakiPenguin | tparcina: wrong port ? |
| 10:35.18 | jgoo | tparcina: reboot all phones, and asterisk. You were making a call, make sure asterisk physically powers off and on, so you know it restarts |
| 10:35.26 | qdk | tparcina: stop that CAPS crap. |
| 10:35.29 | jmls | angryuser: exten => _006XXXXXXXX,2,NoOp($AVAILCHAN) is wrong. You are missing a {}. Should be exten => _006XXXXXXXX,2,NoOp(${AVAILCHAN}) |
| 10:35.30 | tparcina | this is asterisk in production, and has stoped to work |
| 10:35.39 | tparcina | no, i don't have firewall |
| 10:35.56 | tparcina | the port is 5060, and it shoudl work |
| 10:36.04 | angryuser | jmls, yes i saw that an instant ago, it does no change the result |
| 10:36.05 | qdk | tparcina: look at tcpdump of the traffic.... set VERY verbose logging in asterisk... |
| 10:36.18 | qdk | tparcina: should, but does it? |
| 10:36.26 | jmls | also exten => _006XXXXXXXX,4,Cut(theChannel=AVAILCHAN,,1) should be exten => _006XXXXXXXX,4,Cut(theChannel=AVAILCHAN,-,1) |
| 10:36.56 | angryuser | hmm |
| 10:37.13 | jgoo | jmls: jsut to clarify, as this wiki talks about so many versions, in 1.2.0, set variable is done with Set(varname="value") |
| 10:37.14 | jmls | or exten => _006XXXXXXXX,4,Cut(theChannel=AVAILCHAN,/,1) |
| 10:37.15 | jgoo | right? |
| 10:37.42 | angryuser | but when i do noop (${availchan}) before i got no chain... |
| 10:37.56 | tparcina | qdk: i have softphone on my laptop, i have ethereal on my laptop, i see only sip requests that my laptop sends, but * doesn't reply |
| 10:38.34 | tparcina | qdk: and on SIP DEBUG IP mylaptop - I don't se any message - that means that ASterisk dosn't get any message |
| 10:38.45 | RoyK | ROTFL. Someone has put up a $200 bounty for adding IPv6 support to asterisk: http://www.voip-info.org/wiki-Asterisk+bounty+IPv6 |
| 10:39.55 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 10:40.10 | tparcina | jgoo: I have rebooted asterisk and several phones - it didn't help |
| 10:41.15 | *** join/#asterisk backblue (n=igor@82.102.1.42) |
| 10:41.15 | qdk | tparcina: ok, so if asterisk doesnt get the message, why are you asking for help in a asterisk-channel? |
| 10:41.50 | jmls | angryuser: I get Zap/3-1 in my ${AVAILCHAN} |
| 10:41.51 | angryuser | http://pastebin.ca/218778 i have remouved errors, the pb persists, after ChanIsavail(Zap/3) $AVAILCHAN contain no string, any ideas? |
| 10:42.03 | jmls | whoa. deja-vu |
| 10:42.05 | jmls | :) |
| 10:42.11 | angryuser | :) |
| 10:42.25 | angryuser | and availstatu got 0 |
| 10:42.31 | jmls | angryuser: what version oif * ? |
| 10:42.52 | angryuser | oif?:) |
| 10:43.07 | jmls | (of) |
| 10:43.10 | tparcina | qdk: because of this - http://pastebin.ca/218783 |
| 10:43.30 | angryuser | asterisk 1.2 |
| 10:43.48 | tparcina | qdk: I'll double chech network |
| 10:45.00 | jmls | angryuser: are you sure that you have zap channels up to 3 ? |
| 10:45.20 | jmls | (try a quick dial(zap/3 ...) |
| 10:45.27 | angryuser | yes i have Zap/1 - Zap/8 |
| 10:45.45 | *** join/#asterisk kio (n=kio@ool-4577ae5e.dyn.optonline.net) |
| 10:45.49 | angryuser | ok il double check |
| 10:45.50 | jmls | zap show channels gives ? |
| 10:46.33 | angryuser | zap show channels gives me 8 channels |
| 10:46.46 | jmls | Zap/1 - Zap/8 ? |
| 10:46.53 | jeremy_g | angryuser:u dont look that angry when you smile |
| 10:46.55 | jeremy_g | :) |
| 10:47.10 | angryuser | hehe i am trying to |
| 10:47.43 | jeremy_g | tparcina:yup it doesnt get any msg or ur laptop ip is wrong |
| 10:47.59 | jmls | angryuser: are you sure that zap/3 is not in use ? If it is, ${CHANAVAIL} will be blank ... |
| 10:48.55 | angryuser | hm, you think zap/3 is stuck or something? |
| 10:49.08 | angryuser | normally it is not used |
| 10:49.15 | angryuser | il try a hard swich |
| 10:49.20 | jmls | try zap/1 or dialling out using zap/3 |
| 10:49.32 | angryuser | *switch |
| 10:49.46 | jmls | always painful, a hard switch |
| 10:49.51 | angryuser | ZAP/1 and Zap/2 woring fine |
| 10:50.07 | jmls | as in CHANAVAIL works ? |
| 10:51.10 | *** join/#asterisk xnon (i=xnon@200.8.30.50) |
| 10:52.25 | tparcina | my asterisk doesn't get sip registration request messages |
| 10:52.30 | jgoo | jmls: so, the channel was Zap/1-1 on my script, and **UNKNOWN** in the voicemail |
| 10:52.35 | tparcina | I'm positive that network is working fine |
| 10:53.00 | jgoo | jmls: I tried __VAR but ${VAR} still didn't find it in the **UNKNOWN** |
| 10:53.01 | tparcina | Asterisk listens 5060 port, and is bint do 0.0.0.0 address |
| 10:53.09 | tparcina | I have static address on my asterisk |
| 10:53.21 | jmls | voicemail cannot have an unknown channel - it's in a channel if it's executing |
| 10:53.22 | Inez | do someone use festival app? |
| 10:53.30 | tparcina | what else I can check to se why asterisk doen't get any SIP message? |
| 10:53.48 | jmls | tparcina: check the settings on the phone to make sure they are ok. |
| 10:54.09 | *** join/#asterisk base_asterisk (n=base_@81.215.73.135) |
| 10:54.28 | base_asterisk | hello |
| 10:54.29 | tparcina | jmls: I have double checked that, they are fine |
| 10:54.47 | jmls | can you ping the * server from the phone (some phones allow you to do this) |
| 10:55.26 | base_asterisk | anyone knows how to debug on Asterisk for human.. |
| 10:55.30 | jmls | when you reboot a phone, what are the messages ? |
| 10:55.40 | jmls | base_asterisk: welcome to the madhouse |
| 10:55.51 | jgoo | jmls: that is the channel that is prints out. |
| 10:55.51 | base_asterisk | :) |
| 10:56.30 | jmls | jgoo: can you paste your dialplan bits ? |
| 10:56.57 | base_asterisk | because whenever i write "sip debug" , the logs are streaming on the screen, i cant see the top of the logs. |
| 10:57.12 | base_asterisk | how can i see the logs in Asterisk |
| 10:57.24 | jgoo | hrm, well, what parts? I forward to voicemail from a script... I have just one inbound that I am testing with |
| 10:57.37 | angryuser | jmls: well i am able to create channels ZAP/1 and ZAP/2 but dial (ZAP/3) give me -unable to create a channe of type zap, cause unknown.... |
| 10:57.56 | jmls | angryuser: ah ha. There's your problem :) |
| 10:58.08 | angryuser | yep |
| 10:58.14 | jmls | base_asterisk: /var/log/asterisk contains the logs |
| 10:58.35 | jmls | base_asterisk: also check logger.conf to ensure that debug messages are turned on in the logs |
| 10:58.55 | *** join/#asterisk Buglouse (n=SourceRa@66.97.121.210) |
| 10:58.58 | jmls | angryuser: what about zap/4 5 6 7 8 ? |
| 10:58.59 | *** join/#asterisk tsurk0 (n=tsurko@80.72.68.86) |
| 10:59.04 | jgoo | jmls: channel is unknown, the place I am putting this output is in vm_email.inc |
| 10:59.47 | tparcina | my Linux machine gets SIP messages, but asterisk doesn't. How to check why? |
| 10:59.54 | PakiPenguin | hmmm |
| 10:59.58 | angryuser | jmls thx a lot for ur help, i got a got for an hour or so,,, |
| 11:00.06 | jmls | got a got ? |
| 11:00.06 | PakiPenguin | :) |
| 11:00.12 | PakiPenguin | got to go :p jmls |
| 11:00.14 | PakiPenguin | hehe |
| 11:00.25 | angryuser | whatever i am latee.. |
| 11:00.33 | jmls | sounds very nice. got a got for an hour. |
| 11:00.34 | base_asterisk | thanks jmls, but it's not contain sip logs, you know (invite, trying,...) i want to see the missing logs on the screen. these files not contains that kind of logs, just print out screen and i cant catch |
| 11:00.42 | *** join/#asterisk xnon_ (n=xnon@200.8.30.50) |
| 11:00.52 | jmls | base_asterisk: also check logger.conf to ensure that debug messages are turned on in the logs |
| 11:00.55 | PakiPenguin | base_asterisk : check logger.conf , enable it for messages |
| 11:00.59 | jmls | I win ! |
| 11:01.14 | PakiPenguin | lol yeah :) |
| 11:03.25 | Inez | can someone help me with festival? |
| 11:03.59 | tparcina | problem solved! I'm not shure what it was, but since It just started to work I asume that problem was in DNS server. |
| 11:04.06 | jgoo | yey |
| 11:04.12 | tparcina | For now, that is only logical explanation |
| 11:04.21 | jgoo | praise be to the DNS gods |
| 11:04.46 | jmls | tparcina: we always use IP addresses in config files for this exact reason |
| 11:05.14 | tparcina | Thank's to jmls, jeremy_g, qdk, jgoo, angryuser and everybody else who have tried to help me |
| 11:05.27 | jgoo | so, jmls, I am stumped. np tparcina |
| 11:05.54 | tparcina | I'm sorry if I have make someone disconfort for writing with capital letters, but please do understand my situation. |
| 11:06.17 | jmls | tparcina: no worries. people just don't like being shouted at :) |
| 11:06.21 | tparcina | this is the first time that working Asterisk has fall down on me, and that I don't know what's the problem. |
| 11:06.54 | jmls | as you said, your DNS went walkabouts |
| 11:07.01 | tparcina | So, it wasn't so big deal that Asterisk didn't respond to SIP messages, it was the problem that I didn't know hwy is that happening. |
| 11:07.01 | jgoo | ok, time to look at a different fastagi server approach |
| 11:07.50 | *** join/#asterisk saftsack (n=oliver@p54A7ED45.dip.t-dialin.net) |
| 11:08.33 | *** join/#asterisk mrg82 (n=na@office.intercea.co.uk) |
| 11:09.37 | mrg82 | Whats the maximum number of simultaneous calls an adsl connection of 256k upload can handle before quality starts dropping off? |
| 11:09.48 | mrg82 | sorry, with the GSM codec |
| 11:09.49 | saftsack | 2 |
| 11:09.58 | saftsack | oh dunno, sry |
| 11:10.35 | tparcina | jmls: I alsoo use IP as much as I can, but sometimes I just have to use DNS name. So, for that reason I need to have srvlookup=yes in my sip.conf. and that is the reason why my asterisk has stop responding to sip register request messages... |
| 11:10.58 | RoyK | mrg82: 20ms packetization on RTP gives an overhead of 16kbps regardless of codec, and GSM is 13,5kbps, so just do the math :) |
| 11:11.13 | jmls | yeah. Asterisk also has a problem starting properly if DNS is not available and you use srvlookup |
| 11:13.31 | tparcina | jmls: I have read about this DNS problems. How did you solve it? |
| 11:14.03 | *** join/#asterisk VibroMax (n=phisto@83.229.70.154) |
| 11:14.54 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 11:15.09 | jgoo | tparcina: the keyword here is //has//. it doesn't sound solved :/ |
| 11:17.05 | jgoo | ok I am sending: SET VARIABLE __TESTA TESTA |
| 11:17.30 | jgoo | and this was my final try, ${TESTA} and ${ENV(TESTA)} are both blank |
| 11:19.58 | jgoo | has anyone written their agi driven voicemail app? |
| 11:20.27 | jgoo | I mean, it isn't a huge app. maybe I should just write my own. *SIGH* |
| 11:22.09 | *** join/#asterisk Marshall16 (n=Marshall@cpe-76-181-166-122.columbus.res.rr.com) |
| 11:24.25 | *** join/#asterisk coppice (n=chatzill@229.166.17.210.dyn.pacific.net.hk) |
| 11:29.12 | tparcina | jgoo: It sin't solved, i know that, but now I'll do something about it. |
| 11:29.39 | jmls | tparcina: I don't - I use ip addresses only. |
| 11:30.04 | tparcina | jgoo: I just havent decaide which way to go. disabling serverlookups, or installing some DNS cache program on my linux |
| 11:30.41 | jgoo | jmls, I am defeated. I don't think this is possible. So I will record the files... and use my own database to save the info, and make my own voicemail app |
| 11:30.43 | jgoo | sucks |
| 11:30.48 | jmls | jgoo: sorry, just never has the need to use AGI. |
| 11:30.55 | jmls | (have had to use) |
| 11:31.10 | jgoo | no worries, very few people seem to use asterisk for more than out-of-the-box stuff |
| 11:31.25 | jmls | hey! we use it for a lot of "out-of-the-box" stuff |
| 11:31.39 | jgoo | we need an agi/*programming (not *-dev) irc channel |
| 11:31.42 | jmls | all our agents are monitored using jabber and presence, for starters :) |
| 11:31.54 | jgoo | that is cool |
| 11:32.02 | jmls | we use func_odbc a *lot* in the dialplans |
| 11:32.14 | jgoo | so their status (loging, logout, dnd) is visible in a list? |
| 11:32.18 | jmls | yup |
| 11:32.29 | jmls | dialling / on call / wrapup etc |
| 11:32.43 | jgoo | how do they set their status to wrapup? |
| 11:32.52 | jgoo | that is the time given, after hangup, riiight |
| 11:33.02 | *** join/#asterisk p0g0 (n=pogo@madwifi/support/p0g0) |
| 11:33.09 | jmls | yup. We monitor when the the call hangs up, start a timer. |
| 11:33.30 | jmls | when they've finished their notes / etc , they "go ready". End of wrapup |
| 11:33.51 | jmls | we set the status using jabber. We've built a jabber client into our application. |
| 11:33.56 | jgoo | nice, I need to do that for my second * project |
| 11:34.23 | jgoo | anyway, donut + coffee time . yey. I should feel guilty, my weekend run, was more of a weekend stroll |
| 11:34.36 | jgoo | check runningahead.com < great site! ok afk |
| 11:34.59 | *** join/#asterisk McLazarus (n=mcallist@pool-72-78-32-109.phlapa.east.verizon.net) |
| 11:35.21 | *** join/#asterisk UlbabraB (n=UlbabraB@host241-43-static.72-81-b.business.telecomitalia.it) |
| 11:37.10 | *** join/#asterisk bigjb (n=nbigjb@195.60.10.114) |
| 11:37.52 | jmls | hey, my 6 year old has just asked me if I have "TuxPaint" on my computer ("you know, the one with the penguin" he added) |
| 11:37.54 | jmls | cool! |
| 11:39.04 | jgoo_ | :-) that is awesome jmls !! |
| 11:39.18 | jgoo_ | actually, educational software, games and enlglish software is something I want to get into! |
| 11:40.46 | *** join/#asterisk AsteriskAlbania (n=info@217.24.244.130) |
| 11:41.36 | *** join/#asterisk UlbabraB (n=UlbabraB@host241-43-static.72-81-b.business.telecomitalia.it) |
| 11:43.19 | AsteriskAlbania | I have a T400P DIGIUM and ASTERISK box, time after time the card seems to reports ERRORS on the E1 from PSTN which after collecting a certain number off errors brings the E1 down. It is needed to reset the E1 manually from the PSTN side, is there any way not to notify for errors ? |
| 11:45.26 | skrusty | anyone know why i'd get warnings about thread blocking and choppy audio in meetme? only head the choppy audio when coming in over a sip trunk, and not from local sip devices |
| 11:49.54 | fourcheeze | jmls: you're lucky. My daughters school has just got a suite of windows rubbish |
| 11:50.03 | fourcheeze | but my kids still like tuxpaint |
| 11:50.08 | *** join/#asterisk stkn (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 11:53.01 | *** join/#asterisk mfroes (n=mfroes@200-162-218-81.corp.ajato.com.br) |
| 11:53.13 | mfroes | hey .. can someone help ? |
| 11:53.46 | angryuser | jmls: i have founs the reason, gsm line was down:) |
| 11:53.59 | angryuser | found* |
| 11:54.01 | mfroes | i always get this error when trying to do a b10 for a busy extension |
| 11:54.04 | mfroes | Spawn extension (default, 074545454 8 ) exited non-zero on 'SIP/phone131-fa2f' |
| 11:55.29 | jgoo_ | jmls: I found another weirdness now |
| 11:55.53 | jgoo_ | I make the recording, and play it back, and give them options to save. However, getVariable cannot retrive RECORDED_FILE |
| 11:58.05 | *** part/#asterisk RayJWPi (n=RayJWPi@pD9E82DB1.dip0.t-ipconnect.de) |
| 11:59.01 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 12:00.12 | mfroes | here is the sip debug http://www.rafb.net/paste/results/hSFUqc16.html |
| 12:00.16 | mfroes | anyone can help ? |
| 12:00.53 | *** join/#asterisk Dibbler (n=Dibbler@zidane.pi-net.net) |
| 12:02.42 | jgoo_ | [TK]D-Fender: hola. So, not sure if this is your wagon of nuts, but I am having trouble with variables. I am setting them and getting them, but they are not seein in Voicemail, as the channel is **Unknown** (as reported) |
| 12:03.13 | mfroes | no one ? |
| 12:03.48 | jmls | angryuser: cool. congrats :) |
| 12:04.33 | [TK]D-Fender | mfroes: "SIP/2.0 401 Unauthorized" |
| 12:04.55 | [TK]D-Fender | mfroes: Looks like your register is set up ok, but your peer/friend entry is not. |
| 12:05.17 | jmls | jgoo_: wagon of nuts ?? |
| 12:05.27 | jgoo_ | kettle of fish just seems to old hat |
| 12:05.31 | jgoo_ | s/to/too |
| 12:06.03 | [TK]D-Fender | jgoo_: : what variables? set, how? used where? Pastebin would be nice.... |
| 12:06.04 | jmls | I like "floats your boat" :) |
| 12:06.09 | coppice | fish in a kettle must really ruin the taste of your tea |
| 12:06.17 | jmls | yeuch. |
| 12:06.22 | [TK]D-Fender | <- Captain of the Titanic swim team |
| 12:06.47 | jmls | depends what's in the kettle. Moonshine, mmmmmmmmm |
| 12:07.10 | coppice | fishy moonshine? |
| 12:07.52 | jmls | ewww. getting too fishy for me |
| 12:08.39 | *** join/#asterisk teknoprep (n=tekon@unaffiliated/teknoprep) |
| 12:08.41 | jgoo_ | [TK]D-Fender: right now, the problem I noted was, the channel reported by the voicemail app is **Unknown**. I set variable __Foo, and try and retreive it as ${Foo}. it is an agi script |
| 12:09.56 | angryuser | jmls: i got ChainIsAvail working on all ZAP/1-2 an Zap/4-8 bun not Zap/3 port is dead?? |
| 12:10.14 | angryuser | tdm400p |
| 12:10.42 | jgoo_ | jmls: warm fish milkshake. |
| 12:11.11 | jmls | angryuser: possibly - it sure looks like a problem with it. |
| 12:11.59 | MikeJ | they let coopice in here? |
| 12:12.44 | [TK]D-Fender | jgoo_: Could be unknown because the caller hung up in there, no? |
| 12:12.46 | MikeJ | I thought this place had standards |
| 12:13.00 | tzanger | heh... md RAID5 resync over USB1... 143kB/sec |
| 12:13.10 | jgoo_ | [TK]D-Fender: no, I don't hang up, I wait and it continues out of voicemail |
| 12:14.01 | jgoo_ | but, right now, I am making my own voicemail, because I debugged this for a while, and decided using record, however, now I cannot grab ${RECORDED_FILE}. again, this is a very fastagi specific question, and may be asterisk-java issue. But I don't know anyone who uses asterisk-javba |
| 12:14.04 | [TK]D-Fender | jgoo_: Well I can |
| 12:14.13 | [TK]D-Fender | 't comment further unless you show me something substantial. |
| 12:14.14 | *** part/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 12:14.53 | jgoo_ | [TK]D-Fender: I can show you the agi script |
| 12:15.04 | jgoo_ | or the output |
| 12:15.17 | [TK]D-Fender | jgoo_: how about everything related from start to finish |
| 12:15.35 | jgoo_ | including the agi call in the dialplan? |
| 12:15.44 | [TK]D-Fender | jgoo_: Would be nice... |
| 12:17.09 | *** join/#asterisk acctor (n=heh@my81-91-204-13.mynow.co.uk) |
| 12:21.06 | jgoo_ | [TK]D-Fender: here is a quick overview of the test, start to finish http://pastebin.ca/218895 |
| 12:21.24 | *** part/#asterisk SpaceBass (n=sp@static-71-251-230-6.rcmdva.fios.verizon.net) |
| 12:23.58 | *** part/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 12:28.16 | [TK]D-Fender | jgoo_: What language? |
| 12:28.28 | [TK]D-Fender | jgoo_: Ahh, java |
| 12:28.34 | [TK]D-Fender | jgoo_: http://asterisk-java.sourceforge.net/apidocs/net/sf/asterisk/fastagi/BaseAGIScript.html#getFullVariable(java.lang.String) |
| 12:28.39 | [TK]D-Fender | jgoo_: See this? |
| 12:28.51 | fourcheeze | Anyone know all about queues here? |
| 12:29.13 | fourcheeze | I'm trying to setup a queue where the destination is a phone which is on pstn dialed out via a sip provider |
| 12:29.43 | backblue | what's wrong with asterisk svn? |
| 12:29.52 | [TK]D-Fender | jgoo_: This seems to clearly imply that it will read the current channel name into a *JAVA* variable. This is NOT a function to retrieve an *ASTERISK* "channel variable" |
| 12:30.33 | fourcheeze | <PROTECTED> |
| 12:30.45 | fourcheeze | I can see how to do it on a direct channel but not over a SIP provider |
| 12:30.59 | *** part/#asterisk hack1 (i=1076@203.199.110.93) |
| 12:31.44 | *** join/#asterisk sizzla (n=jvanitou@LAubervilliers-151-11-30-4.w193-251.abo.wanadoo.fr) |
| 12:31.49 | [TK]D-Fender | jgoo_: Saw a sample somewhere else..... could be wrong, but it doesn't lok like its intended purpose. |
| 12:33.32 | *** join/#asterisk wow (n=zataz@80.92.66.182) |
| 12:33.33 | wow | hi |
| 12:34.46 | jgoo_ | [TK]D-Fender: what? Yes I am using getFullVariable to read variables. I have that working, but in two instances, variables are not. |
| 12:35.24 | jgoo_ | [TK]D-Fender: I am not confused with the API, but getVariable doesn't work for RECORDED_FILE and setVariable's aren't read in voicemail application :-( as it appears the channel is different and even with __ they don't work |
| 12:35.54 | *** join/#asterisk stoffell (n=stoffell@fw.catsanddogs.com) |
| 12:37.21 | *** join/#asterisk brif8 (n=brif8@67.78.24.178) |
| 12:37.40 | brif8 | Hi All anyone using an external FXO gateway |
| 12:38.42 | [TK]D-Fender | jgoo_: Where does it tell you that it sets this variable in the first place? |
| 12:42.34 | stoffell | if polycom phones do a reinvite, do they use the RTP ports as defined in * or does it go on different ports? |
| 12:44.22 | *** join/#asterisk akoch (n=chatzill@mail.gk-soft.de) |
| 12:44.38 | inspired | has anyone used Dial with the d option? It doesn't jump out of the extension if it captures a dtmf digit like it should. |
| 12:45.27 | akoch | hello after install 1.2.13 I get the error --> |
| 12:45.29 | akoch | == ISDN1: Answering for 33 |
| 12:45.30 | akoch | <PROTECTED> |
| 12:45.32 | akoch | Ouch ... error while writing audio data: : Broken pipe |
| 12:45.35 | akoch | Segmentation fault |
| 12:45.49 | akoch | going back to 1.2.12.1 works, the system works again |
| 12:46.19 | [TK]D-Fender | stoffell: The endpoints should renegotiate based on their mutual capabilities, not that of the transferring point. |
| 12:46.34 | *** join/#asterisk ShipHead (i=ShipHead@gateway/tor/x-905ae506914f2065) |
| 12:48.31 | *** join/#asterisk Deeewayne (n=dwayne@ool-44c0d56e.dyn.optonline.net) |
| 12:48.32 | jgoo_ | [TK]D-Fender: I never thought this sets the variable, I had a test earlier trying to set a variable, im many many ways, but now I am trying to get a variable |
| 12:49.20 | [TK]D-Fender | jgoo_: But what tells you the variable even exists? If it doesn't then returning null sure makes sense, doesn't it? |
| 12:49.51 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 12:49.58 | jgoo_ | [TK]D-Fender: that is why I am here, the docs says that RECORDED_FILE gives the last file using Record |
| 12:50.07 | jgoo_ | but, it doesn't seem to here :/ |
| 12:50.52 | jgoo_ | my whole issue is: I need to know the voicemail that was left, to embedd it onto a page. I also want to edit the subject of the voicemail to include an auto reply token to help people reply to these voicemail pages/emails |
| 12:51.06 | [TK]D-Fender | jgoo_: But you aren't using "Record" you are using another Java function, not the Asterisk proper app |
| 12:51.38 | jgoo_ | I am using AGI... hrm, but you are right, what is to say that the AGI function should call the Record() app and behave the same way |
| 12:51.50 | *** join/#asterisk Wall (n=mnose@host96.201-253-161.telecom.net.ar) |
| 12:51.54 | jgoo_ | this Java function is merely an implementation of the API though :-/ |
| 12:51.59 | stoffell | [TK]D-Fender; oh, okay, so gotta go have a look in the polycom docs then |
| 12:52.02 | Wall | hola |
| 12:52.07 | jgoo_ | it does actually record, and playback, when I use them |
| 12:52.12 | Wall | alguna entiende español ? |
| 12:52.18 | Wall | alguno entiende español ? |
| 12:52.30 | jmls | Wall: no senor |
| 12:52.38 | [TK]D-Fender | jgoo_: I didn't say it doesn't record to a file. I'm saying you have to call the REAL "Record" app in the dialplan to get the variable. |
| 12:52.40 | jgoo_ | (segnor) |
| 12:52.58 | Wall | jmls, oks sorry |
| 12:53.01 | tzanger | completely off-topic but perhaps someone here knows... are there any Indian holidays or festivities going on? I haven't seen any of the people from India that I converse with lately |
| 12:53.10 | jgoo_ | [TK]D-Fender: you know that for sure? or a hunch? |
| 12:53.21 | jmls | Wall: I was joking with you. |
| 12:53.30 | *** join/#asterisk myiagy (n=myiagy@200.175.61.250.static.gvt.net.br) |
| 12:53.41 | [TK]D-Fender | jgoo_: Use some common sense here. |
| 12:53.45 | jmls | jgoo_: don't mess with the [TK]D-Fender |
| 12:53.59 | *** part/#asterisk Wall (n=mnose@host96.201-253-161.telecom.net.ar) |
| 12:54.10 | jgoo_ | [TK]D-Fender: just asking if this is common knowledge, or a suspicion and I should dig deeper |
| 12:55.26 | jgoo_ | back to square one. fastagi isn't very useful as it is righ tnow. |
| 12:56.10 | *** join/#asterisk solomac (n=albert@tdev225-46.codetel.net.do) |
| 12:56.18 | jmls | jgoo_: why do you need to use AGI anyway ? What can't you do in the dial plan ? |
| 12:56.42 | jgoo_ | jmls: I have a dev environment here with 3500 + classes |
| 12:57.02 | jgoo_ | if I can handle everything in a simple agi thread, I can make very cool apps. |
| 12:57.23 | jgoo_ | if I have to hack in dialplan scripting, I can make very unconnected apps |
| 12:57.31 | solomac | hello i need help conecting a h323 terminal to asterisk (trixbox) |
| 12:58.08 | *** join/#asterisk ambriento (n=ambrient@201-27-80-82.dsl.telesp.net.br) |
| 12:59.12 | jmls | depends on what you want to achieve. we have a cool app, linked with and integrated with asterisk, but using simple jabber, astmanproxy and dial scripts. works like a charm. |
| 12:59.30 | [TK]D-Fender | jgoo_: No, I'm not at all experienced in this, but I call it like I see it. Everything about these functions indicates they are 100% independent of each other and you should not assume anything about them is the same beyond the fact they both allow you to record a file |
| 12:59.53 | [TK]D-Fender | solomac: .... |
| 12:59.56 | [TK]D-Fender | ~trixbox |
| 13:00.11 | jbot | i heard trixbox is NOT supported here! People using it should join #trixbox or #freepbx (FreePBX is the new name of AMP) |
| 13:00.14 | [TK]D-Fender | Any ops awake here? |
| 13:00.38 | jgoo_ | [TK]D-Fender: and that is a great suggestion for me to start looking into. if the agi calls are different from dialplan calls, that would make me *tear*. |
| 13:00.57 | jgoo_ | #trixbox doesn't exist... |
| 13:01.09 | [TK]D-Fender | jgoo_: You know the filename. You can set a channel variable yourself if you want to and so far I doubt you even have a need. |
| 13:01.17 | sizzla | Hello, |
| 13:01.18 | sizzla | <PROTECTED> |
| 13:01.18 | sizzla | Which codec G729 do I have to use? |
| 13:01.34 | sizzla | Here the CPU information of my server |
| 13:01.41 | ambriento | omg |
| 13:01.42 | jgoo_ | [TK]D-Fender: why would you doubt? I set 3 chan variables for the voicemail, I included them in vm_email.inc - but they were blank |
| 13:01.44 | sizzla | VIA Esther processor 2000MHz |
| 13:01.52 | jgoo_ | because, the channel voicemail runs in is a different channel |
| 13:02.14 | *** part/#asterisk solomac (n=albert@tdev225-46.codetel.net.do) |
| 13:02.16 | akoch | has someone a Idea for my proble described above with the 1.2.13 ??? |
| 13:02.18 | sizzla | thank you for your help |
| 13:03.11 | hegemoOn | sizzla: give thanks and praise |
| 13:03.39 | [TK]D-Fender | jgoo_: Where? I don't recall seeing them. |
| 13:05.07 | *** join/#asterisk Pazzo (n=thomas@dialin-225136.rol.raiffeisen.net) |
| 13:05.46 | jgoo_ | [TK]D-Fender: that was a previous issue I was working on, jmls was helping be debug it, I was calling api methods, then called SET VARIABLE __FOO "lol" directly, still ${FOO} was blank in the voicemail channel |
| 13:06.16 | jgoo_ | but, I think that is an issue with the channels being different, not the api failing. =[ |
| 13:06.53 | jgoo_ | (because I could set and get to verify this working, and get other asterisk variables) |
| 13:07.17 | [TK]D-Fender | jgoo_: You failed to show me a complete sample that doesn't work... |
| 13:07.59 | jgoo_ | [TK]D-Fender: I don't need to show you an example. I am not asking for your help on that, and I have established findings that show what is not working. I didn't fail at anything :-) |
| 13:08.21 | angryuser | hi fender:) |
| 13:08.54 | *** join/#asterisk hi365 (n=hi365@bzq-167-158.dsl.bezeqint.net) |
| 13:08.55 | angryuser | any resource on how-to implement billing in asterisk? |
| 13:09.01 | [TK]D-Fender | jgoo_: You showed only an attempt to retrieve a variable from a function that you assumed made one. this does not show me anything to help with this new problem. |
| 13:09.03 | jgoo_ | anyway, [TK]D-Fender: drop it, I am not going to argue over the issues here, I am just curious if anyone has AGI experience with variables and voicemail, nothing more |
| 13:09.14 | *** join/#asterisk Ebola (n=Ebola@host86-136-130-111.range86-136.btcentralplus.com) |
| 13:09.20 | jeremy_g | trying to test my * box, need a hand with sipsak. i want to register with username:a password:b and then want sip sak to send a 480 response to invite received from * box |
| 13:09.23 | jeremy_g | how to do that?? |
| 13:09.33 | jmls | jgoo_: you won't get anywhere by pissing people off. We're all trying to help here |
| 13:09.34 | [TK]D-Fender | angryuser: Not my field, sorry. I have heard the name "a2billing" thrown around though. Start there maybe |
| 13:09.43 | angryuser | ok thx |
| 13:09.45 | jgoo_ | Yes, as I mentioend [TK]D-Fender: that was a previous issue. I can show you older code, but, I think I put this older (new) issue to bed, but if you wanna take a look, I'll pastebin it |
| 13:09.58 | *** join/#asterisk hi365 (n=hi365@bzq-167-158.dsl.bezeqint.net) |
| 13:10.39 | [TK]D-Fender | jgoo_: If you want someone elses input on it sure. I'm not looking for an argument here, just trying to help. And yeah that means you have to acutally provide the useful stuff ;) |
| 13:11.08 | jgoo_ | [TK]D-Fender: I appreciate that, I gotta work out what useful in a lot of the time, as my frame of reference is probably different |
| 13:11.10 | jgoo_ | :p |
| 13:12.05 | hi365 | really off topic here but how do i automiticaly do: /msg NickServ IDENTIFY <password> every time i log in? |
| 13:12.05 | hi365 | in mIRC |
| 13:12.05 | *** join/#asterisk ShipHead (i=ShipHead@gateway/tor/x-140b2b3223938e67) |
| 13:12.21 | [TK]D-Fender | hi365: in the "perform" section of your connection. |
| 13:13.03 | [TK]D-Fender | hi365: Just don't bother to add "/join #asterisk", because it always reorders the lines so the join happens BEFORE the MSG therefore getting you rejected. |
| 13:13.13 | [TK]D-Fender | mIRC sucks that way..... |
| 13:13.35 | [TK]D-Fender | mind you Chatzilla doesn't show me enter/leave messages so I should WELCOME that problem again... |
| 13:13.58 | hi365 | well i set auto join via the favorites button on the toolbar |
| 13:16.26 | jgoo_ | Without going to far, I ran many set variables, including "SET VARIABLE __TESTA TESTA" ... in vm_email.inc I placed emailsubject= ${CHANNEL} A${TESTA}. The channel name printed out, the variable didn't. The channel was **Unknown** versus Zap/1-1 as my channel. |
| 13:16.31 | jgoo_ | pastebin: [TK]D-Fender: http://pastebin.ca/218979 |
| 13:17.07 | jgoo_ | that is a test-case of the issue I am experiencing prior to the Recording one (which I suspect you are correct in that the agi call doesn't behave the same as the dialplan app) |
| 13:18.32 | *** join/#asterisk hi365 (n=hi365@bzq-167-158.dsl.bezeqint.net) |
| 13:18.48 | hi365 | [tk] it worked but didnt work |
| 13:18.57 | hi365 | i.e. login after /j |
| 13:18.58 | *** join/#asterisk Eliran_Itzhak (n=eliran@bzq-88-152-190-223.red.bezeqint.net) |
| 13:19.36 | hi365 | is there no wait command or a way to force the order? |
| 13:20.57 | jgoo_ | hi365: start giving out flyers, build connections, earn influence, and rise to the top, then you can force the order. That is my 5 year plan. |
| 13:21.34 | *** join/#asterisk Mercestes (n=Merceste@rrcs-71-41-157-70.sw.biz.rr.com) |
| 13:21.58 | [TK]D-Fender | jgoo_: Have you considered that Voicmail does not fully parse variables and is only looking at a very fixed list of valid values? |
| 13:22.53 | [TK]D-Fender | jgoo_: Considering that the vaiables it DOES use are not at your disposal upon exit... |
| 13:23.17 | tzanger | wtf |
| 13:23.33 | tzanger | is that shit about bkw being asked to leave true? |
| 13:24.37 | [TK]D-Fender | tzanger: Yes, well corroborated by both sides. Learn the lesson from his being kicked out last night and let sleeping dogs lie.... |
| 13:24.46 | jgoo | [TK]D-Fender: hrm. It can read ENV(variables) I think... I assumed that ${} lookups was standard code. Curse me for assuming that! jmls thinks it is because of the channel name difference because it is an **Unknown** channel, the inheritance breaks |
| 13:25.11 | hi365 | jgoo uve got it all figured out, eh? |
| 13:25.17 | hi365 | k. im waiting... |
| 13:25.24 | [TK]D-Fender | jgoo: Assuming anything in * is "solid" or "consistant" sounds kinda silly doesn't it? :) |
| 13:25.26 | tzanger | [TK]D-Fender: wow. |
| 13:25.30 | jgoo | no.... *tear* |
| 13:26.05 | jgoo | ok, so, I will write my own voicemail app. w00t! I am glad I get paid for doing this, I would cry if I spent all day coming to this conclusion on my own time muahhaha |
| 13:26.44 | jgoo | Thanks for being a ear for my troubles [TK]D-Fender |
| 13:27.06 | acctor | are ATA-18x firmware versions 3.x ok to use with asterisk? I remember reading on voip info at one point that they recommended 2.x |
| 13:27.09 | jgoo | you raised some very interesting points, I will try and think less like a developer when wondering why things don't work :p |
| 13:27.10 | acctor | that seems to be gone now though |
| 13:27.45 | *** join/#asterisk pingwin[work] (i=pingwin@gateway/tor/x-0782487513d61761) |
| 13:28.17 | jgoo | heh, and thanks jmls, I think you were right about the channels too, good help. jmls: can I grok some of your dialplan? it sounds sufficiently complex that I may learn something from it. |
| 13:28.31 | *** join/#asterisk af_ (n=af@ip-170-152.sn1.eutelia.it) |
| 13:29.35 | *** join/#asterisk DarKnesS_WolF (n=wolf@62.114.197.195) |
| 13:30.33 | DarKnesS_WolF | hello i'm trying to install phpagi but i don't know what should i do ? i movied phpagi.example.php to /etc/asterisk where i should place the php files ?? /var/lib/asterisk/agi-bin/ ?? |
| 13:31.14 | *** join/#asterisk trelane_ (n=trelane@unaffiliated/trelane) |
| 13:31.54 | *** join/#asterisk dasenjo (n=dasenjo@208.195.215.211) |
| 13:32.26 | *** join/#asterisk SpaceBass (n=sp@static-71-251-230-6.rcmdva.fios.verizon.net) |
| 13:33.30 | *** join/#asterisk |stefan| (n=stefan@119cable89.soderhamn-net.com) |
| 13:33.42 | |stefan| | in what app_ module is the GROUP() function located ? |
| 13:33.48 | *** join/#asterisk xezz (n=xez@serial.trust-it.gr) |
| 13:34.19 | *** join/#asterisk CunningPike (n=CunningP@204.239.8.149) |
| 13:37.05 | *** join/#asterisk DeeJay[2] (n=deejay2@office.abi.ca) |
| 13:37.25 | DeeJay[2] | Is there any T.38 support module for asterisk out there? |
| 13:38.11 | *** join/#asterisk caciano (n=caciano@200.198.105.46) |
| 13:40.27 | |stefan| | in what app_ module is the GROUP() function located ? ?? |
| 13:41.28 | *** join/#asterisk hi365 (n=hi365@bzq-167-158.dsl.bezeqint.net) |
| 13:42.28 | pingwin[work] | is there a better app, preferabally php app, that reports the CDR reports? would like it to include Queue stats as well |
| 13:43.12 | *** join/#asterisk gerhard7 (n=gerhard@a213-84-7-87.adsl.xs4all.nl) |
| 13:43.45 | [TK]D-Fender | pingwin[work]: there are all sorts of links on the WIKI. Search there |
| 13:44.20 | [TK]D-Fender | pingwin[work]: When in doubt, search for yourself |
| 13:44.27 | [TK]D-Fender | pingwin[work]: Or make your own. |
| 13:44.35 | pingwin[work] | [TK]D-Fender: I have, and the best I found was asterisk-stats and I'd just like that, but would like something with queue stats |
| 13:44.45 | coppice | DeeJay[2]: what kind of T.38 support are you looking for? |
| 13:44.45 | pingwin[work] | [TK]D-Fender: i have already searched, and I can make my own |
| 13:44.58 | [TK]D-Fender | pingwin[work]: Queuemetrics <- |
| 13:45.10 | *** join/#asterisk scurb_ (n=scurb_@dsl253-055-082.dfw1.dsl.speakeasy.net) |
| 13:45.18 | *** join/#asterisk dsfr (i=dsfr@pdpc/sponsor/digium/dsfr) |
| 13:45.23 | *** join/#asterisk _cmach (n=caciano@200.198.105.46) |
| 13:45.24 | pingwin[work] | [TK]D-Fender: cool, I'll check that out, thanks |
| 13:46.23 | wow | hum, with 1.4 if I say with-odbc i got checking for SQLConnect in -lodbc... no |
| 13:46.50 | wow | but i have : rpm -qa | grep -i odbc |
| 13:47.00 | wow | unixODBC-devel-2.2.11-1.RHEL4.1 |
| 13:47.08 | CunningPike | pingwin[work]: We use Astrisk Guru's queue_stats: http://www.asteriskguru.com/tools/queue_stats.php |
| 13:47.19 | CunningPike | pingwin[work]: We're very happy with it |
| 13:47.23 | wow | and sql.h is into : /usr/include/sql.h |
| 13:47.24 | pingwin[work] | CunningPike: thanks! I'm going to check that out |
| 13:47.46 | *** join/#asterisk Muzkur (n=muzkur@200-206-138-117.dsl.telesp.net.br) |
| 13:47.59 | CunningPike | pingwin[work]: It does queue stats only - we rolled our own CDR reports |
| 13:48.07 | Muzkur | anyone can help me to configure an ata planet vip-156? |
| 13:48.57 | pingwin[work] | CunningPike: that's alright, we're pretty happy with Asterisk-Stats CDR reports, just need some queue stats also |
| 13:49.34 | CunningPike | pingwin[work]: Yup - queue_stats should work fine for you then |
| 13:50.04 | _cmach | could anyone help me with a "one way audio" problem on a multi homed asterisk |
| 13:50.27 | _cmach | without nat between the endpoints |
| 13:50.30 | tzanger | _cmach: verify that it's not sending audio out the wrong IP |
| 13:50.43 | tzanger | wait: multihomed as in multihomed IP space or multihomed as in multiple IPs |
| 13:51.06 | _cmach | the "rtp debug" of the asterisk console shows the streams in both ways |
| 13:51.34 | _cmach | but one side has no sound |
| 13:51.48 | tzanger | verify with tcpdump |
| 13:52.29 | _cmach | tcpdump too, show the rtp ok |
| 13:52.39 | _cmach | :( |
| 13:53.20 | _cmach | i'm using an old version of asterisk |
| 13:53.24 | _cmach | 1.2.0 beta |
| 13:54.38 | _cmach | beta1 |
| 13:56.19 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 13:56.19 | *** mode/#asterisk [+o mog] by ChanServ |
| 14:00.52 | [TK]D-Fender | bbiab |
| 14:03.41 | |stefan| | in what app_ module is the GROUP() function located ? ?? |
| 14:05.03 | *** join/#asterisk Cresl1n (n=matt@dsl253-055-082.dfw1.dsl.speakeasy.net) |
| 14:05.03 | *** mode/#asterisk [+o Cresl1n] by ChanServ |
| 14:06.05 | *** join/#asterisk Deeewayne (n=dwayne@ool-44c0d56e.dyn.optonline.net) |
| 14:10.31 | *** join/#asterisk coteyr (n=rcotey@rrcs-71-40-173-130.se.biz.rr.com) |
| 14:13.09 | coteyr | well I thought I would try here. But I'm new to asterisk so be kind. I am trying to connect an asterisk machine to my avaya System. I want to use it to record calls. For right now I am useing an X100P card. It is supposed to DIAL the avaya system. Dial a piticular extention. Then record untill something happens. The something is not yet defined. What I am tryign to define the something as is "when the Caller Record changes" |
| 14:13.27 | coteyr | the carrer record I mentioned before is not callerid as far as I can tell |
| 14:13.57 | coteyr | on the Avaya phones (the 6408D+ for sure) it's the information displayed on the LCD screen of the phone |
| 14:14.32 | coteyr | is there anyway known to get this information. I tried useing the CALLERID function but it returns null |
| 14:15.35 | *** join/#asterisk Cresl1n (n=matt@dsl253-055-082.dfw1.dsl.speakeasy.net) |
| 14:15.36 | *** mode/#asterisk [+o Cresl1n] by ChanServ |
| 14:17.03 | *** join/#asterisk mfroes (n=mfroes@200-162-218-81.corp.ajato.com.br) |
| 14:17.13 | xezz | hello , when i call my queue i can see the asterisk CLI output : -- Playing 'queue-thankyou' (language 'en') |
| 14:17.16 | mfroes | do anyone know a sip client for linux ? |
| 14:17.20 | mfroes | ekiga is not working |
| 14:17.27 | *** join/#asterisk hi365 (n=hi365@bzq-167-158.dsl.bezeqint.net) |
| 14:17.29 | xezz | is there a way to change that ? |
| 14:17.38 | hi365 | anyone have a grandstream 2000? |
| 14:17.45 | hi365 | im hearing a crazy echo |
| 14:18.09 | *** join/#asterisk compueater (n=eonblu_e@63.247.107.130) |
| 14:19.17 | xezz | i have a grandstream 102 and im hearing a crazy echo to hi365 |
| 14:19.52 | Muzkur | anyone can help me to configure an ata planet vip-156? |
| 14:20.25 | Muzkur | ive onfigured in sip.conf and in ata webconfig but asterisk give me a authentication error |
| 14:21.29 | *** join/#asterisk QbY (n=Kelvin@66.236.241.67.ptr.us.xo.net) |
| 14:21.43 | QbY | my boss is an asshole. |
| 14:22.05 | QbY | I'm offering a bounty for anyone who has a working configuration for a Cisco 7971G that works with Asterisk. |
| 14:22.10 | compueater | my asterisk keeps going down -- itll be fine for a little while and then the phones will just stop working |
| 14:22.22 | compueater | i get some fatal error regarding wfxo or something |
| 14:23.45 | backblue | compueater: www.buysipphones.com |
| 14:24.29 | compueater | backblue i have some cisco 7960s |
| 14:24.35 | compueater | are those no good? |
| 14:25.07 | backblue | yes, they are. |
| 14:26.37 | compueater | when i do an init 1 and init 3 the phones come back up |
| 14:26.37 | *** join/#asterisk sb_mx (n=sb_mx@200.78.229.18) |
| 14:26.41 | compueater | so something is failing |
| 14:26.50 | compueater | if i stop and start asterisk it doesnt fix the problem |
| 14:26.56 | compueater | i have to init 1 init 3 |
| 14:27.04 | compueater | what else runs for asterisk to start |
| 14:27.40 | *** join/#asterisk salvatore_ (n=sal@217-133-51-177.b2b.tiscali.it) |
| 14:28.05 | salvatore_ | helo |
| 14:28.12 | backblue | compueater: zaptel? |
| 14:28.27 | salvatore_ | j use zaptel, yes |
| 14:29.15 | *** join/#asterisk xnon (i=xnon@200.8.30.50) |
| 14:29.17 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 14:29.47 | [TK]D-Fender | So.... any ops awake now? |
| 14:29.50 | compueater | is zaptel something that would fail periodically? |
| 14:30.42 | backblue | compueater: yes |
| 14:30.44 | salvatore_ | j have some troubles using bristuff and qozap in p2p mode.Can Someone help or suggest me something? |
| 14:30.57 | *** join/#asterisk tsurk0 (n=tsurko@vpn-pppoe-213-240-242-237.megalan.bg) |
| 14:31.06 | backblue | compueater: i sugest you to restart zaptel and asterisk, every day, in a none-working hour. |
| 14:31.17 | backblue | put a cronjob or something. |
| 14:31.28 | salvatore_ | thanks blackblue |
| 14:31.41 | compueater | hmm -- ok |
| 14:32.10 | compueater | does Trixbox come with an ircd installed by default? |
| 14:32.13 | compueater | i see an ircd service |
| 14:32.24 | *** join/#asterisk wunderkin (i=kev@ip72-208-3-42.ph.ph.cox.net) |
| 14:32.58 | salvatore_ | but j see the lines do not work, and this happens more than 1 time in a day |
| 14:33.51 | salvatore_ | the telco tells me * has too much errors, so NTs go bad |
| 14:34.05 | *** join/#asterisk Ox0F0-0FF (n=pierre@201.38.238.66) |
| 14:34.18 | *** join/#asterisk juanjoc (n=juanjoc@201.216.212.113) |
| 14:34.37 | compueater | wcfxs not found |
| 14:34.39 | compueater | what does that mean |
| 14:35.37 | salvatore_ | are you speeaking with me? |
| 14:38.32 | [TK]D-Fender | compueater: Not a module you should be caring about |
| 14:40.16 | compueater | ok |
| 14:41.02 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 14:41.02 | *** mode/#asterisk [+o anthm] by ChanServ |
| 14:42.21 | *** join/#asterisk Qwell (n=north@unaffiliated/qwell) |
| 14:42.21 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 14:43.25 | [TK]D-Fender | anthm: Hey, care to do a small service for us and put the #freepbx warning back into the channel topic for us? Would be greatly appreciated... |
| 14:43.51 | mishehu | freepbx warning? what was it warning? |
| 14:44.50 | [TK]D-Fender | mishehu: About this channel not being a place for support on it. |
| 14:45.25 | QbY | does anyone have a better suggestion for debugging 401 Unauthorized errors? than sip debug? |
| 14:45.32 | mishehu | [TK]D-Fender: ah |
| 14:45.38 | mishehu | as I said, I know nothing about that channel ;-) |
| 14:45.59 | mishehu | QbY: there's that or tcpdumping |
| 14:46.15 | [TK]D-Fender | QbY: Not much to say. user/pass/realm is not correct. Thats all.... |
| 14:46.20 | mishehu | I personally know of no better way to debug sip |
| 14:46.31 | [TK]D-Fender | QbY: Means double check everything and when in doubt pastebin it up. |
| 14:46.47 | mishehu | QbY: your URL might be wrong, especially if you're using macro std-exten |
| 14:47.24 | QbY | Well, I'm just trying to register a phone, and its saying 401 Unauthorized.. User and Pass are correct.. I'd like to know why Asterisk thinks differently |
| 14:48.06 | mishehu | QbY: ah, well, donno... I'd do either sip debug or tcpdump |
| 14:48.09 | [TK]D-Fender | QbY: I seriously doubt its correct..... pastebin your sip.conf and the failed call attempt with SIP debug enabled |
| 14:48.19 | mishehu | tcpdump might just be a l ittle easier to browse thru in wireshark or something |
| 14:48.28 | [TK]D-Fender | QbY: It doesn't lie about this stuff..... |
| 14:48.52 | mishehu | [TK]D-Fender is always a patient person |
| 14:48.53 | QbY | give me one sec. |
| 14:49.16 | anthm | does that really help? |
| 14:49.55 | Inez | do anyone use L(...) feature for Dial cmd? |
| 14:49.57 | coppice | If he's always a patient, he needs all the help he can get |
| 14:50.39 | mishehu | coppice: not as much as you need with your jokes sometimes heh |
| 14:50.44 | *** join/#asterisk amdtech (n=amd011@ss-5-100.shsu.edu) |
| 14:50.49 | [TK]D-Fender | coppice: Why is it that when I see a doctor its at his "practice"? Dammit I want someone with EXPERIENCE! |
| 14:50.51 | mishehu | and I thought I made some corny ones, but you far surpass me on that |
| 14:51.48 | coppice | [TK]D-Fender: you sound like the kinda guy who would never fly with Virgin |
| 14:52.14 | mishehu | fly with a virgin girl? hmmmm |
| 14:52.15 | *** join/#asterisk Ng (n=cmsj@mairukipa.tenshu.net) |
| 14:52.18 | Ng | hi folks |
| 14:52.27 | |stefan| | in what app_ module is the GROUP() function located ? ?? |
| 14:52.36 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 14:52.46 | Ng | is there a way to control who can dial an extension? I want to make a meetme conference call that only registered users can connect to |
| 14:53.10 | mishehu | [TK]D-Fender: mutha fucka stupid ass snakes on a mutha fucking plane ? |
| 14:53.31 | [TK]D-Fender | mishehu: Yes... it WAS stupid. Worth a rental, but not box-office price... |
| 14:54.01 | mishehu | [TK]D-Fender: I'm doubtful it's even worth downloading it from a torrent or gnutella |
| 14:54.10 | coppice | why rent that, when you could rent something worth watching? |
| 14:54.10 | [TK]D-Fender | Ng: You can control who can do what an which way you want. Thats what extensions.conf is for. |
| 14:54.29 | jgoo | [TK]D-Fender: is flight sim 2001 *the* best flight sim so far? I am interested in spending a few hours. I heard some raves about flightgear... |
| 14:55.00 | coppice | The best flight simulators are redifussion and link-miles |
| 14:55.05 | [TK]D-Fender | jgoo: I'm betting you completely missed my terribly off-colour joke..... |
| 14:55.14 | jgoo | you mean the '2001' :-) |
| 14:55.25 | jgoo | I was going to say, that is the one with the towers still in it |
| 14:55.40 | [TK]D-Fender | jgoo: You learn quickly my young Jedi.... |
| 14:55.46 | jgoo | and the 'crash course' no I saw it, but I am dehydrated, and over sugared |
| 14:56.04 | Ng | [TK]D-Fender: ah ok, so I guess I can use a GotoIf()? If so, would you happen to know what I should be reading about for testing if a user is registered? |
| 14:56.12 | *** join/#asterisk palyza (n=pj@ip-85-160-12-70.eurotel.cz) |
| 14:56.54 | [TK]D-Fender | Ng: Why would you put UNAUTH'd calls into the same context as those used by registered devices? Very unhealthy practice... |
| 14:57.13 | [TK]D-Fender | Ng: You need to rethink your concept of contexts.... |
| 14:57.40 | coppice | I read my son a book from the library about the guy who tightrope walked bewteen the towers. It seemed odd to read "once upon a time 2 towers stood here" |
| 14:57.50 | QbY | [TK]D-Fender.. Per your request: http://pastebin.ca/219125 |
| 14:58.32 | *** join/#asterisk Ebola (n=Ebola@host86-136-130-111.range86-136.btcentralplus.com) |
| 14:58.57 | Ng | [TK]D-Fender: it's an inherited config, at the moment everything is just in the default context, which does make sense for the setup as-was. Is there a way I can do this without refactoring the entire config? |
| 14:59.54 | [TK]D-Fender | Ng: You clearly need to make seperate contexts. Time to get to work... |
| 14:59.57 | *** join/#asterisk DarKnesS_WolF (n=wolf@62.114.187.108) |
| 15:00.02 | Ng | bah ;) |
| 15:01.21 | [TK]D-Fender | QbY: Kill the username in sip.conf, and reload and retry |
| 15:01.36 | *** join/#asterisk SuPrSluG (n=SuPrSluG@pool-71-241-169-13.buff.east.verizon.net) |
| 15:01.48 | SuPrSluG | hello |
| 15:02.05 | salvatore_ | hello |
| 15:02.06 | DeeJay[2] | coppice : I just want to be able to send and receive fax thru a linksys SIP device.. |
| 15:02.10 | DeeJay[2] | (routers) |
| 15:02.25 | DeeJay[2] | nomatter how... i just want it to work. |
| 15:02.25 | salvatore_ | iaxmodem? |
| 15:04.19 | coppice | DeeJay[2]: that doesn't really answer my question. I assume the linksys is suppsoed to support T.38. Where is the other end? Do you just need T.38 to pass through *, or do you need a gateway to the PSTN? |
| 15:04.50 | DeeJay[2] | I need to send it via a PRI |
| 15:04.56 | DeeJay[2] | which is connected to asterisk |
| 15:07.54 | coppice | DeeJay[2]: then you are out of luck right now. passthrough, with some limitations for the real world, is in the * 1.4 betas. other modes of T.38 are only in my stuff, and not integrated with * |
| 15:08.21 | QbY | [TK]D-Fender.. No Change. |
| 15:08.26 | salvatore_ | j have some troubles using qozap in p2p mode.Can Someone help or suggest me something? |
| 15:08.41 | [TK]D-Fender | QbY: Try it with another phone. |
| 15:08.55 | QbY | i have a ton of other phones reigstered.. |
| 15:08.58 | QbY | its only this stupid 7971 that my boss "had to have" |
| 15:10.01 | palyza | anyone here knows, about status of Q.SIG in zaptel/asterisk? especially if "caller id name" transfer/display is supported |
| 15:10.43 | palyza | I would like to connect asterisk to siemens Hipath using PRI/Q.SIG and I need caller id name transfer between IP phone and "legacy" phone behind siemens pbx |
| 15:11.28 | [TK]D-Fender | QbY: I'd then lay bets the phones config just isn't working like you expect it to. Got other Cisco's that work fine? |
| 15:11.49 | QbY | yeah, all my ciscos are running good |
| 15:12.01 | QbY | that's why i offered a bounty for anyone who could provide me a working 7971G config |
| 15:12.08 | QbY | why does he have to make my life miserable |
| 15:12.58 | *** join/#asterisk mrg82 (n=na@office.intercea.co.uk) |
| 15:13.07 | backblue | anyone can sugest me a good wiki sip phone? |
| 15:13.23 | mutilator | no |
| 15:13.28 | Qwell | wiki phone? |
| 15:14.09 | *** join/#asterisk techie (n=techie@c-67-182-22-174.hsd1.ca.comcast.net) |
| 15:14.10 | mrg82 | I have an account with a sip provider that gives me a geographical number. When I call i want the the outgoing CID to be my mobile number? Is this not possible? |
| 15:14.16 | *** part/#asterisk techie (n=techie@c-67-182-22-174.hsd1.ca.comcast.net) |
| 15:14.18 | *** part/#asterisk amdtech (n=amd011@ss-5-100.shsu.edu) |
| 15:14.21 | backblue | ups, wifi phone |
| 15:14.22 | backblue | :) |
| 15:14.51 | [TK]D-Fender | QbY: <contact>7b452e87-4496-4762-e11f-b26751a1884b</contact> |
| 15:15.05 | [TK]D-Fender | QbY: <contact>sip username</contact> |
| 15:15.12 | [TK]D-Fender | QbY: Try setting them the same |
| 15:15.15 | QbY | k |
| 15:15.19 | QbY | i did earlier |
| 15:15.22 | QbY | will try again |
| 15:15.51 | Qwell | mrg82: depends on the provider |
| 15:16.04 | *** join/#asterisk marv[work] (n=timr@br0.asteriasgi.com) |
| 15:16.32 | *** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin) |
| 15:16.34 | PakiPenguin | hello :) |
| 15:17.15 | *** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin) |
| 15:18.56 | compueater | is caller id data something that needs to be allowed at the service provider end or is it a setting i can set? |
| 15:19.32 | [TK]D-Fender | compueater: The answer was 2 lines up.... go get some coffee |
| 15:20.04 | PakiPenguin | :p [TK]D-Fender i need coffee too |
| 15:20.22 | *** join/#asterisk syzygyBSD (n=chatzill@66.226.228.204.cpe.speedyquick.net) |
| 15:20.29 | syzygyBSD | Morning all |
| 15:20.38 | *** join/#asterisk dr0ck (i=dr0ck@nat/digium/x-64cbdc142645ad98) |
| 15:20.54 | [TK]D-Fender | load chan_na_masala.so :O |
| 15:21.24 | PakiPenguin | :p |
| 15:21.25 | PakiPenguin | nope |
| 15:21.35 | PakiPenguin | load chan_kheer.so :) |
| 15:21.57 | *** join/#asterisk wulfy814 (n=lorentz@216.48.0.4) |
| 15:22.13 | wulfy814 | I'm having trouble with call parking in trunk |
| 15:22.24 | wulfy814 | I'm using Polycom 430's and 601's |
| 15:23.38 | wulfy814 | when I tranfer the call to 700, it reads back the parking slot, but when hitting transfer again it doesn't complete the action |
| 15:23.49 | wulfy814 | it stops the music on hold for the parked party and that's it |
| 15:23.56 | wulfy814 | any ideas? |
| 15:26.47 | *** part/#asterisk sulex (n=sulex@pdpc/supporter/active/sulex) |
| 15:27.50 | *** join/#asterisk _alex_mx_ (n=alex@200.78.229.18) |
| 15:30.12 | *** join/#asterisk xezz (n=xez@serial.trust-it.gr) |
| 15:30.49 | *** join/#asterisk alerios (n=alerios@190.24.98.181) |
| 15:30.50 | *** join/#asterisk Koshatul (n=evangeli@ppp188-33.static.internode.on.net) |
| 15:31.01 | compueater | i've had a problem where sometimes coming off of hold the person cannot hear me but i can hear them |
| 15:32.33 | wulfy814 | quit |
| 15:36.39 | *** join/#asterisk Koshatul (n=evangeli@ppp188-33.static.internode.on.net) |
| 15:37.37 | *** join/#asterisk p0g0_ (n=pogo@madwifi/support/p0g0) |
| 15:38.33 | *** join/#asterisk RoyK (n=roy@ti211310a080-4407.bb.online.no) |
| 15:38.59 | *** join/#asterisk kristalino (n=kristali@cl-157.dub-01.ie.sixxs.net) |
| 15:41.41 | xezz | <PROTECTED> |
| 15:43.47 | *** join/#asterisk wangster (n=wangster@static-64-201-170-129.ptr.terago.ca) |
| 15:44.01 | *** join/#asterisk dasenjo (n=dasenjo@63.245.86.106) |
| 15:44.28 | wangster | In "Zap show channel", there is a field "Fax Handled:", when does this get set? |
| 15:44.58 | wangster | Even though our zap channels are saying "Fax detected", that field is staying set as "no". |
| 15:45.32 | brif8 | Does the CP-101 come in black or are all clipcomm phones white ? |
| 15:46.59 | [TK]D-Fender | xezz: What * version are you running? |
| 15:49.32 | xezz | 1.2.10 |
| 15:50.03 | [TK]D-Fender | xezz: Then I highly suggest you to upgrade IMMEDIATELY. There is a remote root exploit that come in through chan_skinny. |
| 15:50.12 | [TK]D-Fender | xezz: You are probably being hacked as we speak. |
| 15:50.40 | xezz | well , i dont think so |
| 15:50.50 | xezz | about the second * |
| 15:50.59 | xezz | cause the second ip was from my network |
| 15:51.01 | [TK]D-Fender | xezz: You using skinny for Cisco phones? Either way I've covor your ass if I were you... |
| 15:51.22 | xezz | yes |
| 15:52.37 | carrar | xezz# ls -la |
| 15:52.39 | carrar | wow |
| 15:52.41 | carrar | I got roto! |
| 15:52.43 | carrar | roto! |
| 15:52.45 | carrar | ack |
| 15:52.47 | carrar | root |
| 15:52.48 | carrar | haha |
| 15:53.04 | xezz | sure |
| 15:53.12 | carrar | heh |
| 15:53.13 | RoyK | carrar: try this: dd if=/dev/urandom of=`mount | grep -w / | awk '{ print $1 }'` |
| 15:53.16 | Corydon-w | Welcome to 1991 |
| 15:53.31 | carrar | RoyK, ok waiting,how long does it take? |
| 15:53.39 | carrar | ..still running |
| 15:53.51 | carrar | heh |
| 15:54.08 | tzafrir | carrar, if you were serious, you don't have root anymore |
| 15:54.19 | [TK]D-Fender | RoyK: ASS |
| 15:54.26 | carrar | For those without humor: I am not serious |
| 15:54.30 | tzafrir | RoyK, not nice |
| 15:54.54 | carrar | Was my lame attempt at glee in the morning |
| 15:54.57 | tzafrir | Some people are known to actually operate upon such advices |
| 15:55.25 | [TK]D-Fender | RoyK: If I had ops, you'd be skidding out of here for suggesting something like that..... |
| 15:55.43 | [TK]D-Fender | RoyK: please don't joke about that kind of stuff.... |
| 15:56.22 | carrar | for RoyK to say that to me is ok, but probably not to anyone else who is not unix savvy |
| 15:58.06 | tzanger | bah |
| 15:58.09 | tzanger | that was funny |
| 15:59.05 | Ng | [TK]D-Fender: ok, I've split things up so we have a default context for incoming stuff, an employees context and a conferences context. I've included the employees context in the default one so anyone can call employees, but I'm not quite sure now how to get the employee and conference contexts hooked up without allowing default context to access it |
| 15:59.10 | Ng | any suggestions? |
| 15:59.18 | Ng | or an example of such a scenario (or similar) I could study |
| 15:59.25 | salvatore_ | j have some troubles using qozap in p2p mode.Does someone know if resetinterval can help in solving this problem? |
| 15:59.32 | Winkie | off topic: what ram do i buy for an amd 4200+ am2 :( |
| 15:59.48 | [TK]D-Fender | Ng: You need to think it over. its all about how you "include" contexts together..... this should be a 1 minute fix. |
| 15:59.50 | carrar | Winkie, try looking it up on memoryx.com |
| 15:59.51 | *** join/#asterisk pifiu (n=someone@216.5.79.1) |
| 16:00.02 | tzanger | Ng: [trusted] include = employees, include = confrerence |
| 16:00.04 | [TK]D-Fender | Winkie: Check your MB |
| 16:00.12 | Winkie | [TK]D-Fender: it accepts 3 speeds :( |
| 16:00.17 | *** join/#asterisk miguel3239 (n=chatzill@ns1.nashuacs.com) |
| 16:00.21 | qdk | Winkie: whatever fit you MB. |
| 16:00.21 | tzanger | Ng: contexts (many of them, simple blocks) are your friend... jus tbuild up bigger and bigger ones and include what you want in each one |
| 16:00.37 | Winkie | oh really? I guess i am too used to matching processor + memory speeds :( |
| 16:00.41 | pifiu | hey everyone |
| 16:00.41 | Ng | tzanger: hmm. at the moment I'm trying to include employees in conference, but they can't see the extensions |
| 16:00.44 | qdk | Winkie: your* |
| 16:01.25 | *** join/#asterisk luke-jr_work (n=luke-jr@2002:4335:4375:0:20d:60ff:fe60:756a) |
| 16:01.35 | qdk | Winkie: your MB will tell you that too... personaly i would go with DDR-677 for single core CPU. |
| 16:01.37 | luke-jr_work | anyone know anything about a Adit 600? |
| 16:01.44 | Winkie | qdk: it's dual core |
| 16:02.04 | Winkie | the ram could easily be the most expensive bit of this :) |
| 16:02.19 | qdk | Winkie: 800mhz then. |
| 16:02.31 | carrar | luke-jr_work, yeah I use those |
| 16:02.34 | qdk | Winkie: it IS. |
| 16:02.34 | Winkie | qdk: fair enough, ta |
| 16:02.35 | luke-jr_work | qdk, wtf is wrong with PC100? =p |
| 16:02.41 | carrar | luke-jr_work, they work awesome with asterisk |
| 16:02.45 | luke-jr_work | carrar, any reason to suspect its the cause of lotsa noise? |
| 16:02.46 | qdk | luke-jr_work: :-) |
| 16:03.02 | pifiu | hey fender |
| 16:03.04 | carrar | interfearance from like radio stations? |
| 16:03.17 | luke-jr_work | carrar, staticish |
| 16:03.25 | luke-jr_work | carrar, trying to bridge a FXS to a FXO |
| 16:03.29 | carrar | ah, sounds like stranded wiring |
| 16:03.39 | carrar | hrmm |
| 16:03.40 | luke-jr_work | both FXS/FXO go through the Adit to a Sangoma and into Asterisk |
| 16:04.10 | carrar | I haven't had static issues with mine |
| 16:04.25 | carrar | make sure to use solid copper |
| 16:04.46 | luke-jr_work | stranded wiring? :| |
| 16:04.55 | luke-jr_work | is the stuff in cat5e solid copper? :) |
| 16:04.56 | carrar | yeah stranded not good for analog |
| 16:05.15 | carrar | cat5e should be solid |
| 16:05.26 | luke-jr_work | how about a punchboard thing? |
| 16:05.27 | carrar | but not always |
| 16:05.51 | carrar | have you tried removing things till th static goes away? |
| 16:05.59 | luke-jr_work | removing what? |
| 16:06.15 | carrar | well plug a phone directly into the adit |
| 16:06.21 | carrar | or the other end |
| 16:06.30 | carrar | bbl, car pool ride is here |
| 16:06.36 | luke-jr_work | adit only has one of those long 8-channel connectors |
| 16:06.41 | luke-jr_work | aww |
| 16:07.32 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 16:10.39 | jeremy_g | this is intersting |
| 16:10.55 | jeremy_g | if u enable sip debug |
| 16:11.31 | jeremy_g | and then set verbose hight, shud it log the sip messages to the vebose log file |
| 16:11.38 | jeremy_g | shud it? |
| 16:11.47 | luke-jr_work | wtf is a verbose log file |
| 16:12.46 | jeremy_g | luke-jr_work:e.g its urass if u add to logger.conf. urass => verbose |
| 16:13.06 | PakiPenguin | :p |
| 16:13.45 | *** part/#asterisk RoyK (n=roy@ti211310a080-4407.bb.online.no) |
| 16:13.46 | jeremy_g | i start getting the sip messages in the verbose log after a minute of enabling the sip debug and set verbose to a high value |
| 16:13.48 | jeremy_g | damn |
| 16:17.59 | *** join/#asterisk hi365 (n=hi365@bzq-167-158.dsl.bezeqint.net) |
| 16:18.04 | *** join/#asterisk arcanine (n=arcanine@203.82.44.179) |
| 16:18.51 | hi365 | anyone using asttapi? |
| 16:19.13 | hi365 | it says its dialing (outlook) but its not. also taked forever to disconect a call |
| 16:22.32 | *** join/#asterisk Greek-Boy (n=grb@193.220.93.162) |
| 16:22.43 | Greek-Boy | Which of u guys use apt-get to maintain an asterisk installation? |
| 16:23.05 | Greek-Boy | I use apt-get to maintain my debian system but always build asterisk/zaptel from source |
| 16:23.57 | mog | thats what most of us do |
| 16:24.14 | Greek-Boy | ok |
| 16:24.18 | Greek-Boy | just checking :) |
| 16:26.12 | *** join/#asterisk hmmhesays (n=hmmhesay@24-117-135-28.cpe.cableone.net) |
| 16:26.20 | hmmhesays | wiki wiki wild wild |
| 16:26.30 | _cmach | <PROTECTED> |
| 16:26.32 | hmmhesays | i love ethereal |
| 16:26.48 | _cmach | without nat between the endpoints |
| 16:26.50 | hmmhesays | if you describe your problem in more detail |
| 16:26.55 | _cmach | <PROTECTED> |
| 16:27.05 | _cmach | ok |
| 16:27.19 | _cmach | two asterisk boxes |
| 16:27.27 | _cmach | without nat between them |
| 16:27.58 | _cmach | a basic configuration, one extension that call an extension on the other side |
| 16:28.03 | *** join/#asterisk xezz (n=xez@serial.trust-it.gr) |
| 16:28.09 | hmmhesays | what are you using for endpoints? |
| 16:28.19 | hmmhesays | softphones? hard phones? |
| 16:28.24 | _cmach | hard phones |
| 16:28.46 | hmmhesays | i'm guessing your reinvites aren't working right |
| 16:29.26 | _cmach | what can happen in a "not working right" reinvite :-) |
| 16:29.27 | _cmach | ? |
| 16:29.35 | hmmhesays | 1 way audio |
| 16:29.46 | [TK]D-Fender | _cmach: You should never be uising reinvites. |
| 16:29.53 | hmmhesays | set your sip.conf entries to "canreinvite=no" |
| 16:30.06 | hmmhesays | if they are on the same network there is no reason not to |
| 16:30.12 | hmmhesays | unless it presents problems |
| 16:32.12 | QbY | is it possible to dump or get asterisk to show the credentials it is being offered by a phone? |
| 16:34.17 | pifiu | hey fender wasup |
| 16:36.30 | hmmhesays | bah there is no ethereal package for x64 |
| 16:36.32 | *** join/#asterisk hohum (n=dcorbe@host-12-195-58-237.iad1.interceltelecoms.net) |
| 16:37.28 | [TK]D-Fender | ~wikis |
| 16:37.29 | jbot | extra, extra, read all about it, wikis is http://www.voip-info.org |
| 16:38.55 | _cmach | canreinvite=no doesn't solved :( |
| 16:38.58 | PakiPenguin | ~meow |
| 16:38.59 | jbot | jbot: woem |
| 16:39.06 | PakiPenguin | ~meowmeow |
| 16:39.12 | PakiPenguin | hehe |
| 16:40.18 | *** join/#asterisk Buglouse (n=SourceRa@66.97.121.210) |
| 16:43.16 | hmmhesays | ~hmmhesays |
| 16:43.17 | jbot | i heard hmmhesays is not really here... |
| 16:43.18 | jeremy_g | sth has gone wrong with my *, its not even registering with my provider |
| 16:43.19 | jeremy_g | darn |
| 16:43.34 | jeremy_g | sip show registry lists 'Request Sent' |
| 16:44.34 | jeremy_g | _cmach:what r ya tryin to solve |
| 16:44.38 | *** join/#asterisk sahafeez (n=sahafeez@67.109.14.227.ptr.us.xo.net) |
| 16:45.09 | _cmach | 1 way audio; no nat; canreinvite=no |
| 16:45.25 | hmmhesays | on both sides? |
| 16:45.27 | *** join/#asterisk Zaw (i=zaw@unaffiliated/zaw) |
| 16:45.30 | _cmach | asterisk 1.2.0beta1 |
| 16:45.42 | hmmhesays | that smell of openwrt |
| 16:45.47 | hmmhesays | *smells of even |
| 16:45.48 | jeremy_g | _cmach:why are using that |
| 16:45.55 | Zaw | what's a decent, low-cost entry level phone that works well with asterisk? i know that Sipura made a couple that i've been recommended before but i'm looking for suggestions |
| 16:45.59 | _cmach | i can't upgrade it now, i want to know the bug that is causing this |
| 16:46.06 | jeremy_g | _cmach:u can be technically fucked, it has serious vulnerbilites too |
| 16:46.25 | jeremy_g | unless u dont have or dont mind people dialing out from ur dash |
| 16:46.28 | [TK]D-Fender | Zaw: Phones worth considering Polcyom, Aastra, Linksys (in that order). |
| 16:46.34 | sahafeez | if i have sip peer setup in sip.conf to a sip provider - does it try to register one the start of asterisk and does it keep trying until it works. my sip provider is telling me that he does not see the request ever.. |
| 16:46.40 | hmmhesays | or you can consider an ata |
| 16:46.45 | CunningPike | Zaw: You could take a look at a couple of the Polycoms - 301, or 430 |
| 16:46.45 | sahafeez | s/one/on |
| 16:46.47 | [TK]D-Fender | Zaw: Model dependent on what your needs / wants are and your budget |
| 16:46.55 | Zaw | [TK]D-Fender / CunningPike: thanks |
| 16:46.56 | sahafeez | sip show registry shows nothing.. |
| 16:47.13 | [TK]D-Fender | Zaw: Polycom IP 430 should only be considered in very specific conditions. |
| 16:47.19 | sahafeez | Zaw: i have the 301, 501, and 600. good phones |
| 16:47.40 | jeremy_g | sahafeez: :) its obvious |
| 16:47.41 | Zaw | polycom 301 sounds good so far.. |
| 16:47.46 | wulfy814 | [TK]D-Fender: why don't you like the 430 ? I've had pretty good results with it |
| 16:48.01 | [TK]D-Fender | Zaw: I have IP 301, 430, 501, and 601. |
| 16:48.01 | sahafeez | it is was i would not ask.. |
| 16:48.11 | wulfy814 | Zaw: The 301's have to be powered by a brick or power injector from polycom |
| 16:48.21 | wulfy814 | the 430 & 601 are PoE compatible |
| 16:48.23 | [TK]D-Fender | wulfy814: I didn't say it wasn't a good phone. Just that if it comes to choosing a model that it might not make the list because of its price-point. |
| 16:48.52 | Zaw | i'll google for some specs on the polycom 301. thanks |
| 16:48.54 | [TK]D-Fender | wulfy814: The IP 501 is only marginally more expensive and give a much bigger & nicer screen, better speakerphone, and more line keys. |
| 16:48.59 | wulfy814 | by the time you factor in UPS for each computer/phone you are better of with PoE models in my opinion |
| 16:49.10 | hmmhesays | fscking bank of america, website been down for days |
| 16:49.15 | [TK]D-Fender | wulfy814: Depends if you are even LOOKING at PoE |
| 16:49.25 | [TK]D-Fender | wulfy814: Which is part of the "depends" I was talking about. |
| 16:49.28 | Zaw | Polycom Soundpoint IP 301 ? |
| 16:49.47 | CunningPike | Zaw: That's the one |
| 16:49.48 | [TK]D-Fender | Zaw: Yes, that is the low end. Great phone that only lacks a speakerphone |
| 16:49.55 | Zaw | very good |
| 16:50.00 | wulfy814 | the lack of speakerphone, no PoE, and lowest res display - but it sounds very good |
| 16:50.05 | wulfy814 | and is easy to deploy |
| 16:50.16 | [TK]D-Fender | wulfy814: Yup, 301 is a great little phone. |
| 16:50.29 | *** join/#asterisk scurb_ (n=scurb_@dsl253-055-082.dfw1.dsl.speakeasy.net) |
| 16:50.37 | wulfy814 | [TK]D-Fender: are you running 1.4 yet, or trunk? |
| 16:51.12 | [TK]D-Fender | wulfy814: If someone were to ask me "suggest me a general purpose phone for 20 employees with PoE", THEN I'd suggest IP 430's. If its for 2 people who can use a power brick I'd much sooner suggest the IP 501. |
| 16:51.20 | wulfy814 | [TK]D-Fender: I'm having issues with call parking with polycoms |
| 16:51.23 | [TK]D-Fender | wulfy814: Nope. I'm not a "beta" person |
| 16:51.44 | wulfy814 | [TK]D-Fender: if they've never seen a 601 and have limited needs I would agree |
| 16:51.58 | [TK]D-Fender | wulfy814: I'm on 1.2 FTP releases only until 1.4 final comes out. Even then I might wait until 1.4.1 comes out 2 days later ;) |
| 16:52.00 | wulfy814 | [TK]D-Fender: after using a 601 or 430 hard to go back |
| 16:52.17 | wulfy814 | I've gotten the presence working for the parking slots |
| 16:52.23 | wulfy814 | which I'm thrilled about |
| 16:52.27 | *** join/#asterisk TedC (n=cabeen@form.chem.ucsb.edu) |
| 16:52.38 | wulfy814 | when someone is parked in 701 it lights up an appearance button on the polycom |
| 16:52.42 | [TK]D-Fender | wulfy814: I know, but its a question of NEEDS vs WANTS. Of course the IP 601 is beter than the rest so why not just get those? Budget is a reason you know... |
| 16:52.44 | wulfy814 | and I can hit that button to pick up the call |
| 16:53.16 | [TK]D-Fender | wulfy814: Easy to do in a MicroBroswer click-to-call script as well. |
| 16:53.32 | [TK]D-Fender | wulfy814: Would be a very minor thing for me to code one day |
| 16:53.35 | wulfy814 | but when I actually park the call I'm stuck using blind transfer |
| 16:53.48 | wulfy814 | Microbrowser is only available on the 601 right? |
| 16:53.54 | [TK]D-Fender | wulfy814: Why is that? You referring to using the Polycom Park feature? |
| 16:54.00 | [TK]D-Fender | wulfy814: Correct |
| 16:54.26 | wulfy814 | if I hit transfer - 700, send (reads back slot) and hit transfer again the MOH stops on the remote end (call being parked) and nothing happens on my pcom |
| 16:54.52 | wulfy814 | if I do blind transfer (watching the console) it reads back the slot to nothing , but it successfully parks the call |
| 16:55.45 | [TK]D-Fender | wulfy814: Thats... "interesting".... |
| 16:55.50 | mutilator | figured out why it costs you $200 to get your macbook painted black: http://www.dansdata.com/black.htm |
| 16:56.04 | *** join/#asterisk coppice (n=chatzill@229.166.17.210.dyn.pacific.net.hk) |
| 16:56.05 | hmmhesays | good lord ethereal is taking FOREVER to compile |
| 16:56.58 | *** join/#asterisk aptura (n=pcsuppor@S010600a0c93f6f7e.vs.shawcable.net) |
| 16:57.11 | sahafeez | 501 is fine for a desk. we use 301 for the lower life forms, 501 for the mgrs and 601 for the conference and front desk. (tounge in check) |
| 16:57.13 | Ng | tzanger: your context suggestion isn't really working, users from the employee context can't see the conference context. I have them both in a separate trusted context, but I don't see why that would work anyway, they will be making calls from the employees context, so the trusted context won't ever be checked, surely? |
| 16:58.27 | Ng | so I'd need to inclue conference in employees, but since employees is included in default so external people can call them, that would imply that external people could also then cascade to the conference extension? |
| 16:59.48 | [TK]D-Fender | Ng: Pastebin your whole dialplan please. |
| 17:00.07 | *** join/#asterisk hoobastooba (n=ckwall@63.149.122.93) |
| 17:00.36 | Ng | [TK]D-Fender: it'll have to be a censored version, but sure. |
| 17:00.49 | *** join/#asterisk miguel3239 (n=chatzill@ns1.nashuacs.com) |
| 17:00.56 | *** join/#asterisk roving_prole (n=Harper@72-254-127-143.client.stsn.net) |
| 17:01.03 | [TK]D-Fender | Ng: Keep it to a minimum |
| 17:01.06 | hoobastooba | I see many programs that allow me to send a fax from email through asterisk. I am looking for one that will allow me to also receive. Any suggestions? I looked at Hylafax and asterfax. mail2fax and others. |
| 17:01.12 | *** join/#asterisk kilobit2001 (n=locid@206-248-152-104.dsl.teksavvy.com) |
| 17:01.25 | kilobit2001 | does asterisk support voicexml? |
| 17:01.26 | [TK]D-Fender | hoobastooba: SpanDSP |
| 17:01.29 | *** join/#asterisk Mportnoy (n=test@201.199.68.150) |
| 17:01.35 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 17:01.38 | [TK]D-Fender | kilobit2001: : not natively |
| 17:01.39 | hoobastooba | [TK]D-Fender: thanks you, googling |
| 17:01.58 | Mportnoy | Hi I need to move all the recordings of October 2006 on the monitor directory to a folder called October, how can I do this ? |
| 17:02.03 | tzanger | Ng: you put them in the trusted context, of course. |
| 17:02.15 | tzanger | Mportnoy: find is your friend |
| 17:02.23 | *** part/#asterisk hoobastooba (n=ckwall@63.149.122.93) |
| 17:02.25 | Mportnoy | tzanger: what is the command ? |
| 17:02.48 | luke-jr_work | kilobit2001, the company I work for has something for that |
| 17:02.52 | tzanger | man find, of course |
| 17:03.10 | Mportnoy | well |
| 17:03.13 | Mportnoy | can you show me the command, the whole line ? |
| 17:04.10 | jeremy_g | kilobit2001:it wud soon, they way jgoo is going |
| 17:05.15 | Ng | tzanger: I duplicate all their extensions from employees to trusted? |
| 17:05.24 | Ng | tzanger: that hardly seems optimal ;) |
| 17:05.26 | tzanger | Ng: f8uck no |
| 17:05.35 | tzanger | THINK about what you're trying to do |
| 17:05.36 | *** join/#asterisk hoobastooba (n=ckwall@63.149.122.93) |
| 17:05.39 | tzanger | holy hell people, THINK |
| 17:06.12 | Mportnoy | well the problem is that man is pretty bad |
| 17:06.12 | Ng | I appreciate you guys helping me and I do have the o'reilly book and voip-info in front of me, I am trying to figure this out on my own ;p |
| 17:06.21 | hoobastooba | i have 2 TE405P cards... Looking to unload them so I can upgrade. Anyone interested. |
| 17:06.23 | tzanger | Mportnoy: no it's not |
| 17:06.35 | tzanger | Ng: you want a trusted context that lets people call employees and conferences |
| 17:06.41 | Ng | tzanger: no |
| 17:06.42 | Mportnoy | tzanger: yes it does not show examples |
| 17:06.43 | tzanger | you already have an [employees] and [conferences] I think |
| 17:07.01 | tzanger | Mportnoy: unless you use -exec stupidly you won't break anything by trying a few options |
| 17:07.02 | Ng | tzanger: I want [default] to be able to call [employees] only and for [employees] to be able to call [conferences] |
| 17:07.09 | Ng | [TK]D-Fender: http://pastebin.com/812698 |
| 17:07.10 | tzanger | Mportnoy: nad I'm *positive* that google got something on this |
| 17:07.23 | tzanger | Ng: good, now code EXACTLY what you wrote |
| 17:07.36 | tzanger | [default] include = employees |
| 17:07.41 | tzanger | [employees] include = conferences |
| 17:07.51 | [TK]D-Fender | Ng: And next time... use pastebin.ca . .com is broken |
| 17:08.05 | Ng | [TK]D-Fender: ok |
| 17:09.09 | *** join/#asterisk _cmach (n=C@200.198.105.46) |
| 17:09.34 | Ng | tzanger: doesn't that mean that [defaults] includes employees which includes conferences, thus [default] ends up including [conferences]? |
| 17:09.38 | *** join/#asterisk insanity5 (n=fewa@216-207-205-36.dia.static.qwest.net) |
| 17:09.57 | insanity5 | What is a good orgination provider? Reliable, no bs, unlimited open lines |
| 17:10.18 | luke-jr_work | insanity5, *maybe* Teliax, but I haven't used them for years |
| 17:10.38 | insanity5 | It seems they come and go... and so do the numbers every few years |
| 17:10.43 | tzanger | Ng: then odn't do that |
| 17:10.52 | tzanger | ng: [defaults] |
| 17:10.56 | tzanger | include = employees |
| 17:11.04 | tzanger | [employees] contains *just* the extensions. that's it |
| 17:11.06 | tzanger | [trusted] |
| 17:11.08 | tzanger | include = employees |
| 17:11.12 | tzanger | include = conferences |
| 17:11.14 | *** join/#asterisk malverian (n=malveria@gentoo/developer/malverian) |
| 17:11.23 | tzanger | and have all your phones use the trusted context |
| 17:12.01 | tzanger | or, if you *must* have your sip phones use the employees context |
| 17:12.07 | tzanger | [defaults] |
| 17:12.11 | *** join/#asterisk sahafeez (n=sahafeez@67.109.14.227.ptr.us.xo.net) |
| 17:12.12 | tzanger | include = valid_extensions |
| 17:12.14 | tzanger | [employees] |
| 17:12.17 | tzanger | include = conferences |
| 17:12.23 | luke-jr_work | get rid of defaults ;p |
| 17:13.10 | tzanger | [valid_extensions] has all the extensions of hte people |
| 17:13.10 | tzanger | and yes, get rid of defaults |
| 17:13.10 | tzanger | NEVER use [default] |
| 17:13.10 | tzanger | NEVER EVER EVER |
| 17:13.10 | Ng | tzanger: aha, changing the context of each phone to trusted makes sense |
| 17:13.11 | sahafeez | jermey_g: thanks. worked. |
| 17:13.11 | Ng | why? |
| 17:13.11 | Ng | what's wrong with [default]? |
| 17:13.11 | tzanger | Ng: because it is a default. That means there may be misconfigured peers or malicious peers trying to access shit in there |
| 17:13.32 | tzanger | my [default] on all my systems is exten => _.,1,NoOp(hit ${EXTEN} in [default], this is NOT GOOD) |
| 17:13.44 | tzanger | because it means something tried to hit [default] and I never use it |
| 17:14.03 | Ng | tzanger: but the only extensions I have in [default] are ones we've specifically put in there. what "shit" are you referring to? |
| 17:14.11 | kilobit2001 | anyone can tell me how many context I need, if I have 5 menus, each with 5 nested menus? |
| 17:14.46 | anthm | and if 2 trains leave the station at the same time? |
| 17:15.01 | [TK]D-Fender | Ng: http://pastebin.ca/219355 |
| 17:15.31 | coppice | anthm: then it can't be British Rail |
| 17:15.37 | [TK]D-Fender | kilobit2001: Each menu is a context. Do the math |
| 17:15.39 | tzanger | Ng: [default] is .. well, the *default* -- it may be referenced in many places in * code. Unless you do a very careful code review you are not assured that nothing will ever end up in there |
| 17:16.05 | hmmhesays | hmmm not good not good |
| 17:16.09 | Ng | tzanger: ok |
| 17:16.11 | Ng | [TK]D-Fender: thanks |
| 17:16.15 | tzanger | there's too much cruft in * for me to trust it to never end up in [default] when [default] is referenced in a ton of sample files |
| 17:16.16 | Ng | tzanger: thanks to you too |
| 17:16.19 | hmmhesays | it seems asterisk is not passing my fax reinvite back through to my terminating gateway |
| 17:16.21 | tzanger | it's just a paranoia thing I guess |
| 17:16.22 | Ng | I reckon this will be ok now |
| 17:16.25 | jeremy_g | :D |
| 17:17.01 | [TK]D-Fender | [default] is a terrible name and should never be used. |
| 17:17.16 | [TK]D-Fender | todays key-word is "EXPLICIT" |
| 17:18.28 | anthm | how bout [segfault] instead? |
| 17:19.35 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 17:20.28 | jgoo | [TK]D-Fender: careful, you might wake some people up shouting words like that ;-) |
| 17:20.55 | [TK]D-Fender | anthm: No, thats a Novemeber even ;) |
| 17:21.24 | [TK]D-Fender | anthm: Though I do find it rather amusing :0 |
| 17:22.20 | *** join/#asterisk Juggie (i=agony@CPE00c049d9f271-CM00137186c8d8.cpe.net.cable.rogers.com) |
| 17:23.24 | wulfy814 | [TK]D-Fender: I think it's my own stupidity that's holding me back not 1.4 |
| 17:23.42 | wulfy814 | I can transfer a call to one of the three extensions I have 8005, 8010, or 8015 |
| 17:23.57 | wulfy814 | but I can't transfer it anywhere else, including 8500 (voicemail) |
| 17:24.05 | wulfy814 | it behaves the same way as the parking |
| 17:24.31 | wulfy814 | it connects me to voicemail instead of sending the user there |
| 17:25.11 | jgoo | anyone heard of openradio? |
| 17:26.16 | jgoo | [TK]D-Fender: I called the Record() application explicitly (yey) from AGI, using exec (not an agi command) and I still didn't get the variable. I am suspecting that, in ANY agi call,a new channel is created or something.... weird |
| 17:28.17 | [TK]D-Fender | jgoo: If it did, then you'd at least see that new channel name. No... I don't think so...... |
| 17:28.50 | [TK]D-Fender | jgoo: Seems pretty clear that either that function doesn't do what you think, or you're calling it wrong |
| 17:28.55 | *** join/#asterisk CharlesR (n=charlesr@adsl-75-24-20-166.dsl.yntwoh.sbcglobal.net) |
| 17:29.55 | *** join/#asterisk jgoo_ (n=e4b80e21@87.202.216.107) |
| 17:30.01 | jgoo_ | damn |
| 17:30.03 | jgoo_ | OK |
| 17:30.05 | jgoo_ | wow |
| 17:30.24 | *** part/#asterisk hoobastooba (n=ckwall@63.149.122.93) |
| 17:30.32 | jgoo_ | DAMN my foolish mind. you need %d in a Record to get the filename. OK so now I see the variable. mentally good. |
| 17:30.45 | jgoo_ | still, bittersweet. I still need to finish this voicemail app |
| 17:31.01 | [TK]D-Fender | jgoo: You going to share it when you're done? |
| 17:31.11 | *** join/#asterisk DanTMG (n=danielga@124-168-3-90.dyn.iinet.net.au) |
| 17:33.03 | jgoo_ | [TK]D-Fender: yes, I hope to have a full range of neat and cool apps. right now.. it is kinda tied to my oracle database, as I use a customer set of data objects. But, it will be simple to make it run on hibernate, and let everyone else choose their own method (it records each voicemail in a database, not just filesystem) |
| 17:33.16 | jgoo_ | s/customer/custom |
| 17:33.33 | jeremy_g | :D |
| 17:33.42 | hmmhesays | So i'm having a problem here, wondering if someone can shed some light on it |
| 17:33.52 | jgoo_ | plus some nice features like a few methods that get choices from people, handle validation and errors in a really robust way |
| 17:34.17 | [TK]D-Fender | jgoo : can ODBC serve that purpose? WOuld allow for better abstraction and you could allow for a local DB w/o process overhead (SQLIte) |
| 17:35.13 | jgoo_ | *cough*jdbc |
| 17:35.15 | jgoo_ | but yes |
| 17:35.24 | hmmhesays | Asterisk 1.2 I have an ata and a pri gateway. when a call comes into the pri gateway it gets sent to the ATA, there is a fax machine connected to the ata. when the ata detects the fax tone it sends and invite back to asterisk to change the codec over to ulaw. This doesn't get forwarded to my pri gateway so it stays using g.729 and obviously the fax fails |
| 17:35.39 | jgoo_ | I will add jdbc, and put some hibernate classes in there for anyone, and you can use a jdbc-odbc bridge if you want :-) |
| 17:35.41 | *** join/#asterisk prttp (i=achi@45.Red-83-50-35.dynamicIP.rima-tde.net) |
| 17:36.11 | hmmhesays | is there any way I can make asterisk pass that invite back through to my pri gateway? |
| 17:36.18 | [TK]D-Fender | jdbc isn't a "bad" idea, just less of a "free" one. |
| 17:36.20 | jgoo_ | [TK]D-Fender: honestly, if agi is sweetened up, and there is a great higher level library, it will be CHILDSPLAY to write the most insanely complicated apps |
| 17:36.47 | [TK]D-Fender | jgoo : Surprised you haven't tried writeing a REAL app for this... |
| 17:37.07 | jgoo_ | [TK]D-Fender: I don't get that comment, jdbc is just how java does :p you can control any DB with jdbc, just grab that driver, if you want to use some odbc driver, use a bridge |
| 17:37.09 | jgoo_ | REAL? :p |
| 17:37.32 | jgoo_ | what do you mean by REAL? ^_^ |
| 17:38.18 | [TK]D-Fender | jgoo : an actual * registered app (there is a template for this). Would reduce overhead even more and open up new functionality |
| 17:39.16 | jgoo_ | registered? you mean... in .c ? I am confuzzled at this juncture. I guess you mean lower level. that is the idea of AGI. I want higherlevel, like ask("question") and stuff, ok I have to go |
| 17:39.20 | jgoo_ | *gone* |
| 17:39.31 | *** join/#asterisk MORRICE (n=ForSaken@morrice.win.mnsi.net) |
| 17:40.20 | MORRICE | Can someone tell me what the copywrite on Asterisk is, i have a consultant refusing to give me access to the box he sold us and including refusing to just give me a copy of our config files |
| 17:40.58 | hmmhesays | anyone anyone? |
| 17:41.39 | MORRICE | I thought this product was like a GNU thing |
| 17:41.39 | Pj_ | MORRICE: Asterisk is GPL |
| 17:41.52 | Pj_ | The login password to your box isn't probably |
| 17:42.02 | Pj_ | and the config files I don't know but I wouldn't be too sure |
| 17:42.08 | macTijn | lol |
| 17:42.18 | Pj_ | I never saw a GPL header up there |
| 17:42.24 | Pj_ | :D |
| 17:42.29 | macTijn | nope |
| 17:42.36 | macTijn | configs are public domain :P |
| 17:42.37 | MORRICE | hmm |
| 17:42.43 | Pj_ | macTijn: yeah ! |
| 17:42.45 | macTijn | haha |
| 17:42.47 | macTijn | no |
| 17:42.50 | [TK]D-Fender | MORRICE: * may be GPL, but in the lowest term, * config files can be construed as "code" to which I suppose he could claim "copyright" over. |
| 17:42.54 | dasenjo | configs colud be closed |
| 17:43.00 | [TK]D-Fender | macTijn: I currently beg to differ |
| 17:43.14 | Pj_ | MORRICE: if he sold you the box it's yours isn't it ? |
| 17:43.17 | macTijn | [TK]D-Fender: about the pub domain ? duh. |
| 17:43.18 | MORRICE | how would one be able to backup and rebuild their ssytem them |
| 17:43.26 | Pj_ | Just reboot the damn thing on a live cd and get the files |
| 17:43.39 | Pj_ | Although maybe you would be -stealing- his config files then |
| 17:43.40 | Pj_ | muahahah |
| 17:43.40 | [TK]D-Fender | MORRICE: And if you "own" the box, and not just a "license to use" you could always just go in and grab them |
| 17:43.44 | MORRICE | Oh I know HOW I could get it.. but its not the point |
| 17:43.55 | MORRICE | Oh we own the server |
| 17:44.13 | Pj_ | The point is the contract you have with that guy |
| 17:44.16 | macTijn | MORRICE: any agreements on usage of the server ? |
| 17:44.19 | MORRICE | and wouldnt those configs be OURS as they are, our way of operating |
| 17:44.24 | Pj_ | But he can totally claim copyright over his config files |
| 17:44.33 | MORRICE | Im pullin those now, but I do not think there is any signed agreement |
| 17:44.36 | macTijn | MORRICE: like a rental contract or so ? |
| 17:44.43 | Pj_ | the answer lies in your contract |
| 17:44.47 | [TK]D-Fender | MORRICE: Ok, if you don't see anything in your contract of sale prohibiting you from doing otherwise, go hack into it and look into the files. if there is a copyright notive then you'll have to consider that moving on. |
| 17:44.51 | dasenjo | MORRICE can own the bix, but can have a support contract that dont let him grab the files or manage the server |
| 17:45.10 | macTijn | my personal business is based on the fact that the config is created by me, therefor owned by me |
| 17:45.22 | *** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin) |
| 17:45.23 | MORRICE | and if you go out of business |
| 17:45.24 | macTijn | and it's certainly not GPL or any other form of open |
| 17:45.35 | dasenjo | macTijn, my case is similar |
| 17:46.18 | macTijn | I might "open up" snippets |
| 17:46.34 | macTijn | but primarily it's mine, and no one else gets to touch it |
| 17:46.46 | MORRICE | I think at that point you might find yourself in breech of the crimmial code |
| 17:46.49 | MORRICE | for data |
| 17:46.56 | macTijn | the what now ? |
| 17:47.14 | macTijn | excuse me |
| 17:47.22 | macTijn | you must confuse me for some dumb american. |
| 17:47.28 | MORRICE | ;p; |
| 17:47.56 | MORRICE | there is a fine line on 'MISCHIEF' and refusing access to own data is one of them |
| 17:48.08 | MORRICE | (d) obstructs, interrupts or interferes with any person in the |
| 17:48.08 | MORRICE | > lawful use of data or denies access to data to any person who is |
| 17:48.08 | MORRICE | > entitled to access thereto. |
| 17:48.09 | jeremy_g | lol |
| 17:48.18 | dasenjo | our config files are readeble, but the client have no administration access to the server |
| 17:48.26 | macTijn | MORRICE: but are you entitled ? |
| 17:48.32 | dasenjo | just through destar, our interface |
| 17:48.38 | jeremy_g | MORRICE:where r ya quoting from |
| 17:48.40 | jeremy_g | reference |
| 17:48.44 | MORRICE | that makes sense.. you dont want them to MODIFY them and have a server blow up |
| 17:48.52 | FuriousGeorge | anyone know if multiparking is scheduled for the 1.4 release cycle? or is that something that is going to have to wait till 1.6 |
| 17:48.54 | MORRICE | but they should have access to them for reference or backup |
| 17:49.23 | macTijn | MORRICE: are you actually entitled to access those files, and on what grounds ? |
| 17:49.23 | dasenjo | yes .. there are backups |
| 17:49.26 | *** part/#asterisk Ng (n=cmsj@mairukipa.tenshu.net) |
| 17:49.29 | *** join/#asterisk Gunde (n=spamyous@82.153.170.213) |
| 17:49.39 | MORRICE | Im referencing that from LAWS on mischief |
| 17:50.02 | jeremy_g | MORRICE:book,manual,url?? |
| 17:50.05 | macTijn | MORRICE: can you please answer my question ? |
| 17:50.18 | MORRICE | macTijn what ? |
| 17:50.29 | MORRICE | jeremy_g one sec |
| 17:50.30 | macTijn | [19:49] <macTijn> MORRICE: are you actually entitled to access those files, and on what grounds ? |
| 17:50.57 | MORRICE | Why would I not be entitled to data on my server for my office for a product that is GNU? |
| 17:51.16 | hmmhesays | no one has any idea huh? |
| 17:51.24 | macTijn | MORRICE: did you config it ? |
| 17:51.36 | *** join/#asterisk diclophis-work (n=jbardin@65.203.37.58) |
| 17:51.37 | Pj_ | MORRICE: 1°) Not everyone is american 2°) It's not because you can use it that you can know how it works, take any proprietary software |
| 17:51.47 | Pj_ | And they're not "illegal" even by american standards |
| 17:51.54 | Pj_ | especially by american standards |
| 17:51.58 | MORRICE | I have done some of the configs |
| 17:52.08 | MORRICE | I have added and change items to work as we go |
| 17:52.19 | macTijn | MORRICE: through a web interface ? |
| 17:52.27 | Pj_ | doesn't matter much |
| 17:52.38 | Pj_ | what counts is "are you entitled to" |
| 17:52.40 | Pj_ | as you said |
| 17:52.48 | macTijn | true |
| 17:52.52 | Pj_ | not if you "can do it", or "did it" already |
| 17:52.54 | macTijn | I don't believe so |
| 17:53.08 | Pj_ | otherwise we're back at square one, just reboot the puter and get the files |
| 17:53.10 | FuriousGeorge | hmmhesays: i PRI has INVITES like sip? or are you talking about sip INVITES? |
| 17:53.44 | justinu|laptop | pri/q.931 calls them SETUPs |
| 17:53.45 | MORRICE | heres the link.. oh and sorry guys I should have also said.. Im Canadian Eh |
| 17:53.47 | MORRICE | http://www.canlii.org/ca/sta/c-46/sec430.html |
| 17:54.00 | FuriousGeorge | i thought pri's for telphony had 24 interfaces which could be made digital or analog (never even seen one personally) |
| 17:54.46 | Pj_ | MORRICE: doesn't answer the fact that you're entitled or not |
| 17:54.49 | *** join/#asterisk lters (n=tech@eg1.ekn.com) |
| 17:55.14 | MORRICE | well thats the question I came in here for lol |
| 17:55.16 | *** join/#asterisk Alric (n=nbowyer@masq.hyperusa.com) |
| 17:55.23 | Pj_ | but it's not because it's _on_ your server that it do belong to you |
| 17:55.40 | *** join/#asterisk srbaker (n=srbaker@S01060017ee01d049.no.shawcable.net) |
| 17:55.43 | Pj_ | Well the answer is not here |
| 17:55.44 | MORRICE | http://security.uwo.ca/CRIMINAL.CODE.html |
| 17:55.44 | lters | any good calculators to compute minutes per meg on gsm or wan? |
| 17:55.50 | *** part/#asterisk Zaw (i=zaw@unaffiliated/zaw) |
| 17:55.51 | Pj_ | Cause config files have nothing to do with the gpl |
| 17:55.59 | Pj_ | so it's all about your contract |
| 17:56.14 | MORRICE | k thanks pj |
| 17:59.05 | *** join/#asterisk Qwell (n=north@unaffiliated/qwell) |
| 17:59.05 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 17:59.27 | MikeJ | gpl has everything to do with copywrite law.. so if it isn't copywritable.. gpl won't apply.. |
| 18:00.01 | MikeJ | also, gpl governs distribution, not use.. once it's in your hands, the gpl makes no use restrictions at all.. only distribution restrictions |
| 18:00.33 | *** join/#asterisk spatulamaan (n=ggilmore@207.188.8.252) |
| 18:00.46 | MikeJ | if a config file is copywritable.. it can be held under the terms of the gpl... if it is distributed with such a license.. |
| 18:00.54 | Pj_ | MikeJ: true |
| 18:01.20 | Pj_ | but he's saying that the guy won't give him the files |
| 18:01.40 | Pj_ | So I don't think it's the kind of person to paste a GPL header at the beginning of the files :) |
| 18:02.25 | MikeJ | I wasn't getting into that conversation.. just the gpl piece of it |
| 18:02.34 | MORRICE | :) |
| 18:02.59 | MikeJ | what exactly is the larger issue at hand/ |
| 18:03.00 | MikeJ | ? |
| 18:03.16 | Pj_ | MikeJ: if MORRICE owns the config files on his server or not |
| 18:03.25 | MikeJ | config files for what? |
| 18:03.29 | Pj_ | * |
| 18:03.44 | MikeJ | it's a server that he owns, that he posesses? |
| 18:03.49 | *** join/#asterisk Arno[Slack] (i=100@master.infinityperl.org) |
| 18:04.03 | MORRICE | I want a copy of the configs.. I do not care so much about modify rights on the live server just I want copies of the configs |
| 18:04.15 | MORRICE | I own server and its in my basement in my server room |
| 18:04.24 | MikeJ | so you own a server, and you paid somone to come install and configure asterisk for you? |
| 18:04.54 | MORRICE | a local company is selling phone systems.. They sell you hardware and support time and install. |
| 18:05.02 | MORRICE | They make it clear they are not selling the software |
| 18:05.06 | MikeJ | ok.. |
| 18:05.43 | MikeJ | and it's put out there as support and configuration.... not they are selling you anything propriatary.. right? |
| 18:06.00 | MORRICE | from our meetings YES |
| 18:06.02 | MikeJ | I suppose the issue is you don't have root access to the box ? |
| 18:06.08 | Pj_ | (contract) |
| 18:06.27 | Pj_ | meetings, oral agreement, "what I thought", what you think is right |
| 18:06.33 | MORRICE | Correct they wont give that over or will void support.. Which is fine. In worse case I know how to get past it |
| 18:06.38 | *** join/#asterisk CunningPike (n=CunningP@dsl253-055-082.dfw1.dsl.speakeasy.net) |
| 18:06.39 | Pj_ | means bullshit ornearly so when dealing with law |
| 18:06.50 | MikeJ | MORRICE.. ok no big deal... |
| 18:06.53 | MikeJ | but you own the box.. |
| 18:06.56 | MikeJ | right? |
| 18:06.57 | MORRICE | YES |
| 18:07.04 | MORRICE | we have big fat bill for hardware |
| 18:07.11 | MORRICE | its ours |
| 18:07.13 | MikeJ | what country are you in? |
| 18:07.19 | Pj_ | canada |
| 18:07.20 | MORRICE | Canada |
| 18:07.51 | MikeJ | ok.. so the legal question here is, would the config file fall under copywrite law in canada.. |
| 18:07.51 | tzafrir | A config file is certainly copyrighttable (and certainly not copy*write*ble). So theoreticallly as the sample config files of Asterisk are released under the same license as Asterisk (never mentioned explicitly in the source tree that they aren't), anything derived from them is under the same license, by the terms of the GPL |
| 18:08.06 | tzafrir | Unless I missed something in the docs of Asterisk |
| 18:08.32 | MikeJ | tzafir, I think you just conflicted yourself. |
| 18:08.40 | Pj_ | yup |
| 18:08.45 | tzafrir | MikeJ, how? |
| 18:08.47 | hmmhesays | bah driving me nuts |
| 18:08.55 | MikeJ | is a configuration file copyrighttable? |
| 18:08.58 | hmmhesays | MikeJ help me |
| 18:09.06 | tzafrir | extensions.conf certainly is |
| 18:09.11 | tzafrir | It is not trivial |
| 18:09.23 | tzafrir | rtp.conf? hmmm.. maybe not |
| 18:09.26 | MikeJ | that's an assertion on your part... |
| 18:09.32 | MikeJ | but I don't know that to be true.. |
| 18:09.49 | MikeJ | especially with differences in different countries laws. |
| 18:09.55 | MikeJ | but that is the crux of things. |
| 18:10.03 | MikeJ | hmmhesays, what's up.. |
| 18:10.09 | *** join/#asterisk ToTo (n=ToTo@host35-167-dynamic.2-87-r.retail.telecomitalia.it) |
| 18:10.33 | clyrrad | I have a question about DTMF - I have phones set to INBAND+INFO, and it works well with Asterisk as well as most IVR systems. Howerver some IVR systems will not accept the DTMF's - Just wondering if anyone has had this issue - or knows of a work around? |
| 18:10.35 | MikeJ | regardless... if it is.. then it is likely derivative work of the original files from asterisk, which may or may not be gpl... |
| 18:10.47 | lters | any good minutes per wav calculators out there? |
| 18:10.49 | hmmhesays | Well I'm having a problem with faxing, not a networking issue but |
| 18:11.14 | Pj_ | I don't think it's "derivative work" neither it's any "work" at all |
| 18:11.17 | MikeJ | the fact of the matter is.. it's your box.... there is no law keeping you from hacking the box and getting the config files that I can think of.. |
| 18:11.18 | tzafrir | The license of Asterisk is well known.. And anyway, Digium is in a position to release those files under a different license |
| 18:11.21 | *** join/#asterisk wulfy814 (n=lorentz@216.48.0.4) |
| 18:11.33 | hmmhesays | I have a pri gateway and an ata. When the ata detects a fax it sends a reinvite to asteirsk with the prefered fax codec, this never makes it to my pri gateway so I end up with one leg ulaw one leg g.729 |
| 18:11.33 | tzafrir | In case clarification is needed |
| 18:11.59 | MikeJ | I am not talking about digiums release anyhow.. I am talking about this guys deal.. |
| 18:12.00 | Pj_ | It would be tantamount saying that if you create a drawing with gimp, the settings you used for your blur would be GPL'ed because it's based on the default setting |
| 18:12.00 | MikeJ | his deal is.. it's his box.. he can hack it and get a copy of the files.. . |
| 18:12.25 | MikeJ | if all you want is to have the files, I can't think of anything stopping you |
| 18:12.36 | Pj_ | MikeJ: same here |
| 18:12.52 | Pj_ | If it's law however, just get you _contract_ |
| 18:12.56 | MORRICE | I understand.. I was just getting frustrated and wanted to understand how someone could say Im not allowed to thm |
| 18:13.01 | tzafrir | The drawing itself (the image) is copyrightable. And so are the files that save it (e.g: .xfc files) |
| 18:13.06 | *** join/#asterisk scurb_ (n=scurb_@dsl253-055-082.dfw1.dsl.speakeasy.net) |
| 18:13.21 | MORRICE | Im waiting for my boss to bring me the file from her office |
| 18:13.28 | MikeJ | I bet they were dumb enough to leave the thing running as root.. so if you have manager access, add system ext to change the root pwd, or to set up an accout with the appropriate access.. |
| 18:13.42 | MikeJ | or to just ftp the files off the box |
| 18:13.58 | Pj_ | tzafrir: I could copyright the drawing, sell it to someone and then I could be forced to say which settings I use ? |
| 18:14.07 | Pj_ | don't think so |
| 18:14.12 | MikeJ | MORRICE.. is your only issue that you want a copy of the files? |
| 18:14.12 | *** join/#asterisk C6Vette (n=info@72-166-37-114.dia.static.qwest.net) |
| 18:14.23 | MikeJ | or do you want to sell them to somone else? |
| 18:14.26 | Pj_ | Hm wait I'm getting silly |
| 18:14.35 | Pj_ | :D |
| 18:14.48 | MikeJ | if you just want a copy for yourself... just take a copy.. |
| 18:15.06 | MORRICE | I just want a copy for reference shake.. Just like all my network related items.. If SAID consultant disappears I still have 2 feet to stand on when I call another consultant |
| 18:15.28 | Pj_ | but still I don't "think" the conf files are GPLed, are they ? do you have a sample case or... ? |
| 18:16.05 | tzafrir | Pj_, the question is different: if you made very special settings, and then I copy them and distribute them al over the world, I may be stepping over the copyrights law |
| 18:16.06 | *** join/#asterisk CunningPike_ (n=CunningP@204.239.8.149) |
| 18:16.09 | MORRICE | I was hopin you folks would have run into this and might know.. but thats ok |
| 18:16.27 | tzafrir | That is: if I'm just copying and not reimplementing |
| 18:16.58 | MikeJ | MORRICE... then no problem.. do you have manager access to the box? |
| 18:17.08 | MORRICE | I have NO access to box.. |
| 18:17.17 | MORRICE | Ill have to boot it from a cd and make changes :) lol |
| 18:17.21 | MikeJ | well.. clearly you can make phone calls to it |
| 18:17.31 | MikeJ | you can't use manager? |
| 18:17.39 | MORRICE | lol yes |
| 18:17.58 | MikeJ | port scan it, what ports it listenting on? |
| 18:17.59 | tzafrir | lters, gsm is 1MB perl 10 minutes, IIRC |
| 18:18.01 | MORRICE | well they have a web interface on top I log into |
| 18:18.06 | tzafrir | s/perl/per/ |
| 18:18.16 | clyrrad | I have a question about DTMF - I have phones set to INBAND+INFO, and it works well with Asterisk as well as most IVR systems. Howerver some IVR systems will not accept the DTMF's - Just wondering if anyone has had this issue - or knows of a work around? |
| 18:18.17 | MikeJ | MORRICE, can you add arbitrary extensions? |
| 18:18.22 | hmmhesays | bah |
| 18:18.26 | lters | tzafrir: I see, thanks |
| 18:18.29 | MORRICE | I can add ext from the web interface |
| 18:18.41 | MikeJ | using asterisk dialplan syntax? |
| 18:18.41 | MORRICE | and from in there I have cli |
| 18:18.44 | tzafrir | yey!, I used s/ for correcting perl :-) |
| 18:18.57 | Qwell | an actual asterisk CLI? |
| 18:19.08 | MORRICE | one sec |
| 18:19.09 | Qwell | MORRICE: type ! |
| 18:19.10 | Pj_ | which version of asterisk are you running ? do you have chan_skinny loaded ? |
| 18:19.12 | Pj_ | muahhahaha |
| 18:19.18 | *** join/#asterisk qdk (n=qdk@0x535eae17.boanxx9.adsl-dhcp.tele.dk) |
| 18:19.19 | *** join/#asterisk ellisdee (n=ellisdee@69.15.174.114) |
| 18:19.22 | clyrrad | can anyone help me out with my DTMF question? I am wondering if there is another DTMF that is better to use? |
| 18:19.25 | Inez | does anyone use res_odbc with pgsql? |
| 18:19.25 | lters | tzafrir: wow, that was neat. |
| 18:19.26 | MikeJ | Pj_.. hehe |
| 18:19.33 | Inez | i have problem to cofnigure odbcinst.ini and odbc.ini |
| 18:19.34 | lters | s/hehe/heh/ |
| 18:19.56 | ellisdee | i have some incorrect paths set to config files. in asterisk 1.2.*: is there a file that has a path for all config files that can be manually modified to reflect the name change of config files? |
| 18:19.58 | tzafrir | lters, it only works on your lines |
| 18:19.59 | Qwell | clyrrad: rfc2833 works well |
| 18:20.08 | lters | aha |
| 18:20.33 | clyrrad | Qwell: does it support more systems than INBAND+INFO? |
| 18:20.34 | Inez | Qwell can you help me tih odbc and pgsql? |
| 18:21.04 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 18:21.04 | *** mode/#asterisk [+o russellb] by ChanServ |
| 18:21.06 | Aurs | Inez: I'm using res_odbc |
| 18:21.11 | Qwell | russellb: ! |
| 18:21.19 | russellb | Qwell: ! |
| 18:21.20 | CunningPike_ | clyrrad: This may or may not help - when using SPA-3000s with IVR, we had to set the SPA to use INBAND and use dtmfmode=inband in sip.conf to get it to work |
| 18:21.24 | russellb | greetings, sir |
| 18:21.35 | lters | s/aha/hmm/ |
| 18:21.44 | [TK]D-Fender | CunningPikeSPA's should use INFO |
| 18:21.59 | Corydon-w | Help me, OB1 Qwell noby... you're my only hope... |
| 18:21.59 | clyrrad | CunningPike - yea we have set INBAND+INFO becase it works well with Asterisk and most IVR's - there are just some IVR's out there that wont get the DTMF's |
| 18:22.28 | clyrrad | I was wondering if this is a case of - it wont work with ALL IVR's? |
| 18:22.43 | clyrrad | it seems if you set one mode to make one IVR happy - the next one does not work anymore... |
| 18:22.46 | Aurs | Inez: this might help you: http://www.asteriskguru.com/tutorials/realtime_pgsql.html |
| 18:22.57 | Aurs | *gone* |
| 18:23.40 | lters | tzafrir: what about wav49 or wav... |
| 18:23.42 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 18:23.42 | *** mode/#asterisk [+o mog] by ChanServ |
| 18:23.46 | Inez | Aurs thanks |
| 18:23.54 | lters | or would there be no reason for it... |
| 18:24.02 | tzafrir | wav49 is gsm |
| 18:24.12 | tzafrir | wav is not gsm |
| 18:24.24 | CunningPike_ | [TK]D-Fender: We tried INFO, but no dice - INBAND worked |
| 18:24.30 | Pj_ | wing ? |
| 18:24.39 | Pj_ | damn, doesn'twork |
| 18:24.42 | CunningPike_ | [TK]D-Fender: I think it's the IVR itself |
| 18:24.50 | clyrrad | yes INFO never worked for us too - we had to set INFO+INBAND |
| 18:25.25 | [TK]D-Fender | CunningPike : if you set the SPA to INFO and your sip.conf entry for it it should work perfectly. Did for me. |
| 18:25.32 | clyrrad | I was thinking its a case of either the IVR can read VOIP DTMF or it cant - becase the IVR works fine from a LAN line |
| 18:25.40 | clyrrad | cant* |
| 18:25.48 | lters | tzafrir: I see, hmm. wonder why * saves vm as all 3. |
| 18:26.13 | *** join/#asterisk drcode (n=chatzill@87.69.59.186.cable.012.net.il) |
| 18:26.15 | drcode | hi all |
| 18:26.18 | drcode | whats u |
| 18:26.20 | drcode | p |
| 18:26.31 | clyrrad | and "Auto" on the SPA's do not seem to work at all |
| 18:26.44 | tzafrir | lters, tell it the format explicitly. Check the sample voicemail.conf |
| 18:26.57 | lters | ok. |
| 18:26.58 | tzafrir | hi drcode |
| 18:27.09 | *** part/#asterisk brif8 (n=brif8@67.78.24.178) |
| 18:27.20 | drcode | hi tzafrir |
| 18:29.36 | *** join/#asterisk Arno[Slack] (i=100@master.infinityperl.org) |
| 18:32.19 | hmmhesays | faxing is going to drive me insane |
| 18:32.45 | anthm | that would make it a fax driver i guess |
| 18:33.01 | *** join/#asterisk sulex (n=sulex@pdpc/supporter/active/sulex) |
| 18:35.18 | *** join/#asterisk dextro (n=dextro@suffrage.itfreedom.com) |
| 18:36.09 | *** join/#asterisk sahafeez (n=sahafeez@67.109.14.227.ptr.us.xo.net) |
| 18:37.03 | hmmhesays | I can't get my endpoints to successfully negotiate a fax codec |
| 18:37.15 | *** join/#asterisk telamon (n=telamon@pac.isn.net) |
| 18:37.16 | *** join/#asterisk sahafeez (n=sahafeez@67.109.14.227.ptr.us.xo.net) |
| 18:37.48 | telamon | Are International phone numbers (from a North American perspective) always the same length? |
| 18:38.51 | Qwell | telamon: no |
| 18:38.55 | SpaceBass | not sure all country codes are 2 digits |
| 18:38.56 | SpaceBass | some are 3 |
| 18:39.00 | [TK]D-Fender | SpaceBass: Very doable. |
| 18:39.06 | Qwell | SpaceBass: some are 1 |
| 18:39.13 | SpaceBass | and each country can have its own format after that |
| 18:39.20 | SpaceBass | USA, case in point...1 |
| 18:39.34 | SpaceBass | [TK]D-Fender, really? on OS X too? |
| 18:39.42 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 18:39.50 | telamon | Ah, that's what I thought. Okay, _011. it is then. |
| 18:40.07 | *** join/#asterisk DanTMG (n=danielga@124-168-3-90.dyn.iinet.net.au) |
| 18:40.13 | SpaceBass | thats what I do... 011. |
| 18:40.29 | Corydon-w | _011XXXXX. is what I use |
| 18:40.34 | denon | telamon: 011XXX. may be a safe bet |
| 18:40.52 | Corydon-w | There are no shorter numbers than 5 long |
| 18:41.04 | denon | Corydon-w: was considering if any country had a 0 that worked from outside |
| 18:41.05 | Corydon-w | Even Vatican City has longer |
| 18:41.07 | denon | 011610 or such |
| 18:41.13 | [TK]D-Fender | SpaceBass: quick hacks : set a print driver to always print to PS and then deposit the files on a polled server in a user folder. those folders would then get polled and imported into a web interface for identification for destination. from there you just tell it to send it on its merry way and you can have Hylafax / SpanDSP spit them out. |
| 18:41.30 | denon | Corydon-w: like for an operator or some kinda service |
| 18:41.43 | Corydon-w | denon: probably not, though I can't say for sure |
| 18:41.53 | denon | nod, thats what I figure too .. but also, I cant say for sure :) |
| 18:41.56 | denon | hence the 011XXX. :) |
| 18:42.01 | SpaceBass | [TK]D-Fender, i did think about something along those lines...using SpanDSP... but I want to be able to avoid the web part |
| 18:42.03 | *** join/#asterisk billwarddc (n=IceChat7@ppp-69-148-16-53.dsl.austtx.swbell.net) |
| 18:42.15 | Corydon-w | denon: what about calling Russian Federation operator? |
| 18:42.16 | SpaceBass | [TK]D-Fender, I was thinking of using postfix and parsing an email where the subject is the number to call |
| 18:42.16 | telamon | Just out of curiosity, how do you determine the country and city code from a number? Since the country codes and local number can be of varying length, it seems rather difficult. |
| 18:42.25 | Corydon-w | denon: that's a single digit country code |
| 18:42.28 | denon | Corydon-w: ok, fine, 011. :) |
| 18:42.38 | *** join/#asterisk yogurt2ungue (n=yogurt2u@24-48-231-201.fibertel.com.ar) |
| 18:43.08 | Corydon-w | Just pointing it out |
| 18:43.39 | [TK]D-Fender | SpaceBass: But taht would mean more interfention on the users side |
| 18:43.42 | denon | nod |
| 18:43.58 | SpaceBass | [TK]D-Fender, yeah...true...save as PS then email... |
| 18:44.18 | [TK]D-Fender | SpaceBass: Doesn't Hylafax have a driver for MacOS? |
| 18:44.24 | SpaceBass | it may |
| 18:44.33 | SpaceBass | i haven't looked into Halyfax in 2 years |
| 18:44.40 | hmmhesays | is it possible to allow or disallow codecs based on an extension dialed? |
| 18:44.43 | [TK]D-Fender | Corydon-w: What country has a single digit country code? |
| 18:45.05 | Corydon-w | NANP and Russian Federation both have single digit country codes |
| 18:45.07 | Qwell | [TK]D-Fender: The US? |
| 18:45.17 | Corydon-w | NANP is 1. Russian Federation is 7. |
| 18:45.25 | [TK]D-Fender | Qwell Really? what is it then? How would one call from say.. France? |
| 18:46.12 | Qwell | dial the france international dial prefix, then 1NXXNXXXXXX |
| 18:46.26 | Qwell | 011 isn't a global thing |
| 18:46.35 | Corydon-w | France's international prefix is 00, right? |
| 18:46.49 | CunningPike | Most European countries are 00 |
| 18:47.40 | [TK]D-Fender | Qwell : So thats _001NXXNXXXXXX for frans to call USA? |
| 18:47.46 | Qwell | something like that |
| 18:47.50 | [TK]D-Fender | France* |
| 18:47.55 | [TK]D-Fender | Qwell : interesting. |
| 18:48.20 | [TK]D-Fender | Qwell : Sounds like an early-dial horror ;) |
| 18:49.41 | drcode | can I use sip phone with 10 video confrence meeting + polycom? |
| 18:50.23 | *** join/#asterisk krondorl (n=krondorl@acid.auricnet.ca) |
| 18:50.25 | xheliox | Any anyone shed some light on what "-- PROGRESS with cause code 127 received" means? I'm receiving this when having a one way audio issue and trying to determine if it's related. |
| 18:52.30 | krondorl | Hi all.. Anyone here use the FOP interface? I'm looking to find out how to run the op_server.pl automatically in gentoo.. if I issue the command in a ssh connection, when I exit, the .pl shuts down. I need it to stay running. |
| 18:52.57 | CunningPike | xheliox: http://www.voip-info.org/wiki/index.php?page=Asterisk+variable+HangupCause |
| 18:53.32 | [TK]D-Fender | krondorl: How would you call it from the CLI? |
| 18:53.45 | CunningPike | krondorl: I think the FOP web site includes init scripts for a variety of distros...... |
| 18:54.14 | krondorl | it's not called from the cli.. the init's from the site are not very clear and do not work in gentoo. |
| 18:54.20 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 18:55.41 | [TK]D-Fender | krondorl: I jsut asked how you were calling it from the Linux CLI, not * |
| 18:56.01 | CunningPike | krondorl: You'll need to take a working init from your distro, take the closest FOP sample and fire up vi |
| 18:57.03 | krondorl | Fender: Opps sorry... /usr/local/op_server.pl -d |
| 18:57.33 | ellisdee | kron, just ru it in a screen session |
| 18:57.38 | krondorl | CunningPike: Ok.. my mind must be fried, I didn't think of that... |
| 18:57.59 | CunningPike | krondorl: :D |
| 18:59.17 | *** join/#asterisk anthonyl (n=anthonyl@dsl253-055-082.dfw1.dsl.speakeasy.net) |
| 18:59.23 | krondorl | Ellisdee: easier said then done as most of our work is remote and to far to travel to get access to the screen.. Not only that. It's another company altogether and we'd need permission to get access. |
| 18:59.28 | *** join/#asterisk FunnyManVA (n=coriley@dsl253-055-082.dfw1.dsl.speakeasy.net) |
| 19:02.39 | *** join/#asterisk ilo_admin (n=ilo_aste@ip-12-30-102-190.hqglobal.net) |
| 19:03.19 | ilo_admin | Hello to everyone and a special hello to CunningPike |
| 19:03.38 | CunningPike | ilo_admin: Worked for ya, then? ;) |
| 19:04.23 | *** join/#asterisk tvile (n=chatzill@204.168.15.5) |
| 19:04.27 | *** join/#asterisk florz (n=florz@2001:1a50:503c:0:0:0:0:1) |
| 19:05.01 | drcode | I can connect into mysql or sqllite sip users? |
| 19:05.18 | anthonyl | are you asking about realtime/ |
| 19:05.21 | *** join/#asterisk FuriousGeorge (n=brian@ool-43536ea8.dyn.optonline.net) |
| 19:05.46 | ilo_admin | Yes CunningPike it worked Thanks |
| 19:05.54 | ilo_admin | I am extra happy today |
| 19:06.14 | CunningPike | ilo_admin: Excellent - welcome aboard |
| 19:06.15 | *** join/#asterisk Cresl1n (n=matt@dsl253-055-082.dfw1.dsl.speakeasy.net) |
| 19:06.15 | *** mode/#asterisk [+o Cresl1n] by ChanServ |
| 19:07.47 | ilo_admin | Thank YOU |
| 19:09.19 | *** part/#asterisk FuriousGeorge (n=brian@ool-43536ea8.dyn.optonline.net) |
| 19:10.03 | [TK]D-Fender | krondorl: Just do "/usr/local/op_server.pl -d &" and that will load it as a daemon that won't close when you logout |
| 19:11.50 | *** join/#asterisk alerios (n=alerios@190.24.98.181) |
| 19:11.54 | *** join/#asterisk kristalino (n=kristali@cl-157.dub-01.ie.sixxs.net) |
| 19:12.58 | CunningPike | ilo_admin: Now that you're here, got a question? :) |
| 19:13.13 | *** join/#asterisk Dibbler_ (n=Dibbler@dsl-217-155-254-174.zen.co.uk) |
| 19:13.38 | krondorl | Fender: Ah so that's what the & symbol is for?? |
| 19:14.56 | [TK]D-Fender | krondorl: YUP |
| 19:15.28 | [TK]D-Fender | krondorl: which is for instance why you do "safe_asterisk &" to start * as a daemon from the CLI |
| 19:15.58 | [TK]D-Fender | krondorl: a Linux "must know" item. I personally know little MORE than that ;) |
| 19:17.21 | ilo_admin | Yes I do |
| 19:18.34 | *** join/#asterisk MRH2 (n=Mr_happy@host-83-146-30-242.bulldogdsl.com) |
| 19:19.03 | ilo_admin | I have installed Asterisk 1.2.13 on top of Ubuntu Dapper Server 6.06 Patched. I wanted to have asterisk startup automatically after booting completes |
| 19:19.06 | MRH2 | hi can anyone confirm that ${CALLERIDNUM} works in 1.4 |
| 19:19.10 | ilo_admin | Suggestions? |
| 19:19.45 | CunningPike | MRH2: It's deprecated in 1.2 and absent in 1.4 - use ${CALLERID(num)} instead |
| 19:19.55 | MRH2 | thanks |
| 19:20.04 | *** part/#asterisk MORRICE (n=ForSaken@morrice.win.mnsi.net) |
| 19:20.35 | hmmhesays | nuts nuts nuts |
| 19:20.57 | hmmhesays | so is there any way I can change the codec i'm using if a fax is detected? |
| 19:21.04 | hmmhesays | on a sip channel |
| 19:21.10 | CunningPike | ilo_admin: Have you tried 'make config'? |
| 19:21.20 | ilo_admin | I run make config under asterisk and it if this was redhat it would work fine |
| 19:21.31 | ilo_admin | To no avail |
| 19:21.49 | CunningPike | ilo_admin: Hmm - you may need to take an existing working init script for your distro and get hacking :) |
| 19:22.00 | ilo_admin | I think I am going to have to custom script but not sure where to begin |
| 19:22.28 | ilo_admin | Yes, the location for config files is not the same on ubuntu as other flavors of linux |
| 19:22.39 | CunningPike | ilo_admin: Why am I not surprised |
| 19:22.55 | *** join/#asterisk angler (i=angler@pdpc/sponsor/digium/angler) |
| 19:22.55 | CunningPike | ilo_admin: Does the script itself work? |
| 19:23.19 | ilo_admin | I have the zaptel config files working fine but not asterisk, the answer to your question is no |
| 19:23.59 | C6Vette | can agi be used outside the dial plan to get status of channels using Asterisk::AGI in perl? |
| 19:24.00 | ilo_admin | You are not surprised of file location in ubuntu why? |
| 19:24.47 | CunningPike | ilo_admin: You've tried running the script directly? What I'm getting at is whether the issue is that the script itself doesn't run, or whether it's simply in the wrong place |
| 19:25.16 | ilo_admin | let me check one moment |
| 19:25.23 | CunningPike | ilo_admin: And I'm not surprised because I am irrationally biased against ubuntu :) |
| 19:25.48 | hmmhesays | is it possible to change the sip codec in the dialplan? |
| 19:26.03 | CunningPike | hmmhesays: I don't believe so..... |
| 19:26.27 | C6Vette | hmmhesays, I have heard others asking for this feature, but last I heard no you cant. |
| 19:26.31 | hmmhesays | i know there is a SIP_CODEC variable |
| 19:26.37 | *** join/#asterisk rva (n=rafael@200.210.51.130) |
| 19:26.57 | CunningPike | hmmhesays: I think it's read only though - transcoder negotiation is carried out at call setup |
| 19:27.10 | hmmhesays | unless a reinvite is sent |
| 19:27.25 | rva | hi...could someone please help me with a TE110P, unicall and a brazilian E1? |
| 19:27.35 | hmmhesays | the problem having is when my ata detects a fax it changes the codec to ulaw, but that doesn't make it to my pri gateway |
| 19:27.46 | CunningPike | hmmhesays: Isn't the presence of a re-invite a function of the result of codec negotiation? |
| 19:27.47 | ilo_admin | CunningPike, it was not created although the make command was issued |
| 19:28.09 | ilo_admin | I checked the init.d directory and I see zaptel but no asterisk |
| 19:28.21 | hmmhesays | you can do many things with a reinvite |
| 19:28.36 | CunningPike | ilo_admin: Interesting - can you pastebin the output from the 'make config' command? |
| 19:28.37 | CunningPike | ~pb |
| 19:28.38 | jbot | rumour has it, pb is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 19:29.00 | ilo_admin | which one? |
| 19:29.31 | CunningPike | ilo_admin: pastebin.ca |
| 19:29.35 | ilo_admin | You would like I run make config under asterisk and what file should I be posting for you to look at? |
| 19:29.49 | MikeJ | jbot... what of these rumors... I want faxes |
| 19:30.05 | hmmhesays | so I'm stuck now |
| 19:30.09 | *** join/#asterisk saftsack (n=oliver@p54A7F487.dip.t-dialin.net) |
| 19:30.10 | saftsack | hi |
| 19:30.21 | hmmhesays | my endpoints won't negotiate a fax codec right, and I can't set it in the dialplan |
| 19:30.22 | saftsack | what codec do i need to contact sip:support@patton.com? |
| 19:30.42 | CunningPike | ilo_admin: The output from the make config |
| 19:30.57 | ilo_admin | it is written to a log file no? |
| 19:31.16 | ilo_admin | Is the log file located under asterisk-version directory? |
| 19:32.17 | *** join/#asterisk Assim (n=Assimila@216.83.78.108) |
| 19:32.18 | *** join/#asterisk LogiForce (n=LogiForc@cc51914-a.groni1.gr.home.nl) |
| 19:33.06 | Assim | I acidently created a dynamic agent in a queue using FOP. How do I log that agent out of that queue? |
| 19:33.14 | CunningPike | ilo_admin: You should get a bunch of stuff on the screen after you enter the command? |
| 19:33.15 | *** join/#asterisk slayer192 (n=slayer19@dsl253-055-082.dfw1.dsl.speakeasy.net) |
| 19:33.20 | CunningPike | Or not? |
| 19:33.29 | slayer192 | anthonyl: you in dallas? |
| 19:33.43 | ilo_admin | I got it. I just thought it write to a file one moment |
| 19:33.54 | jmls | if anyone is in the mood, have a prod at #8216 - I've added a couple of new features to the app_queue application. Specifically Queue stat variables and the option to run a macro on member connect. |
| 19:34.04 | jmls | needs some testing |
| 19:35.16 | CunningPike | Assim: 'agent logoff' ? |
| 19:35.44 | Assim | well that is how I log off normal agents, but they don't show in there. I'll try with the full uid |
| 19:36.32 | QbY | I'm still offering a Bounty for a working 7971G configuration.. |
| 19:36.39 | ilo_admin | CunningPike just upload the text from running make config |
| 19:36.43 | Assim | SIP/106 (dynamic) (Not in use) has taken no calls yet |
| 19:36.44 | Assim | .... Agent Logoff SIP/106 does not work. |
| 19:36.54 | CunningPike | ilo_admin: Got a link for me? |
| 19:37.15 | [TK]D-Fender | jmls: aS A QUEUE USER : THANK YOU |
| 19:37.57 | jmls | whoo! don't shout :) |
| 19:38.50 | jmls | hope it's useful. Within the macro, you can now use the variables to "do things" - for example, we are sending queue stats via jabber |
| 19:39.57 | saftsack | can somebody tell me please if alaw, ulaw or gsm are working on support@patton.com? |
| 19:40.13 | ilo_admin | http://pastebin.ca/219588 this is so cool |
| 19:41.23 | *** join/#asterisk CelticLord2112 (n=CelticLo@69.15.174.114) |
| 19:41.54 | hmmhesays | well you can set the SIP_CODEC variable |
| 19:42.02 | CelticLord2112 | has anyone heard of a voicemail to email bug on Asterisk 1.2.4 running on gentoo? |
| 19:43.08 | tzafrir | CelticLord2112, why not use up-to-date packages? |
| 19:44.55 | CunningPike | ilo_admin: Where is your successful zaptel init file located? |
| 19:45.23 | hmmhesays | turns out you can change the codec in the dialplan |
| 19:45.29 | CunningPike | hmmhesays: Do share! |
| 19:45.40 | ilo_admin | one moment let me check to make sure I write the correct thing here |
| 19:45.41 | hmmhesays | as long as you haven't answered the call yet |
| 19:45.54 | CunningPike | hmmhesays: Ah - makes sense |
| 19:46.00 | hmmhesays | thats all I need |
| 19:46.19 | hmmhesays | now I need someone to send me a fax |
| 19:46.25 | CunningPike | hmmhesays: Ya - I was jumping ahead in my thoughts to after the call was answered |
| 19:46.50 | Assim | Usage: remove queue member <channel> from <queue> |
| 19:47.01 | Assim | there is the answer |
| 19:47.09 | ilo_admin | the zaptel file is located in /etc/init.d/ |
| 19:48.13 | CunningPike | ilo_admin: What folder are you in when you run 'make config'? |
| 19:48.24 | CelticLord2112 | tzfrir: have several systems deployed on 1.2.4....need to find out if there is a pressing need to upgrade them all or not |
| 19:48.38 | CunningPike | ilo_admin: You should be in wherever your Asterisk source is |
| 19:48.42 | *** join/#asterisk scurb_ (n=scurb_@dsl253-055-082.dfw1.dsl.speakeasy.net) |
| 19:48.46 | *** join/#asterisk slayer192 (n=slayer19@dsl253-055-082.dfw1.dsl.speakeasy.net) |
| 19:49.11 | CelticLord2112 | also trying to determine if it is an asterisk issue or an MTA issue on the server |
| 19:50.09 | tzafrir | CelticLord2112, can you send any other message through the sendmail interface (/usr/sbin/sendmail )? |
| 19:50.33 | PakiPenguin | hi tzafrir |
| 19:50.35 | PakiPenguin | how are you |
| 19:50.39 | tzafrir | Hi |
| 19:50.59 | CelticLord2112 | yes i can... |
| 19:51.41 | CelticLord2112 | problem is the voicemail.conf file is set up...even matches what I have on systems with later packages (1.2.10) |
| 19:51.51 | ilo_admin | Cunning are you writing to me that the zaptel file you are asking about its location should be in the usr/src/asterisk-version directory |
| 19:52.07 | *** join/#asterisk aptura (n=sales@S010600a0c93f6f7e.vs.shawcable.net) |
| 19:52.12 | CelticLord2112 | but asterisk never seems to hand the voicemail .wav file off to the MTA |
| 19:52.25 | slayer192 | anyone here sitting at the Westin at the moment? |
| 19:52.32 | CelticLord2112 | have not even been able to establish that is is calling the MTA |
| 19:53.01 | CelticLord2112 | my challenge is that these 1.2.4 systems are a legacy I have inherited, and i have zero documentation on how the confiugrations were developed |
| 19:53.07 | *** join/#asterisk MAttH (n=MattH@cloud2.chilitech.net) |
| 19:53.18 | *** join/#asterisk saftsack (n=oliver@p54A7F487.dip.t-dialin.net) |
| 19:53.49 | MAttH | Hi... I have two machines tied togethor using IAX over the Internet. It seems that sometimes the Internet connection at this one location will get high latency from time to time and when that happens the jitterbuffer in IAX just dies and the audio stops working... any thoughts on fixing this? I basically get 'frames out of seuqnece' errors and the audio never fixes iteself |
| 19:54.31 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 19:54.31 | *** mode/#asterisk [+o russellb] by ChanServ |
| 19:55.04 | *** join/#asterisk stkn (i=nobody@gentoo/developer/pdpc.active.stkn) |
| 19:55.25 | [TK]D-Fender | russellb: Hey, care to do me (/us) a quick favour? |
| 19:55.28 | CunningPike | ilo_admin: 'make: Warning: File `.depend' has modification time 2.8e+08 s in the future' - what is your system's date and time? |
| 19:55.42 | russellb | [TK]D-Fender: mayyyyyyyyyybe |
| 19:55.51 | Qwell | /topic :p |
| 19:55.56 | Qwell | [TK]D-Fender: I could've done that, fyi |
| 19:56.04 | [TK]D-Fender | russellb: lol... the usual... adding the #freepbx notice back to the channel topic :) |
| 19:56.06 | russellb | yeah, i knew it'd be about the topic |
| 19:56.18 | russellb | Qwell can do it ... i'm too lazy |
| 19:56.28 | Qwell | What did it say before? |
| 19:56.40 | [TK]D-Fender | Qwell : I've asked a few times today, but you didn't notice and russellb has helped me a few times on this before. |
| 19:56.47 | russellb | join #freepbx for freepbx/8 other names support |
| 19:56.58 | Qwell | well |
| 19:57.12 | [TK]D-Fender | Qwell : yeah, much like that. |
| 19:57.12 | ilo_admin | one moment |
| 19:57.14 | Qwell | is it really appropriate to send people there who are having trixbox OS issues? |
| 19:57.36 | scurb_ | Anyone know a good sipclient for windows mobile 5? |
| 19:57.38 | russellb | i really don't see a problem with those people asking here |
| 19:57.58 | [TK]D-Fender | Qwell : Better there than here regardless. Maybe a general note for OS issues go to your distro's support channel. |
| 19:57.59 | scurb_ | for square screens. |
| 19:58.01 | Qwell | russellb: with OS issues? |
| 19:58.08 | ilo_admin | This is the first time I checked this it is Sun Jan 11 17:29:40 EST 1998 |
| 19:58.12 | russellb | Qwell: no, freepbx |
| 19:58.22 | russellb | Qwell: but #freepbx is probably more helpful |
| 19:58.33 | *** join/#asterisk [Outcast] (n=bill@222-154-47-223.jetstream.xtra.co.nz) |
| 19:59.05 | *** topic/#asterisk by Qwell -> Asterisk: The Open Source PBX -=- http://www.asterisk.org -=- New Asterisk 1.0, 1.2, 1.2-netsec, 1.4beta, Asterisk-addons 1.2, 1.4beta, Zaptel 1.2, 1.4beta, and Libpri 1.2 releases now available! (Oct. 18, 2006) Join #freepbx for freepbx/trixbox support. |
| 19:59.14 | ilo_admin | I checked on SER box and the same thing. Interesting during installation I set the proper date, time based on region |
| 19:59.32 | MikeJ | hmm |
| 19:59.39 | MikeJ | what is 1.2-netsec? |
| 20:00.01 | russellb | works with ranch networks firewall stuff |
| 20:00.16 | hmmhesays | ack this is driving me nuts |
| 20:01.03 | *** join/#asterisk pifiu-laptop (n=someone@216.5.79.1) |
| 20:01.10 | *** join/#asterisk b11d (n=noway@234-200-29-134.hcc.mnscu.edu) |
| 20:01.11 | *** join/#asterisk pids (n=pids@dsl081-072-084.sfo1.dsl.speakeasy.net) |
| 20:01.12 | b11d | hello all |
| 20:02.17 | slayer192 | anyone here at Astricon today? |
| 20:02.25 | b11d | i am |
| 20:02.28 | CunningPike | ilo_admin: Try setting the proper date and time and trying again |
| 20:02.29 | b11d | not in attendance |
| 20:02.39 | slayer192 | Looking to start a Artiecon 2006 beer-league |
| 20:02.57 | slayer192 | heh, looks like I may have started early |
| 20:03.01 | hmmhesays | so I set my sip codec, but the damn thing |
| 20:03.10 | tzafrir | Qwell, write the explicit versions? |
| 20:03.11 | b11d | im so there.. |
| 20:03.22 | ilo_admin | cool will do give me a moment |
| 20:03.28 | Qwell | tzafrir: ? |
| 20:03.29 | b11d | i want on the Asterisk International Beer Quaffing Team |
| 20:03.30 | *** join/#asterisk FunnyManVA (n=coriley@dsl253-055-082.dfw1.dsl.speakeasy.net) |
| 20:03.47 | tzafrir | Qwell, in the topic |
| 20:04.00 | slayer192 | I hear its a tough team to join |
| 20:04.13 | b11d | I'm from Northern Ontario, Canada.. where men can drink at age 8 |
| 20:04.23 | b11d | so i've got a good background.. and a strong constitution |
| 20:04.35 | slayer192 | CunningPike: Going to the CodeZone event later? |
| 20:04.38 | b11d | lets get drinking some practice kegs |
| 20:04.38 | aptura | b11 im here in bc. |
| 20:04.40 | trelane_ | b11d, so do I but I drink until I lose |
| 20:04.50 | b11d | ahh nice ;) bc is enjoyable |
| 20:04.54 | aptura | CunningPike what are you doing this week? |
| 20:05.11 | CunningPike | slayer192: Probably not - I think C is a vitamin |
| 20:05.23 | CunningPike | aptura: At Astriconm |
| 20:05.24 | slayer192 | lol |
| 20:05.30 | b11d | I need to figure out how to allow people to dial 2621111 and reach 1111 :/ |
| 20:05.31 | aptura | ahh flying there then |
| 20:05.45 | CunningPike | aptura: Bit far for the scooter |
| 20:05.50 | b11d | can this be done with exten => _XXX1111 ? |
| 20:06.36 | aptura | Cunning yea :) |
| 20:06.43 | CunningPike | b11d: exten => _XXX1111,1,Dial(${EXTEN:-4} |
| 20:07.00 | b11d | ahhh.. very nice |
| 20:07.12 | b11d | im going to give that a whirl |
| 20:07.34 | FunnyManVA | b11d: add a close ) to what CunningPike said so it is exten => _XXX1111,1,Dial(${EXTEN:-4}) |
| 20:07.52 | CunningPike | FunnyManVA: That was a test! ;) |
| 20:08.03 | PakiPenguin | i am getting [Oct 24 15:54:09] == Connect attempt from '127.0.0.1' unable to authenticate a lot on my screen , how do i check which usernamed is trying to connect |
| 20:08.08 | b11d | yeah, I figured that :) |
| 20:08.15 | FunnyManVA | Figured it was just a typo, but just in case he cuts and pastes |
| 20:08.16 | PakiPenguin | i cant seem to find what is trying to connect to my system |
| 20:08.20 | aptura | CunningPike you worked with dids and multi line configs for the polycom? |
| 20:08.33 | CunningPike | PakiPenguin: Running FOP or anything? |
| 20:08.33 | jmls | PakiPenguin: it's an astmanproxy or fop or something like that |
| 20:08.37 | jmls | damn! |
| 20:08.45 | PakiPenguin | i know ... |
| 20:08.57 | PakiPenguin | jmls: but i need to know what is connecting to the manager interface |
| 20:09.01 | PakiPenguin | i cant seem to find what is trying |
| 20:09.01 | PakiPenguin | hehe |
| 20:09.10 | b11d | hmm.. it works, but now i cant just dial "1111" |
| 20:09.13 | jmls | astmanproxy ? |
| 20:09.22 | jmls | fop ? |
| 20:09.22 | b11d | i need 1112222 and 2222 to both work.. |
| 20:09.23 | Mercestes | What is your guys reaction to the Polycom Sip 2.0.1? Worth updating to? |
| 20:09.36 | CunningPike | aptura: Yes - we have DIDs and multi-lines....... |
| 20:09.37 | jmls | they both connect to the manager interface |
| 20:10.12 | [TK]D-Fender | Mercestes: Yup. faster performance, better NAT handling, enhanced platform support, etc. |
| 20:10.12 | PakiPenguin | jmls: nope , havent got any thing like them |
| 20:10.13 | hmmhesays | Oct 24 15:05:29 DEBUG[2457]: chan_sip.c:3695 process_sdp: Oooh, we need to change our formats since our peer supports only 0x100 (g729) and not 0x4 (ulaw) |
| 20:10.24 | hmmhesays | I don't understand that, I have g729 and ulaw on both sides |
| 20:10.51 | MikeJ | Oooh...! |
| 20:11.00 | MikeJ | great message.. |
| 20:11.04 | justinu|laptop | heh |
| 20:11.05 | FunnyManVA | b11d: exten => 1111,1,Dial(${EXTEN}) beefore the above will work for that case. Is it always 111 4 digit exten or just 4 digit exten? |
| 20:11.14 | russellb | it's not an error |
| 20:11.15 | MikeJ | you may have it on both sides in *.. but what is that end actually sending |
| 20:11.16 | russellb | it's a DEBUG message |
| 20:11.31 | slayer192 | PakiPenguin: Can you run a 'netstat -nap | grep 127' and look for the process that is either in TIME_WAIT or ESTABLISHED ? |
| 20:11.34 | ilo_admin | Cunning we are set for GMT correct for North America, correct? |
| 20:11.44 | MikeJ | russelb.. very good... ? |
| 20:11.47 | *** join/#asterisk Cresl1n (n=matt@dsl253-055-082.dfw1.dsl.speakeasy.net) |
| 20:11.48 | *** mode/#asterisk [+o Cresl1n] by ChanServ |
| 20:11.48 | hmmhesays | MikeJ: both sides are sending g729 and ulaw as capbabilites |
| 20:11.54 | b11d | if I just dial 1111, then it wont go to the correct voicemail box.. |
| 20:11.55 | Mercestes | [TK]D-Fender: Sweet..thanks. |
| 20:12.02 | hmmhesays | which is why I'm confused |
| 20:12.03 | b11d | unless I can figure out how to alias multiple extensions to the same voicemail box |
| 20:12.06 | CunningPike | ilo_admin: I set the clock to UTC and then set the appropriate time zone |
| 20:12.29 | MikeJ | hmmhesays, confirmed in the packets? |
| 20:12.32 | FunnyManVA | if so, then you want exten => _111XXXX,1,Dial(${EXTEN:-4}) and then exten => _XXXX,1,Dial($EXTEN) |
| 20:12.35 | b11d | hhmm.. maybe I need to rethink that whole thing.. |
| 20:12.44 | Qwell | wait, what? |
| 20:12.45 | b11d | thanks for the advice |
| 20:12.46 | hmmhesays | MikeJ: yep |
| 20:12.52 | jmls | PakiPenguin: you using freepbx / trixbox / something like that ? |
| 20:12.57 | Qwell | FunnyManVA: If you Dial(${EXTEN}), you'll get...a funky loop |
| 20:13.07 | CunningPike | b11d: Set the vm box to be 1111 - then, in your _XXX1111 dialplan, you need to go to Voicemail(${EXTEN:-4}@default) |
| 20:13.16 | b11d | yep.. thats what im thinking now |
| 20:13.22 | PakiPenguin | slayer192: check pvt |
| 20:14.31 | anthonyl | ihey slayer192 @ astricon? |
| 20:15.07 | slayer192 | anthonyl: oh yeah |
| 20:15.18 | slayer192 | got my very own seat in the back |
| 20:15.19 | FunnyManVA | Yeah, sorry about that. That needs to be what you want the XXXX exten to do, like SIP/1111 |
| 20:15.32 | anthonyl | are you at the developers summit? |
| 20:15.34 | FunnyManVA | slayer192, i'm in the back too |
| 20:15.54 | PakiPenguin | lol , FunnyManVA wave :) |
| 20:15.55 | hmmhesays | bah |
| 20:15.57 | b11d | my god that works just tits.. thanks chaps. |
| 20:15.58 | hmmhesays | wtf is going on here |
| 20:16.07 | b11d | wtf is right.. |
| 20:16.16 | b11d | lets get drunk as fuck next week hmmhesays |
| 20:16.21 | b11d | we can meet in Bemidji :) |
| 20:16.29 | anthonyl | everyone in #asterisk at astricon standup! |
| 20:16.32 | anthonyl | right now! |
| 20:16.35 | *** join/#asterisk rustyb (i=rustyb@68-235-135-252.atlsfl.adelphia.net) |
| 20:16.35 | slayer192 | anthony: yeah |
| 20:16.38 | CunningPike | ilo_admin: Did you get anywhere? |
| 20:16.46 | slayer192 | noone is standing |
| 20:16.56 | anthonyl | i know |
| 20:17.04 | anthonyl | well what room are you all in? |
| 20:17.12 | FunnyManVA | this is why the wifi AP is overloaded. We're all on IRC. |
| 20:17.16 | anthonyl | ya |
| 20:17.27 | b11d | heh |
| 20:17.39 | slayer192 | Im in the Dev Sumit |
| 20:17.42 | anthonyl | me too |
| 20:17.46 | ilo_admin | Yes I am still working on it. |
| 20:17.47 | CunningPike | Ah yes, IRC - that well-known bandwidth hog.......... |
| 20:17.49 | FunnyManVA | copy that here |
| 20:17.51 | ilo_admin | One moment |
| 20:17.51 | anthonyl | im in the front left next to mog |
| 20:17.58 | jmls | anyone figured out how to make an IRC client give the user an electric shock ? |
| 20:18.07 | jmls | that would get someone's attention ... ;) |
| 20:18.12 | *** join/#asterisk clive- (n=pirch@dsl-145-25-55.telkomadsl.co.za) |
| 20:18.14 | CunningPike | jmls: Use DCC current |
| 20:18.14 | slayer192 | ok.... back right corner |
| 20:18.28 | anthonyl | i think im on the right as well |
| 20:18.32 | PakiPenguin | lol |
| 20:18.39 | FunnyManVA | back left corner here |
| 20:18.48 | anthonyl | sweet |
| 20:18.50 | anthonyl | in the blue? |
| 20:18.52 | jmls | I'm working on trying to get a headset build up a charge so that the team leader can zap em if they aren't doing well '< |
| 20:19.06 | slayer192 | ha, found on |
| 20:20.50 | PakiPenguin | lol jmls i would love to buy it |
| 20:21.03 | PakiPenguin | hehe tons of market here |
| 20:23.08 | Inez | Do anyone use L option for Dial command? |
| 20:23.17 | *** join/#asterisk akoch (n=chatzill@mail.gk-soft.de) |
| 20:23.17 | Inez | or G option? |
| 20:23.28 | clive- | inez what for? |
| 20:23.42 | Inez | for limit call duration |
| 20:23.53 | akoch | hello, how can I install mISDN on a 64bit system? |
| 20:24.04 | hmmhesays | can I get someone to try send me a fax? |
| 20:24.13 | akoch | I get /usr/lib64/gcc-lib/x86_64-suse-linux/3.3.3/../../../../x86_64-suse-linux/bin/ld: device.o: relocation R_X86_64_32 can not be used when making a shared object; recompile with -fPIC |
| 20:24.22 | hmmhesays | pretty please |
| 20:24.38 | *** join/#asterisk xnon (i=xnon@200.8.30.50) |
| 20:24.43 | clive- | Inez then use L , or S I think also does the trick |
| 20:25.56 | slayer192 | anthonyl: Im the guy right in front and to the left of the guy in blue.... |
| 20:26.05 | Inez | clive- L dont good work |
| 20:26.23 | Inez | if calleed hangup then all party are disconnected, but I want get back in to dialplan by xcaller. |
| 20:26.27 | Inez | to make another connection |
| 20:27.14 | b11d | i can do it |
| 20:27.16 | b11d | hmmhesays |
| 20:28.08 | slayer192 | astricon meeting at the door! |
| 20:28.08 | anthonyl | stand up! |
| 20:28.13 | Inez | ;] |
| 20:28.17 | PakiPenguin | lol |
| 20:28.21 | Inez | clive- do you knwo what is my problem |
| 20:28.30 | PakiPenguin | how many in the channel are at astricon right now ? |
| 20:28.31 | anthonyl | how far in the front? |
| 20:28.39 | anthonyl | thats it im going to wave |
| 20:28.41 | b11d | just start shouting nick's |
| 20:28.52 | b11d | someone yell "I am the liquor" -- it'll be funny |
| 20:29.18 | anthonyl | hhehe |
| 20:29.19 | ilo_admin | Cunning this is a little nightmarish to set the date correctly I am still working on it |
| 20:29.50 | CunningPike | ilo_admin: man date ;) |
| 20:29.55 | slayer192 | I AM THE LIQOUR! |
| 20:30.03 | anthonyl | well anyone want to go to the gas station with me to buy smokes? |
| 20:30.04 | CunningPike | ilo_admin: I'll be here...... |
| 20:30.11 | anthonyl | because im about too |
| 20:30.13 | anthonyl | to |
| 20:30.13 | *** join/#asterisk wrmem (n=monnin@monnin-win.ci.uiuc.edu) |
| 20:30.32 | slayer192 | I got a spare pack if you need them, carton in the car somwehere |
| 20:30.50 | slayer192 | ok, I'll be by the door |
| 20:30.56 | akoch | something about the mISDN 64bit error? |
| 20:30.56 | slayer192 | battery is low |
| 20:32.21 | clive- | inez for a calling card? |
| 20:32.55 | b11d | oh damn |
| 20:33.02 | *** join/#asterisk Buglouse (n=SourceRa@66.97.121.210) |
| 20:33.07 | b11d | did Slayer actually yell that at Astricon? |
| 20:33.18 | Inez | clive- yes |
| 20:33.59 | PakiPenguin | lol |
| 20:34.47 | pifiu-laptop | i have a question |
| 20:35.19 | clive- | inez try astcc |
| 20:35.35 | pifiu-laptop | if you do a line such as exten => 123454678,1,Dial(IAX2/TEST/12345678) |
| 20:35.41 | pifiu-laptop | where should that word "TEST" be at also? |
| 20:36.07 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 20:36.07 | *** mode/#asterisk [+o anthm] by ChanServ |
| 20:36.53 | hmmhesays | ok when I pick up and make asterisk answer I have no problems changing the codec |
| 20:38.27 | Inez | clive- astcc is not good solution for me needs |
| 20:38.39 | *** join/#asterisk h3x0r (n=hex@ip68-224-236-92.lv.lv.cox.net) |
| 20:40.13 | PakiPenguin | Inez: what are your needs |
| 20:40.24 | clive- | Inez what solution are you trtying to do |
| 20:40.38 | *** join/#asterisk bobby1234 (i=grondsy@h678631.serverkompetenz.net) |
| 20:41.06 | *** join/#asterisk fourcheeze (n=rich@82.153.23.79) |
| 20:41.06 | bobby1234 | hello |
| 20:41.09 | bobby1234 | anyone there? |
| 20:41.12 | fourcheeze | no |
| 20:41.23 | bobby1234 | okie |
| 20:41.23 | Inez | In Asterisk I am receivin gcall from SIP client, I want connect him to next SIP client via dial command, but settings call max duration, and after duration exceed or after called party hangup, then i want to call second user |
| 20:41.42 | hmmhesays | i'm still getting no compatible codecs |
| 20:41.44 | hmmhesays | which is nuts |
| 20:41.58 | fourcheeze | hmmhesays: what are you trying to do? |
| 20:42.11 | hmmhesays | get these faxes to work properly |
| 20:42.14 | clive- | Inez I do that with astcc, just modified it a bit |
| 20:42.29 | hmmhesays | trying to get the endpoints to use a certain codec when I dial my fax extension |
| 20:42.42 | Inez | clive- astcc you modify? |
| 20:42.46 | Inez | can you tell me more on prv? |
| 20:43.00 | fourcheeze | hmmhesays: ahh yeah |
| 20:43.05 | fourcheeze | that sort of thing sucks |
| 20:43.07 | Inez | clive- and after call you return to primary context and allow caller to make another call? |
| 20:43.32 | fourcheeze | faxes in general suck anyway |
| 20:44.46 | clive- | inez, its open source, just dive in and chaneg it how you like |
| 20:45.35 | Inez | yes, but id you did it already, maybe can you give me it? |
| 20:49.32 | hmmhesays | so when I don't set the ${SIP_CODEC} to ulaw in the dialplan I get "no compatible codecs |
| 20:49.33 | hmmhesays | " |
| 20:50.20 | fourcheeze | hmmhesays: why wouldn't you set it like that? |
| 20:51.28 | Inez | clive- astcc works good with 1.4 asterisk? |
| 20:51.43 | *** join/#asterisk blebleble (i=godie@caesar.godie.net) |
| 20:52.30 | sahafeez | hey, if my provider is seeing a 407 back is it my nat setup? |
| 20:52.37 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 20:52.48 | sahafeez | i never seen anything on the asterisk console re:dubug |
| 20:57.05 | ilo_admin | Okay Cunning I changed the date and still get the same error |
| 20:57.29 | fourcheeze | What's polycom digitmap language for 3 or more digits? |
| 20:57.43 | fourcheeze | xxx+ ? |
| 20:57.55 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 20:57.56 | CunningPike | ilo_admin: Hmm - OK - let me have another think.... |
| 20:58.15 | ilo_admin | Yeah that was fun, give me more |
| 20:58.47 | hmmhesays | I tihnk I finally rigged this thing up to send faxes |
| 20:58.53 | hmmhesays | mabe |
| 20:58.57 | hmmhesays | *mebbe |
| 20:59.29 | hmmhesays | what a freaking codec balancing act |
| 20:59.49 | fourcheeze | hmmhesays: when you achieve nirvana I'd be interested to see how |
| 20:59.55 | hmmhesays | i need a fax test though |
| 21:00.15 | hmmhesays | I have mediatrix 2102's and I'm specifying one port for faxing |
| 21:00.24 | hmmhesays | on those i'm manually setting the codec to ulaw |
| 21:00.48 | hmmhesays | if someone could shoot me a fax that would be cool |
| 21:01.55 | *** join/#asterisk macTijn (i=martijn@linda.net.insecure.nl) |
| 21:03.09 | pids | Anyone have a clue of where I could hook into have asterisk append a dtmf to a variable? |
| 21:03.24 | hmmhesays | huh? |
| 21:03.31 | *** join/#asterisk pollohawk (n=pollohaw@mmail.picksend.com) |
| 21:03.58 | Inez | clive- ? |
| 21:04.08 | fourcheeze | pids: try asking that another way |
| 21:04.13 | pollohawk | How can I configure my TE110P to accept 12 voice lines and 12 data lines over my T1 line? |
| 21:04.34 | h3x0r | rtfm |
| 21:04.48 | *** join/#asterisk Strom_C (n=strom@netblock-66-159-243-60.dslextreme.com) |
| 21:04.57 | clive- | Inez, I guess it will work on 1.4, sorry, just distracted |
| 21:05.20 | ilo_admin | Cunning I have to go but I will be back later, Thanks for everything |
| 21:05.34 | Inez | clive- can you show me your changes, i am looking at file, but is too compilcated |
| 21:05.38 | pids | fourcheeze, what I mean is that currently asterisk only sends a dtmf event to the channel if your looking for it. What I want is to put every dtmf that shows up into a variable. |
| 21:05.44 | *** part/#asterisk sahafeez (n=sahafeez@67.109.14.227.ptr.us.xo.net) |
| 21:05.49 | aptura | trying to recall to reduce eco tx should be reduced or increase but how much |
| 21:06.03 | hmmhesays | pids, show application read |
| 21:06.40 | CunningPike | ilo_admin: OK - later |
| 21:06.48 | aptura | looks like 8.2 for rxgain |
| 21:07.08 | Strom_C | aptura: that seems like an unusually high rxgain value |
| 21:07.26 | pids | hmmhesays, I have to be looking for a dtmf in that case. |
| 21:07.41 | hmmhesays | yeah |
| 21:07.50 | aptura | Strom okay what do you sugest. |
| 21:07.55 | pids | what I am looking to do is have asterisk notify me when a dtmf shows up in the stream |
| 21:08.11 | aptura | Iv got a little feedback right now so need to qwench it. |
| 21:08.30 | fourcheeze | pids: can you do what you want with a feature? |
| 21:08.37 | *** join/#asterisk macTijn (i=martijn@linda.net.insecure.nl) |
| 21:09.03 | pids | fourcheeze, buffering dtmf input |
| 21:09.37 | fourcheeze | are you doing something like playing a message and recording all the dtmf during playback? |
| 21:10.25 | pids | as an example if you wanted to check you bank account you could push the account number and id number all in one long string and the system could then parse that out. |
| 21:10.44 | fourcheeze | anyone know how to implement a call queue where the destination is a phone on the end of a pstn line? |
| 21:10.48 | *** join/#asterisk damanivu (n=damanivu@ip68-4-207-173.oc.oc.cox.net) |
| 21:10.55 | fourcheeze | pids: can't you just dial that number? |
| 21:11.12 | fourcheeze | I mean |
| 21:11.30 | pids | well not if you want to have the input change based upon menu contect |
| 21:11.33 | fourcheeze | have a very long timeout on getting digits to dial |
| 21:12.00 | fourcheeze | and then have an entry for _X. where you can parse it later |
| 21:12.25 | *** join/#asterisk Buglouse (n=SourceRa@66.97.120.212) |
| 21:13.38 | pids | doesnt work, if your moving from one menu to another and asterisk isnt listening for a dtmf then it just ignores the dtmf and does not put it in _.X |
| 21:13.38 | pids | It sees it, just ignores it. |
| 21:13.38 | pids | If you have iax debug on it shows on the console but never gets sent up to the channel. |
| 21:13.38 | fourcheeze | I'm just thinking like an IVR |
| 21:13.38 | fourcheeze | those work I'm sure |
| 21:13.39 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.216.2) |
| 21:13.53 | fourcheeze | that's basically where you dial an extension during a message |
| 21:15.13 | pids | right but thats beacuse asterisk is off looking for a dtmf, its basically in a giant while loop. however if you are doing something that does not expect dtmf input then asterisk just ignores what comes in on the channel |
| 21:15.22 | pids | Its not event driven. |
| 21:15.27 | fourcheeze | so what are you doing that doesn't expect dtmf? |
| 21:15.38 | pids | agi |
| 21:15.42 | fourcheeze | ahh |
| 21:15.53 | fourcheeze | can't you avoid that? |
| 21:16.05 | fourcheeze | return back to the dialplan? |
| 21:16.16 | aptura | what do you people set your zap rx/tx to? |
| 21:16.57 | blebleble | think mine is +3.6 or so |
| 21:17.15 | pids | What I want is it to say "Hey heres a DTMF on the IAX channel, I'll append it to this channel variable" Then I can look at that variable for any dtmf digits that arrived while I was out in the agi doing whatever. |
| 21:17.32 | pids | fourcheeze, not if they are doing something in the agi. |
| 21:17.41 | fourcheeze | ok |
| 21:17.45 | fourcheeze | I can't think of a way around it |
| 21:19.55 | fourcheeze | do the polycom's support a stun server? |
| 21:20.04 | fourcheeze | s/polycom's/polycoms |
| 21:20.16 | fourcheeze | if so how? |
| 21:20.26 | *** join/#asterisk ManxPower (n=manxpowe@124.sub-70-216-103.myvzw.com) |
| 21:20.40 | *** join/#asterisk inv_Arp (i=junya@c-71-206-88-100.hsd1.fl.comcast.net) |
| 21:22.09 | *** join/#asterisk xpasha (n=pavel@static-host.10-252-30-217.kgts.ru) |
| 21:22.14 | xpasha | hello |
| 21:22.23 | xpasha | anybody tested 1.3beta3 |
| 21:22.28 | xpasha | I have strange problem |
| 21:23.04 | *** join/#asterisk dextro (n=dextro@64.25.11.250) |
| 21:23.12 | xpasha | called phone still ringing when calling hangs up |
| 21:23.21 | xpasha | both sides on SIP |
| 21:23.51 | b11d | I didnt know there was a 1.3 |
| 21:24.00 | xpasha | 1.4 |
| 21:24.03 | b11d | OH |
| 21:24.03 | b11d | :) |
| 21:24.07 | xpasha | sorry :) |
| 21:25.38 | b11d | <-- jackass |
| 21:25.38 | b11d | hmm.. so how long does it ring? |
| 21:25.38 | xpasha | hmmm very long |
| 21:25.38 | xpasha | I did not wait for timeout |
| 21:25.38 | b11d | are you sure you arent putting the call on hold when you are "hanging up" ? |
| 21:25.38 | b11d | thus creating the illusion that the call was destroyed.. |
| 21:25.38 | xpasha | just hang up called side too |
| 21:25.38 | b11d | so, despite the fact that you hang up both sides, it continues to ring? |
| 21:25.38 | xpasha | I have some kind of lab |
| 21:25.38 | fourcheeze | anyone using polycoms behind nat with stun ? |
| 21:25.38 | xpasha | both sides at my desk :) |
| 21:25.39 | fourcheeze | or without stun ? |
| 21:25.39 | b11d | yeah i hear you on that xpasha.. |
| 21:25.49 | b11d | well.. thats pretty fucked up eh |
| 21:25.49 | *** part/#asterisk LogiForce (n=LogiForc@cc51914-a.groni1.gr.home.nl) |
| 21:25.52 | b11d | I guess thats why its beta.. i've not heard of that happening before. |
| 21:25.59 | b11d | fourcheese, im doing it. |
| 21:26.11 | xpasha | one example |
| 21:26.11 | b11d | oh wait.. sorry.. I misread your question |
| 21:26.15 | xpasha | just a moment |
| 21:26.28 | aptura | from voipjet - Dear Customers, |
| 21:26.28 | aptura | You asked for it, you got it! |
| 21:26.28 | aptura | 1- Our new Level(3) bandwidth premium NYC server is here |
| 21:26.37 | aptura | just got the email from them. |
| 21:26.42 | b11d | and this pleases you? |
| 21:27.29 | [TK]D-Fender | fourcheeze : Polycom's do not support STUN, and I have had them working from behind NAT just fine |
| 21:27.38 | aptura | looks like voipjet is expanding also into california. |
| 21:27.47 | xpasha | exten => 997960,1,Dial(SIP/084322997828@217.107.x.2|10) |
| 21:28.05 | *** part/#asterisk mixi (n=mixi@the.one.and.only.iammixi.de) |
| 21:28.09 | fourcheeze | [TK]D-Fender: are there any secrets - I have them working fine just here but at a customer's site they seem not to register reliably |
| 21:28.14 | xpasha | in this case called side phone still ringing even when I hang up |
| 21:28.26 | xpasha | calling side is also SIP |
| 21:28.29 | fourcheeze | [TK]D-Fender: also I keep seeing that they support stun |
| 21:28.36 | fourcheeze | but then nowhere to configure |
| 21:28.48 | [TK]D-Fender | fourcheeze : Why are you asking if you're so sure? Where do you see this? |
| 21:28.56 | b11d | I've got to go.. sorry... ttyl all |
| 21:29.13 | fourcheeze | [TK]D-Fender: I mean their dealers seem to advertise it - however I'll take your word that they don't |
| 21:29.23 | fauxalliance | Verdammt: has anyone successfully configured an avaya 4612. (no sip, only H.323) it connects to tftp and such, looking for configuration guidelines. |
| 21:29.30 | [TK]D-Fender | fourcheeze : if you have multiple behind a NAT I'd suggest setting each ones port to something different |
| 21:29.47 | [TK]D-Fender | fourcheeze : Show me. |
| 21:30.22 | aptura | [TK]D-Fender is there any impedence measuring echo tools avaible that you know of? |
| 21:30.45 | fourcheeze | http://store.voxilla.ca/product.php?productid=16159&cat=0&page=1 |
| 21:31.00 | xpasha | So It's very amazing situation that digium produce even third version of beta with so huge bugs |
| 21:31.04 | fourcheeze | [TK]D-Fender: for instance - not that I'm in .ca |
| 21:31.05 | xpasha | like dscribed |
| 21:31.22 | justinu|laptop | xpasha: why is it amazing? |
| 21:32.06 | xpasha | so my friend who is linux kernel developer was shocked with digium's style of making releases |
| 21:32.33 | xpasha | the make releases with such problems |
| 21:32.46 | [TK]D-Fender | fourcheeze : ok, well its not mentioned in the admin guide. |
| 21:32.56 | fourcheeze | yeah, so I see |
| 21:33.09 | xpasha | that sometimes it's possible to guess that they make releases from SVN by random |
| 21:33.14 | xpasha | not testing |
| 21:33.20 | xpasha | no any checking |
| 21:33.42 | justinu|laptop | maybe that's why zaptel never made it into the kernel sources? |
| 21:33.43 | denon | xpasha: there's testing for stable releases |
| 21:33.48 | xpasha | I can guess they do it to force people to buy business version |
| 21:34.06 | xpasha | hehe :) |
| 21:34.08 | denon | xpasha: you dont have to check out the bleeding edge, you can choose a stable version |
| 21:34.39 | xpasha | the same situation with 1.2 branch |
| 21:35.17 | xpasha | when I take the new "stable" release I usually find some problems with even old functions |
| 21:35.18 | *** join/#asterisk srbaker (n=srbaker@S01060017ee01d049.no.shawcable.net) |
| 21:35.46 | xpasha | of so called "stable" release made with bugs like :: |
| 21:35.55 | xpasha | of so called "stable" release made with bugs like "one way audio" |
| 21:36.06 | justinu|laptop | don't forget rfc2833 issues |
| 21:36.10 | xpasha | of=or |
| 21:36.14 | *** part/#asterisk srbaker (n=srbaker@S01060017ee01d049.no.shawcable.net) |
| 21:36.18 | denon | xpasha: you sure it's not user error? improper codecs etc |
| 21:39.37 | xpasha | I can find in digium news the meassage about stable release with such bug |
| 21:39.37 | *** join/#asterisk sahafeez (n=sahafeez@67.109.14.227.ptr.us.xo.net) |
| 21:39.38 | *** join/#asterisk yogurt2ungue (n=yogurt2u@24-48-231-201.fibertel.com.ar) |
| 21:39.38 | xpasha | and new release 1.2.x.1 after that |
| 21:39.38 | [TK]D-Fender | fourcheeze : And confirming with their datasheets, the mention NAT support, but not STUN |
| 21:39.38 | denon | xpasha: and of course, the linux kernel is flawless .. |
| 21:39.38 | denon | xpasha: your linux dev friend should quit laughing at asterisk, and start learning from the bsd developers |
| 21:39.38 | xpasha | how has it been "tested" if release consisted so fatal problem? |
| 21:39.38 | denon | better use of his time |
| 21:39.38 | xpasha | it's not important but linux kernels almost never consist fatal errors in stable releases |
| 21:39.58 | denon | fatal is relative .. you classify a fatal kernel error as something that wont boot? |
| 21:39.58 | xpasha | so this is the reason when my friend was shocked starting work with asterisk |
| 21:40.05 | denon | what if I have some crazy network card that doesnt work .. |
| 21:40.10 | denon | or some weird raid controller .. that's fatal |
| 21:40.19 | xpasha | denon SIP one way audio is fatal error :) |
| 21:40.23 | denon | it's very difficult to test each release of saterisk with every sip device out there |
| 21:40.33 | denon | xpasha: not to a zap user it's not |
| 21:40.42 | sahafeez | exit |
| 21:40.43 | sahafeez | exit |
| 21:41.32 | *** join/#asterisk Buglouse (n=SourceRa@66.97.120.212) |
| 21:41.50 | xpasha | so don't you think that even beta version with this bug that I desribed have any rights to be released? |
| 21:41.55 | *** part/#asterisk Gunde (n=spamyous@82.153.170.213) |
| 21:42.23 | denon | xpasha: I'll agree, there's always plenty of room for improvement .. and there will always be a few mistakes, and I encourage you to help in the dev process if you feel you can fill a need |
| 21:42.32 | xpasha | and I dont believe that digium's developers responsible for releases did not know about it |
| 21:42.54 | justinu|laptop | lol, as if the dev process was open to outsiders |
| 21:44.35 | festr__ | i think there should be more shorter stable releases. |
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| 21:45.28 | xpasha | the same situation with cisco IOS |
| 21:46.10 | xpasha | when you install new version you take a risk that anything old and working all the time good can immediately stop to work with new version |
| 21:47.13 | denon | xpasha: as with any application |
| 21:47.21 | hmmhesays | yep |
| 21:47.24 | denon | especially if you're using it in a n unsupported way |
| 21:47.40 | denon | it's always the admin's job to test it in his environment before rolling to production |
| 21:47.43 | xpasha | you mean bug report? |
| 21:47.55 | xpasha | so i did not find chan_sip in the list |
| 21:48.12 | denon | xpasha: you're welcome to cooperate in the testing effort .. |
| 21:48.21 | denon | anything beyond offering to help is really just trolling |
| 21:50.06 | xpasha | I will take SVN and test if this still has this bug |
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| 22:00.43 | xpasha | shiiitt |
| 22:00.54 | xpasha | the current SVN still has this bug |
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| 22:09.57 | |stefan| | in what app_ module is the GROUP() function located ? ?? |
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| 22:14.55 | gmfm | |stefan| funcs/func_groupcount.c looks to be the one |
| 22:15.41 | epac | on our old phone system we could use the "link" button to forward calls, and with another set of keys, bridge the two calls (to have 3 people talking together). how is that called? (so i can search the web for how to configure use that feature? |
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| 22:17.13 | DrkShdw | epac: sounds like you mean conferencing |
| 22:17.33 | epac | well, i've seen the stuff about the "meetme" conferences... |
| 22:17.39 | epac | but that's not what i'm looking for |
| 22:17.39 | DrkShdw | yeah |
| 22:17.44 | CunningPike | epac: Yup - 3-way calling or conferencing |
| 22:18.05 | CunningPike | epac: It's conferencing on the phone, not via Asterisk - like 3-way calling on a cell |
| 22:18.07 | epac | all i find when i look for conference is the "conference/meetme" extension setup |
| 22:18.20 | epac | CunningPike : yep |
| 22:18.33 | epac | i've got polycom phones if that helps... |
| 22:18.50 | CunningPike | epac: It Just Works(tm) |
| 22:18.55 | epac | :o) |
| 22:19.15 | aptura | ive been given crap once on voice quality by bridging pstn with cell once. |
| 22:19.17 | epac | don't need to setup keys or anything on the phone? |
| 22:19.41 | CunningPike | epac: Negatory |
| 22:19.46 | aptura | it was possible transcoding issues and my sip provider. |
| 22:20.16 | epac | so it's as simple a "call user1", Press flash, call user2, press conference ? |
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| 22:24.59 | sexyken | Hey everyone! I just bought a WIP330 -- I'm curious to know if it's possible to configure asterisk so that I can set my callerid through my phone? |
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| 22:30.07 | sexyken | Anyone? |
| 22:38.19 | carrar | sexyken, you could dial a 20 digit phone number when you dial out :) |
| 22:38.25 | carrar | and have Asterisk parse it out :) |
| 22:39.06 | carrar | 1-10 extract out and make the callerid |
| 22:39.11 | carrar | 11-20 dial too number |
| 22:39.20 | carrar | heh |
| 22:40.39 | sexyken | Hrm. |
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| 23:05.59 | gmfm | can asterisk send text messages to polycom phones? |
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| 23:14.02 | rpm | does anyone have any experience working with broadsoft carrier grade pbx? |
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| 23:22.30 | lters_ | any cepstra users here with sucess stories? |
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| 23:27.44 | [hC] | [TK]D-Fender: you alive? |
| 23:28.59 | [hC] | anyone use a polycom 601, especially with an expansion module? |
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| 23:52.35 | gmfm | anyone know why this happens occasionally? Oct 24 16:51:57 WARNING[23009]: chan_zap.c:7926 zt_pri_error: Call Reference Length not supported: 0 |
| 23:52.36 | gmfm | Oct 24 16:51:57 WARNING[23009]: chan_zap.c:9056 pri_dchannel: Received NOTIFY on unconfigured channel 255/255 span 1 |
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