| 00:00.58 | Jedirl | |Vulture|: what is XO? |
| 00:02.11 | |Vulture| | mikewho2: I used shop4t1.com where I started my search |
| 00:02.11 | AgiNamu | syle, I'm not opposed to hiring someone at all |
| 00:03.06 | AgiNamu | just so long they are in a position to actually support my needs :) |
| 00:03.06 | AgiNamu | That's why I am looking at Digium's support/biz-ed + some big vendor |
| 00:03.26 | |Vulture| | Jedirl: just a phone company offering lots of new-type of voice/data soultions |
| 00:03.36 | ariel_ | |Vulture|, I don't either but due to some prices and dealer I have had to use them. |
| 00:03.36 | ariel_ | AgiNamu, digium's not that big. |
| 00:03.36 | DaLion | so no one got experienc e setting up ast with 2 nucs ? |
| 00:03.40 | geesus | any aussies here? |
| 00:03.46 | *** join/#asterisk JerJer[mobile] (~jj@ip68-103-26-140.ks.ok.cox.net) |
| 00:03.58 | Jedirl | |Vulture|: ok |
| 00:04.34 | AgiNamu | no, they aren't. but they make the cards :\ |
| 00:04.34 | AgiNamu | if someone can get a machine, stress test it, and ensure it works |
| 00:04.34 | AgiNamu | I'd be willing to pay for that |
| 00:04.34 | ariel_ | AgiNamu, your name remindes me of a friend of mine in the Philippines are you from there? |
| 00:04.46 | AgiNamu | but it'd have to be from a supported vendor |
| 00:04.47 | AgiNamu | nope, not at all |
| 00:04.47 | AgiNamu | im a white canadian cracka |
| 00:04.58 | mikewho2 | or does anyone know a website that helps me find available PRI's in my area? |
| 00:05.40 | syle | are you sure this g729 will only work on intel processors? |
| 00:06.05 | syle | gonna try some different optimization in the code maybe , give it one more go |
| 00:06.10 | ariel_ | argh network split. |
| 00:06.26 | *** join/#asterisk valence (~valence@Quebec-HSE-ppp230300.qc.sympatico.ca) [NETSPLIT VICTIM] |
| 00:06.30 | ariel_ | syle, like I said before I have not used the open one. |
| 00:06.31 | Jedirl | syle: if it's C code it should work anywhere |
| 00:06.43 | Jedirl | syle: but if it contains assembler code... |
| 00:09.01 | syle | well they aren;t releasing the source code so no way of knowing i guess |
| 00:09.12 | Jedirl | wasn't it opensource? |
| 00:09.24 | syle | the ipp libraries? |
| 00:09.28 | *** join/#asterisk shido6 (~greg@d57-87-253.home.cgocable.net) |
| 00:09.55 | *** join/#asterisk Delvar (~irc@83.146.53.34) [NETSPLIT VICTIM] |
| 00:09.56 | Jedirl | I tought you were talking about the opensource G729 codec |
| 00:10.01 | *** join/#asterisk JohnJacob (~JohnJacob@the.med-web.com) |
| 00:10.09 | *** join/#asterisk darkskiez (~darkskiez@usergc137.dsl.pipex.com) [NETSPLIT VICTIM] |
| 00:10.25 | syle | http://www.readytechnology.co.uk/open/g729/ |
| 00:10.42 | syle | ftp://download.intel.com/software/products/ipp/downloads/l_ipp_ia32_itanium_p_4_1.tar |
| 00:10.59 | syle | this bitch came as one big install binary |
| 00:11.27 | syle | with license key |
| 00:11.34 | ariel_ | I got to make some coffee....(Espreso anyone?) |
| 00:12.03 | *** join/#asterisk DEEZED (deezed@adsl-065-006-189-182.sip.bct.bellsouth.net) |
| 00:12.31 | *** join/#asterisk Sedorox (brandon@sedorox.staff.smartserv) [NETSPLIT VICTIM] |
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| 00:12.32 | *** join/#asterisk A-Tuin|work (~A-Tuin@nat.office.legend.net.uk) [NETSPLIT VICTIM] |
| 00:12.32 | syle | <PROTECTED> |
| 00:12.32 | syle | Ouch ... error while writing audio data: : Broken pipe |
| 00:12.32 | syle | Illegal instruction |
| 00:13.20 | *** join/#asterisk file[mac] (~jcolp@mctn1-3494.nb.aliant.net) |
| 00:13.30 | tzanger | syle: don't compile for a processor you don't have |
| 00:13.37 | syle | well here is the problem i guess |
| 00:13.37 | tzanger | if you're on a cyrix processor use i586 |
| 00:13.39 | tzanger | not i686 |
| 00:13.40 | syle | # For Athlon (untested) |
| 00:13.40 | syle | #OPTIMIZE= -O6 -march=athlon -mcpu=athlon -ffast-math -fomit-frame-pointer |
| 00:13.40 | syle | #IPPCORE=a6 |
| 00:13.51 | tzanger | syle: those are commented |
| 00:13.58 | syle | that one crashed my machine |
| 00:14.31 | Juggie | syle, there are builds with gcc |
| 00:14.33 | tzanger | syle: do you have an athlon? |
| 00:14.38 | Juggie | use the pentium2 build or something |
| 00:15.08 | syle | trying PII |
| 00:15.15 | tzanger | syle: what processor do you have |
| 00:15.32 | syle | AMD processor at 800 mhz |
| 00:15.40 | file[mac] | a Duron? |
| 00:15.49 | pussfeller | there some command line way to delete all my voicemails? |
| 00:16.09 | tzanger | syle: likely a duron then? |
| 00:16.22 | tzanger | do you not know the exact processor? |
| 00:16.25 | Juggie | try the pentium or pentium 2 build |
| 00:16.32 | Juggie | the gcc build |
| 00:16.53 | syle | owww |
| 00:16.59 | syle | it liked PII |
| 00:18.00 | syle | omg what a pain in the ass that was |
| 00:18.23 | syle | that would have been hell it the system was ext2 |
| 00:18.36 | syle | no journaling on all those crashes lol |
| 00:18.43 | syle | yes this is just one of my older comps at home |
| 00:19.00 | *** join/#asterisk iswm (iswm@iswm.user) [NETSPLIT VICTIM] |
| 00:19.16 | syle | 800 mhz i figure is still good for some things |
| 00:20.18 | file[mac] | let's do it like they do on the discovery channel! |
| 00:20.54 | Jedirl | I have 50 P-II where I work |
| 00:21.06 | Jedirl | IBM hardware with RAID SCSI hard-disks |
| 00:21.09 | enots | syle, just convert ext2 to ext3 :) |
| 00:21.25 | syle | i would never use less than a 2.6 ghz processor for production |
| 00:21.47 | Jedirl | DNS servers just work in a P-II 400, believe me |
| 00:21.48 | Jedirl | :D |
| 00:22.15 | syle | dns servers get no load though hehe |
| 00:22.36 | file[mac] | except when you get them to reload their zones |
| 00:22.41 | file[mac] | and you have LOTS of zones |
| 00:22.43 | file[mac] | then it's FUN |
| 00:22.49 | syle | well depends, if you link the bind code with mysql like i did , you can have some overhead on the database, but about it |
| 00:22.54 | *** join/#asterisk newbien (~e@147.241.33.65.cfl.res.rr.com) |
| 00:23.11 | Jedirl | I use djbdns |
| 00:23.23 | syle | yeah i use that for internal dns to |
| 00:23.37 | Jedirl | internal? |
| 00:23.40 | syle | just not public dns |
| 00:23.44 | Jedirl | why not on public ones? |
| 00:24.01 | enots | =) |
| 00:24.06 | enots | secret dns! |
| 00:24.08 | syle | cause djbdns didn;t have any database drivers |
| 00:24.31 | syle | and ldap is not a database i don;t care what anyone says hehe |
| 00:24.37 | Jedirl | I wouldn't trust mysql for a DNS |
| 00:24.38 | Jedirl | ever |
| 00:24.46 | file[mac] | not this again. |
| 00:24.59 | syle | lol |
| 00:25.01 | Jedirl | hehehe |
| 00:25.04 | syle | dejavu |
| 00:25.11 | Jedirl | this has been discussed before? |
| 00:25.20 | ariel_ | file[mac], don't worry it's like well you know as.... eveyone has one. |
| 00:25.22 | syle | find i won;t even argue with jedirl about what he just said then |
| 00:25.41 | Jedirl | yup, just my opinion |
| 00:26.18 | syle | i;ve been using mysql since it first came out, its how you code the applications to go with the database is what matter , not so much the database |
| 00:26.40 | Jedirl | well, I have seen so many "mysqld got signal 11" |
| 00:26.48 | Jedirl | too many |
| 00:26.49 | file | next you'll be saying you'll never be caught using a Mac |
| 00:27.11 | syle | jedirl then your optimizing or doing something wrong |
| 00:27.35 | syle | that message is usually a result of bad memory block in your server |
| 00:27.39 | Jedirl | syle: I don't think I'm doing anything wrong... and I always use mysql.com binaries |
| 00:27.53 | syle | change out your ram |
| 00:28.01 | ariel_ | I have see error with mysql, sql and even on db2 there all well bad and good. |
| 00:28.03 | Jedirl | not a ram problem :) |
| 00:28.17 | enots | compile by hand |
| 00:28.26 | xeet2 | jedirl: are you sure? have you run memtest? |
| 00:28.33 | ariel_ | just one word backups |
| 00:28.47 | Jedirl | xeet2: no memtest for sun hardware I guess |
| 00:29.08 | syle | fine then don;t , happy crashing lol |
| 00:29.09 | xeet2 | just because you don't see any issues with other apps doesn't mean it isn't a memory issue |
| 00:29.09 | syle | sun hardware |
| 00:29.13 | xeet2 | is the memory ecc? |
| 00:29.17 | syle | waste of money |
| 00:29.30 | xeet2 | syle: agreed, at least now |
| 00:29.30 | Jedirl | syle: I didn't buy it, it was there when I arrived |
| 00:29.51 | Jedirl | syle: I agree sun hardware is a waste of money in many cases |
| 00:30.59 | syle | well can;t help you |
| 00:31.01 | enots | hoorror.. waste of money for sun hardware and running mysql binary on it.. |
| 00:31.13 | syle | when i worked for companies with sun hardware i used oracle cause they could afford it |
| 00:31.44 | Jedirl | enots: what makes you think your compiler is better than mysql's? |
| 00:31.51 | ariel_ | argh oracle is also another one that is suppose to be very good which it is. But still has problems. |
| 00:31.54 | syle | veritas and fiber array channels, now there were the days lol |
| 00:32.28 | enots | jed, my mysql dont have SIGSEGV |
| 00:33.01 | Jedirl | ender: mine have had |
| 00:33.49 | ariel_ | ok back to asterisk stuff.... |
| 00:34.09 | enots | compile it from sources.. i think this is will solve ur problem |
| 00:34.09 | *** join/#asterisk bofh42 (~bofh42@p54822C2A.dip0.t-ipconnect.de) |
| 00:34.38 | Jedirl | no, really... I don't know why people think their compiler is better than the rest of the world's :) |
| 00:34.45 | ariel_ | Anyone here know the settings for the qos on the polycom. Is the number like from 1 to 7 higher is better or lower is? |
| 00:34.54 | syle | sure if your thinking its a library linking problem, but i doubt it, i beleive all their binaries are statically linked |
| 00:35.05 | enots | jed, im not only about compiler.. |
| 00:35.06 | Jedirl | I don't know why people think they do "./configure; make" better than the ones who coded the software |
| 00:35.20 | enots | =) |
| 00:35.25 | Jedirl | enots: if they provide binaries which are targeted to your *exact* platform... |
| 00:35.29 | enots | i know |
| 00:36.21 | enots | coz some coders dont know a lot about my os. :) but configure and gcc, ld know better |
| 00:36.48 | Jedirl | :) |
| 00:40.16 | *** join/#asterisk Sato1 (~rauleli@209.12.201.200) |
| 00:40.59 | enots | anyway its strange to hear about -11. mysql is pretty stable. so its can be hardware fault (you dont trust this idea) or binary uncompatibility... |
| 00:42.03 | fugitivo | use postgresql! it's a real database |
| 00:42.53 | Sato1 | whats make mysql not a real database? |
| 00:42.56 | syle | i still thinks its her ram |
| 00:42.59 | ariel_ | oh no not that one..... |
| 00:43.24 | Sato1 | i like mysql too |
| 00:43.33 | syle | throw out the sun crap get some dell hardware for mysql |
| 00:44.02 | syle | dell poweredge 2650's works great |
| 00:44.17 | syle | only cost me about 8k |
| 00:44.45 | *** join/#asterisk d-tech (~dtc@node-423a1ebb.cle.onnet.us.uu.net) |
| 00:44.57 | syle | 8 gigs of ram, 6 scsi drives, dual 3.2 cpu's |
| 00:45.04 | Sato1 | there are nice sets of multi procesors using opteron, quite reliable and not that expensive |
| 00:46.18 | Juggie | isnt skinny=sccp? |
| 00:46.28 | enots | yep |
| 00:46.32 | *** join/#asterisk ben_no (~chatzilla@ras-c5800-2-55-239.dialup.wisc.edu) |
| 00:46.39 | enots | new flash in PR :) |
| 00:46.44 | Juggie | then chan_skinny.c |
| 00:46.45 | Juggie | in channels |
| 00:48.57 | Sato1 | whats the best code for fax in asterisk? or at least the must common |
| 00:49.21 | Sato1 | spandsp is giving me a couple troubles, cant get a fax even the dealing begins |
| 00:49.58 | ariel_ | Sato1, what codec are you using or service to get the call in? |
| 00:50.31 | Sato1 | from pstn to zap |
| 00:50.49 | *** join/#asterisk iswm (iswm@iswm.user) |
| 00:50.49 | enots | juggie, i think there some features/differences... XXI century in da world :) |
| 00:51.25 | ben_no | I installed 1.0 on a 350Mhz with 128Mb and on first reboot after fresh install, booting got stuck on "GRUB". (This is my first time trying asterisk) Any reason why? |
| 00:52.06 | ben_no | I think it installed alright, but the loader didn't work |
| 00:52.47 | ben_no | Is there a LiveCD of Asterisk? |
| 00:52.47 | fugitivo | ben_no: what did you install? |
| 00:52.57 | ben_no | Asterisk 1.0 |
| 00:53.01 | syle | i think i've earned a joint |
| 00:53.10 | fugitivo | ben_no: what linux distribution? |
| 00:53.20 | ariel_ | ben_no, which OS? |
| 00:53.24 | ben_no | None(?) |
| 00:53.37 | ben_no | I thought it could be loaded by itself? |
| 00:53.49 | ben_no | It's just a program? |
| 00:53.51 | ariel_ | hehehehe |
| 00:53.55 | fugitivo | ben_no: asterisk is a program, that runs under an operating system |
| 00:54.36 | enots | but ben_no "installed" it! |
| 00:54.38 | *** join/#asterisk JerJer[mobile] (~jj@ip68-103-26-140.ks.ok.cox.net) |
| 00:54.41 | ariel_ | if you had 256mg ram you could try Asterisk@home which is a iso that installs everything you need to get started. |
| 00:54.57 | ariel_ | question how do you install asterisk without an os? |
| 00:55.06 | JerJer[mobile] | so like what would cause a: SIP/2.0 488 Not Acceptable Here (codec error) |
| 00:55.26 | ariel_ | wrong codec picked |
| 00:55.29 | JerJer[mobile] | m=audio 1100 RTP/AVP 0 |
| 00:55.30 | JerJer[mobile] | a=ptime:20 |
| 00:55.30 | JerJer[mobile] | a=rtpmap:0 PCMU/8000 |
| 00:55.34 | JerJer[mobile] | from the sending sdp |
| 00:55.40 | syle | because asterisk@home includes an os |
| 00:55.42 | JerJer[mobile] | allow=ulaw is in asterisk |
| 00:55.48 | JerJer[mobile] | well sip.cnf |
| 00:55.52 | JerJer[mobile] | +o |
| 00:55.53 | ben_no | sorry, I didn't realize asterisk was a seperate program too http://asteriskathome.sourceforge.net/ |
| 00:56.14 | ariel_ | asterisk@home has the os. But it needs 256mg ram |
| 00:56.17 | *** join/#asterisk _-Jon-_ (jon@hs-wolfedale-216048166226.3web.net) |
| 00:56.22 | syle | does centos or whatever have the 2.6 kernel? |
| 00:56.31 | _-Jon-_ | Hey everyone |
| 00:56.37 | ariel_ | 2.4 |
| 00:56.48 | _-Jon-_ | Anyone here unfortunate enough to have a Budgetone product die on them? |
| 00:57.06 | ariel_ | at least the one with AAH it's based on 3.4 the version 4 is 2.6 based. |
| 00:58.13 | ben_no | does anyone here use Asterisk@home? |
| 00:58.18 | geesus | used to |
| 00:58.20 | geesus | dont any more |
| 00:58.25 | ben_no | why not? |
| 00:58.47 | geesus | bbs ill tell you in a min |
| 01:01.21 | _-Jon-_ | Anyone know what the warranty is on a HT-286? |
| 01:01.44 | *** join/#asterisk doolph (doolph@200.46.148.35) |
| 01:05.02 | *** part/#asterisk Nuxi (~nuxi@cust-sdsl-204-250-82-202.bzn-co-i1000-01.bridgeband.net) |
| 01:07.52 | *** join/#asterisk outsidefactor (~blah@203-217-79-71.dyn.iinet.net.au) |
| 01:08.46 | AgiNamu | syle, the open G729 will only work on Intel. it uses intel-specific library |
| 01:08.56 | ariel_ | ben_no, I use asterisk@home on some systems. |
| 01:09.16 | geesus | aight back, I stopped using a@h cause the other labs im looking to hook up to use debian and conf wise it would make it easier if we were running the same thing |
| 01:09.34 | doolph | HI |
| 01:11.36 | syle | agi then i guess you;ll be surprised to know after some code changes i made it works |
| 01:12.42 | *** join/#asterisk iguy (~iguy@dsl093-197-234.mke1.dsl.speakeasy.net) |
| 01:15.00 | *** join/#asterisk kimo_sabe (nick@zappa.azrackspace.net) |
| 01:15.02 | meppl | guten nacht |
| 01:15.32 | meppl | good night |
| 01:15.48 | DaLion | RNING[1219]: chan_sip.c:778 __sip_xmit: sip_xmit of 0x86ee41c (len 780) to 192.168.1.22 returned -1: Permission denied |
| 01:15.50 | DaLion | nice |
| 01:17.39 | *** join/#asterisk mikewho2 (~Schnappi@ip68-105-227-82.lu.dl.cox.net) |
| 01:20.10 | Qwell | file: You messaged me earlier? |
| 01:20.40 | *** join/#asterisk constfilin (~cf@204.147.182.37) |
| 01:20.51 | constfilin | greetings |
| 01:21.31 | constfilin | I've got a problem with Zap TDM card |
| 01:21.54 | *** join/#asterisk Pete_Largo (~Pete_Larg@adsl-65-71-225-121.dsl.rcsntx.swbell.net) |
| 01:21.59 | tzanger | constfilin: excellent. call digium support :-) |
| 01:22.05 | geesus | hah |
| 01:22.35 | *** part/#asterisk K9DI_BSD_WrkStn (~k9bsd@207-246-185-168.EastVillage.ResNet.wiu.edu) |
| 01:22.36 | constfilin | tzanger: well, I'll do it, but may be someone here has seen somethinkg like this can can help me sooner. |
| 01:23.22 | *** join/#asterisk riksta (~rick@host81-155-216-40.range81-155.btcentralplus.com) |
| 01:23.41 | tzanger | constfilin: well you best be explaining your problem |
| 01:24.49 | DaLion | anyone knot avout this ? |
| 01:24.50 | DaLion | p.c:778 __sip_xmit: sip_xmit of 0xbf99eed8 (len 382) to 192.168.1.22 returned -1: Permission denied |
| 01:25.32 | constfilin | ok, I have a TDM card and when someone calls it, the first call comes through all right. |
| 01:28.28 | DaLion | ok how much for help with nat and ipw ? |
| 01:28.31 | DaLion | ipfw |
| 01:28.55 | constfilin | Now if during that time a second call comes to the TDM card, the caller hears the long rings but the call nevers coems through. |
| 01:30.07 | *** part/#asterisk makhtar (~ageller@mail.bulletinnews.com) |
| 01:30.09 | *** join/#asterisk Beirdo_ (~gjhurlbu@beirdo.user) |
| 01:30.27 | *** join/#asterisk ixx (foobar@cpe-70-113-47-137.austin.res.rr.com) |
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| 01:35.03 | *** join/#asterisk tengulre (~tengulre@61.185.238.166) |
| 01:36.58 | *** join/#asterisk constfilin (~cf@204.147.182.37) |
| 01:37.32 | constfilin | greetings, sorry I had to get out to get a few more details of the problem. |
| 01:37.53 | constfilin | tzanger: are you still here? |
| 01:39.42 | tzanger | constfilin: what card is this |
| 01:40.35 | constfilin | I have a Wildcard TDM400P card, and the very first port on the card works just fine, the calls are coming in to asterisk. However the calls to second, 3rd and 4th ports are not showing up in asterisk. Can this be somethign that I misconfigured in /etc/zaptel.conf or /etc/asterisk/zapata.conf or it looks more like a hardware problem? |
| 01:41.11 | Qwell | constfilin: They're all FXO? |
| 01:41.19 | *** part/#asterisk outsidefactor (~blah@203-217-79-71.dyn.iinet.net.au) |
| 01:41.31 | tzanger | constfilin: you need to make sure /etc/zaptel.conf and /etc/asterisk/zapata.conf are configured for all ports |
| 01:41.46 | tzanger | just off the top of my head it sound slike you're only configuring the first port |
| 01:44.17 | constfilin | thanks for replying. Here's my /etc/zaptel.conf: |
| 01:44.27 | constfilin | span=1,1,0,esf,b8zs |
| 01:44.27 | constfilin | #fxoks=1-24 |
| 01:44.27 | constfilin | bchan=1-23 |
| 01:44.27 | constfilin | dchan=24 |
| 01:44.27 | constfilin | fxsks=25-28 |
| 01:44.40 | Qwell | its a PRI? |
| 01:44.55 | Qwell | erm, also a pri |
| 01:45.28 | constfilin | and /etc/asterisk/zapata.conf |
| 01:45.29 | constfilin | [trunkgroups] |
| 01:45.29 | constfilin | [channels] |
| 01:45.29 | constfilin | context=pots_default |
| 01:45.29 | constfilin | switchtype=national |
| 01:45.30 | constfilin | signalling=fxo_ls |
| 01:45.31 | constfilin | rxwink=300 ; Atlas seems to use long (250ms) winks |
| 01:45.32 | constfilin | usecallerid=yes |
| 01:45.32 | pussfeller | these docs that come with the sipura 2100 suck hard |
| 01:45.34 | constfilin | hidecallerid=no |
| 01:45.36 | constfilin | callwaiting=yes |
| 01:45.39 | constfilin | usecallingpres=yes |
| 01:45.40 | constfilin | callwaitingcallerid=yes |
| 01:45.42 | constfilin | threewaycalling=yes |
| 01:45.44 | constfilin | transfer=yes |
| 01:45.45 | Qwell | ~pastebin |
| 01:45.46 | jbot | i guess pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca |
| 01:45.46 | constfilin | cancallforward=yes |
| 01:45.48 | constfilin | callreturn=yes |
| 01:45.50 | constfilin | echocancel=yes |
| 01:45.52 | constfilin | echocancelwhenbridged=yes |
| 01:45.54 | constfilin | rxgain=0.0 |
| 01:45.56 | constfilin | txgain=0.0 |
| 01:45.58 | constfilin | group=1 |
| 01:46.00 | constfilin | callgroup=1 |
| 01:46.02 | constfilin | pickupgroup=1 |
| 01:46.05 | constfilin | immediate=no |
| 01:46.06 | constfilin | immediate=no |
| 01:46.08 | constfilin | context=pri_default |
| 01:46.10 | constfilin | switchtype = national |
| 01:46.12 | constfilin | signalling = pri_cpe |
| 01:46.14 | constfilin | group = 1 |
| 01:46.16 | Qwell | oh come on |
| 01:46.16 | constfilin | channel = 1-23 |
| 01:46.18 | constfilin | immediate=mo |
| 01:46.20 | constfilin | context=tdm_default |
| 01:46.22 | constfilin | switchtype=national |
| 01:46.24 | constfilin | signalling=fxs_ks |
| 01:46.26 | constfilin | group = 2 |
| 01:46.28 | constfilin | callerid=asreceived |
| 01:46.30 | constfilin | channel = 25-28 |
| 01:46.32 | constfilin | faxdetect=both |
| 01:46.34 | constfilin | I have a PRI/TDM combination. PRI is 1023 channel, TDM 25-28 channels |
| 01:46.48 | constfilin | jbot: thanks for the tip, I a newbie in IRC chatting. |
| 01:46.48 | jbot | no worries, constfilin |
| 01:46.49 | Juggie | ooookkkay, use pastebin please |
| 01:46.50 | Qwell | ...don't EVER do that again |
| 01:48.00 | constfilin | sorry guys, will do next time. Looking at http://pastebin.ca. Meanwhile, what do you think. Is is a misconfiguration? (BTW, jsut called Digum paid support - no one is avail at this time) |
| 01:49.04 | tzanger | well you don't need a switchtype for the TDM card but that wount' be causing issue |
| 01:49.28 | tzanger | do you have call waiting on the POTS line (i.e. is the telco providing call waiting?) |
| 01:49.44 | constfilin | ok, I've got what pastebin is. Thanks! |
| 01:52.12 | constfilin | no, these are business lines, no call waiting. |
| 01:52.28 | Qwell | Do the lines work with just a regular phone? |
| 01:52.32 | Qwell | remove the obvious problems |
| 01:53.08 | *** join/#asterisk Veto (mdkuser@cpe-66-69-38-192.satx.res.rr.com) |
| 01:53.32 | Jedirl | 1023 channels in PRI? |
| 01:53.36 | Qwell | 1-23 |
| 01:53.42 | *** join/#asterisk lilo (lilo@levin-pdpc.staff.freenode) |
| 01:53.43 | Jedirl | wow |
| 01:53.44 | Jedirl | =D |
| 01:53.48 | tzanger | Jedirl: they're short conversations :-) |
| 01:54.12 | tzanger | constfilin: so if you're on a call on line 1 and a call comes in on line 2 what happens? |
| 01:54.14 | Qwell | give each call a millisecond at a time |
| 01:54.29 | Jedirl | or less |
| 01:54.29 | Jedirl | xD |
| 01:57.04 | darwin35 | cvs-head = fubard |
| 01:57.45 | darwin35 | <=== not a happy camper |
| 01:59.45 | constfilin | the line itself is ok, because if I plug it in another PBX (3COM), it rings all right |
| 01:59.49 | Qwell | fubar how? |
| 01:59.54 | Jedirl | by the way |
| 01:59.58 | Juggie | theres a patch, just apply it |
| 02:00.09 | Jedirl | anyone here thinks he's going to need the new digium DS3 card? |
| 02:00.10 | darwin35 | ? |
| 02:00.25 | Qwell | Jedirl: I plan on needing one eventually |
| 02:00.42 | Jedirl | such a qty of channels |
| 02:01.18 | Jedirl | put SS7 on such a card and you have a quite powerful switch |
| 02:01.36 | pussfeller | there something I am missing about connecting to this spa-2100? its pingable but no web interface |
| 02:02.02 | tzanger | Jedirl: we use MaxTNTs for our DS3 terminations |
| 02:02.14 | Jedirl | what is MaxTNTs? |
| 02:02.25 | tzanger | Lucent MaxTNT |
| 02:02.32 | tzanger | 672 modems per chassis |
| 02:02.42 | Jedirl | don't know what is it |
| 02:02.49 | Qwell | massive channel bank? |
| 02:02.50 | Jedirl | I'm quite new to this business |
| 02:03.37 | Jedirl | tzanger: can you separate a E3 card in "blocks of" E1s? |
| 02:03.56 | tzanger | Jedirl: that's called an M13 |
| 02:04.04 | Jedirl | am |
| 02:04.04 | tzanger | or a DS3 Mux |
| 02:04.22 | Jedirl | asterisk's new card does it? |
| 02:04.36 | tzanger | we used one to take a DS3 and split it out into its 28 DS1s and feed pairs to AS5248s before we got into the MaxTNTs |
| 02:04.39 | tzanger | Jedirl: no |
| 02:04.52 | tzanger | Jedirl: the DS3000P just takes a DS3 in and gives you 672 DS0s |
| 02:05.04 | Jedirl | oh |
| 02:05.11 | tzanger | much the same way as a TE110P takes a DS1 and gives you 24 or 32 DS0s |
| 02:05.19 | Qwell | DS0? |
| 02:05.20 | *** join/#asterisk remmo (~rem@smack.isp.net.au) |
| 02:05.26 | Qwell | sorry, yes, I do realize how dumb a question that is |
| 02:05.37 | tzanger | DS0 = 64kbps channel |
| 02:05.47 | tzanger | 24xDS0 = DS1 (for T1) |
| 02:05.52 | Jedirl | I know |
| 02:05.58 | constfilin | ok, I am back. |
| 02:05.59 | tzanger | 7xDS1 = DS2 (not really seen in the real world) |
| 02:06.02 | Qwell | So, * would see it as a 672 single channels, as a T1 is 24 single channels? |
| 02:06.03 | tzanger | 4xDS2 = DS3 |
| 02:06.23 | tzanger | Qwell: usually you run PRI over them, so you'd have 28 T1 spans |
| 02:06.34 | constfilin | basically, asterisk picks up a call on the first port, but not on other ports. The cal does come in from the TELCO |
| 02:06.36 | Qwell | tzanger: with the ds300p I mean, of course |
| 02:06.38 | tzanger | but you would typically run NFAS over it so you don't eat up so many D channels |
| 02:06.59 | tzanger | constfilin: ok, with no lines active, dial line 2 with your cell phone |
| 02:07.02 | Qwell | erm, ds3000p |
| 02:07.05 | tzanger | does asterisk answer? |
| 02:07.38 | Jedirl | what is NFAS? |
| 02:07.40 | constfilin | tzanger: diales, long rigns, asterisk is not picking up |
| 02:07.44 | Qwell | ~nfas |
| 02:07.45 | jbot | from memory, nfas is "Non-Facility Associated Signaling" FixMe: saves a D channel on PRI's orsomethingorother |
| 02:07.45 | tzanger | constfilin: I'm trying to determine if asterisk sees it ring or not |
| 02:07.51 | tzanger | hahaha |
| 02:07.54 | Qwell | nice |
| 02:07.55 | Jedirl | hehe |
| 02:07.56 | Jedirl | ok |
| 02:08.03 | constfilin | it doesn't matter if there is a call on line 1 or not. If you call line 2 it is not picked up |
| 02:08.06 | tzanger | NFAS is a way to share a D channel amongst several PRIs in order to gain timeslots |
| 02:08.07 | Qwell | it helps reduce the total number of D channels you need, when you have multiple |
| 02:08.19 | *** join/#asterisk forrestc{hm} (~forrestc@iMach.com) |
| 02:08.27 | forrestc{hm} | Hello everyone. |
| 02:08.27 | tzanger | you can (apparently, I've never seen it this high) share 28 T1s with a single D channel |
| 02:08.39 | tzanger | constfilin: do you see the line ringing in the asterisk console? |
| 02:08.41 | Qwell | I thought somebody was saying 6-7 or something |
| 02:08.52 | *** join/#asterisk Inv_arp (junya@adsl-8-230-54.mia.bellsouth.net) |
| 02:09.06 | tzanger | yes we typically do 7 T1s to a single D chan. |
| 02:09.17 | Qwell | tzanger: yeah, I think it was you saying that before |
| 02:09.17 | constfilin | tzanger: When I call line 1, I see May 19 19:02:17 VERBOSE[25198]: -- Starting simple switch on 'Zap/25-1' |
| 02:09.19 | forrestc{hm} | tzanger: do you do a backup D? |
| 02:09.44 | tzanger | forrestc{hm}: nah, it's all on the same physical wire. I can't see ever needing a backup since if the main D is down chances are the backup will be too |
| 02:09.46 | constfilin | But I can't get it to show somethign like "Starting simple switch on 'Zap/26-1" |
| 02:10.00 | tzanger | constfilin: physically swap the lines going to port 1 and 2 on the TDM card |
| 02:10.09 | forrestc{hm} | tzanger: So, really you have 1D on a DS3. |
| 02:10.11 | tzanger | and ring the original line 1 -- do you see Zap/26 ringing? |
| 02:10.19 | tzanger | forrestc{hm}: no, we have 4 |
| 02:10.28 | tzanger | forrestc{hm}: 1 for every 7 DS1s |
| 02:10.49 | tzanger | I think bkw_ claimed to have set up a system with 1 D for the entire 28 DS1s on a DS3 |
| 02:10.54 | forrestc{hm} | that makes sense... |
| 02:11.02 | Qwell | tzafrir: 2005-04-28 07:05:19 <bkw_> you gain 28 more channels on a DS3 if you use NFAS |
| 02:11.23 | Qwell | looks like he used 0 D channels. :P |
| 02:11.26 | tzanger | I don't doubt him, I just have never seen it, and would suspect that you'd have congestion problems if a ton of lines suddently changed state |
| 02:11.34 | tzanger | Qwell: well you gain 1 B for every D you knock out |
| 02:11.34 | constfilin | tzanger: how do I physically swap the lines on TDM card? |
| 02:11.43 | Qwell | tzanger: You wouldn't knock out 28 D's |
| 02:11.43 | tzanger | constfilin: unplug them out of the back and swap them |
| 02:11.47 | Qwell | probably a typo on his part. :) |
| 02:11.50 | tzanger | Qwell: well you couldn't |
| 02:11.58 | tzanger | you could in theory knock out 27 of them |
| 02:12.13 | Qwell | 7 sounds like a nice round number |
| 02:12.15 | tzanger | when you're running 10:1+ line ratios every B counts |
| 02:12.25 | tzanger | Qwell: well 7 DS1s to a DS2, it fits nicely |
| 02:12.28 | Qwell | sorry, 10:1+ ratios? |
| 02:12.36 | tzanger | Qwell: dialup internet |
| 02:12.47 | Qwell | ? |
| 02:12.56 | tzanger | no sane internet provider has a modem for every user |
| 02:13.00 | Qwell | right |
| 02:13.00 | tzanger | they run x users to 1 modem |
| 02:13.00 | constfilin | tzanger: we did that. There was this test: there is a line coming from TELCO. Wehn we first plugged this line into first port, asterisk picked it up. When we plugged that line into 2nd port, asterisk didn't pick it up |
| 02:13.17 | forrestc{hm} | 10:1 is about what we used to run. |
| 02:13.18 | tzanger | constfilin: sounds like a hardware or lack of configuration issue |
| 02:13.24 | Qwell | 10 subscribers to 1 modem? Only 1 subscriber can use a modem at once, correct? |
| 02:13.25 | tzanger | you did run ztcfg -v, right? |
| 02:13.31 | tzanger | Qwell: correct |
| 02:13.35 | Qwell | So, if more then 10% try to get on...you're pretty much screwed? |
| 02:13.41 | Qwell | or rather, they are |
| 02:13.43 | forrestc{hm} | Yes, but it never happens. |
| 02:14.03 | tzanger | forrestc{hm}: well we actually run higher now I think... when you start packing multiple DS3s the ratios can get crazy and you don't get the busies you did when running a single DS1 (I think we did 6.5 or 7:1 way back in those days) |
| 02:14.04 | forrestc{hm} | 10:1 is about right. |
| 02:14.14 | tzanger | Qwell: correct |
| 02:14.24 | constfilin | tzanger: the result of ztcfg is at http://pastebin.ca/12202 |
| 02:14.36 | forrestc{hm} | I think 8:1 was the original ratio, but 10:1 seems about right. |
| 02:14.43 | tzanger | Qwell: it's teh same thing as the telco providing your DSL does not have 4MB for every subscriber |
| 02:14.45 | forrestc{hm} | Of course... if you are running DS3's. |
| 02:14.45 | Qwell | So, each line still has to go into a modem? Thats kinda lame. |
| 02:14.51 | tzanger | it's called oversubscription and it happens in any market |
| 02:14.57 | forrestc{hm} | Qwell: No, you use a box with a whole bunch of DSP's in it. |
| 02:15.10 | Qwell | like the aforementioned maxtnt? |
| 02:15.16 | forrestc{hm} | We're still using the USR total control rack. |
| 02:15.20 | forrestc{hm} | And yah, the maxtnt. |
| 02:15.27 | tzanger | Qwell: well remember the MaxTNT has a pair of coaxial cables comign into it from the telco -- 672 calls can theoretically terminate over that connection |
| 02:15.37 | tzanger | then the chassis itself has 672 modems in it |
| 02:15.41 | forrestc{hm} | er to a tnt. |
| 02:15.56 | Qwell | and pukes |
| 02:16.22 | forrestc{hm} | We had 60 USR courier V.everythings plugged into a pair of terminal servers probably 8 years ago. |
| 02:16.23 | Qwell | So, not only would you be oversubscribed on the modems, you'd be oversubscribed on the bandwidth? |
| 02:16.27 | tzanger | Qwell: you're thinking phsyical modems with boxes and cables |
| 02:16.34 | Qwell | tzanger: oh |
| 02:16.42 | forrestc{hm} | What a freakin mess. |
| 02:16.43 | *** join/#asterisk cmantito (~gphreak@pcp09481621pcs.medfrd01.nj.comcast.net) |
| 02:16.47 | tzanger | IIRC a single modem card on a MaxTNT has like 24 modems on one circuit board |
| 02:16.50 | Qwell | so, its just a bunch of modems on a single card? |
| 02:16.54 | Qwell | I see |
| 02:17.02 | tzanger | Qwell: a bunch of DSPs on a single card, yes |
| 02:17.04 | kimo_sabe | eww, modems |
| 02:17.27 | forrestc{hm} | The USR Total Control rack had 24 per slot and could hold like 15-20 cards. |
| 02:17.29 | Qwell | how much bandwidth would you typically have with a DS3? |
| 02:17.35 | forrestc{hm} | All in about 5 U's. |
| 02:17.39 | Qwell | I mean...you'd likely oversubscribe, right? |
| 02:17.40 | forrestc{hm} | 45Mb.s |
| 02:17.42 | cmantito | need some quick advice. how much bandwidth does a SIP connection from a softphone to an asterisk server use on average? |
| 02:17.48 | forrestc{hm} | Peak. |
| 02:17.50 | Qwell | forrestc{hm}: That was the answer I expected...but not what I meant |
| 02:17.53 | tzanger | and the MaxTNTs allow you to use those DSPs as v92 modems or as voice processors (same thing really) -- instead of turning squeaks and squaks into data and back it's converting hmms and haws itno g729 :-) |
| 02:17.55 | forrestc{hm} | cmantio: depends. |
| 02:18.01 | cmantito | forrestc{hm}: on? |
| 02:18.02 | tzanger | constfilin: that looks perfect |
| 02:18.04 | Qwell | bandwidth to the internet, not with the DS3 |
| 02:18.15 | forrestc{hm} | cmantito the codec |
| 02:18.25 | Qwell | You would have a seperate connection to the net, right? |
| 02:18.28 | tzanger | constfilin: next question: what does zap show channels (from asterisk CLI) show? |
| 02:18.34 | tzanger | this really shoudln't be that difficult |
| 02:18.37 | forrestc{hm} | Qwell: it's aggregate. |
| 02:18.40 | cmantito | forrestc{hm}: I don't even have it configured yet, is there a codec you reccomend for limited bandiwdth environments? |
| 02:18.48 | Qwell | forrestc{hm}: care to explain? |
| 02:18.53 | tzanger | constfilin: actually first things first -- you loaded the wctdm driver AFTER you loaded teh T1 card driver, right? |
| 02:18.56 | forrestc{hm} | cmantito: depends on what you mean by limited bandwidth. |
| 02:19.28 | forrestc{hm} | qwell: If I give 3 people a T1's worth of bandwidth peak, and they don't use it all at the same time, then I don't need 3xT1's upstream. |
| 02:19.30 | Qwell | tzanger: Thats how it works? It literally depends on the order the stuff was modprobed? |
| 02:19.34 | cmantito | forrestc{hm}: for example, say it's a DSL connection. how could you fit the most SIP connections over your DSL line? |
| 02:19.40 | Qwell | forrestc{hm}: I understand that part |
| 02:19.53 | constfilin | tzanger: http://pastebin.ca/12203 |
| 02:20.09 | forrestc{hm} | cmantito: it also depends on what acceptable call quality is. |
| 02:20.17 | Qwell | forrestc{hm}: How much bandwidth typically would you have (to the internet, NOT with the DS3), for those 672 channels? |
| 02:20.23 | cmantito | forrestc{hm}: there isn't one. you have to be able to hear the other person, and that's it. |
| 02:20.29 | forrestc{hm} | Qwell: Oh. |
| 02:20.32 | forrestc{hm} | Qwell: now I see. |
| 02:20.35 | Qwell | :) |
| 02:20.48 | Qwell | I asked wrong...I knew I did |
| 02:20.53 | forrestc{hm} | Qwell: Let me look at the stats on a pool. |
| 02:21.53 | DaLion | i need a pool |
| 02:21.54 | DaLion | ;) |
| 02:22.26 | tzanger | Qwell: of course |
| 02:22.30 | tzanger | how else would order be determined? |
| 02:22.39 | forrestc{hm} | Qwell: looks like I've got a pool with a peak usage of about 30 modems recently, and the peak bandwidth was about 256K. |
| 02:22.56 | tzanger | constfilin: looks good, now what about the module load order |
| 02:23.04 | constfilin | tzanger: my /etc/modprobe.conf is at http://pastebin.ca/12204. Does it answer your question? |
| 02:23.07 | tzanger | when your system starts up, is wctdm loaded first or is wct1xxp loaded first |
| 02:23.33 | tzanger | constfilin: unfortunately not |
| 02:23.36 | forrestc{hm} | cmantito: g.711u takes the most bandwidth and has the best quality. It also degrades best when you start getting congested. |
| 02:23.51 | tzanger | constfilin: stop asterisk, rmmod wctdm and wct1xxp and zaptel |
| 02:24.00 | tzanger | then modprobe wct1xxp and wctdm in that order |
| 02:24.08 | tzanger | (use wcte1xxp if that's the card you have) |
| 02:24.09 | forrestc{hm} | cmantito: g.729 is awesome, takes very little bandwidth, but can be expensive, since you have to license it. |
| 02:24.21 | Qwell | forrestc{hm}: So, about 3:2? |
| 02:24.27 | cmantito | forrestc{hm}: hmm...is there a free low bandwidth one? =P |
| 02:24.30 | tzanger | forrestc{hm}: $10/transcode is not expensive |
| 02:24.33 | forrestc{hm} | cmantito: but it also degrades quickly to unacceptable. |
| 02:24.39 | tzanger | cmantito: yes, gsm is nice |
| 02:24.42 | cmantito | hm |
| 02:24.47 | tzanger | ilbc is more CPU intensive and more bandwidth than gsm |
| 02:24.53 | cmantito | and generally how much bandwidth does gsm use? |
| 02:24.54 | tzanger | lpc10 is VERY low bitrate but it sounds like shit |
| 02:25.02 | tzanger | cmantito: about 3.4kB/sec/conversation |
| 02:25.11 | cmantito | ok |
| 02:25.14 | constfilin | tzanger: also the output of lsmod is at http://pastebin.ca/12207 |
| 02:25.21 | cmantito | thanks guys, tzanger, forrestc{hm} |
| 02:25.22 | cmantito | night. |
| 02:25.23 | forrestc{hm} | Let me see if I can find a document you may want to read. |
| 02:25.26 | forrestc{hm} | before you run off. |
| 02:25.36 | cmantito | k |
| 02:25.47 | Qwell | tzanger, forrestc{hm}: Thanks, you guys have been a big help |
| 02:25.48 | tzanger | constfilin: it *looks* like wctdm was loaded second (whcih is good) but I'm not sure if the order in lsmod is always the order it actually got loaded in |
| 02:25.57 | Qwell | tzanger: it is |
| 02:26.03 | Qwell | at least, on Linux |
| 02:26.16 | tzanger | in that case, it should be working |
| 02:26.19 | Qwell | reverse order, that is |
| 02:26.21 | constfilin | tzanger: I am dialing, not answer |
| 02:26.26 | constfilin | sdfdfdf |
| 02:26.27 | constfilin | dfl;kgdkgj |
| 02:26.35 | tzanger | ztcfg won't let you define a channel that doesn't exist |
| 02:26.37 | tzanger | hahahaha |
| 02:26.58 | *** join/#asterisk constfilin (~cf@204.147.182.37) |
| 02:27.03 | forrestc{hm} | cmantito: this might not be exactly related to what you are doing, but it discusses some of the bandwidth/quality issues you may want to consider. |
| 02:27.14 | constfilin | tzanger: I am dialing the second line, there is no answer |
| 02:27.15 | cmantito | anything is appreciated. |
| 02:27.21 | tzanger | ztcfg won't let you define a channel that doesn't exist |
| 02:27.27 | tzanger | so that means that the 4 FXO modules are loaded |
| 02:27.33 | tzanger | er not loaded but installed and being detected correctly |
| 02:27.49 | forrestc{hm} | cmantito: see: http://motorola.canopywireless.com/library/index.php?page=resources&rel_dir=bb1XIfDHDFv9df5hyXGPpOx0xUZ8cLyNMuA= |
| 02:27.54 | tzanger | at this point I feel that you either have a defective card or there's some other issue I just can't think of offhand |
| 02:27.57 | tzanger | sorry |
| 02:28.06 | cmantito | thanks forrestc{hm} |
| 02:28.07 | constfilin | does it mean that it is a hardware problem? |
| 02:28.12 | forrestc{hm} | On that page click on the voip whitepaper. |
| 02:28.14 | constfilin | ah, you were first :) |
| 02:28.23 | cmantito | forrestc{hm}: k |
| 02:28.24 | tzanger | constfilin: or, as I said, a problem I am just not seeing |
| 02:28.31 | forrestc{hm} | cmantito: did the link come through ok. |
| 02:28.38 | *** join/#asterisk Corydon76-home (beige@pcp08665860pcs.500ash01.tn.comcast.net) |
| 02:28.39 | cmantito | yes |
| 02:28.48 | forrestc{hm} | ok. Sometimes the long links get mangled... you know... |
| 02:28.52 | cmantito | yeah |
| 02:28.59 | constfilin | tzanger: ok, thanks for looking into it. Hopefully, Digium support will call be back soon. |
| 02:29.07 | cmantito | while I'm asking stupid questions tonight, what does asterisk suck along the line of system resources? |
| 02:29.21 | forrestc{hm} | cmantito: it depends ;) |
| 02:29.24 | constfilin | BTW, I am going to try outgoing calls through the 2nd port, see if it goes through. |
| 02:29.39 | cmantito | forrestc{hm}: I hate that line now. :P on? |
| 02:29.44 | tzanger | cmantito: depends entirely on what you're asking it to do |
| 02:29.46 | forrestc{hm} | cmantito: the codec. |
| 02:29.51 | Qwell | Did swapping the cable from line 2 to line 1 work? |
| 02:29.53 | cmantito | hmm |
| 02:29.56 | cmantito | ok |
| 02:29.57 | cmantito | thanks |
| 02:30.00 | tzanger | cmantito: I had 1FXS and 1FXO running on a P90 (with no MMX) quite nicely |
| 02:30.00 | Qwell | port 2 to port 1 that is |
| 02:30.11 | tzanger | it worked great so long as I didn't ask it to do MOH or use the iLBC codec |
| 02:30.13 | forrestc{hm} | cmantito: it doesn't take a lot unless you are doing a lot. |
| 02:30.20 | cmantito | tzanger: no hardware for me, just softphones. |
| 02:30.22 | cmantito | forrestc{hm}: good to hear. |
| 02:30.25 | cmantito | thanks again guys |
| 02:30.29 | tzanger | people run asterisk on the crappy little ARM processors in WRT54s |
| 02:30.38 | forrestc{hm} | cmantito: g.711 doesn't need a lot of cpu, but some of the more agressive codecs like ilbc are CPU pigs. |
| 02:30.54 | cmantito | cool. |
| 02:30.55 | xeet2 | tzanger: you can run asterisk on those? |
| 02:31.13 | tzanger | xeet2: google is your friend |
| 02:31.25 | xeet2 | thats crazy |
| 02:31.27 | forrestc{hm} | That is assuming you are transcoding... but I don't want to go down that road. |
| 02:31.38 | xeet2 | ya know what would be cool is to get asterisk to run on a tivo |
| 02:31.50 | xeet2 | its linux after all |
| 02:31.56 | *** join/#asterisk Moc[Toronto] (~mochouina@142.131.66.185) |
| 02:32.02 | forrestc{hm} | You know it sounds like microsoft is thinking about something similar.... |
| 02:32.08 | forrestc{hm} | on the Xbox360 |
| 02:32.14 | forrestc{hm} | or it's successor. |
| 02:32.45 | forrestc{hm} | But I have to admit seeing the WRT54GS running * at VON was cool. |
| 02:34.05 | forrestc{hm} | Is there a reason I'm missing why * isn't logging sip debug to the messages file? |
| 02:34.08 | Sedorox | cmantito: I think its file that has it running on all his macs |
| 02:34.25 | cmantito | really. |
| 02:34.26 | cmantito | wow. |
| 02:34.34 | Sedorox | yea.. think the mini too |
| 02:34.35 | Sedorox | lol |
| 02:34.43 | forrestc{hm} | You know, I bet it would run on OSX... |
| 02:34.57 | forrestc{hm} | (not saying people haven't....) |
| 02:35.20 | forrestc{hm} | Hmmm... maybe that would be a good excuse for me to buy a mini |
| 02:35.29 | Sedorox | lol |
| 02:35.42 | forrestc{hm} | You know the mac... not the car. |
| 02:35.45 | *** join/#asterisk newmember (~newmember@S010600a0c93dce87.cg.shawcable.net) |
| 02:35.47 | Sedorox | I wanted to try it.. but I haven't had a chance to get Tiger running in PearPC yet |
| 02:35.48 | constfilin | tzanger: Outgoing calls go through the second line just fine. It is only a problem with incomign calls that I have |
| 02:35.49 | forrestc{hm} | :) |
| 02:36.34 | cmantito | my os X runs like crap. |
| 02:37.34 | tzanger | constfilin: ??! |
| 02:38.18 | forrestc{hm} | constfilm: I came in late... quickly, what probably were you having? |
| 02:38.41 | forrestc{hm} | quickly = be brief. .... not "in a hurry". |
| 02:38.47 | cmantito | anyone know how to convince OS X to boot to a console and not aqua? |
| 02:38.54 | cmantito | (off topic, sorry) |
| 02:39.23 | cmantito | heh |
| 02:39.27 | cmantito | better than me |
| 02:39.31 | kimo_sabe | cmantito: google it, but I think it's something like CMD-CTRL-S |
| 02:39.38 | cmantito | thanks kimo_sabe |
| 02:39.51 | forrestc{hm} | But again, I'm a unix (primarily FreeBSD) guy. |
| 02:40.01 | constfilin | I have a Wildcard TDM400P card. The first port on this card works all right. The 2nd,3rd and 4th ports seem to only pass outgoing calls, no incoming calls are possible through them. Asterisk just doesn't pick them up |
| 02:40.03 | forrestc{hm} | With unfortunately a bit of windows .. |
| 02:40.55 | forrestc{hm} | constfilin: that sounds fun. |
| 02:40.56 | tzanger | forrestc{hm}: he's got a T100P and TDM04B. incomign calls on the TDM04 port 1 * sees ring, but not the other ports. swapping lines 1 and 2 does not swap behaviour. outgoing calls work on all ports. |
| 02:40.56 | geesus | im half and half |
| 02:41.14 | forrestc{hm} | Bad module? |
| 02:41.19 | constfilin | we checked /etc/zaptel.conf and /etc/asterisk/zapate.conf - those seem all right |
| 02:41.22 | tzanger | forrestc{hm}: 3 bad modules? |
| 02:41.26 | geesus | linux server, linux desktop, windows\dos work machines, windows terminal services for database entry |
| 02:41.31 | forrestc{hm} | That is a good point... |
| 02:41.35 | tzanger | constfilin: try putting the line that # does see ring in port 3 or 4 and trying it |
| 02:41.58 | cmantito | well you guys have been a great help, and I'd love to stay and talk, but I have 34 xchat tabs open and I need to get rid of some. |
| 02:42.00 | cmantito | thanks agasin. |
| 02:42.03 | cmantito | again, even. |
| 02:42.40 | forrestc{hm} | from the pastebin it looks like he's also got a T1 card in there? |
| 02:42.44 | constfilin | tzanger: I have a line from TELCO, the calls comes on this line. I the line in port 1, then asterisk sees it, if it is in 2,3 or 4 it does not see it. |
| 02:42.46 | cmantito | aww heck |
| 02:42.57 | *** part/#asterisk cmantito (~gphreak@pcp09481621pcs.medfrd01.nj.comcast.net) |
| 02:43.09 | constfilin | tzanger: Or did you mean something else? |
| 02:43.24 | tzanger | constfilin: just on a lark, are all four modules green coloured or are three of them red? |
| 02:44.00 | constfilin | tzanger: one sec. |
| 02:44.41 | *** join/#asterisk santiago (~santiago@63.245.86.248) |
| 02:44.49 | forrestc{hm} | YES!!!! it's failing again. |
| 02:45.07 | forrestc{hm} | sorry.... trying to troubleshoot an intermittent is a pain. |
| 02:45.25 | Inv_arp | how does one cancel a broadvoice account? |
| 02:45.40 | tzanger | Inv_arp: stop paying them? |
| 02:45.53 | Inv_arp | tzanger: they take it automatically from cc card |
| 02:46.02 | tzanger | Inv_arp: put a stop payment on it |
| 02:46.09 | *** join/#asterisk iq|laptop (~iq@207-224-101-46.omah.qwest.net) |
| 02:46.12 | tzanger | Inv_arp: this is assuming you tried the nice way first |
| 02:46.22 | Inv_arp | tzanger: yep... |
| 02:46.26 | sudhir492 | send them an email |
| 02:46.32 | Inv_arp | i did |
| 02:46.40 | tzanger | then tell your CC company you are contesting the charges from them |
| 02:46.43 | tzanger | that will get their attention |
| 02:46.48 | ariel_ | did you call them on the phone???? |
| 02:46.49 | Inv_arp | tzafrir: np thx |
| 02:47.03 | Inv_arp | ariel_: never could get through |
| 02:47.08 | xeet2 | Inv_arp: there is a cancel option on their site |
| 02:47.40 | constfilin | tzanger: all four lights are green |
| 02:47.40 | ariel_ | it's cc calling time then. |
| 02:47.40 | xeet2 | Inv_arp: but yeah they rarely answer their phone |
| 02:47.40 | tzanger | constfilin: not lights |
| 02:47.40 | tzanger | the physical card itself |
| 02:47.40 | Inv_arp | xeet2: hmmm where? |
| 02:47.40 | tzanger | it will be a blue card |
| 02:47.40 | tzanger | and there will be four modules. are they all green or are they green and red |
| 02:47.41 | tzanger | at least one of them must be green |
| 02:47.41 | xeet2 | Inv_arp: under the billing section I believe, I just canceled 2 =) |
| 02:47.58 | Inv_arp | xeet2: ima go to iax.cc |
| 02:48.11 | tzanger | Inv_arp: they're really not that much better from my experience |
| 02:48.18 | constfilin | I can't unplug the box right now, but AFAIK they were all red |
| 02:48.26 | xeet2 | Inv_arp: what are you looking for? |
| 02:48.26 | tzanger | constfilin: they can't be all red |
| 02:48.31 | tzanger | red = FXS = you plug a phone into them |
| 02:48.37 | tzanger | gree = FXO = you plug a telco line into them |
| 02:48.46 | Inv_arp | xeet2: DID incoming basically |
| 02:48.55 | xeet2 | Inv_arp: what area? |
| 02:49.01 | constfilin | tzanger: ok, it will take longer for me to check, but I'll be back |
| 02:49.05 | *** join/#asterisk gambolputty (~gambolput@cblmdm72-240-241-108.buckeyecom.net) |
| 02:49.06 | Inv_arp | Inv_arp: north miami |
| 02:49.08 | Inv_arp | fl |
| 02:49.19 | xeet2 | mmm... existing number? |
| 02:50.00 | Inv_arp | xeet2: nah... i dont care any local number |
| 02:50.58 | xeet2 | wish I could help ya, I can't get new florida did's but I can port existing. might want to try selectfone.com |
| 02:51.00 | ariel_ | tzanger, red were FXO and Green were FXS..... |
| 02:52.03 | forrestc{hm} | I found it!!! YES!!! Off to #asterisk-dev |
| 02:52.08 | forrestc{hm} | brb |
| 02:52.15 | ariel_ | FXS — Foreign Exchange Station (Green Modules for TDM400P) |
| 02:52.15 | ariel_ | FXO — Foreign Exchange Office (Red Modules for TDM400P) |
| 02:53.28 | *** join/#asterisk starsoldier (~joey@pcp04633390pcs.gambrl01.md.comcast.net) |
| 02:54.53 | *** join/#asterisk thrash_ (~bcrochet@cpe-069-134-067-181.carolina.res.rr.com) |
| 02:56.55 | *** join/#asterisk dpb9 (~dpb@lexx.daves.net) |
| 02:57.48 | constfilin | tzanger: all 4 modules are red! |
| 02:58.24 | *** join/#asterisk shido6 (~greg@d57-87-253.home.cgocable.net) |
| 02:58.35 | ariel_ | constfilin, you are suppose to plug incoming phone lines to the red ones. Is that what you have plugged into them? |
| 02:59.06 | constfilin | ariel_: absolutely. I have 4 imcoing lines and all of htem are plugged into 4 red modules |
| 02:59.19 | *** join/#asterisk dos000 (~dos000@66.11.173.123) |
| 02:59.46 | dos000 | hi ... my head is about to blow trying to interface asterisk to this sip adapter ! |
| 02:59.50 | xeet2 | uhm, aren't red fxs modules? the ones you plug -phones- into? |
| 03:00.09 | dos000 | read sip ata adapter from azatel .. anyone familiar ? |
| 03:00.15 | xeet2 | dos000: welcome to open source |
| 03:00.29 | xeet2 | and protocols that every vendor does differently |
| 03:00.39 | ariel_ | red are fxo |
| 03:00.58 | ariel_ | look at the top lines: http://www.digium.com/index.php?menu=fxsvfxo |
| 03:01.26 | ariel_ | constfilin, what is the error your getting? |
| 03:02.03 | tzanger | constfilin: I was mistaken |
| 03:02.07 | tzanger | red = fxo green = fxs |
| 03:02.09 | tzanger | I had it backward |
| 03:02.13 | xeet2 | mmm, ok, didn't remember which one it was |
| 03:02.14 | constfilin | ariel_: |
| 03:02.15 | xeet2 | hehe |
| 03:02.20 | constfilin | ariel_: I have a Wildcard TDM400P card. The first port on this card works all right. The 2nd,3rd and 4th ports seem to only pass outgoing calls, no incoming calls |
| 03:02.20 | constfilin | +are possible through them. Asterisk just doesn't pick them up |
| 03:02.29 | dos000 | xeet2, it looks like the adapter is getting rejected because the way the registration is done. it uses md5. what should i see in the sip request fields ? i see a responce field that keeps on changing ! |
| 03:02.54 | xeet2 | dos000: try plaintext first |
| 03:03.00 | ariel_ | when you call the 2nd one what does the cli display? |
| 03:03.18 | dos000 | xeet2, i cant ... this device cannot be changed easily ! |
| 03:03.25 | constfilin | ariel_: exactly nothing, asterisk jsut doesn't see the call |
| 03:03.31 | xeet2 | dos000: why can't it? |
| 03:04.10 | constfilin | when the first line is called, asterisk displays "Starting simple switch on Zap/25-1" |
| 03:04.10 | xeet2 | Zap/25? what else is in this box? |
| 03:04.19 | constfilin | ariel_: because the first 24 channels belong to a PRI in the same box |
| 03:04.22 | xeet2 | ah |
| 03:04.23 | ariel_ | zap/25? what else do you have in the box. |
| 03:04.26 | xeet2 | hahaha |
| 03:04.36 | xeet2 | so you have a pri, and 4 pots lines |
| 03:04.39 | xeet2 | hmmm |
| 03:04.42 | constfilin | correct |
| 03:05.01 | ariel_ | what is the zapata.conf file look like? do you have them on pastebin.ca? |
| 03:05.13 | xeet2 | you have the extra molex power on the tdm card connected to power on the pc right? |
| 03:05.51 | dos000 | xeet2, i do not have access to the web interface ... its a chicken and egg ... it wont allow the web interface unless it is registerd .. and cant get registered if the settings are not changed or debugging the wire and making asterisk comply ! |
| 03:05.53 | ariel_ | if he is using fxo ports it's not needed. the power is only for ring voltage on the fxs |
| 03:06.19 | xeet2 | ariel: really? digium said its needed no matter what, something about extra return for incoming ringing voltage |
| 03:06.31 | constfilin | the result of ztcfg is at http://pastebin.ca/12202 |
| 03:06.47 | xeet2 | constfilin: do you have that extra power connector plugged in? |
| 03:06.50 | ariel_ | constfilin, the is only part of what I want to see. |
| 03:06.53 | constfilin | the result of "zap show channels" is at http://pastebin.ca/12203 |
| 03:07.08 | ariel_ | xeet2, they say that just to make sure everyone plugs it in. |
| 03:07.23 | constfilin | my /etc/modprobe.conf is at http://pastebin.ca/12204 |
| 03:07.32 | xeet2 | ariel: it fixed a problem I was having with one of those boards with 4 fxs modules on it |
| 03:08.01 | xeet2 | er, modules that provide fxs signalling, the red ones |
| 03:08.10 | syle | [22:04] <constfilin> when the first line is called, asterisk displays "Starting simple switch on Zap/25-1" |
| 03:08.19 | syle | isn;t that what it is suppose to do |
| 03:08.23 | syle | he has all fxs |
| 03:08.39 | ariel_ | xeet2, wow I have 2 small via boxes that don't have the plugs and the red boards for pots lines and it works..... |
| 03:08.41 | constfilin | xeet2: not sure, doesn't look so |
| 03:09.09 | xeet2 | ariel: it was an intermittent problem of not seeing incoming calls |
| 03:09.12 | ariel_ | syle, no |
| 03:09.27 | *** part/#asterisk darwin35 (~darwin35@24.3.226.147) |
| 03:09.43 | ariel_ | syle, sorry it says that for both. |
| 03:09.46 | constfilin | xeet2: I have never had any luck calling line 2 3 or 4, it is not intermittent |
| 03:10.08 | xeet2 | constfilin: I dunno, if you can't find anything with the config, for the hell of it try connecting that to power (power down first of course) |
| 03:10.15 | ariel_ | constfilin, is there any way to see your zapata.conf and zaptel.conf files? |
| 03:10.21 | xeet2 | but don't until you've checked out all your configs |
| 03:10.26 | shido6 | . |
| 03:10.35 | shido6 | pastebin.ca |
| 03:10.49 | constfilin | the outgoing calls seem to go throut ports 2 3 and 4 all right, except that when I hanup the called phone, asterisk doesn't get the HANGUP signal. I have to hangup the calling phone manually |
| 03:11.19 | xeet2 | constfilin: that sounds familiar |
| 03:12.05 | ariel_ | constfilin, also do a cat /proc/interrupts |
| 03:12.29 | constfilin | xeet2: the box is down now. My sysadmin is getting it up. Will be a few minutes |
| 03:12.52 | xeet2 | constfilin: have him plug in the power |
| 03:12.57 | xeet2 | constfilin: if its already powered down |
| 03:13.47 | syle | i call them tech rats unless they can compile their own kernels :) |
| 03:14.01 | kimo_sabe | syle: squeek! |
| 03:15.45 | ariel_ | syle, what do you call them if they can compile there own kernels? |
| 03:15.59 | syle | sys admin |
| 03:16.43 | constfilin | xeet2: yes, additinal power supply is going to the TDM card |
| 03:16.50 | ariel_ | syle, I think it just means they can read and follow directions. |
| 03:16.59 | dos000 | how do i enable debugs of sip msg in asterisk ? |
| 03:17.08 | ariel_ | sip debug |
| 03:17.13 | syle | it means they took the time to learn their job |
| 03:17.19 | syle | and like it possibly |
| 03:17.50 | *** join/#asterisk morris (~turntabli@ACD54FBF.ipt.aol.com) |
| 03:18.03 | morris | *yawn* |
| 03:18.30 | syle | i;ve had colocations trying to explain to them how to do something on my server, and they ask me what is root? etc |
| 03:18.34 | syle | now that is frusterating |
| 03:19.21 | Sedorox | damn |
| 03:22.28 | ariel_ | syle, root hummm not su ??? ah I see well are you running it as root! |
| 03:24.07 | ariel_ | well it's been fun. Time to go to bed. See you all in the morning (maybe)...... |
| 03:24.58 | starsoldier | this doesn't exactly relate to asterisk, but how do i place a call on hold with a phone that doesn't have a hold button (its a 900mhz cordless plugged into a spa-1001) |
| 03:26.41 | ariel_ | starsoldier, you don't but you can flash and put it to parking. Good night. |
| 03:27.31 | starsoldier | ah, thanks |
| 03:27.35 | starsoldier | and goodnight :P |
| 03:28.05 | *** join/#asterisk justnulling2 (justnullin@ool-18bab443.dyn.optonline.net) |
| 03:29.41 | constfilin | whoa, after the server power cycle the ports began working all right. |
| 03:29.47 | syle | fun to put someone on hold with flash button |
| 03:29.47 | constfilin | can't believe it |
| 03:29.54 | syle | gives them music lol |
| 03:35.35 | *** join/#asterisk Kumbang (~ecvs@167.205.24.4) |
| 03:36.32 | Kumbang | guys, i got problem with txfax |
| 03:36.52 | Kumbang | Start tx page |
| 03:36.53 | Kumbang | Start tx page |
| 03:38.07 | *** join/#asterisk IsMe (~me@218.111.224.205) |
| 03:39.07 | Kumbang | that line always came up twice |
| 03:39.07 | Kumbang | and then the channel hangup |
| 03:40.30 | IsMe | hi any done CID/GSM SMS callback before? |
| 03:43.05 | *** join/#asterisk CoaxD (coax@shell1.cornernet.com) |
| 03:45.16 | *** join/#asterisk newmedian (~crowlther@Quebec-HSE-ppp230300.qc.sympatico.ca) |
| 03:45.50 | shido6 | what did u think , newmedian |
| 03:46.08 | newmedian | Finally, something we can be proud of. |
| 03:46.27 | newmedian | Excellent. Definitely worth seeing. |
| 03:47.48 | newmedian | I now forgive him for the two previous films. ;) This is an excellent book-end and gives meaningful closure. |
| 03:48.05 | Nivex | good to hear. |
| 03:48.16 | Nivex | I'm going to wait for the crowds to dissipate a bit then go see it. |
| 03:48.35 | newmedian | Was not a full house this evening, so everyone may have the same idea. |
| 03:49.30 | newmedian | Saw it in analog. Going in a few days and will see it in DLP. I hear (and suspect) the DLP will be well worth it. |
| 03:50.58 | kimo_sabe | newmedian: I saw EPII in DLP, it was nice. No twitching |
| 03:51.29 | Juggie | i have a DLP projector |
| 03:51.59 | Juggie | they keep playing Lord of the rings, the two towers on HD |
| 03:52.04 | Juggie | in 1900x1080i |
| 03:52.07 | Juggie | it looks so awesome |
| 03:52.13 | kimo_sabe | Juggie: where?!?!?! |
| 03:52.25 | Juggie | on a high def channel i have on cable ;) |
| 03:52.33 | Juggie | TMNHD |
| 03:52.38 | kimo_sabe | Juggie: sweet |
| 03:52.52 | Juggie | my projector only does 1024x768 |
| 03:52.55 | Juggie | but its 2300 lumens |
| 03:52.58 | Juggie | so it loooks nice |
| 03:53.31 | Juggie | kimo_sabe, http://www.themovienetwork.ca/programming/hd.php |
| 03:53.32 | newmedian | I have an Audio Authority 9A65. I do XBOX in High Def |
| 03:54.15 | Juggie | i have hi-def cable |
| 03:54.18 | Juggie | which is nice |
| 03:54.22 | Juggie | about 30 channels in HD |
| 03:54.27 | kimo_sabe | Juggie: I don't think my 20" TV is up to it ;) |
| 03:54.36 | newmedian | TV=evil. ;) sucks brain. braaainnsss |
| 03:55.16 | newmedian | What's nice about the 9A65 is it allows me to send output to a computer monitor. Computer monitors are cheap, even big ass ones. :) |
| 03:55.22 | shido6 | WHOAAAAAAAAAAA |
| 03:55.26 | shido6 | who's got the xbox? |
| 03:55.37 | shido6 | lets chat |
| 03:56.12 | Juggie | kimo_sabe, i get TMNOD too |
| 03:56.18 | Juggie | so like 25-30 movies on demand each month |
| 03:56.21 | Juggie | (for free) |
| 03:58.03 | forrestc{hm} | Well, I finally tracked down the bug I've been trying to squash for quite a while. |
| 03:58.15 | forrestc{hm} | Now just to see if my fix fixed it. |
| 03:58.58 | *** join/#asterisk sandnigg0r (~niggerplz@66-55-197-254.gwi.net) |
| 04:00.45 | *** join/#asterisk docelmo (~me@116-39.202-68.tampabay.res.rr.com) |
| 04:04.09 | Juggie | kimo_sabe, http://www.themovienetwork.ca/hdtv/tmnhd.pdf |
| 04:04.20 | Juggie | 24hrs allways in HD |
| 04:04.21 | Juggie | rarr |
| 04:04.36 | docelmo | Are there any issues with the variable DIALSTATUS? |
| 04:05.58 | newmedian | Any comments on this article? http://www.sfgate.com/cgi-bin/article.cgi?file=/news/archive/2005/05/19/national/w080235D40.DTL&type=printable |
| 04:06.45 | *** join/#asterisk bewest (~bewest@h48.204.255.206.cable.srcy.cablelynx.com) |
| 04:09.44 | Legend | any suggestions for a pairgain device? i need to carry two POTS lines over one pair for about 100 feet |
| 04:09.53 | Legend | need something under $500ish |
| 04:10.08 | Qwell | 100 feet isn't much for two pairs, is it? |
| 04:10.17 | Qwell | hell, cat5 is rated at what, 100 meters? |
| 04:10.44 | Legend | Qwell: errr, i only have one pair run, and for various reasons, i can't run any additional wiring |
| 04:10.56 | Qwell | 1 pair, or 1 cable? |
| 04:11.01 | Legend | need to carry two POTS lines over on pair, i know ive seen stuff that will do it, but i can't seem to find anything |
| 04:11.04 | Legend | one _pair_ |
| 04:11.06 | Qwell | err |
| 04:11.51 | Qwell | running an additional pair wouldn't be much hassle if one is already there... |
| 04:11.58 | Qwell | but I conceed, you have your reasons |
| 04:12.30 | Legend | if you must know, its a condo building, conduit is blocked ;-) |
| 04:12.38 | Legend | http://www.charlesindustries.com/main/te_dualn.html |
| 04:12.48 | Legend | something like this, but in the form factor of a small modem |
| 04:12.56 | Qwell | why not pay to have it done? |
| 04:13.09 | Qwell | surely the hoa will let a professional do it |
| 04:13.13 | Qwell | and it'd cost less then $500 |
| 04:13.28 | Legend | the professionals are the ones who said the conduit is blocked |
| 04:13.37 | Qwell | So tell them to unblock it. :p |
| 04:13.42 | *** join/#asterisk {sean} (~sean@adsl-69-214-130-169.dsl.lgtpmi.ameritech.net) |
| 04:13.47 | Legend | doesn't work quite like that ;-D |
| 04:13.48 | Qwell | I've seen wiring put through 2' of cement in a hotel |
| 04:14.10 | {sean} | has anybody ever seen a really cheesey narrated video that explains how packets/networks/firewalls work? |
| 04:14.10 | Legend | and? |
| 04:14.30 | Qwell | Legend: cut the line |
| 04:14.34 | Qwell | They'll have to unblock it to repair it |
| 04:14.40 | Qwell | and it'll STILL cost less then $500 |
| 04:14.50 | Legend | Qwell: the building is concrete |
| 04:14.53 | Legend | the conduit is in concrete |
| 04:14.55 | Qwell | So? |
| 04:15.03 | Qwell | as I said, I've seen wiring put through 2' of cement |
| 04:15.06 | newmedian | Use your coax for cable tv drop and use it for voice instead. ;) |
| 04:15.08 | Qwell | concreate, whatever |
| 04:15.14 | Legend | uhm |
| 04:15.17 | Legend | nevermind |
| 04:15.23 | Qwell | fire shield in a hotel ceiling |
| 04:15.30 | Qwell | or whatever the hell they're called |
| 04:15.40 | Legend | firewalls |
| 04:15.58 | newmedian | Do an exterior wiring run outside the building to the next floor down that is unblocked. |
| 04:16.06 | Legend | heheheh |
| 04:16.15 | Legend | none of those options will work, i've been down those roads |
| 04:16.30 | Juggie | why do you need two phone lines |
| 04:16.30 | Legend | just casually asking if anyone has used or heard of a simple device to do it ;-) |
| 04:16.32 | *** join/#asterisk kb1_kanobe (~krisbouti@d154-20-43-155.bchsia.telus.net) |
| 04:16.48 | newmedian | point-to-point laser link. Secure Wifi A channel. |
| 04:17.14 | Flav | mux and demux :) |
| 04:17.25 | bewest | does anyone use mplayer to play mp3s on *? |
| 04:17.30 | Legend | Flav: thats the idea |
| 04:17.48 | PTG123 | Legend, what are you trying to do? |
| 04:18.02 | Juggie | PTG123, thats what i asked |
| 04:18.05 | Juggie | why do you need a second line |
| 04:18.10 | Juggie | seems one pots line should be plenty |
| 04:18.14 | PTG123 | i just woke up in t his conversation |
| 04:18.31 | newmedian | 640k should be enough... |
| 04:19.00 | Legend | PTG123: scroll up |
| 04:19.03 | Flav | anyone got a recommendation for cheap(ish) voip service with an 800 number DID? |
| 04:19.12 | Legend | i need to carry two ports lines over a single copper pair for about 100 feet |
| 04:19.20 | kb1_kanobe | newmedian: 640k, eh? That's what the designers of the XT said... |
| 04:19.53 | PTG123 | legend: convert to voip? run the two coper as 10bt ethernet |
| 04:20.01 | tessier | Does asterisk use a script somewhere to email voicemails? I would like to use the same script to email recorded calls... |
| 04:20.16 | Flav | newmedian: they still show up on ebay |
| 04:20.22 | PTG123 | tessier, its built into the vm module, not a script |
| 04:20.34 | Legend | PTG123: ethernet needs two pairs, not one, and i need small simple equipment, because the room is not ACed, and physical security is a potential problem |
| 04:20.36 | newmedian | I'm just sorry I got rid of my LA34 Decwriter. :( |
| 04:20.47 | kimo_sabe | newmedian: there's a company that makes replicas |
| 04:20.48 | Legend | looking for something about the size of an ATA or an external modem |
| 04:20.49 | PTG123 | Legend, wrong :) |
| 04:20.54 | tessier | PTG123: Bummer. Ok. |
| 04:21.00 | PTG123 | Legend, 10bt uses 2 wires, 100bt uses 4 |
| 04:21.12 | Legend | PTG123: uhm |
| 04:21.14 | PTG123 | Legend, so if you convert it to 10bt, you can route it over the wires |
| 04:21.23 | PTG123 | 10bt uses one send and one receive wire |
| 04:21.29 | PTG123 | 100bt uses 2 send and 2 receive wires |
| 04:21.46 | Legend | PTG123: so in a 10bt system, where is the ground? |
| 04:21.49 | Juggie | Legend, why do you need two pots. |
| 04:21.51 | newmedian | Yeah those DEC terminals were monsters, but lovable. They made the most precious crunchcrunch noise as you did a keypress. And the head would move to the right, then move back again. Mmm. |
| 04:21.55 | doolph | whre can i download a free iax phone |
| 04:22.02 | PTG123 | Legend, no ground in ethernet |
| 04:22.09 | Legend | Juggie: because i do, what kind of question is that? |
| 04:22.29 | Juggie | well.... you are in a voice over ip channel here |
| 04:22.31 | Legend | PTG123: heh, so the TX and RX are in reference to what? |
| 04:22.36 | Juggie | is it possible you could replace a pots line with voip :) |
| 04:22.48 | Legend | Juggie: thats not an option |
| 04:22.53 | PTG123 | Legend, one sec let me find you a wiring diagram |
| 04:23.04 | PTG123 | I only know this b/c i had to figure it out for POE |
| 04:23.21 | Legend | i'd love to see this |
| 04:23.26 | Qwell | interesting, I always thought 10bt needed 2 pairs |
| 04:23.42 | PTG123 | I KNOW 100bt is 2 send and 2 receive :) |
| 04:23.54 | PTG123 | one sec i am seeing if i can find it on google |
| 04:24.00 | PTG123 | 10bt i am 75% positive on |
| 04:24.08 | Qwell | I never really knew...I always just assumed i guess |
| 04:24.15 | Legend | Qwell: 10bt is two pairs as well, PTG123 just doesn't know that yet |
| 04:24.27 | PTG123 | in POE there is 2 send, 2 receive, 2 positive and 2 negative power lines |
| 04:24.29 | Flav | The IEEE Specification for Ethernet 10BaseT requires that two twisted pairs be used |
| 04:24.54 | PTG123 | well thinnet is defintely 2 wires, so thats an option, but one sec |
| 04:25.27 | Juggie | there are those phone line networking gear you can get |
| 04:25.28 | Legend | RX+, RX-, TX+, TX- |
| 04:25.39 | Flav | i don't think coax is going to be any more of an option for Legend than running another pair is |
| 04:25.39 | PTG123 | you could be right, but you could always use thinnet |
| 04:25.44 | Legend | Juggie: no, voip is not an option, period |
| 04:25.49 | PTG123 | i think 10bt |
| 04:25.55 | Qwell | meh, just do what whoever said, and just run a cable outside. :p |
| 04:25.56 | PTG123 | is RX/TX+ and RX/TX- |
| 04:26.02 | Juggie | Legend, are the lines for voice? |
| 04:26.10 | Legend | Juggie: yes |
| 04:30.12 | PTG123 | just use thinnet |
| 04:30.15 | PTG123 | i am too lazy to search any more :) |
| 04:30.30 | Legend | PTG123: but that is ethernet, i need pots |
| 04:30.38 | PTG123 | convert to voip |
| 04:30.51 | *** join/#asterisk Pete_Largo (~Pete_Larg@adsl-65-71-225-121.dsl.rcsntx.swbell.net) |
| 04:30.52 | Legend | [00:25] <Legend> Juggie: no, voip is not an option, period |
| 04:31.00 | Legend | and that assumes i have coax, as well, which i don't |
| 04:31.08 | PTG123 | well what did you want us to do, change the laws of physics? :) |
| 04:31.20 | PTG123 | nah you wouldn't need coax, on a short distance |
| 04:31.22 | newmedian | it might help if you stamped your foot? ;) |
| 04:31.26 | Legend | PTG123: no, pairgain devices exist, i've seen them used in the past |
| 04:31.38 | Legend | was just asked for suggestions on a model or brand that someone might have used before |
| 04:31.42 | PTG123 | yah but the expense of them, it would be cheaper to use voip |
| 04:31.50 | Legend | ok |
| 04:31.50 | Legend | shit |
| 04:31.55 | PTG123 | just use the hole and put a laser beam through |
| 04:31.56 | Legend | when i say VOIP is _not an option_ |
| 04:32.00 | Legend | i mean, its _not an option_ |
| 04:32.03 | Juggie | Legend, i was just thinking |
| 04:32.10 | Juggie | two pairs of copper is enough for T1 |
| 04:32.16 | PTG123 | why not use those $200 embedded pcs on either end :) |
| 04:32.24 | Juggie | you could bring 24 lines to each side :) |
| 04:32.31 | Legend | erm, with $500 T1 cards? |
| 04:32.32 | Flav | Legend: fwiw, you are asking a POTS/switched question on a voip/packet channel :) |
| 04:32.43 | Qwell | Flav: yeah, really |
| 04:32.46 | Juggie | Legend, there is no cheap soliution |
| 04:32.52 | Pete_Largo | what's the question? I missed it, I just joined. |
| 04:33.00 | Qwell | Pete_Largo: don't even bother... |
| 04:33.06 | Pete_Largo | lol, ok then |
| 04:33.09 | Legend | Flav: fwiw, people here are in to telephony, it was a simple off the cuff question |
| 04:33.24 | newmedian | I think we were trying to be helpful |
| 04:33.29 | Legend | Flav: everyone seemed to jump on me with solutions to a different question |
| 04:33.41 | Legend | Flav: didn't mean to create a mess |
| 04:33.44 | PTG123 | Flavor Flav |
| 04:33.47 | Flav | Legend: not trying to criticize, just trying to point out that on such a channel, you're going to get the "if all you have is a hammer (voip), everything looks like a nail" situation |
| 04:33.48 | *** join/#asterisk xbmodder_lappy (~xbmodder@dsl092-015-136.sfo1.dsl.speakeasy.net) |
| 04:33.48 | Qwell | flava |
| 04:33.54 | PTG123 | whateva |
| 04:33.54 | PTG123 | :) |
| 04:33.56 | Qwell | ;] |
| 04:34.15 | Legend | Flav: and btw, * does TDM as well ;-) |
| 04:34.21 | Legend | so this isn't purely a voip channe |
| 04:34.28 | xbmodder_lappy | Hey y'all |
| 04:34.40 | Flav | Legend: heh, true :) |
| 04:35.04 | xbmodder_lappy | WIll packet8 SIP work with * |
| 04:35.11 | xbmodder_lappy | I don't want to use a adapter |
| 04:35.22 | kimo_sabe | xbmodder_lappy: it's agains't the TOS I think |
| 04:35.23 | Flav | xbmodder_lappy: the voip says packet8 only supports their locked-to-packet8 ata |
| 04:35.24 | Qwell | xbmodder_lappy: its against their terms |
| 04:35.25 | Qwell | heh |
| 04:35.31 | Qwell | oh wait, packet8 |
| 04:35.37 | Qwell | nevermind, I instantly thought Vonage. |
| 04:35.48 | Flav | s/the voip/the voip-info.org wiki/ |
| 04:36.04 | Qwell | xbmodder_lappy: its best to call them |
| 04:36.14 | Qwell | vonage does not allow it. packet8 may |
| 04:37.11 | PTG123 | xbmodder_lappy, why usethem when their are plenty of voip companies dedicated to asterisk |
| 04:37.17 | xbmodder_lappy | packet8 xdoes not allow it. |
| 04:37.23 | Qwell | xbmodder_lappy: There you go then |
| 04:37.29 | xbmodder_lappy | PTG123: name some other than broadvoice |
| 04:37.47 | xbmodder_lappy | that are good. |
| 04:37.59 | PTG123 | xbmodder_lappy, what kind of package you looking for? |
| 04:38.44 | xbmodder_lappy | PTG123: infinite calling to canada+USA |
| 04:38.51 | xbmodder_lappy | under 25 bucks |
| 04:39.01 | blitzrage | xbmodder_lappy: mixnetworks.com |
| 04:39.35 | xbmodder_lappy | and number porting |
| 04:39.37 | Flav | what's the cheapest voip provider that would let me have an 800 number at home? |
| 04:39.49 | PTG123 | try teliax |
| 04:39.51 | PTG123 | they have that i believe |
| 04:40.04 | Qwell | nufone has 8xx too, if teliax doesn't |
| 04:40.31 | xbmodder_lappy | does mix lock the devices? |
| 04:40.35 | xbmodder_lappy | mixnet |
| 04:40.39 | PTG123 | yah teliax has em |
| 04:40.48 | PTG123 | cheapest right now is probably livevoip |
| 04:40.53 | Flav | Qwell: yeah, i couldn't tell from their site how it would be priced, though - is it an additional one-time charge (the $25 vanity) alone, or is a 800# DID cost something per month? |
| 04:40.56 | PTG123 | that may change soon enough |
| 04:40.56 | PTG123 | :) |
| 04:41.04 | Qwell | 2c/min incoming |
| 04:41.08 | Qwell | on top of the 2c/min outgoing |
| 04:41.15 | PTG123 | usually dids have a couple bucks a month ch arge |
| 04:41.19 | PTG123 | and a per minute fee |
| 04:42.36 | xbmodder_lappy | PTG123: another? |
| 04:43.08 | PTG123 | another what? :) |
| 04:43.25 | PTG123 | xbmodder_lappy, how many minutes a month could youpossibly do? |
| 04:43.50 | xbmodder_lappy | 1000000 |
| 04:43.54 | blitzrage | xbmodder_lappy: what devices? |
| 04:43.57 | Qwell | Then you need a per minute providder |
| 04:43.59 | Qwell | period |
| 04:44.13 | PTG123 | xbmodder_lappy, you do more then 1500 on any of those providers, they will give you the boot |
| 04:44.36 | *** join/#asterisk scardinal (~supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 04:45.05 | xbmodder_lappy | more like 1000 |
| 04:45.17 | Qwell | Then a per minute provider would be cheaper. |
| 04:45.28 | Qwell | funny how that works out |
| 04:45.43 | PTG123 | yah exactly |
| 04:45.46 | PTG123 | per minute would be cheaper |
| 04:45.48 | Qwell | on a $29.99 plan, you break even at 1499.5 minutes |
| 04:46.05 | PTG123 | qwell: with livevoip, it would take 2200 minutes or some shit :) |
| 04:46.08 | Qwell | they shut you off at 1500 though :D |
| 04:46.13 | Qwell | PTG123: well, still |
| 04:46.25 | Qwell | most I've seen are 2c/min |
| 04:46.52 | xbmodder_lappy | see the problem is here: I do over 1700 minutes on my cell phones |
| 04:46.59 | xbmodder_lappy | what will my home phone do? |
| 04:47.07 | xbmodder_lappy | A minimum bill price is around $20.50 a month. Packet8 investiages accounts with a useage over 3000-4000 min. |
| 04:47.10 | PTG123 | its unlikely you would exceed 2k |
| 04:51.08 | *** join/#asterisk loud (~ariel@omfg.wtf.no) |
| 04:51.50 | *** join/#asterisk joaovianna (naturalvoi@node-40247a6a.ewr.onnet.us.uu.net) |
| 04:52.50 | xbmodder_lappy | is it pretty easy? |
| 04:53.05 | PTG123 | is what pretty easy |
| 04:53.32 | joaovianna | Anyone has documentation about 911 for VOIP ??? |
| 04:54.00 | Juggie | i dont think there is a 100% solid solution |
| 04:54.35 | joaovianna | Thanks Juggie, do you think the FCC will give us the solution ? |
| 04:54.42 | Juggie | no |
| 04:54.46 | Juggie | but theres no easy answer |
| 04:54.51 | xbmodder_lappy | who cares |
| 04:55.02 | PTG123 | your wholesaler has to implement the solution |
| 04:55.03 | loud | fcc, 'just do it'. |
| 04:55.06 | PTG123 | L3 already has one |
| 04:55.08 | joaovianna | PSAP's dont work !!! |
| 04:55.10 | Juggie | personally, i thought myself it should be the responsibility of the end device |
| 04:55.12 | PTG123 | so it needs to be suppoirted down the line |
| 04:55.31 | Juggie | so a phone should make me put in information before i can use it |
| 04:55.43 | joaovianna | Juggie, please answer my question... When we buy some DID's, it have to came with 911 or we have to provide ? |
| 04:55.55 | Juggie | joaovianna, i dont live in the USA |
| 04:55.59 | Juggie | therefore, i do not care |
| 04:56.04 | PTG123 | joaovianna, technically the person selling to the end user will have to provide it |
| 04:56.17 | Juggie | you have to provide it, or a link to regular 911 |
| 04:56.20 | joaovianna | Juggie, FCC change the rules today. |
| 04:56.26 | Juggie | joaovianna, i live in danada |
| 04:56.28 | Juggie | *canada |
| 04:56.30 | Juggie | no FCC here |
| 04:56.35 | joaovianna | OK |
| 04:56.48 | xbmodder_lappy | i am going to buy service from voipex |
| 04:57.11 | xbmodder_lappy | I want a ATA because i don't want to rewire the house. |
| 04:57.17 | PTG123 | xbmodder_lappy, never heard of them |
| 04:57.26 | PTG123 | xbmodder_lappy, you may want to rethink that :) |
| 04:57.26 | xbmodder_lappy | If I get it |
| 04:57.41 | xbmodder_lappy | Can I still want to use my own device? |
| 04:57.46 | Juggie | newmedian, i think a solid solution is everytime your ip changes (registered with your provider) force the user to login and update contact information |
| 04:57.55 | Juggie | then when you dial 911 , you know where to route the call |
| 04:57.58 | newmedian | Juggie: ouchie |
| 04:58.03 | PTG123 | xbmodder_lappy, whats their web address? |
| 04:58.13 | xbmodder_lappy | https://www.voipexinc.com/signup.php |
| 04:58.21 | Juggie | i mean how else do you implement it, you cant get solid information from ARPA |
| 04:58.24 | newmedian | Juggie I know people whose IP address changes every few hours on ADSL. :| |
| 04:58.26 | Juggie | you can only know who owns the IP |
| 04:58.47 | Juggie | newmedian, then perhaps it could be based on MAC |
| 04:58.55 | joaovianna | Juggie, but the VOIP provider can provide the address right ? |
| 04:59.21 | PTG123 | xbmodder_lappy, looks fly by n ight to me :) |
| 04:59.52 | xbmodder_lappy | ? |
| 05:00.02 | newmedian | joaovianna Unless you tell the VOIP provider what your address is, they don't know it. And if you go down the street with your ATA or call from another city with your Soft Phone, unless you've manually updated your location.... they won't know it. |
| 05:00.09 | Pete_Largo | has anyone seen the actual order? I've only seen the press release |
| 05:00.23 | *** join/#asterisk jeremywhiting (~jeremy@70-56-99-134.slkc.qwest.net) |
| 05:00.38 | newmedian | or if you call through a VPN.. or other secure tunnel. You could be anywhere in the world. |
| 05:00.51 | Juggie | newmedian, thats why i said the only 100% solution is to force the user to provide that information |
| 05:01.04 | PTG123 | order won't be public knowledge for a few months |
| 05:01.12 | PTG123 | up to now its just speculation |
| 05:01.12 | joaovianna | newmedian: From the voip provider point of view we should force the customer to register their address and route the address to a 911 database... |
| 05:01.32 | newmedian | Juggie: I'm not disagreeing with you, but that's going to be yucky. I suppose if their IP changes then the next call they make you could give them an auto recording saying they need to log in and update or their E911 won't work. |
| 05:01.35 | kimo_sabe | Juggie: that's still not 100%. E911 just has to accept the reality that you can't have perfect info all the time |
| 05:01.37 | PTG123 | the problem is the 911 call centers sometimes can't even be dialed with a 10 digit # |
| 05:02.12 | joaovianna | newmedian: Using their phone # as key is much better then MAC address... |
| 05:02.25 | PTG123 | you can't get someones mac address |
| 05:02.43 | newmedian | Juggie mentioned MAC |
| 05:02.54 | Juggie | i mentioned mac as a way to tell when a device moved networks |
| 05:03.03 | Juggie | to force a user to re-enter location information |
| 05:03.42 | newmedian | So if IP or MAC changes, force a login. But IP can change every few hours. If their IP changed and the next call they want to make is 911, then ... |
| 05:03.53 | joaovianna | FCC: Now every voip company is require to have 911 to their customers... |
| 05:04.12 | Juggie | newmedian, thats why use mac as well, and dont block any calls to 911 |
| 05:04.21 | joaovianna | http://www.fcc.gov/realaudio/mt051905.ram |
| 05:04.28 | Juggie | just like when you have a phone line in a house which doesnt have phone service |
| 05:04.30 | Juggie | 911 isnt blocked |
| 05:04.40 | PTG123 | only the internal switch knows a devices mac, you can't get it over the internet |
| 05:04.57 | Juggie | its not the device mac i'm talking about |
| 05:04.59 | newmedian | Time for firmware upgrades... |
| 05:05.31 | Juggie | i'm talking in the situation where adsl ip changes often, you could use the mac of the gateway device, be it the adsl modem, to know that even tho the ip changed, the phone was on the same network |
| 05:05.45 | newmedian | So better to be a DID/PSTN->VOIP provider than a Termination services provider at this point in time. |
| 05:06.08 | joaovianna | But I can have a aplication in * (AGI) that trigger the address from my customer... |
| 05:06.19 | Juggie | there is no way to get address information from an ip |
| 05:06.24 | Juggie | that is accurate |
| 05:06.25 | newmedian | Juggie: as long as you were able to read that MAC and knew definitively that it was the last HOP before the ISP |
| 05:06.42 | Juggie | new, the voip device can tell the mac easily |
| 05:06.44 | Juggie | and can report it |
| 05:07.02 | newmedian | It can report the next upstream hop, not necessarily the last hop off the network |
| 05:07.16 | newmedian | well, arping, yes |
| 05:07.29 | Juggie | its going to report the mac of the router, adsl modem (if its a softphone) etc |
| 05:08.04 | joaovianna | I think you don't need to look for the MAC... Why ? If ATA register in my * for a particular phone # or account #, I can retrieve their address in my database... I'm a wrong ? |
| 05:08.06 | newmedian | Damned needless stifling of technology. |
| 05:08.09 | tengulre | anyone know install webmin for asterisk? |
| 05:08.24 | Juggie | joaovianna, because thr ATA can be anywhere not necesairly at the billing address |
| 05:08.31 | newmedian | Anywhere in the world. |
| 05:08.40 | Juggie | and there inlies the problem |
| 05:09.19 | joaovianna | Juggie, yes... But he have to update their information (address) when he move. |
| 05:09.47 | Juggie | joaovianna, but they may take the device on vacation to a relatives |
| 05:09.48 | newmedian | joa person could be using a WiFi hotspot. Could go to a hotel. Be using a Wifi on a plane. |
| 05:09.55 | Flav | cell phones sending GPS coordinates with the 911 call seem like a nice option |
| 05:09.57 | Juggie | so that people can call their local number to get them |
| 05:10.23 | Juggie | a voip device is not guaranteed to be at the billing location and that is why its an issue |
| 05:10.28 | Juggie | if it was it would be easy |
| 05:10.31 | joaovianna | Juggie, well in this case if not our fault... We can provide acurate information if they update it. |
| 05:10.44 | newmedian | They could be using a Soft Phone for a Pocket PC on a GPRS/EDGE connection anywhere in the world, constantly changing cell sites, even country boundaries. |
| 05:11.05 | Juggie | joaovianna, hence why i said, implement a system to tell to the best of your ability when a device/phone changes networks |
| 05:11.14 | Juggie | and when this occurs, force the user to update address information |
| 05:12.10 | joaovianna | Juggie. You right... But by law, now we have to offer the possibility to route their call to 911. |
| 05:12.25 | Juggie | joaovianna, exactallty.... |
| 05:12.26 | newmedian | Well, you do, for now. And maybe us later. |
| 05:12.29 | Juggie | so you have to know where they are |
| 05:13.11 | joaovianna | Vonage was sued 2 or 3 times when a customer tried to rich 911 and fail. |
| 05:13.50 | newmedian | Here's a nice paranoid angle... just tie it all into TRW / credit rating system; Profile everyone, update their every move. Tie it into Real ID in the US. och! |
| 05:13.55 | blitzrage | I wish people would fuckin' realize that VoIP should NOT be your primary line |
| 05:14.06 | blitzrage | and if it is, you take the responsibility of it not working when you need it |
| 05:14.13 | Juggie | blitzrage, but companies are marketing it as being yor primary line. |
| 05:14.17 | Juggie | so they have to be accountable |
| 05:14.23 | blitzrage | then they should be shot |
| 05:14.32 | blitzrage | because it should never be marketed like that |
| 05:14.35 | Juggie | my cable company is comming out with a solid system |
| 05:14.43 | Juggie | which will not be a problem for them |
| 05:14.50 | Juggie | because their ATA is going to accept COAX |
| 05:14.53 | blitzrage | thats fine - but it better work when the power is out, and I better be able to reach 911 at all time |
| 05:14.59 | Juggie | and be tied into their cable network |
| 05:15.11 | Juggie | it will also plug into the wall, and have a battery backup built inside |
| 05:15.12 | joaovianna | blitzrage: Yes, it works well in 2005, but I think all the 4 millions landlines pstn will be VOIP in 10 years... (BIG BUSINESS TO US !!!) |
| 05:15.23 | Juggie | so your phone will work when the power is out |
| 05:15.31 | newmedian | But also the collective sheeple is not exactly Mensa material, particularly when it comes to technology. Just look at people who buy things from email and perpetuate the Spam cycle. Or who don't update their virus & malware defs and become part of a zombie botnet. :( |
| 05:15.38 | Juggie | of course this requires the cable infrastructure be backed up, and they are working on that |
| 05:16.23 | justnulling2 | anyone has 7960 version 3 or 4 of sip image? |
| 05:16.26 | newmedian | I'm sure Ted Rogers is keeping an eye on the market. |
| 05:16.38 | blitzrage | keeping an eye? |
| 05:16.41 | Juggie | well doing it over coax |
| 05:16.43 | blitzrage | he's right in the middle of it |
| 05:16.48 | Juggie | i think they have thier shit together |
| 05:16.50 | joaovianna | Asterisk could have a database like (DUNDI) with the adress for provide 911 services... |
| 05:16.54 | blitzrage | who do you think the CRTC is helping? |
| 05:16.59 | Juggie | put the device down by your dmark point |
| 05:17.07 | Juggie | plug bell into that, it has b attery backup |
| 05:17.10 | Juggie | seems solid |
| 05:17.29 | Juggie | i'd get it if it was cheap... then ted rogers will be getting me for phone/cell/internet/hi def tv |
| 05:17.30 | Juggie | everything |
| 05:17.44 | Juggie | but if he can pull it off and bundle it to save me money, i'll buy in |
| 05:17.45 | blitzrage | if its a big market provider, I probably don't want them :) |
| 05:18.07 | blitzrage | well, he just bought Sprint, so thats already going to cut a lot of minutes away from Bell |
| 05:18.16 | Juggie | really... he bought sprint? |
| 05:18.19 | Juggie | i had no idea |
| 05:18.19 | newmedian | woah |
| 05:18.28 | Juggie | got a link to an article? |
| 05:18.33 | Juggie | he bought fido some time ago |
| 05:18.35 | blitzrage | Juggie: news.google.ca |
| 05:19.07 | Juggie | damn |
| 05:19.12 | Juggie | i have my tickets for super7 for friday |
| 05:19.14 | Juggie | whats the jackpot/ |
| 05:19.16 | newmedian | A nice lottery win would go a long way to sitting on a beach near blue water, with someone offering a strawberry marg. |
| 05:19.37 | blitzrage | when you guys win, send me $40,000 so I can pay off my school loans |
| 05:19.39 | newmedian | Haven't checked the jackpot. But just about anything would be appreciated. Hope /dev/random is listening. |
| 05:19.46 | blitzrage | thats all I want |
| 05:20.27 | joaovianna | I'm thinking... DNS works fine, DUNDI works fine, ENUM works fine... Why is so difficult to create a centralized database information for voip customers ??? (TO ALL DEVELOPERS) |
| 05:20.48 | newmedian | Because the information doesn't CHANGE so much. |
| 05:20.56 | blitzrage | joaovianna: then do it :) |
| 05:20.58 | Juggie | newmedian, i had 5 of the 7 super 7 numbers last year |
| 05:20.59 | Juggie | and the bonus |
| 05:21.03 | Juggie | so 6 of 8 |
| 05:21.06 | Juggie | and all i got was 150$ |
| 05:21.11 | joaovianna | newmedian, so it is easier... |
| 05:21.11 | blitzrage | ouch |
| 05:21.16 | Juggie | one more number and i would have had 100k |
| 05:21.19 | newmedian | oooh Juggie! argh... horrible! you just have been... happysad |
| 05:21.32 | Juggie | yes, pissed |
| 05:21.36 | blitzrage | ok... bedtime |
| 05:21.39 | Juggie | for 5/7 bonus doesnt count |
| 05:21.55 | Juggie | 6/7 is 1000, 6/7+bonus is 100k |
| 05:22.12 | newmedian | I was part of an office pool that won the lottery. Several million. Except that the woman whose turn it was to buy the tickets forgot to buy them. And our numbers came up. |
| 05:22.33 | newmedian | She was afraid to show up to work. |
| 05:22.58 | joaovianna | Intrado charge $70,000.00 to route calls to E911... |
| 05:23.06 | newmedian | ack! |
| 05:23.52 | newmedian | So this E911 requirement may just kill the little guy. |
| 05:23.53 | joaovianna | I'm very worry about it... If you guys can listen to the meeting in FCC today... |
| 05:24.12 | joaovianna | http://www.fcc.gov/realaudio/mt051905.ram |
| 05:24.24 | joaovianna | Now It is law in EUA. |
| 05:27.19 | *** join/#asterisk hellop (~hellop@cpe-70-93-40-171.hawaii.res.rr.com) |
| 05:27.54 | joaovianna | All voip provider has 120 days to do that by law. |
| 05:28.34 | newmedian | Where's Bulworth when we need him. |
| 05:30.51 | newmedian | http://www.imdb.com/title/tt0118798/ |
| 05:33.41 | loud | i think voipjet just died. |
| 05:33.42 | hellop | why is 911 a problem? If a caller has no ability to make any local calls, then they should have to be able to call 911. |
| 05:33.58 | hellop | why is 911 a problem? If a caller has no ability to make |
| 05:34.25 | hellop | god dammit.. |
| 05:34.37 | *** join/#asterisk Corydon76-home (brown@pcp08665860pcs.500ash01.tn.comcast.net) |
| 05:34.38 | newmedian | E911 is what cell phones are for. These days just about everyone has a cell phone. If they get rid of their landlines, they should use a cell phone. And keep VOIP for non emergency calls. Even Microsoft tells you not to use their software for mission critical applications. |
| 05:40.35 | hmodes | oh my |
| 05:41.31 | newmedian | lions, tigers and bears? |
| 05:41.38 | hmodes | verily |
| 05:46.29 | loud | voipjet went satellite, 64 bytes from 216.118.117.46: icmp_seq=0 ttl=46 time=788.738 ms |
| 05:49.06 | Juggie | hah |
| 05:49.16 | PTG123 | thats b/c they skimp on quality :) |
| 05:49.27 | PTG123 | why pay for an internet connection when a cable modem will do :) |
| 05:51.07 | *** join/#asterisk Evanrude (~david@wsip-68-15-251-34.dl.dl.cox.net) |
| 05:51.08 | loud | how can i ignore a 'Registration from xxx' besides blocking it at a router level ? |
| 05:52.43 | *** join/#asterisk jonas (jonas@jonas.probe-networks.host.freenode) |
| 05:53.28 | xbmodder_lappy | why doesn't skype use SIP |
| 05:53.34 | xbmodder_lappy | that'd be the pimp then |
| 05:54.34 | *** join/#asterisk Inv_arp (junya@adsl-8-230-54.mia.bellsouth.net) |
| 06:04.11 | *** join/#asterisk Trickyphillips (~Trickyphi@adsl-68-124-61-95.dsl.irvnca.pacbell.net) |
| 06:07.41 | *** join/#asterisk sandnigg0r (~niggerplz@66.55.197.254) |
| 06:12.54 | *** join/#asterisk Evanrude (~david@wsip-68-15-251-34.dl.dl.cox.net) |
| 06:12.54 | *** join/#asterisk dant (~dan@81-86-69-213.dsl.pipex.com) [NETSPLIT VICTIM] |
| 06:12.54 | *** join/#asterisk kore (kore@mindwipe.org) [NETSPLIT VICTIM] |
| 06:14.43 | *** join/#asterisk JerJer[mobile] (~jj@ip68-103-26-140.ks.ok.cox.net) |
| 06:21.50 | JerJer[mobile] | the way i read the ruling is that all current VoIP operators are grand-fathered in |
| 06:21.57 | *** join/#asterisk RestLessGemini (~umairbari@202.142.189.86) |
| 06:21.57 | JerJer[mobile] | for E-911 |
| 06:22.18 | JerJer[mobile] | meaning that whoever they are interconnected with currently must provide them with an E-911 solution |
| 06:22.38 | JerJer[mobile] | granted the cost structure might not be pretty, but this also opens up some competition in this area |
| 06:23.02 | JerJer[mobile] | </delayed response> |
| 06:24.39 | *** part/#asterisk forrestc{hm} (~forrestc@iMach.com) |
| 06:26.12 | Dovid | i make all clients have a pots line for 911 |
| 06:28.06 | Pete_Largo | where did you get that 70k number joaovianna ? |
| 06:28.33 | Qwell | JerJer[mobile]: doesn't seem too bad then, no? |
| 06:29.04 | Pete_Largo | JerJer have you seen a copy of the ruling? I've only seen the press release |
| 06:30.02 | PTG123 | yah here is a great marketing gimic |
| 06:30.31 | PTG123 | "replace your phone lines with cheaper voip with better features, oh shit wait, actually scratch that, you need to keep your phone line to have a phone line with us, so forget saving money, pay more :)" |
| 06:41.32 | *** join/#asterisk lilo (lilo@levin-pdpc.staff.freenode) |
| 06:44.01 | *** join/#asterisk cmk (~cmk_@p54A3E497.dip.t-dialin.net) |
| 06:53.58 | *** join/#asterisk indego (~chris@floyd.gms.lu) |
| 07:06.36 | *** join/#asterisk wiseguy_ (~chivilis@85.206.9.173) |
| 07:06.39 | wiseguy_ | hellow |
| 07:06.47 | wiseguy_ | anyone using quadBRI? |
| 07:08.54 | *** join/#asterisk wildgoose (~edward@xunil.mail.wildgooses.com) |
| 07:12.53 | *** join/#asterisk scanna (~scannachi@81-174-16-211.f5.ngi.it) |
| 07:19.14 | *** join/#asterisk Rez (lorez@lorez.staff.freenode) |
| 07:23.26 | *** join/#asterisk Veesmooth (~veegee@vpn26-99.njit.edu) |
| 07:26.13 | wiseguy_ | anyone using quadBRI? |
| 07:26.39 | Veesmooth | any apache or lxr experts in here? |
| 07:29.32 | indego | wiseguy_: I hope to be soon, what problems are you having? |
| 07:30.14 | Veesmooth | well |
| 07:30.22 | Veesmooth | i got my apache up and running |
| 07:30.29 | Veesmooth | but im trying to get my lxr running |
| 07:30.33 | Veesmooth | im having problem with that |
| 07:30.38 | Veesmooth | u know any thing about lxr |
| 07:30.49 | *** join/#asterisk vandien (~stephan@p5091A039.dip0.t-ipconnect.de) |
| 07:31.07 | vandien | hi |
| 07:31.09 | DFT | exit |
| 07:31.19 | *** part/#asterisk xeet2 (~xeet3@pbx1.istx.net) |
| 07:31.43 | vandien | i suppose this is mainly an english channel hm? |
| 07:31.52 | Veesmooth | any answers to the question out there in asterisk land |
| 07:33.06 | indego | vandien: What language would you like? |
| 07:33.27 | *** join/#asterisk kapejod (~kapejod@p54BCCF5A.dip.t-dialin.net) |
| 07:33.30 | indego | Veesmooth: Personaly, I have never used it. |
| 07:33.47 | Veesmooth | ok tanks, any lxr users out there |
| 07:34.20 | vandien | indego: german would be great |
| 07:34.34 | indego | vandien: I kann ein bisschen |
| 07:34.43 | vandien | hehe |
| 07:34.47 | indego | i = ich :) |
| 07:35.00 | vandien | wellmaybe than my english is better than your german :p |
| 07:35.36 | vandien | so... i'm quite new to asterisk. Can anyone tell me whats the zapata stuff for? |
| 07:35.52 | kapejod | vandien: the zapata library is obsolete now |
| 07:36.06 | indego | vandien: Meine Frau ist Deutsche, deshalb habe ich Deutsch gelearnt. Aber English ist meine Mutter Sprache. |
| 07:36.27 | vandien | obsolete sounds good... |
| 07:36.35 | vandien | indego: hehe na dann :) |
| 07:38.34 | indego | kapejod: I am also new to Asterisk, if Zap is obsolete, what is in it's place. I am using CAPI so I did not need the Zap stuff, but for future reference. |
| 07:38.41 | vandien | okay, my asterisk is connected to a Siemens HiPath. The * is on an extension line (german: Nebenstrelle). i'm using i4l to connect my Fritz!Card2.0 and i can call other normal phones on the hipath. But when i dial the msn the * is connected to nothing hppens... i get timeout :-/ |
| 07:39.14 | vandien | i'll upload my exten, modem, sip.conf |
| 07:39.43 | kapejod | indego: zap is not obsolete , just the zapata library |
| 07:39.55 | indego | ok, thanks |
| 07:40.03 | kapejod | vandien: better use capi :) |
| 07:40.39 | indego | Yes, the capi stuff is lots better. More support and everything is going in that direction. |
| 07:40.42 | kapejod | i4l is not really suitable for voice. |
| 07:41.16 | *** join/#asterisk Jas_Williams (~Jason@host217-43-100-176.range217-43.btcentralplus.com) |
| 07:41.23 | vandien | hm, i feared that answer |
| 07:41.41 | vandien | im not the Linux pro so this seems to be quite hard fpr me |
| 07:41.52 | kapejod | vandien: use SuSE then |
| 07:42.01 | kapejod | it comes with capi drivers for your card |
| 07:42.03 | vandien | i'm using debian sarge an already found some tutorials |
| 07:42.16 | vandien | hmm naa i dont like suse :p |
| 07:42.27 | wiseguy_ | anyone using quadBRI? |
| 07:42.38 | vandien | well, the tuorials seemed to be possible ;) |
| 07:42.57 | kapejod | wiseguy_: yes |
| 07:43.07 | vandien | so i'll try to get CAPI to run... |
| 07:43.40 | drumkilla | what do you mean the zapata library is obsolete? |
| 07:43.46 | drumkilla | that is a totally bogus statement |
| 07:43.58 | kapejod | drumkilla: i mean, you dont need it for * anymore. |
| 07:44.14 | kapejod | i remember times where you had to download zapata too |
| 07:44.20 | vandien | uhm what is zapata normally used for? |
| 07:44.29 | drumkilla | vandien: hardware |
| 07:44.39 | drumkilla | and it's also used as a timing interface |
| 07:44.44 | kapejod | that is zaptel |
| 07:44.47 | kapejod | not zapata |
| 07:44.55 | kapejod | more bogus statements ;) |
| 07:45.07 | vandien | so tell us the truth :D |
| 07:45.07 | drumkilla | oh, bah |
| 07:45.14 | drumkilla | i'm so tired ... |
| 07:45.18 | drumkilla | :) |
| 07:45.20 | kapejod | hehe |
| 07:45.21 | indego | Could someone point me in the right direction. |
| 07:45.21 | indego | I have a working setup (just about), that is incoming and outgoing on the ISDN works. I have 2 SIP extentions that can call one another and out over ISDN. |
| 07:45.21 | indego | My boss is wanting to to have to dial a prefix for outgoing calls ie 9XXX. When I setup the outgoing to look for 5 or more digits for an external call, incoming calls also match and try to call back out on the ISDN line. With a prefix it works. Any ideas of how best to approch this? |
| 07:45.38 | drumkilla | ok, I'll shut up now ... just wanted to make sure there weren't any misconceptions here |
| 07:45.57 | drumkilla | :) |
| 07:46.00 | drumkilla | g'night folks |
| 07:46.05 | kapejod | n8 mr. drumkilla , sir :) |
| 07:46.33 | Jas_Williams | indego: set your out going extensions as exten => _9.,1,Do something |
| 07:46.51 | indego | vandien: I am using CAPI with debian, I will dig out my notes if you like. |
| 07:47.26 | vandien | indego: sounds great |
| 07:50.31 | *** join/#asterisk kilobit (~sam@203.123.20.12) |
| 07:51.34 | *** part/#asterisk Deryl (pgpkeys@static-141-149-128-140.buff.east.verizon.net) |
| 07:52.17 | indego | Jas_Williams: Yep, that I have working. But is with a prefix. Is it possible to check which interface the call is coming from, ie: _XXXXX.,1,GotoIf(${INTERFACE} = ${CAPI}?2:5) |
| 07:53.29 | indego | vandien: do you support file transfer via irc? |
| 07:55.56 | indego | vandien: If so I will send you my small notes file. |
| 07:56.36 | vandien | dunno, never tried it here. can you send an email? ill query you |
| 07:56.53 | *** join/#asterisk hellop (~hellop@cpe-70-93-40-171.hawaii.res.rr.com) |
| 07:57.14 | hellop | hi |
| 07:57.20 | nine76 | wb hellop |
| 07:57.25 | indego | vandien: sure |
| 07:59.44 | wiseguy_ | anyone using quadBRI? |
| 08:03.24 | vandien | but anyways, it would be great if someone could have a look at my config with i4l. my confs are at www.php-support.net/nopaste.php?id0426 |
| 08:04.08 | *** join/#asterisk Sato1 (~rauleli@sato1.wizardteam.com) |
| 08:05.07 | vandien | the problem is that i cant call the asterisk... |
| 08:05.27 | wiseguy_ | i have problem with quadBRI |
| 08:05.28 | wiseguy_ | http://pastebin.ca/12223 |
| 08:06.39 | *** join/#asterisk Zeeek (~icechat5@Zeeek.sustaining.supporter.pdpc) |
| 08:06.51 | hellop | so, if I have an X100p card, then usbuhci should not be running right? |
| 08:07.29 | *** join/#asterisk Mike_TK (~Mike_@bell.yes.net.ua) |
| 08:07.41 | hellop | Because my card should be the timing source, not the USB controller. |
| 08:08.23 | indego | wiseguy_: Is that i4l or CAPI, if capi have you applied the capi patch? |
| 08:09.00 | wiseguy_ | it is quadBRI |
| 08:09.14 | kapejod | indego: the quadbri uses zaptel drivers |
| 08:09.38 | vandien | indego: can i use a std debian kernel-imageß |
| 08:09.39 | vandien | ? |
| 08:10.56 | *** join/#asterisk macTijn (martijn@linda.net.insecure.nl) |
| 08:11.01 | hellop | vandien, yes, sarge regular kernel |
| 08:11.24 | vandien | would be 2.6.8 |
| 08:11.33 | hellop | vandien, there's a great howto for it |
| 08:11.51 | vandien | where? .> |
| 08:12.45 | *** join/#asterisk onkeltimm (~chatzilla@213-84-102-203.adsl.xs4all.nl) |
| 08:13.47 | hellop | shit I have the link on another PC not at this location.. |
| 08:14.15 | hellop | umm.. I got to it from one of the install guides listed on the wiki I think. |
| 08:14.56 | hellop | ahh here: http://asterisk.net.au/intro/0/ |
| 08:16.01 | onkeltimm | with a setup like [SIP phone 1] => [* box] <=IAX=> [* box] <= [SIP phone 2], can I do call pickup? if yes, how? |
| 08:17.42 | vandien | hellop: well i dont see anything about capi |
| 08:18.18 | hellop | I thought were were talking about a kernel. |
| 08:18.24 | hellop | debian |
| 08:18.38 | vandien | well, activating CAPI on debian ;) |
| 08:19.36 | vandien | hm, i found a german howto. ill try that one, but thank you anyway :) |
| 08:19.41 | *** join/#asterisk RoyK (~roy@179.80-203-29.nextgentel.com) |
| 08:19.42 | hellop | Well, I'm all tapped out for advice. ;) |
| 08:21.28 | kapejod | ~lart RoyK |
| 08:21.34 | kapejod | :) |
| 08:23.30 | *** join/#asterisk rajo (~rajo@matrix.cs.uni-sb.de) |
| 08:23.37 | indego | vandien: you need to make sure that the CAPI stuff is intalled in the kernel. If a stock kernel has it, great. I normaly build my own. It is not that hard. To check for CAPI "grep -i capi /boot/config-2.4.*" |
| 08:23.38 | *** join/#asterisk rajo (~rajo@matrix.cs.uni-sb.de) |
| 08:24.28 | vandien | theres an m |
| 08:24.35 | vandien | means module? |
| 08:25.10 | vandien | indego: does that mean i can load it? |
| 08:25.19 | *** join/#asterisk rajo (~rajo@matrix.cs.uni-sb.de) |
| 08:25.35 | indego | vandien: yes, it does. |
| 08:26.05 | vandien | hm... HOW? *g* |
| 08:26.10 | indego | vandien: I had dissabled the i4l stuff so as to have no possible conflict. |
| 08:26.16 | indego | modprobe |
| 08:26.41 | vandien | its loaded |
| 08:26.42 | vandien | wow |
| 08:26.45 | vandien | thats it? |
| 08:26.52 | indego | # /etc/modules: kernel modules to load at boot time. |
| 08:27.04 | indego | capiinit |
| 08:27.18 | indego | capiinfo |
| 08:27.32 | *** join/#asterisk elric (~kavit@ppp114-10.static.internode.on.net) |
| 08:27.58 | vandien | done |
| 08:28.17 | *** join/#asterisk rajo (~rajo@bfs.cs.uni-sb.de) |
| 08:28.50 | indego | did you get info back from capiinfo? |
| 08:28.56 | vandien | hmm so i alredy have capi support? what else do i need? fritzcard driver |
| 08:29.14 | vandien | no... |
| 08:29.15 | indego | yes, that is the next bit in my doc |
| 08:29.28 | hellop | Is the second part of the Asterisk docs blank, or am I dumb? |
| 08:29.33 | vandien | # capiinit: command not found |
| 08:29.51 | vandien | wtf |
| 08:30.10 | indego | start again after 3. Build and install kernel |
| 08:30.19 | vandien | ha, didnt install capiutils |
| 08:30.24 | indego | :) |
| 08:31.05 | *** join/#asterisk cmk (~cmk_@p54A3E497.dip.t-dialin.net) |
| 08:31.20 | vandien | urgs i got several No such device infos |
| 08:33.33 | vandien | www.php-support.net/nopaste.php?id=427 |
| 08:34.31 | indego | vandien: did you download and install the driver? Read my doc again from step |
| 08:34.40 | indego | setp 4. |
| 08:35.08 | vandien | well i skipped 1-3, did 4 and tried capiinfo |
| 08:35.11 | *** join/#asterisk cmk (~cmk_@p54A3E497.dip.t-dialin.net) |
| 08:35.21 | vandien | so i need the drivers before i can do capiinfo.. kk |
| 08:35.29 | indego | yes |
| 08:35.57 | vandien | sorry, i'm quite impatient... |
| 08:38.03 | indego | vandien: I thought my doc was shot enough ;) |
| 08:38.09 | file | argh |
| 08:38.37 | *** join/#asterisk Fabe (~spamhere@pD95B0BF7.dip0.t-ipconnect.de) |
| 08:42.59 | vandien | grrr i hate compiling things... |
| 08:46.05 | vandien | www.php-support.net/nopaste.php?id=428 indego |
| 08:46.37 | vandien | line 8 |
| 08:46.50 | vandien | no rule to create "modules" |
| 08:47.00 | vandien | what does it mean? |
| 08:49.11 | vandien | wait... .ko files are for 2.6 kernels? |
| 08:49.20 | vandien | grr |
| 08:50.26 | vandien | installing kernel 2.6.8.. |
| 08:51.43 | indego | you might need the kernel source |
| 08:52.04 | vandien | hm theres no source package, only headers |
| 08:52.08 | indego | download the kernel-source counterpart to the kernel image |
| 08:52.16 | indego | ok, grab them |
| 08:52.23 | vandien | i already have |
| 08:52.37 | vandien | i think i read that the never drivers are for 2.6 only |
| 08:52.42 | indego | humm, that might have been why I built the kernel from source |
| 08:52.43 | vandien | i triad with a 2.4 |
| 08:52.47 | vandien | hrhrhr |
| 08:53.04 | RoyK | talking to me?? huuh? talking to me??? |
| 08:53.05 | RoyK | :) |
| 08:53.35 | Zeeek | you're on everyone's ignore list, so, no |
| 08:53.53 | RoyK | :) |
| 08:53.57 | Zeeek | hi |
| 08:54.09 | RoyK | Zeeek: why so arrogant today? |
| 08:54.23 | Zeeek | no I was being jovial |
| 08:55.45 | indego | BTW I just got my logic right, now I can do prefixless routing :) |
| 08:55.58 | Zeeek | sounds exciting |
| 08:56.01 | indego | exten => _XXXXX.,1,GotoIf($[${CHANNEL:0:3} = CAPI]?incoming,s,1) |
| 08:56.30 | indego | I was having the problem that incoming calls on the ISDN were being routed back out on the ISDN |
| 09:00.05 | *** join/#asterisk morris (~morris@pcworkshop.plus.com) |
| 09:00.38 | morris | Zeeek, i got zaptel working last night in the end.. i had done something stupid with zapata.conf ;p |
| 09:02.20 | *** join/#asterisk tessier (~treed@203.210.216.1) |
| 09:02.47 | Zeeek | what was it, morris? |
| 09:03.13 | Zeeek | you forgot planet=earth ? |
| 09:03.58 | morris | i had placed another [channel] directive to the end of the config.. it didnt occur to me at the time that there would be another.. |
| 09:04.00 | RoyK | "a small planet in the vicinity of betelgause" |
| 09:04.07 | morris | commented out the existing one in the config |
| 09:04.10 | morris | and it was fine |
| 09:06.16 | tessier | Anyone know why, when using nat=no and canreinvite=yes, my xten and Cisco 7912 can talk directly to each other for rtp (on same network behind nat) but when I hang one of the phones up it sends a bye directly to asterisk instead of to the other phone. This seems contrary to the diagram on page 12 of RFC3261. |
| 09:06.23 | Zeeek | OMG |
| 09:07.02 | Zeeek | so morris, status report: where are you in your mater plan? |
| 09:07.11 | Zeeek | master plan |
| 09:07.44 | Zeeek | a great fisherman must be a master bater of course |
| 09:09.48 | hellop | Can I not have singl digit extensions? |
| 09:10.05 | Zeeek | why of course you may! |
| 09:10.59 | Zeeek | exten => 0,1,Dial(ZAP/1) |
| 09:11.04 | vandien | indego: hmmm make and make install ran trough without problems.... but i still cant load it... any idea? |
| 09:11.25 | hellop | fo instant this doesnt work: exten => 1,2,Record(/tmp/arecording:gsm) |
| 09:11.37 | hellop | Bu if I replace the 1 with 12 it works. |
| 09:12.02 | Zeeek | dialplan problem |
| 09:12.16 | Veesmooth | any apache guru out there |
| 09:12.17 | Veesmooth | ? |
| 09:12.25 | hellop | Zeeek, where should I look to fix it? |
| 09:12.36 | Zeeek | ask (not a guru but have the book) |
| 09:12.36 | RoyK | Veesmooth: perhaps #apache might help :P |
| 09:12.44 | Zeeek | Veesmooth what? |
| 09:13.03 | Zeeek | hellop the wiki has about 5 pages on this |
| 09:13.13 | Zeeek | I wish I had the links |
| 09:13.15 | Veesmooth | he he, well i just got a simple apache question to ask |
| 09:13.20 | RoyK | hellop... as in hell operator? |
| 09:13.24 | Zeeek | ASK already |
| 09:13.49 | Veesmooth | well i can access my server with no problem |
| 09:13.58 | Veesmooth | which is the default folder |
| 09:14.03 | Veesmooth | but when i create sub folders |
| 09:14.07 | Veesmooth | it cant access them |
| 09:14.13 | hellop | RoyK, no as in Hello, I'm peeing! |
| 09:14.20 | hellop | uhhh no you're right |
| 09:14.21 | Veesmooth | only restricted to show the default folder of webpages |
| 09:14.23 | Veesmooth | hmmm |
| 09:14.33 | Zeeek | the word "folder" scares me a little |
| 09:14.45 | hellop | Zeeek, ok I'll keep reading. |
| 09:15.00 | hellop | It's just, someone keeps calling. And.. like, I feel sorry for them. |
| 09:15.10 | Zeeek | hellop the point is, what happens when you dial 1? |
| 09:15.24 | indego | vandien: did you update the /etc/isdn/capi.conf file? |
| 09:15.34 | *** join/#asterisk pranav (~dawda_pra@202-63-174-250.exatt.net) |
| 09:15.35 | hellop | Zeeek, it waits for the second digit, then timesout, and continues. |
| 09:15.39 | Zeeek | Veesmooth the answer is in httpd.conf |
| 09:15.51 | Veesmooth | hmm, ok |
| 09:15.56 | Zeeek | but there are so many variable that this really isn't the place to find the answer |
| 09:15.57 | Veesmooth | so that should control it huh |
| 09:16.05 | pranav | hello everyone |
| 09:16.09 | Zeeek | well that cotrols Apache behaviour |
| 09:16.11 | Veesmooth | ok, ill keep diging and see if i can come up with something |
| 09:16.36 | vandien | indego: yes, everything commented but fcpci |
| 09:16.36 | Zeeek | Veesmooth most of the O'Reilly book is online somwhere so you should be able to look that up |
| 09:16.48 | Veesmooth | oreilly on appache right |
| 09:16.52 | Veesmooth | he wrote one on it? |
| 09:16.54 | pranav | can someone tell me i have my asterisk set up and i want to make a call to us pstn |
| 09:17.06 | pranav | is it possible |
| 09:17.09 | morris | Zeeek, currently i am reading slashdot.. once i am done there i am getting back with asterisk. The plan is relatively simple really, send 1 standard analogue line down the internet to another shop which will have adsl and a hardware sip phone. that is the long and the short of it really. aparently it would cost my boss £5000 GBP to have 5 lines installed and 2 year contract with NTL to link our third store up with centrex line our other 2 stores. |
| 09:17.09 | pranav | ? |
| 09:17.30 | Veesmooth | thanks zeek, ill see if i can get an oreily book |
| 09:17.35 | Zeeek | morris that's what I do between home and office |
| 09:17.56 | Zeeek | Veesmooth really if there weren't so many possibilities, I'd be happy to help |
| 09:18.07 | morris | Zeeek, How reliable have you found it ? |
| 09:18.27 | Zeeek | works pefectly with a cheap BT100 |
| 09:18.55 | morris | ah thats awesome |
| 09:18.55 | Zeeek | the only glitches are regarind problems when I don't have asterisk on a fixed ip and even that is solved |
| 09:19.06 | Zeeek | I hate my ISP! |
| 09:19.08 | morris | lol |
| 09:19.14 | morris | yea my home one is dynamic too |
| 09:19.24 | Zeeek | DSL is so cheap here it's incredible but there is zero serious service |
| 09:19.24 | morris | lukcily here they are static at no extra charge |
| 09:19.41 | morris | you from the US then ? |
| 09:19.49 | Zeeek | ironically my home one is static and the astersk is at the moment on a dynamic ip |
| 09:20.03 | morris | i see |
| 09:20.05 | Zeeek | France: DSL 2048/256 for like $20/mo |
| 09:20.11 | morris | ok to swear ? |
| 09:20.12 | morris | lol |
| 09:20.16 | morris | awesome |
| 09:20.23 | Zeeek | hotline = "Dancing with Imbeciles" |
| 09:20.31 | Zeeek | and it costs $0.50/min |
| 09:20.37 | morris | rofl |
| 09:20.38 | RoyK | hi |
| 09:20.40 | indego | vandien: have you installed isdnactivecards ? |
| 09:21.16 | RoyK | is it possible to setup asterisk so a proxy (SER or another) can do all the authentication and then for asterisk to only do call handling? |
| 09:21.23 | Zeeek | in fact, now they've rolled out up to 20Meg DSL for $30 - but I want 100M/100M fiber to the home! |
| 09:21.28 | indego | vandien: even though this is a passive card you need it |
| 09:21.34 | RoyK | as in "allow all calls forwarded by proxy xxx.something.com" |
| 09:21.44 | RoyK | if so, how can I know if calls are forwarded from that proxy? |
| 09:21.57 | Zeeek | RoyK I think that's exactly what one of out voip providers is doing |
| 09:22.03 | morris | Zeeek, that is every geeks dream ! |
| 09:22.22 | RoyK | Zeeek: do you know how I can configure asterisk to do this? |
| 09:22.24 | Zeeek | 100M§100M sounds pretty good - we'll have to see what their peeing is like |
| 09:22.40 | Zeeek | RoyK sorry, no but I think it's been discussd on the ml |
| 09:23.04 | morris | Zeeek, something i just wondered.. is your first language french ? |
| 09:24.03 | Zeeek | no |
| 09:24.18 | Zeeek | my first language is Vulcan |
| 09:24.37 | Zeeek | I'm Minnesota born and bred |
| 09:24.38 | morris | rofl |
| 09:25.18 | pranav | can someone tell me , if i am in a country and i want to make calls to another country using asterisk is it possible? |
| 09:25.31 | pranav | i have sipura configured |
| 09:25.33 | vandien | www.php-support.net/nopaste.php?id=429 indego |
| 09:25.41 | tessier | pranav: no |
| 09:25.49 | tessier | pranav: The person on the other end needs a voip phone also. |
| 09:25.59 | tessier | pranav: Or you need a voip to pstn gateway. |
| 09:26.01 | vandien | kobject register failed for fcpci (-17) |
| 09:26.03 | pranav | well but cant asterisk work as a gatekeeper |
| 09:26.06 | vandien | grrr |
| 09:26.46 | pranav | so how to get this gatewayvoip to pstn gateway |
| 09:27.46 | tessier | pranav: pay someone like nufone some money to gateway for you |
| 09:27.48 | pranav | ok so if a person on the other end is having a voip phone then he also needs to be registered with asterisk |
| 09:27.53 | tessier | www.nufone.net |
| 09:28.02 | tessier | pranav: yes |
| 09:28.24 | Sato1 | how can i redirect calls to asterisk from a FXO port to asterisk in an addpac? it uses the very same way to configure as a cisco |
| 09:28.46 | pranav | thanks tessier |
| 09:29.03 | Zeeek | pranav pranav here are two good intro articles for you: |
| 09:29.04 | Zeeek | http://www.onlamp.com/pub/a/onlamp/2003/07/03/asterisk.html |
| 09:29.04 | Zeeek | http://www.onlamp.com/pub/a/onlamp/2004/01/22/asterisk2.html |
| 09:29.05 | Sato1 | whenever a call goes, the addpac gives its local dialtone waiting for an extension |
| 09:29.35 | pranav | thanks zeeek to you as well |
| 09:29.41 | Zeeek | np |
| 09:29.45 | pranav | i'll go through it |
| 09:30.20 | pranav | actually i have asterisk ready with sipura phones configured on it |
| 09:30.25 | vandien | indego: seems likes hisax modules are the the prob. i'll try to get them unloaded... |
| 09:30.41 | onkeltimm | anyone ever did door answering in asterisk? like we have a Siedle Module that can be connected to our legacy pbx? how could I handle this in *??? |
| 09:30.41 | Zeeek | pranav those articles talk about providers etc. |
| 09:30.55 | pranav | ok |
| 09:31.10 | Zeeek | onkeltimm see the mailing list there have been a coupe of threads |
| 09:31.18 | onkeltimm | ok |
| 09:33.37 | indego | vandien: OK, sounds like a good idea. I disabled all non-capi isdn stuff before compiling the kernel, so I would have not seen this issue |
| 09:35.47 | *** join/#asterisk timur (~timur@pD95795BC.dip.t-dialin.net) |
| 09:36.54 | timur | hello |
| 09:38.09 | timur | is it possible to achieve a kind of DID with 1 SIP number ? |
| 09:38.34 | timur | like 42342-0 as main number, and the rest is done by the asterisk |
| 09:39.06 | morris | ioctl(ZT_LOADZONE) failed: Inappropriate ioctl for device |
| 09:39.06 | morris | Failed to register zone 'United States / North America': No data available |
| 09:39.15 | morris | Zeeek, anything i should be concerned about ? |
| 09:40.11 | Zeeek | hmmm |
| 09:40.40 | Zeeek | you're missing something in zaptel or zapata |
| 09:40.48 | morris | okeys |
| 09:40.49 | morris | thanks |
| 09:41.05 | timur | hey Zeeek, u know what i am talking about ? |
| 09:41.09 | timur | :) |
| 09:41.14 | morris | btw, i have a call going via analogue to sip fine now ;) |
| 09:41.19 | morris | i need to setup dialing rules back out |
| 09:41.23 | morris | but weeeeeeeeee quite happy ;p |
| 09:41.31 | Zeeek | timur no I don't see it yet? |
| 09:41.53 | timur | i wonder if i need to register like 60 numbers at a VoIP provider |
| 09:42.15 | timur | like 60 numbers that gets routed to my asterisk |
| 09:42.24 | timur | or if i can get 1 number |
| 09:42.42 | Zeeek | explain in real life what you want |
| 09:42.50 | Zeeek | a scenario |
| 09:43.02 | timur | ok a company got numbers like XXXXX-0 for the secretary |
| 09:43.27 | timur | and all numbers like XXXXX-345 are done by the PBX |
| 09:43.34 | timur | i want to do this via VoIP |
| 09:43.56 | *** join/#asterisk PandemiK (~PandemiK@62.2.255.42) |
| 09:44.02 | PandemiK | hi |
| 09:44.02 | timur | in germany u can get a PSTN number from a VoIP provider |
| 09:44.37 | Zeeek | <PROTECTED> |
| 09:44.49 | timur | my VoIP phone rings |
| 09:44.58 | toot | you can in edinburgh also -> nplusone.net |
| 09:45.01 | timur | cuz its registerd at the VoIP sip |
| 09:45.13 | toot | although i don't see how that works for 999 calls? |
| 09:45.57 | Zeeek | timur still too vague - sure an incoming PSTN call can ring a voip phone. Is that all you need? |
| 09:46.06 | timur | no |
| 09:46.15 | timur | ok imagine my PSTN number is 73434 |
| 09:46.27 | toot | thats been confusing me recently - i thought that you needed a geographic number for emergency services? |
| 09:46.35 | timur | what happens if somebody calls 7343452 |
| 09:46.51 | Zeeek | nothing |
| 09:46.53 | timur | whats happening with the 52 ? |
| 09:46.58 | timur | :( |
| 09:47.06 | PandemiK | Does someone here use a grandstream BT ? |
| 09:47.12 | timur | i thought i can catch this with my asterisk |
| 09:47.38 | Zeeek | they need to call 73.... and then enter more digits after hearing a message like "if you know the extension.... please dial it" |
| 09:47.54 | timur | yea like a voicemenu |
| 09:48.20 | Zeeek | This is called IVR and you will now be asked to go read about that on the wiki or in http://asteriskdocs.org |
| 09:48.25 | timur | i know this option, but its not working in every-day-businesss |
| 09:48.31 | Zeeek | where there is considerable info |
| 09:48.39 | timur | thank you Zeeek |
| 09:48.44 | timur | IVR - gogo |
| 09:48.46 | timur | :) |
| 09:48.53 | timur | go pikachu go |
| 09:48.58 | Zeeek | timur if it isn't workable why do 99% of the businesses I call use them? |
| 09:49.18 | Zeeek | I can't remember the last time any business answered the phone directly :) |
| 09:49.32 | morris | we do Zeeek ;p but we are small ;/ |
| 09:49.33 | Zeeek | except the plumber - those guys are in the 19th century |
| 09:49.40 | morris | lol |
| 09:49.41 | Zeeek | we do too and we are smal |
| 09:49.50 | timur | u hear voicemenus everyday ? |
| 09:50.00 | Zeeek | yes and many are very obnoxious |
| 09:50.08 | timur | call 1 for Mr.X and 2 fpr Mrs. Y |
| 09:50.11 | timur | rofl |
| 09:50.12 | timur | funny |
| 09:50.19 | timur | i wouldnt mind |
| 09:50.27 | Zeeek | my favorite is when you are asked to enter your phone number and then when you get connected they ask again, PLUS they have callerid anyway! |
| 09:50.30 | timur | but business men do here |
| 09:50.37 | morris | Zeeek, do you have a tip for me to setup a dialplan so that. Anything dialed from the SIP phone is simply routed and mirrored to the zap interface? |
| 09:50.45 | indego | timur: If you have ISDN with DID then you can do what you want. Your main number would be 73434 and you can have the phone company pass the extra digits through. |
| 09:50.55 | timur | indego - i know |
| 09:51.05 | timur | just need an NT mode or summin |
| 09:51.15 | timur | DID |
| 09:51.22 | indego | Yes, and active ISDN card |
| 09:51.24 | Zeeek | morris you don't want ANY internal extension or services ? that's pretty limited, but if intentional can be done |
| 09:51.36 | timur | [11:38] <timur> is it possible to achieve a kind of DID with 1 SIP number ? |
| 09:51.37 | morris | nothing more is needed |
| 09:51.51 | Zeeek | sorry timur, I was talking analog |
| 09:51.57 | morris | and yea.. rather boring.. i will do the rest i want from my home one now i have a better understanding of asterisk |
| 09:52.10 | indego | timur: sorry, didn't read the entire history. |
| 09:52.12 | timur | np |
| 09:52.18 | timur | all fine peeps |
| 09:52.22 | timur | thx |
| 09:52.44 | morris | i have ALWAYS wanted to setup a, IVR system ever since i played a wheel of fortune premium rate game lol |
| 09:52.57 | *** join/#asterisk darwin35 (~darwin35@24.3.226.147) |
| 09:52.59 | *** join/#asterisk ellvis (~ellvis@adsl-flat-basic-105.84-47-116.telecom.sk) |
| 09:53.04 | ellvis | hi people |
| 09:53.09 | timur | IVR - go picachu go :) |
| 09:53.19 | Zeeek | morris there are two ways to go. If it's you the user, just dial a ZAP line. Set immediate=yes in zapata |
| 09:53.44 | Zeeek | that will call the context you setup and you can get a dialton right away |
| 09:53.48 | darwin35 | anyone know the new context for inbound calls is it |
| 09:53.51 | Zeeek | is that what you want? |
| 09:53.54 | darwin35 | <darwin35> like Dial(SIP/#####|s|1) |
| 09:53.54 | darwin35 | <darwin35> or Goto(inbound|o|1) |
| 09:54.07 | darwin35 | my dialplan died |
| 09:54.12 | morris | zeek, i have a dialtone .. however when i dial a number.. it rings untill the person answers then asterisk hangs up :( |
| 09:54.16 | PandemiK | I have a serious : when I call my Bugtetone, there's no sound but when I'm calling with it, everything if fine, can somone help me ? |
| 09:54.42 | darwin35 | you need s r in your options |
| 09:54.50 | Zeeek | morris I don't quite follow. How do you have dialtone? The local one from your SIP shone? |
| 09:54.50 | darwin35 | to pass the rining |
| 09:55.20 | Zeeek | PandemiK what is NAT situation? |
| 09:55.29 | PandemiK | Zeeek: I'm calling on the same subnet |
| 09:56.00 | morris | zeek, currently i made an extention 2000, when calling that from SIP it sends the call direct to ZAP/1.. so then my sip phone will just sit with a dialtone.. alowing me to dial any number after the call is attatched. i dial say.. 2002 DTMF and it will ring an extention in the office on the analogue system.. soon as the call is picked up it cuts the SIP and ZAP call off |
| 09:56.05 | vandien | indego: CAPI is running :D |
| 09:56.08 | PandemiK | Zeeek: hmm, not exactly, two diffrent subnets (asterisk is 192.168.1.11, BT102 is 192.168.2.17 and X-lite is 192.168.2.13) |
| 09:56.21 | Zeeek | PandemiK I'd be tempted to test the same ip with a softphone then |
| 09:56.26 | PandemiK | Zeeek: and wih a router wetween them but no NAT |
| 09:56.48 | Zeeek | still, test the same two boxes with a softphone |
| 09:56.55 | morris | Zeeek, altho that is not what i want |
| 09:57.07 | PandemiK | Zeeek: good idea ! I'll try right now |
| 09:57.23 | Zeeek | morris you have a pbx ? |
| 09:57.42 | morris | not one we have access too, our calls are already routed and managed by NTL |
| 09:57.49 | Zeeek | oh calling ZAP FXS phone? |
| 09:57.49 | darwin35 | anyone ? |
| 09:57.56 | darwin35 | I am going insane |
| 09:58.12 | Zeeek | downgrade to STABLE |
| 09:58.18 | morris | [privileged] |
| 09:58.19 | morris | exten => 2000,1, Dial(ZAP/1,10,t) |
| 09:58.24 | morris | is all i have |
| 09:58.28 | morris | right now |
| 09:58.32 | morris | which establishes a connection |
| 09:58.35 | Zeeek | what ois ZAP/1? an FXO connected to PSTN? |
| 09:58.38 | morris | yea |
| 09:58.40 | morris | x100p |
| 09:58.42 | morris | clone |
| 09:58.55 | Zeeek | and dialing 2000 on the pstn should do? |
| 09:59.05 | RoyK | hm... is it possible to offload _all_ authentication to a SER box? |
| 09:59.42 | morris | Zeeek, nothing was just for testing.. i couldnt work out how to get anything dialed from sip to route to phone |
| 09:59.53 | Zeeek | wow |
| 10:00.20 | morris | the ideal way is just to pick the SIP phone up, dial ANY number and asterisk feeds that exact number to ZAP/1 |
| 10:00.24 | morris | is that doable? |
| 10:00.34 | Zeeek | is you dial ZAP/1 you have the PSTN dialtone and can dial any number that you can on a phone |
| 10:00.39 | morris | yea |
| 10:00.41 | morris | that is right |
| 10:00.45 | morris | i get a direct dialtone |
| 10:00.49 | Zeeek | I don't see what dialing 2002 was about |
| 10:00.54 | morris | ok.. |
| 10:01.00 | Zeeek | dial a local PSTN number |
| 10:01.01 | morris | now when i call 2000 from my softphone |
| 10:01.03 | morris | yea |
| 10:01.04 | morris | tahts right |
| 10:01.07 | morris | it will call another line |
| 10:01.10 | morris | on the pstn network |
| 10:01.17 | *** join/#asterisk IkeKrull (~ikekrul1@203-79-103-167.paradise.net.nz) |
| 10:01.21 | Zeeek | and then it hangs up? |
| 10:01.23 | morris | yea |
| 10:01.27 | morris | soon as some one asnwers |
| 10:01.29 | morris | it rings |
| 10:01.31 | Zeeek | is this an ebay card? |
| 10:01.32 | morris | untill they accept the call |
| 10:01.36 | morris | yes it is |
| 10:01.43 | Zeeek | *end of discussion* |
| 10:01.53 | morris | ok |
| 10:01.56 | Zeeek | try another card in the batch :) |
| 10:02.05 | morris | only got one bitch ;p |
| 10:02.06 | Zeeek | and if in UK, forget it |
| 10:02.26 | morris | when dialing asterisk from the pstn network |
| 10:02.26 | Zeeek | [or do extensive mailing search] |
| 10:02.44 | morris | asterisk routes the call to sip |
| 10:02.45 | morris | perfectly |
| 10:02.54 | morris | there is noticable delay however |
| 10:02.56 | IkeKrull | hi there, does anyone know if it is possible to make asterisk abort its extension-processing sequence when the line it is connected to goes 'off-hook'? |
| 10:03.04 | Zeeek | morris although.... knowing where you are in your config, you might look around for how to configure the X100, that may be it too |
| 10:03.24 | morris | okeys, thanks Zeeek .. |
| 10:03.26 | morris | :) |
| 10:03.42 | Zeeek | morris: look for busydetect for example |
| 10:03.48 | Zeeek | just a sec I'll find the keys |
| 10:03.49 | Jas_Williams | morris: what is working or not now ? |
| 10:03.56 | IkeKrull | that is, i have asterisk acting as a glorified answerphone, but once it detects the ring, it waits then answers the call even if i have picked up the phone on the same line |
| 10:04.38 | Jas_Williams | IkeKrull: Asterisk does not like that get an ATA and have asterisk in ful controll of you phones |
| 10:04.52 | IkeKrull | right, so asterisk cant do this? |
| 10:05.06 | Jas_Williams | IkeKrull: it is not the way it was designed |
| 10:05.14 | Zeeek | morris search wiki and mailing list for : callprogress busydetect busycount |
| 10:05.33 | Zeeek | the truth may lie in one of those or you may have a shitty card |
| 10:06.30 | Zeeek | IIRC the number one unintuitive, unguessable change is "calrprogress=no" |
| 10:06.32 | morris | Jas_Williams, Current scenario is: I have a x100p clone card in this asterisk machine. It is connect to a pstn and when it recieves a call it routes the call to a SIP extention SoftPhone.. communication seems fine. (however a small delay) what i would like to do now is, place a call from SIP and route it via ZAP. Now current i have made a tempory rule for dialing extention 2000 from SIP will connect SIP to ZAP creating a dialtone.. then usin |
| 10:06.32 | morris | g DTMF dial from there *not how i want it done really* then u can call any line via the ptsn.. no problem.. HOWEVER! When the recipient picks the phone up on the pstn side.. both SIP and ZAP are hungup. |
| 10:06.41 | morris | thanks Zeeek |
| 10:06.50 | PandemiK | Zeeek: everything is fine with a softphone, with the same IP and SIP user, so it's from my BT102 config |
| 10:07.20 | Jas_Williams | morris: post your zapata.conf to pastebin.ca |
| 10:07.25 | PandemiK | Zeeek: I'll make it back to initial config, thx |
| 10:07.25 | morris | okeys |
| 10:07.26 | Zeeek | PandemiK so.... BT100 audio.... try changing the RTP port beginning at 10000 for fun |
| 10:07.57 | Jas_Williams | morris where are you located |
| 10:08.21 | morris | I am in the UK |
| 10:08.35 | Jas_Williams | So am I tht helps ;-) |
| 10:08.48 | morris | ah sweet lol |
| 10:08.53 | morris | (where abouts?) |
| 10:08.57 | morris | in in surrey |
| 10:09.10 | Jas_Williams | Somerset (20miles south of Bristol) |
| 10:09.28 | Zeeek | add UK to the mailing list search then :) |
| 10:09.36 | Zeeek | you guys have no luch with BT |
| 10:09.42 | morris | http://pastebin.ca/12228 |
| 10:09.49 | PandemiK | YEah, it seems ;-) |
| 10:09.54 | morris | zeek ok |
| 10:10.00 | Jas_Williams | Zeeek: It works gerat for me most of the time :) |
| 10:10.00 | PandemiK | in my case, everything was working a few days ago |
| 10:10.06 | morris | Jas_Williams, cool.. i was in bristol raving a couple of weeks back ;) |
| 10:10.32 | Zeeek | Jas_Williams as you know, there have been many UK-specific issues on the ml, BT is the Broadvoce of Europe :) |
| 10:11.36 | Jas_Williams | :) |
| 10:11.55 | Zeeek | ringing capacitors indeed! |
| 10:12.05 | fenlander | Most of the UK problems are TDM400p related... |
| 10:12.10 | Zeeek | muhahahaha |
| 10:12.31 | morris | Jas_Williams, http://pastebin.ca/12228 ;p |
| 10:12.52 | Jas_Williams | morris: zapata.conf looks fine lets get your extensions.conf fixed properly, do you wish to use a prefix for outbound calls ? |
| 10:12.53 | hellop | Aha, one digit extensions are implemented by setting the DigitTimeout to 1 or 0. |
| 10:13.41 | Jas_Williams | hellop: No just having 1 digit extensions as the only options rather than many |
| 10:13.58 | PandemiK | Zeeek: setting RTP port to 10000 doesn't change anything |
| 10:14.11 | morris | jas.. the answer in this instance is no. altho, we do naturaly prefix all outbound calls with a 9 normaly. our internal lines in the ptsn all start with a 2.. so either way is suitable.. i think i would prefere using extention rules |
| 10:14.13 | hellop | But, how does it know when to stop? was my next question. |
| 10:14.21 | PandemiK | Is there a way to return the BT102 to manufactory settings ? |
| 10:14.47 | Zeeek | There must be but I don't remember it |
| 10:14.58 | hellop | Jas_Williams, so, if I have 11 and 111 in extensions.conf, you're saying it will wait until digit time out before matching 11? |
| 10:15.15 | Jas_Williams | hellop: yes |
| 10:15.34 | indego | vandien: Cool, good to know |
| 10:15.40 | hellop | ditto |
| 10:16.30 | Jas_Williams | morris: ok what do you have for extension 2000 currently ? |
| 10:16.43 | morris | shall i upload my extentions.conf ? |
| 10:16.55 | morris | ill explain tho anyway.. |
| 10:17.03 | morris | that 2000 extention |
| 10:17.06 | Jas_Williams | yes to pastebin so you don't flood the channel |
| 10:17.07 | morris | was a theoretical number |
| 10:17.18 | morris | just used for testing |
| 10:17.26 | morris | i would prefere to do this properly ignoring what i had setup |
| 10:17.30 | hellop | experimental? |
| 10:17.31 | Jas_Williams | ok |
| 10:18.20 | Jas_Williams | morris: will users be dialling full pstnn numbers with no prefix such as 01749XXXXXX |
| 10:18.23 | morris | Jas_Williams, http://pastebin.ca/12229 |
| 10:19.14 | *** join/#asterisk mrkyr (bviitanen@h24-207-81-82.cst.dccnet.com) |
| 10:19.18 | morris | pstn numbers will always be dialed as 901234 |
| 10:19.23 | morris | since the rest of the network uses this system |
| 10:19.32 | morris | we also have internal numbers it needs to route too |
| 10:19.45 | morris | and they are 2000 2001 2002 2003.. up to 2010 |
| 10:20.05 | Jas_Williams | when the rest of the system routes the call through asterisk the 9 will still be there ? |
| 10:21.16 | *** join/#asterisk rLg (~umairbari@202.142.189.86) |
| 10:21.44 | morris | yes |
| 10:22.39 | Jas_Williams | morris: sorry for all of the questions, is your sip phone in the context tikka ? |
| 10:23.56 | Jas_Williams | morris: add this to [privileged] |
| 10:25.19 | morris | i created a context called tikka in my zapata.. then i renamed the section demo in extentions.conf to tikka |
| 10:25.24 | morris | i think ;p |
| 10:25.47 | *** join/#asterisk t0p (t0p@tech-mgr.chatri.com) |
| 10:25.52 | Jas_Williams | add this to [privileged] exten => _9.,1,Dial(Zap/g2/${EXTEN:1}) |
| 10:26.18 | Jas_Williams | opps should be this add this to [privileged] exten => _9.,1,Dial(Zap/g1/${EXTEN:1}) |
| 10:27.07 | t0p | Jas_Williams: Hi, how is everything? |
| 10:29.00 | Jas_Williams | morris: also you need to add group=1 to your zapata.conf before the channel=>1 |
| 10:29.47 | Jas_Williams | morning t0p fixed all of your issues yet ? |
| 10:29.58 | t0p | Jas_Williams: I got my broken X100P fixed two days ago |
| 10:30.20 | morris | Jas_Williams, .. i had tested before adding group=1 to my zapata.. i will do that now... as i got this error : http://pastebin.ca/12230 |
| 10:30.21 | morris | hehe |
| 10:31.37 | morris | Jas_Williams, only 2 issues exist currently.. |
| 10:32.01 | *** join/#asterisk kapejod (~kapejod@p54BCCF5A.dip.t-dialin.net) |
| 10:32.35 | morris | 1: i must prefix the number with 9901372813418 as an example... as i am aware the 9 is truncated from the outgoing number .. that IS an issue.. as it must be just one 9.. but for now im quite happy.. the other issue is.. as soon as the line is answered.. SIP and ZAP both hangup |
| 10:33.00 | morris | so im much closer to having it sorted |
| 10:33.03 | morris | so im reallly please |
| 10:33.03 | morris | d |
| 10:33.07 | Jas_Williams | morris: you will need to restart asterisk after the zapata |
| 10:33.14 | fenlander | morris: is your x100p connected to a bt line directly or another pbx? |
| 10:33.16 | PandemiK | wow |
| 10:33.33 | PandemiK | the manipulation described by grandstream fot factory settings doesn't work |
| 10:33.35 | Jas_Williams | morris: remove the :1 after the EXTEN to stop the 9 being removed |
| 10:33.38 | morris | direct to the (NTL) phone line |
| 10:34.30 | Jas_Williams | morris: if it is direct to an NTL phone line why do you need to send them a 9 ? |
| 10:34.51 | morris | because it is a centrex system |
| 10:34.56 | morris | we have exentions |
| 10:34.56 | Jas_Williams | ah |
| 10:34.56 | morris | etc |
| 10:35.32 | Jas_Williams | morris: remove the :1 from the ${EXTEN:1} then and the 9 will not be removed |
| 10:35.48 | morris | thanks 9 is removed |
| 10:36.05 | morris | erm |
| 10:36.06 | morris | i mean |
| 10:36.07 | morris | lol |
| 10:36.11 | morris | sorted that extention bit out |
| 10:36.19 | Jas_Williams | ok |
| 10:36.39 | morris | so thats awesome ;) |
| 10:36.59 | Zeeek | PandemiK what do they say? |
| 10:37.01 | Jas_Williams | can you turm on sip debug and place the call again and may be I can see why the call is hanging up |
| 10:37.10 | morris | right ok.. so we have incomming and out going.... just the problem.. when calling from SIP to ZAP .. as soon as the call is picked up.. it hangs up ;/ |
| 10:37.18 | morris | ok cool |
| 10:37.27 | PandemiK | Zeeek: I must select --reset-- in the menu then enter the encoded MAC address |
| 10:37.33 | Jas_Williams | post the results to pastebin.ca |
| 10:37.54 | PandemiK | Zeeek: I mean "encoded" because it's 222 for b (like an SMS) |
| 10:38.21 | PandemiK | Zeeek: everything is ok, the MAC address can be entered but the phoen doesn't reboot |
| 10:39.36 | Zeeek | PandemiK I know there is an issue to revert from the newest firmware. but I assumed you didn't have the very newest? |
| 10:39.49 | darwin35 | May 20 05:34:01 WARNING[79015] chan_sip.c: sip_xmit of 0x821141c (len 446) to 65.67.76.53 returned -1: Bad file descriptor |
| 10:39.59 | darwin35 | anyone know what this is |
| 10:39.59 | PandemiK | Zeeek: Program-- 1.0.6.3 I have :-( |
| 10:40.04 | Zeeek | Pandemik isn't there a factory defaults in the web interface? |
| 10:40.10 | morris | Jas_Williams, http://pastebin.ca/12231 |
| 10:40.15 | PandemiK | Zeeek: no, I looked very well |
| 10:40.18 | morris | it just says bye from what i can tell hehe |
| 10:41.02 | tessier | Man...Cisco does not support STUN. |
| 10:41.07 | tessier | That is a major weakness. |
| 10:42.01 | PandemiK | yeah, but Cisco is mainly using H323 |
| 10:42.42 | PandemiK | STUN is essentially for SIP |
| 10:42.42 | Jas_Williams | morris: asterisk is receiving a bye message from the sip device ? RECEIVE << 192.168.1.2:5060 |
| 10:42.42 | Jas_Williams | BYE sip:2000@192.168.1.7:5060 SIP/2.0 |
| 10:42.46 | tessier | But nobody uses H323 with asterisk unless they are either clueless or insane. But lots of people use Cisco phones with asterisk. |
| 10:42.58 | tessier | So how are they doing it? |
| 10:43.02 | PandemiK | How can I know the lastest version for BT102 ? |
| 10:43.09 | Jas_Williams | morris: so it hangs up, what type of sip phone is this ? |
| 10:43.20 | tessier | Probably just using nat=yes and canreinvite=no but that means you have a very inflexible and inefficient voip network. |
| 10:43.24 | tessier | With asterisk proxying all of the traffic |
| 10:43.44 | Jas_Williams | tessier: or a SIP aware router :) |
| 10:44.11 | tessier | Jas_Williams: I doubt people are installing sip aware routers into their customers homes/offices are they? |
| 10:44.14 | tessier | Talk about a lot of hassle. |
| 10:45.03 | Jas_Williams | tessier: thats why iax is so cool. port forwarding works fine with cisco as well as it can work out its external IP address |
| 10:46.20 | tessier | Jas_Williams: Unfortunately nobody has IAX phones. They all have SIP phone. |
| 10:46.28 | morris | Jas_Williams, it seems asterisk is sending bye to the softphone (x-lite) and x-lite is replying with bye also.. |
| 10:46.50 | tessier | I think we are going to have to start pushing Snom phones. |
| 10:46.55 | Zeeek | PandemiK look on their site |
| 10:46.58 | fenlander | morris, jas: could be a codec mismatch - what are you allowing? |
| 10:47.10 | PandemiK | Zeeek: yeah, that's what I'm doing but I cannot find anything |
| 10:47.17 | tessier | And perhaps Grandstreams for home use. I think they support STUN also. |
| 10:47.39 | morris | fenlander, would the asterisk commandline notify me of any such problem ? |
| 10:48.01 | Jas_Williams | morris: post your sip.conf to pastebin |
| 10:48.03 | Zeeek | PandemiK just a sec I'm busy but if you wait I'll find the link |
| 10:48.05 | morris | okeys |
| 10:48.11 | fenlander | morris: probably - might need a high verbose level, can't rememver |
| 10:48.15 | PandemiK | Zeeek: No problem, I'll continue seraching |
| 10:48.18 | PandemiK | Zeeek: thx :-) |
| 10:48.31 | *** join/#asterisk gambolputty (~gambolput@cblmdm72-240-241-108.buckeyecom.net) |
| 10:49.01 | morris | fenlander, i did asterisk -vvvvc to start her so i have high verbosity.. (does that word exist? lol) hmm.. it says nothing about it.. *confused* hehe |
| 10:49.50 | morris | Jas_Williams, and.. anyone else ;p sip.conf: http://pastebin.ca/12232 |
| 10:49.51 | Jas_Williams | morris: in the cli type sip debug and get the debug from there I did not notice this was from X-lite |
| 10:51.24 | Jas_Williams | morris: in sip.conf [2000] add allow=alaw |
| 10:51.27 | morris | would u like a paste bin of it |
| 10:51.51 | Jas_Williams | try the change to sip.conf first |
| 10:53.00 | Zeeek | Jas_Williams you prolly have a quick link to grandstream firmware ? |
| 10:53.03 | morris | ok changed |
| 10:53.06 | morris | its still hanging up :( |
| 10:53.12 | Jas_Williams | sip reload ? |
| 10:53.22 | morris | i had restarted asterisk |
| 10:53.29 | morris | stop now, sudo asterisk -vvvvc |
| 10:54.00 | morris | i will try using reload commands instead in future tho ;p |
| 10:54.01 | Jas_Williams | do a sip debug now from asterisk and post to pastebin |
| 10:54.10 | hellop | is callerid info handled by a module? Mine worked before I reinstalled... |
| 10:54.55 | PandemiK | Zeeek: I found what was going wrong, I changed lot of options in web interface and it works ! Now, i'll check which option it is |
| 10:55.06 | PandemiK | Zeeek: and thx for your help |
| 10:56.51 | morris | Jas_Williams, i hope this helps ;/ http://pastebin.ca/12234 |
| 10:57.05 | Zeeek | PandemiK heh |
| 10:57.53 | morris | enjoy ur lunch Zeeek |
| 10:59.10 | Zeeek | thx |
| 10:59.19 | *** part/#asterisk Zeeek (~icechat5@Zeeek.sustaining.supporter.pdpc) |
| 11:00.46 | Jas_Williams | morris: Zap is hanging up :-( I cannot tellyou why |
| 11:01.15 | morris | *wants to cry* |
| 11:01.16 | morris | ahaha |
| 11:01.51 | *** join/#asterisk zotz (~zotz@208.196.247.140) |
| 11:02.33 | morris | you know, ubuntu linux is quite nice |
| 11:02.39 | morris | not sure why i like it tho |
| 11:03.20 | RoyK | May 20 13:01:56 NOTICE[18951]: chan_sip.c:8084 sip_poke_peer: Still have a call... |
| 11:03.20 | RoyK | May 20 13:02:00 NOTICE[18951]: chan_sip.c:8053 sip_poke_noanswer: Peer '' is now UNREACHABLE! |
| 11:03.20 | RoyK | May 20 13:02:10 NOTICE[18951]: chan_sip.c:6683 handle_response: Peer '' is now REACHABLE! |
| 11:03.25 | RoyK | sounds reasonable |
| 11:03.27 | RoyK | :P |
| 11:04.21 | *** join/#asterisk fafnir (~hello@tdds-gw.Moscow.gldn.net) |
| 11:06.19 | Jas_Williams | morris: turn off sip debug and post the console output when you place a call. |
| 11:07.54 | morris | ok |
| 11:08.06 | PandemiK | hey : does someone hear of problem with brandstream BT with fixed RTP port in firware 1.0.6.3 ? When I set to anything, I don't have any sound when I'm called, and when set to use radom RTP port, everything is fine. |
| 11:08.53 | morris | jas man: http://pastebin.ca/12236 |
| 11:11.11 | Jas_Williams | morris: remove the qualify=500 from [2000] and sip reload, see if that has any effect |
| 11:12.57 | morris | still exiting my friend :( |
| 11:14.28 | Jas_Williams | morris: :-( so connect a phone to the phone line as well as the X100P and see wht you hear on the line ? |
| 11:14.44 | Jas_Williams | when a call is made |
| 11:14.47 | morris | what should i expect |
| 11:15.10 | Jas_Williams | dialtone then dtmf tomes then ringing then answer |
| 11:16.12 | morris | its as soon as they answer |
| 11:16.13 | morris | it dies |
| 11:16.18 | morris | like erm |
| 11:16.27 | morris | i had it before so that.. |
| 11:16.33 | morris | i ring 2000 from sip |
| 11:16.37 | morris | it would activate the line |
| 11:16.40 | morris | and give me a dialing tone |
| 11:16.43 | morris | that was perfect |
| 11:16.45 | morris | i could hear that |
| 11:16.55 | morris | soon as i sent a phone number down.. it would ring as one would expect |
| 11:16.59 | morris | but.. as soon as its answered |
| 11:17.01 | morris | it all hangs up |
| 11:17.03 | morris | *sobs* |
| 11:17.03 | morris | lol |
| 11:17.23 | morris | so i dont know what to listen out fore |
| 11:17.25 | morris | for* |
| 11:17.27 | morris | its definatly calling out |
| 11:17.51 | morris | right im gonna thank you for your help and get some food |
| 11:17.52 | morris | heheh |
| 11:17.58 | morris | thanks, i really apprecaite the support |
| 11:18.02 | morris | been really nice and helpful |
| 11:18.08 | Jas_Williams | I would test this on a non centrex line as it may be doing something strange with the line |
| 11:18.12 | morris | hmm |
| 11:18.13 | morris | i see |
| 11:18.15 | morris | i didnt think of that |
| 11:18.16 | morris | i will do that |
| 11:18.17 | morris | now |
| 11:18.52 | Jas_Williams | Better put back the ${EXTEN:1} as a normal line will not like the 9 :) |
| 11:19.49 | morris | hhe |
| 11:19.49 | morris | thanks |
| 11:19.52 | morris | i just found that |
| 11:19.52 | morris | rofl |
| 11:20.59 | morris | Jas_Williams, your bloody right.. its the centrex line causing me shit |
| 11:21.13 | morris | something i noticed |
| 11:21.17 | morris | there is an echo :( |
| 11:21.40 | Jas_Williams | I suspect the line is doing a polarity inversion on connect which asterisk is detecting as hangup |
| 11:21.53 | morris | ahh |
| 11:21.54 | morris | iojk |
| 11:21.55 | morris | ok* |
| 11:21.56 | morris | thats shit |
| 11:21.56 | morris | lol |
| 11:22.20 | morris | right ok.. i feel much happier knowing what is up |
| 11:22.22 | morris | thank you again |
| 11:22.25 | morris | im gonna get some food |
| 11:22.26 | morris | try again |
| 11:22.28 | morris | brb |
| 11:22.31 | Jas_Williams | to get rid of echo play with gain settings and echo cancelling |
| 11:22.38 | darwin35 | 10.1.3.0 would have a netmask of 255.255.255.0 right |
| 11:22.46 | morris | ah okeys |
| 11:22.48 | morris | thanks |
| 11:22.48 | morris | ehe |
| 11:24.46 | morris | u have no idea how chuffed i am ;p |
| 11:24.46 | morris | brb |
| 11:29.51 | RoyK | Jas_Williams: the SIP polarity is extremely hard to tell |
| 11:31.50 | Jas_Williams | RoyK: sip polarity ? |
| 11:32.01 | Ahrimanes | <PROTECTED> |
| 11:32.19 | Jas_Williams | Ahrimanes: look in the sip debug it may help |
| 11:32.22 | *** join/#asterisk Blissex (~Blissex@82-69-39-138.dsl.in-addr.zen.co.uk) |
| 11:32.36 | Jas_Williams | Ahrimanes: normally inconsistent userid and password |
| 11:34.15 | Ahrimanes | Jas_Williams: ok, but i 4x checked password and username.. guess i should sniff traffic |
| 11:34.32 | Jas_Williams | Ahrimanes: sip debug in the cli will give most information |
| 11:35.33 | Ahrimanes | Jas_Williams: ok, thx |
| 11:37.55 | Ahrimanes | hm |
| 11:39.34 | hellop | I used to get callerid information, after reinstall I don't. Any pointers as to where to start? |
| 11:39.37 | Ahrimanes | it tries WWW-Authenticate: Digest realm="asterisk", nonce="7cf016e7" and gets 403 forbidden |
| 11:39.54 | hellop | To fix it... |
| 11:41.02 | Ahrimanes | Jas_Williams: hm maybe the phone is doing wrong digest or something like that? |
| 11:42.41 | morris | ah i got food... and customers grrr |
| 11:42.46 | darwin35 | what would cause a cisco 7940 to auth its exten but not the pri nmbr |
| 11:43.04 | darwin35 | all the phones are active |
| 11:43.26 | *** join/#asterisk cmk (~cmk_@p54A3E497.dip.t-dialin.net) |
| 11:44.18 | darwin35 | they all keep timing out |
| 11:44.28 | darwin35 | and failing to register |
| 11:44.36 | darwin35 | they register onsce |
| 11:44.47 | darwin35 | I have timeout 200 ms |
| 11:49.45 | *** join/#asterisk meppl (mephisto@p54AAF807.dip.t-dialin.net) |
| 11:50.28 | Jas_Williams | hellop: what type of connection to the pstn |
| 11:53.39 | morris | Jas_Williams, want some chips ? |
| 11:54.27 | Jas_Williams | yes put some mayonaise on them tho |
| 11:54.36 | morris | feck |
| 11:54.38 | morris | i cant believe |
| 11:54.43 | morris | i forgot mayonaisse |
| 11:54.48 | morris | i got ketchup tho ;/ |
| 11:54.55 | Jas_Williams | :) |
| 11:55.12 | morris | these things are a big greasy |
| 11:57.03 | morris | its like taking mouthfuls of the juice left in a chip fryer ;p |
| 11:58.38 | Jas_Williams | Awk!! |
| 11:59.09 | morris | that said, im sure i can manage ;p |
| 11:59.42 | hellop | Jas_Williams, X100P. Maybe my caller ID box is sucking it up. |
| 12:00.20 | hellop | Jas_Williams, I have the the 2 callerid lines in zaptel.conf.. |
| 12:00.33 | Jas_Williams | hellop: are you running the same version of asterisk you were running before or a new model some of the configuration may have changed |
| 12:01.20 | hellop | Jas_Williams, well, fresh CVS, the old one was about 4 months ago. same hardware. |
| 12:01.45 | hellop | It's supposed to just pop up in the CLI right? |
| 12:01.48 | darwin35 | May 20 06:59:47 WARNING[79411]: app.c:613 ast_play_and_record: No audio available on SIP/10.1.3.56-081ba000?? |
| 12:02.24 | hellop | To configure caller id, you add usecallerid=yes, and callerid=asreceived, is there any thing else? |
| 12:03.40 | Jas_Williams | hellop: there are some new options for cli |
| 12:03.42 | Jas_Williams | ; |
| 12:03.42 | Jas_Williams | ; Type of caller ID signalling in use |
| 12:03.42 | Jas_Williams | ; bell = bell202 as used in US, v23 = v23 as used in the UK, dtmf = DTMF as used in Denmark, Sweden and Netherlands |
| 12:03.42 | Jas_Williams | ; |
| 12:03.42 | Jas_Williams | ;cidsignalling=bell |
| 12:03.44 | Jas_Williams | ; |
| 12:03.46 | Jas_Williams | ; What signals the start of caller ID |
| 12:03.57 | Jas_Williams | ; ring = a ring signals the start, polarity = polarity reversal signals the start |
| 12:03.58 | Jas_Williams | ; |
| 12:04.00 | Jas_Williams | ;cidstart=ring |
| 12:04.02 | Jas_Williams | sorry for the flood |
| 12:04.09 | hellop | in zaptel? |
| 12:04.10 | hellop | .conf |
| 12:04.45 | Jas_Williams | in zapata.conf you now need cidsignalling=bell cidstart=ring for us |
| 12:04.59 | darwin35 | http://pastebin.ca/12239 here is a issue calls drop after ring one and no audio on vm |
| 12:05.20 | *** join/#asterisk dalfz (~dman@officewall.workzone.no) |
| 12:06.44 | hellop | Jas, I get "1" now instead of "". We're making progress! |
| 12:07.55 | *** join/#asterisk newbien (~e@147.241.33.65.cfl.res.rr.com) |
| 12:08.23 | hellop | Jas_Williams, ok, the stock .conf files. I deleted it. ok, back on track now, thanks. |
| 12:10.26 | vaewyn | darwin35: not sure on your problem... but this line is still either wrong context or wrong comment: |
| 12:10.27 | vaewyn | include => night|09:00-16:00|sat|*|* ; OPEN w00t! |
| 12:10.32 | vaewyn | :} |
| 12:10.38 | Jas_Williams | darwin35: can you post the cli output on a call that rings once and then stops |
| 12:12.33 | *** join/#asterisk [TK]D-Fender (~joe@4.67.252.216.dsl1.colba.net) |
| 12:15.23 | [TK]D-Fender | !seen ariel_ |
| 12:16.24 | vaewyn | ~seen ariel_ |
| 12:16.45 | jbot | ariel_ is currently on #asterisk (14h 34m 6s). Has said a total of 92 messages. Is idling for 8h 50m 4s |
| 12:17.05 | [TK]D-Fender | Or that ;) |
| 12:18.10 | darwin35 | http://pastebin.ca/12241 ok here is the updated with cli |
| 12:18.28 | darwin35 | its shows the lings ring but no audio |
| 12:18.35 | darwin35 | and yu dont hear the ting |
| 12:18.39 | darwin35 | ring |
| 12:19.08 | dalfz | anyone know if this is a bug? apparently, my asterisk proxy cannot talk sip/2.0 with the external sip server. WARNING[31749]: chan_sip.c:5242 check_via: Don't know how to respond via 'SIP/2.0/udp' |
| 12:19.37 | darwin35 | sorry |
| 12:19.42 | darwin35 | missed a line |
| 12:22.37 | *** join/#asterisk bjohnson (~bjohnson@jecinc.tor.istop.com) |
| 12:25.01 | darwin35 | http://pastebin.ca/12242 |
| 12:25.54 | *** join/#asterisk doughecka (~Tad@doughecka.user) |
| 12:26.33 | darwin35 | any idea |
| 12:27.08 | [TK]D-Fender | vaewyn : Just got quoted 327$CAD for Polycom IP600 (roughly 261$USD). Things are looking good for Poly now.... |
| 12:27.41 | vaewyn | :} Polycoms rock... and they are definately way cheaper for the features |
| 12:28.11 | darwin35 | I am beginning to hate cisco phones |
| 12:28.19 | darwin35 | they keep timing out |
| 12:29.49 | *** join/#asterisk fantomax1 (~fanto@81.208.114.250) |
| 12:31.14 | *** join/#asterisk Ridgeback (~Ridgeback@ppp200-187.lns1.adl1.internode.on.net) |
| 12:32.26 | Ridgeback | hello! hey anyone ever have problems with one switch going UNREACHABLE with dundi show peers? |
| 12:32.48 | *** join/#asterisk exonic (~exonic@209.172.11.54) |
| 12:33.13 | exonic | Hello all |
| 12:33.20 | Ridgeback | hello! |
| 12:33.24 | exonic | I am a strange issue with 2 asterisk servers connected via IAX |
| 12:33.31 | Ridgeback | looks like were it at the moment... |
| 12:33.47 | Ridgeback | whats wrong? |
| 12:34.12 | morris | i wanna pull my eyes out |
| 12:34.16 | morris | grr @ centres |
| 12:34.17 | exonic | I am attempting to take action based on the ${CALLINGPRES} variable, after I send the call to the other server, the ${CALLINGPRES} variable becomes 0 |
| 12:34.19 | morris | centrex |
| 12:35.43 | Ridgeback | exonic, hmmmm could you explain the $CALLINGPRES var please? |
| 12:35.46 | darwin35 | http://pastebin.ca/12242 stops rining and when vm picks up no audio |
| 12:35.53 | Ridgeback | morris, centrex? whats wrong? |
| 12:36.03 | darwin35 | sorr falls off |
| 12:36.25 | exonic | Ridgeback, it's a hex code which states Caller ID Presentation, like if someone dials *67 ${CALLINGPRES} = '35' |
| 12:36.32 | vaewyn | darwin35: http://pastebin.ca/12243 try ditching all those @#@#$ timed includes and go simple like this and see if it fixes your issue |
| 12:37.12 | Ridgeback | exonic, oh ok |
| 12:37.17 | darwin35 | like what |
| 12:37.24 | fantomax1 | does anyone know how to use absolute timeout feature ? |
| 12:37.24 | exonic | One the server which the call comes in on the variable is correct, after it goes over IAX it becomes 0, I really only need a way to get that variable across the IAX channel. |
| 12:37.25 | vaewyn | see my pastebin |
| 12:37.41 | Ridgeback | exonic, asterisk seems to be stripping that var away huh? |
| 12:37.51 | Ridgeback | exonic, perhaps it's local only? |
| 12:38.16 | Jas_Williams | exonic: variables are not passed along IAX |
| 12:38.50 | exonic | Jas_Williams, I know, That kinda of operation makes sense. But it's odd how ${CALLINGPRES} over IAX is set to 0, when it's valid on server1. |
| 12:38.52 | Ridgeback | fantomax1, i believe you set it at the top of your priority i.e. exten => 555,1,AbsoluteTiemoute(60) etc.... |
| 12:39.18 | Ridgeback | hello! hey anyone ever have problems with one switch going UNREACHABLE with dundi show peers? |
| 12:39.39 | vaewyn | Ridgeback: only if the network goes loopy :P |
| 12:39.56 | fantomax1 | ok as firt step ? |
| 12:40.25 | fantomax1 | never used till now , but call stay on for hours otherwise .. i need to close them at 30 min at most |
| 12:40.26 | Ridgeback | vaewyn, lol, true. wierd thing is, it was working fine, then I had to reboot the firewall. I checked the portforwarding stuff, 4520 and 4569 are all there as UDP toward the correct server. |
| 12:40.47 | morris | Ridgeback, Using a x100p clone analogue modem in asterisk... i have a SIP (soft)phone setup which can take calls from the analogue line perfectly fine.. all incomming calls reach sip.. great... |
| 12:40.54 | [TK]D-Fender | vaewyn : Yeah Poly does seem cheaper (since my last Cisco quote was double-discounted). I just want to know that I can do the kind of interactive services I'd like on it and that all of ht normal features are supported in *. no MOH, xfer, multi-line issues, etc.... |
| 12:41.05 | morris | Ridgeback, problem is.. outbound.. i place a call from sip.. as soon as the destination answers.. zap hangs up |
| 12:41.08 | [TK]D-Fender | vaewyn : should be getting samples of IP 500 & 600 in today |
| 12:41.08 | *** join/#asterisk makhtar (~ageller@mail.bulletinnews.com) |
| 12:41.29 | morris | Ridgeback, with BT line this doesnt happen.. it will work normaly.. centrex provided by NTL just hangs up ;/ |
| 12:42.03 | Ridgeback | morris, ah perhaps your fxs_ls or fxs_gs or fxs_ks need to be changed? |
| 12:42.09 | Ridgeback | morris, i.e. signalling? |
| 12:42.10 | morris | i dotn know |
| 12:42.16 | morris | i have it set to ks atm |
| 12:42.35 | vaewyn | [TK]D-Fender: congrats :} yeah... MoH, transfer, conference... etc.. all work great... and the 600s the microbrowser rocks :} |
| 12:42.39 | Ridgeback | morris, try ls, it's what i use out here |
| 12:42.45 | vaewyn | can't wait for the 500s to have the browser |
| 12:42.48 | darwin35 | ay 20 07:42:14 WARNING[88714]: pbx.c:2415 __ast_pbx_run: Timeout, but no rule 't' in context 'night' |
| 12:42.54 | morris | May 20 13:42:37 ERROR[10662]: chan_zap.c:6472 mkintf: Signalling requested is FXS Groundstart but line is in FXS Kewlstart signalling |
| 12:42.57 | Ridgeback | vaewyn, i second that, my IP 600 is fantastic |
| 12:43.18 | morris | May 20 13:43:15 ERROR[10685]: chan_zap.c:6472 mkintf: Signalling requested is FXS Loopstart but line is in FXS Kewlstart signalling |
| 12:43.24 | morris | poo ;p |
| 12:43.35 | Ridgeback | morris, you also need to cahnge it in the other zap config file, one is in /etc/ the other is /etc/asterisk |
| 12:44.25 | morris | i have a symbolic link from /etc/ to /etc/asterisk |
| 12:44.34 | Ridgeback | morris, no! |
| 12:44.46 | morris | am i being stupid |
| 12:44.47 | morris | ;/ |
| 12:44.55 | Ridgeback | morris, well maybe.. you have both a zaptel.conf and a zapata.conf, two different files! |
| 12:45.04 | *** join/#asterisk docelm0 (~docelm0@67.106.194.90.ptr.us.xo.net) |
| 12:45.06 | morris | loadzone=uk |
| 12:45.06 | morris | defaultzone=uk |
| 12:45.06 | morris | fxsks=1 |
| 12:45.17 | morris | fsxfln=1 ? |
| 12:45.22 | morris | change to that |
| 12:45.28 | *** join/#asterisk kape (~kapejod@p54BCCCED.dip.t-dialin.net) |
| 12:45.38 | Ridgeback | fxsls=1 |
| 12:45.40 | morris | ls rather |
| 12:45.42 | morris | thanks |
| 12:45.57 | morris | hmm |
| 12:45.59 | morris | still fails |
| 12:45.59 | Ridgeback | in /etc/zaptel.conf |
| 12:46.03 | Chuji | vaewyn : I think they are End of Life on the IP500's |
| 12:46.12 | Ridgeback | then check /etc/asterisk/zapata.conf |
| 12:46.15 | docelm0 | My ultimate question for the E911 ruling is.. If someone provide BYOD like Asterisk does the E911 still apply? |
| 12:46.16 | Chuji | vaewyn : Should have the 501 out soon |
| 12:46.23 | morris | May 20 13:45:51 ERROR[10732]: chan_zap.c:6472 mkintf: Signalling requested is FXS Loopstart but line is in FXS Kewlstart signalling |
| 12:46.33 | Ridgeback | morris, signalling=fxs_ls in zapata.conf |
| 12:46.37 | vaewyn | Chuji: already do... 301 and 501... but still... same phone... they just beefed the memory |
| 12:47.00 | morris | already set :( |
| 12:47.17 | Ridgeback | morris, then do ztcfg -vv |
| 12:48.53 | morris | line 0: Unable to open master device '/dev/zap/ctl' |
| 12:48.53 | [TK]D-Fender | vaewyn : Is MOH local on the phone or to *? |
| 12:48.53 | darwin35 | is it ,s,1 ot |s|1 ? |
| 12:48.53 | morris | 1 error(s) detected |
| 12:49.31 | [TK]D-Fender | vaewyn : and can you proved cfg files for me to speed up testing? :D |
| 12:49.36 | Ridgeback | morris, hmm check that both files are aligned in configs... there are a plenty of areas that could still need checking |
| 12:50.55 | morris | i may reboot |
| 12:51.02 | morris | not sure why |
| 12:51.03 | morris | but |
| 12:51.03 | morris | <PROTECTED> |
| 12:51.24 | morris | no point really |
| 12:51.26 | morris | im just confused |
| 12:51.37 | Ridgeback | ok give it a try |
| 12:53.43 | fantomax1 | Ridgeback .. do i need only to put that entry for any extension i want to limit in time ? |
| 12:54.19 | Ridgeback | fantomax1, huh? |
| 12:54.37 | fantomax1 | AbsoluteTimeout |
| 12:54.44 | Ridgeback | fantomax1, oh right |
| 12:54.48 | fantomax1 | i need to close a call atmost at 30 min |
| 12:54.58 | Ridgeback | fantomax1, yes that is per extension/call |
| 12:55.04 | fantomax1 | ok thansk |
| 12:55.07 | fantomax1 | thanks a lot |
| 12:55.14 | Ridgeback | fantomax1, no problemo :) |
| 12:55.45 | vaewyn | [TK]D-Fender: * MoH... no need for conf files... webpage on the phones :} and the sip.conf only special is dtmfmode=rfc2833 |
| 12:56.01 | Ridgeback | anyone having issues with dundi showing INVALID packets during a dundi debug? |
| 12:58.50 | *** join/#asterisk Fabe (~spamhere@pD95B0BF7.dip0.t-ipconnect.de) |
| 12:58.54 | *** join/#asterisk morris (~morris@pcworkshop.plus.com) |
| 12:59.00 | morris | wooo Ridgeback |
| 12:59.02 | morris | its bloody sorted |
| 12:59.04 | morris | thank you |
| 12:59.06 | morris | right so.. |
| 12:59.13 | morris | Jas_Williams, thanks for your help.. dude i owe you |
| 12:59.15 | Ridgeback | morris, hey great! |
| 12:59.18 | morris | Ridgeback, same again... dude i owe you |
| 12:59.30 | Ridgeback | morris, no problemo :) |
| 12:59.35 | morris | and.. im sure theres many others that made this possible ;p |
| 12:59.36 | morris | heheh |
| 12:59.42 | morris | Ridgeback, the reboot seemed to fix it |
| 12:59.43 | morris | no idea why |
| 12:59.49 | *** join/#asterisk theHub (~karl@69.177.93.36) |
| 12:59.51 | morris | fxsls=1 |
| 12:59.56 | Ridgeback | probably your zaptel module was stuck |
| 12:59.58 | morris | instead of ks solved it |
| 13:00.08 | morris | ah im soo fucking pleased |
| 13:00.09 | *** join/#asterisk rg1 (~rg1@mail.airlinksystems.com) |
| 13:00.10 | morris | thanks gain |
| 13:00.10 | morris | heheh |
| 13:00.12 | morris | again* |
| 13:00.23 | Ridgeback | lol |
| 13:00.25 | Ridgeback | anytime |
| 13:00.29 | morris | :) |
| 13:00.36 | morris | any suggestion on where to add asterisk to startup ? |
| 13:00.48 | morris | ;/ |
| 13:00.53 | morris | debian |
| 13:00.56 | morris | ubuntu |
| 13:01.01 | RoyK | morris: /etc/init.d/ perhaps? |
| 13:01.05 | morris | okeys |
| 13:01.09 | morris | i said that to my mate |
| 13:01.11 | morris | and he said no |
| 13:01.12 | morris | so.. |
| 13:01.15 | morris | i got worried ;p |
| 13:01.24 | Ridgeback | morris, hmmmm probably local.start somewhere in /etc/ |
| 13:01.39 | RoyK | Ridgeback: don't |
| 13:01.44 | rg1 | quick question - my asterisk server is "stopped" because I killed the process that keeps trying - I changed my zaptel.conf to be correct. Without rebooting, what is the command to try to start Asterisk again? |
| 13:01.46 | RoyK | Ridgeback: use the normal init scripts |
| 13:01.49 | RoyK | Ridgeback: always |
| 13:02.14 | Ridgeback | RoyK, does asterisk come with Gentoo comaptible init scripts? |
| 13:02.24 | RoyK | Ridgeback: perhaps |
| 13:02.43 | RoyK | Ridgeback: otherwise it's not really hard to take a generic script and just modify it to start safe_asterisk |
| 13:02.45 | rg1 | jbot - newbie here - sorry - just hate to keep rebooting because I can't find the command - |
| 13:02.47 | RoyK | which should bve enough |
| 13:03.05 | RoyK | rg1: usually /etc/init.d/asterisk start |
| 13:03.07 | RoyK | rg1: but |
| 13:03.11 | morris | Ridgeback, yea gentoo its easy env-update blah |
| 13:03.16 | syle | rg1 |
| 13:03.18 | RoyK | rg1: if you're experiencing crashes etc |
| 13:03.28 | RoyK | rg1: start it in the foreground using asterisk -gvvvvvvvvvcdn |
| 13:03.28 | syle | rg1: asterisk -vvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvp |
| 13:03.35 | RoyK | syle: not -p |
| 13:03.38 | RoyK | syle: that's stupid |
| 13:03.47 | Jas_Williams | Howabout safe_asterisk in rc.local |
| 13:03.58 | rg1 | checking all this out..... |
| 13:04.23 | syle | whats wrong with -p? |
| 13:04.23 | *** join/#asterisk Nuxi (~nuxi@cust-sdsl-204-250-82-202.bzn-co-i1000-01.bridgeband.net) |
| 13:05.16 | syle | i wouldn;t argue more with your -g |
| 13:05.22 | RoyK | Jas_Williams: better create an init script for it |
| 13:06.01 | rg1 | ok, another dumb question - i changed zaptel.conf - so i need to reload my drivers - anyone know the command to do that? |
| 13:06.32 | rg1 | the module is wctellxp |
| 13:06.56 | Jas_Williams | modprobe wcte11xp -r ? |
| 13:07.05 | rg1 | trying that..... thx |
| 13:07.11 | Jas_Williams | removes |
| 13:07.23 | Jas_Williams | then modprobe it back again |
| 13:08.41 | morris | ah.. |
| 13:08.48 | rg1 | hmmm |
| 13:08.58 | Jas_Williams | ho! |
| 13:09.20 | morris | in init.d there is already a bash script for starting asterisk (called asterisk). but u must edit /etc/default/asterisk needs editing first tho. |
| 13:09.32 | rg1 | jas-williams - that worked - green lights everywhere |
| 13:10.02 | morris | brb |
| 13:11.51 | *** join/#asterisk tzafrir_laptop (~tzafrir@bzq-179-40-134.cust.bezeqint.net) |
| 13:14.10 | *** join/#asterisk SuPrSluG (~SuPrSluG@pool-70-16-33-249.buff.east.verizon.net) |
| 13:14.26 | SuPrSluG | hia ll |
| 13:19.48 | hellop | what ups uperslu gone col done for you |
| 13:23.27 | darwin35 | still having issues with inbound |
| 13:23.35 | darwin35 | outbound is working fine |
| 13:23.36 | rg1 | hey, got my first call to go through - loopstart -> ADIT 600 -> FXO card -> T1 -> wcte11xp -> context-trunk/auto attendant! |
| 13:24.01 | darwin35 | you can dial 4792010150 and see |
| 13:24.19 | darwin35 | 0r 479 2010151 |
| 13:24.55 | rg1 | That was a normal POTS line - now I have a DID-trunk that SBC says is configured as Startdial=delay, remote battery/remote battery. Anyone know what any of that means? |
| 13:25.05 | rg1 | I need to configure my ADIT 600 for it and I don't have a clue |
| 13:29.06 | darwin35 | == Spawn extension (day, s, 3) exited non-zero on 'SIP/10.1.3.56-081f3000' |
| 13:29.13 | darwin35 | this is where it dies |
| 13:29.37 | *** join/#asterisk RoyK (~roy@179.80-203-29.nextgentel.com) |
| 13:29.42 | darwin35 | every time |
| 13:30.17 | *** join/#asterisk km- (~pgrace@brdgw1.rttx.com) |
| 13:30.27 | *** join/#asterisk RoyK (~roy@179.80-203-29.nextgentel.com) |
| 13:30.42 | *** part/#asterisk km- (~pgrace@brdgw1.rttx.com) |
| 13:30.53 | *** join/#asterisk km- (~pgrace@brdgw1.rttx.com) |
| 13:31.09 | *** join/#asterisk astoria (~haydenth@66.235.201.217) |
| 13:31.26 | syle | lol |
| 13:31.34 | syle | checking out asterisk's authors's webcam |
| 13:31.43 | syle | he's using fedora core :) |
| 13:31.56 | [TK]D-Fender | syle : Still looking for hardphones? |
| 13:32.07 | syle | pretty much |
| 13:33.10 | darwin35 | http://pastebin.ca/12250 |
| 13:33.13 | syle | i did find a guy that will get me the cisco 7960 |
| 13:33.18 | syle | for the end of the month |
| 13:33.20 | darwin35 | netweb401 is a good phone |
| 13:33.40 | [TK]D-Fender | Tigerdirect quoted me 337$CAD for Polycom IP 600's |
| 13:33.51 | [TK]D-Fender | Will tell more about later. |
| 13:33.59 | syle | ok |
| 13:34.14 | darwin35 | grr this is not making sense |
| 13:34.25 | indego | [TK]D-Fender: Tigerdirect - The MacOS Tiger developers ^_^ |
| 13:35.07 | syle | 6 line internet phone hmmm |
| 13:35.31 | syle | so you can create 6 different sip accounts for that phone on there basically? |
| 13:36.12 | syle | the prices kinda suck right now, watch in a years time, everyone will have iax phones lol |
| 13:37.32 | darwin35 | http://pastebin.ca/12251 |
| 13:37.53 | fantomax1 | is there anyone familiar with the use of AbsoluteTimeout feature ? |
| 13:41.05 | Jas_Williams | fantomax1: it sets the absolute timeout for the call what do you whant to know |
| 13:42.08 | *** join/#asterisk ariel_ (~Ariel@adsl-068-157-125-248.sip.mia.bellsouth.net) |
| 13:42.11 | Jas_Williams | fantomax1: it has been depreciated in head |
| 13:43.47 | fantomax1 | didn't get Jas |
| 13:43.56 | fantomax1 | I need to limit my calls to 30 mins |
| 13:44.00 | fantomax1 | can I use it ? |
| 13:44.09 | *** join/#asterisk robin_sz (~robin@212.243.40.130) |
| 13:44.52 | Jas_Williams | yes AbsoluteTimeout(1800) |
| 13:45.09 | fantomax1 | in the extension.conf for any extension ? |
| 13:45.43 | fantomax1 | exten => xxx,1,AbsoluteTimeout(1800) ? |
| 13:45.47 | fantomax1 | is it ok ? |
| 13:45.54 | Jas_Williams | yes |
| 13:46.04 | fantomax1 | so the call will be cut at 30 mins ? |
| 13:46.14 | fantomax1 | no additional settings ? |
| 13:46.26 | Jas_Williams | set it at 10 seconds and test :) |
| 13:46.32 | *** join/#asterisk JerJer[mobile] (~jj@ip68-103-26-140.ks.ok.cox.net) |
| 13:46.34 | fantomax1 | yes right :) |
| 13:46.42 | fantomax1 | thanks a lot .. :) |
| 13:47.30 | darwin35 | jas any idea on my issue |
| 13:47.46 | darwin35 | http://pastebin.ca/12252 |
| 13:48.12 | darwin35 | I also get no audio after the first ring |
| 13:49.03 | *** join/#asterisk anthm (~anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 13:49.03 | *** mode/#asterisk [+o anthm] by ChanServ |
| 13:49.30 | blitzrage | does anyone else have an "add indication" in CVS HEAD? |
| 13:49.42 | blitzrage | or was that removed? |
| 13:50.12 | darwin35 | hey blitz |
| 13:50.18 | Jas_Williams | darwin35: no |
| 13:50.35 | darwin35 | doe sit look right |
| 13:51.11 | Jas_Williams | darwin35: yes it looks right |
| 13:51.48 | vaewyn | blitzrage: I don't have it... 4 day old head |
| 13:51.57 | Jas_Williams | blitzrage: I have no application add indication in my cvs Head |
| 13:52.05 | blitzrage | thanks guys! |
| 13:52.33 | *** join/#asterisk christo (~chris@office.enovi.com) |
| 13:53.40 | darwin35 | thnks blitz |
| 13:54.01 | *** join/#asterisk RoyK (~roy@179.80-203-29.nextgentel.com) |
| 13:54.27 | blitzrage | for what? |
| 13:55.08 | darwin35 | anyone having audio issues with head |
| 13:55.16 | darwin35 | with inbound calls |
| 13:55.40 | bkw_ | nope |
| 13:55.48 | bkw_ | darwin35, are you on Linux? |
| 13:56.02 | darwin35 | yes this is a linux bix |
| 13:56.09 | darwin35 | salck 10.1 |
| 13:56.15 | darwin35 | slack 10.1 |
| 13:56.24 | darwin35 | cvs head |
| 13:56.24 | bkw_ | what kind of issues? |
| 13:56.27 | bkw_ | did you make clean |
| 13:56.29 | bkw_ | did you add any modules? |
| 13:56.48 | darwin35 | it rings once then no audio but the cli says the lines are ringing |
| 13:56.53 | RoyK | rm -rf /var/lib/asterisk/modules/ && make clean valgrind |
| 13:57.02 | bkw_ | valgrind isn't going to help him |
| 13:57.03 | vaewyn | bkw_: it's really screwy... he has it ring everything and that goes fine... but it dumps the call when he hits voicemail |
| 13:57.15 | *** join/#asterisk exonic (~exonic@209.172.11.54) |
| 13:57.18 | bkw_ | unless you wanna tell him how use valgrind too |
| 13:57.19 | RoyK | bkw_: make valgrind is nice anyway. makes the stuff debuggable |
| 13:57.28 | bkw_ | its debuggable without that |
| 13:57.37 | bkw_ | ;) |
| 13:57.43 | bkw_ | just not as debuggable |
| 13:57.49 | RoyK | bkw_: a backtrace can be pretty fscked up without make valgrind.... |
| 13:57.56 | exonic | Does anyone know about CallerIDPres, ${CALLINGPRES} variable in the asterisk dialplan. |
| 13:57.56 | bkw_ | nope |
| 13:58.03 | bkw_ | my backtrace always has the correct info in it |
| 13:58.15 | bkw_ | exonic, go to google |
| 13:58.22 | darwin35 | its 2 issues in all |
| 13:58.23 | exonic | bkw_, I have, I can't find anything |
| 13:58.37 | *** join/#asterisk fishboy1669 (proxyuser@62.69.81.129) |
| 13:58.38 | vaewyn | bkw_: http://pastebin.ca/12242 is darwin35's issues |
| 13:58.41 | RoyK | exonic: there's an article or two detailing it on the wiki |
| 13:58.53 | bkw_ | OMG Dial(SIP/1001&SIP/1002&SIP/1003&SIP/1004&SIP/1005&SIP/1010&SIP/1007&SIP/1008&SIP/1009&SIP/1010 |
| 13:58.56 | exonic | !google CALLINGPRES over IAX |
| 13:58.58 | darwin35 | 12252 now |
| 13:58.59 | bkw_ | ever hear a of a flippin call queue |
| 13:59.11 | RoyK | ~callingpres |
| 13:59.24 | RoyK | ~google callingpres |
| 13:59.25 | bkw_ | all those timed includes are crazy |
| 13:59.54 | darwin35 | ok |
| 13:59.59 | vaewyn | bkw_: I had him rewrite the includes to http://pastebin.ca/12252 but still issues |
| 14:00.08 | bkw_ | does ONE phon work |
| 14:00.09 | fishboy1669 | hi any uk bods here? |
| 14:00.12 | RoyK | ~google callingpres site:voip-info.org |
| 14:00.16 | fishboy1669 | if so u may be interested in |
| 14:00.20 | fishboy1669 | http://news.bbc.co.uk/1/hi/business/4555697.stm |
| 14:00.20 | bkw_ | if you have exactly 1 phone in the dial |
| 14:00.22 | bkw_ | does it work |
| 14:00.26 | fishboy1669 | bbc2 7pm |
| 14:00.27 | exonic | RoyK, I know but it says it just works, it says nothing about ${CALLINGPRES} being available over an IAX channel. I even looked into the source of the Dial() app and it has code to send it over the channel. Now what gets me is it's 0 by the time it goes over th IAX channel. I have seached a lot and now seek here for help! |
| 14:00.34 | darwin35 | right now any line you dial rings once and then no audio |
| 14:00.39 | bkw_ | you can't pass vars over IAX2 |
| 14:00.42 | bkw_ | sorry move along |
| 14:00.56 | RoyK | bkw_: no, but you can prefix the number on one end |
| 14:01.08 | bkw_ | well I think iax does calling pres already |
| 14:01.10 | RoyK | bkw_: there still are no IAX variable passing methods in head? |
| 14:01.26 | bkw_ | nope |
| 14:01.30 | RoyK | bkw_: in stable or head? or both? |
| 14:01.38 | Jas_Williams | both |
| 14:02.09 | bkw_ | #define IAX_IE_CALLINGPRES 38 /* Calling presentation (u8) */ |
| 14:02.13 | exonic | Well the thing is it's not trying to "PAss a variable" over a channel, CALLINGPRES is already a variable associated with the channel |
| 14:02.15 | bkw_ | iax will pass calling pres |
| 14:02.17 | bkw_ | NEXT!!! |
| 14:02.18 | darwin35 | bkw look in #996 |
| 14:02.35 | exonic | bkw_, it is passing it, it's coming across as 0! if it wern't passing it it would be "". |
| 14:02.52 | anthm | you can pass vars over iax with my nifty pass vars over iax2 patch =p |
| 14:02.55 | ariel_ | morning everyone |
| 14:03.12 | bkw_ | anthm, yep.. but it was REJECTED... |
| 14:03.14 | bkw_ | haha |
| 14:03.24 | bkw_ | depression.. rejection... the live of an asterisk developer |
| 14:03.40 | fishboy1669 | lol |
| 14:03.40 | RoyK | bkw_: why was it? |
| 14:03.50 | bkw_ | RoyK, take a good guess |
| 14:04.01 | RoyK | because the code looked like mark had written it? |
| 14:04.05 | syle | callingpres interesting |
| 14:04.10 | bkw_ | DOH!!! |
| 14:04.16 | vaewyn | bwahahaha |
| 14:04.18 | anthm | maybe if everyone comes to cluecon |
| 14:04.19 | bkw_ | RoyK, I should slap you for that! |
| 14:04.25 | RoyK | :) |
| 14:04.32 | exonic | well there has got to be a way to do this, the only method I can think of that I know will work is to call DIAL(XXXXXX-${CALLINGPRES}) |
| 14:04.34 | bkw_ | RoyK, you going to come to cluecon? |
| 14:04.37 | syle | if i could find something to change my analog phone numbers callerid that would be great |
| 14:04.50 | vaewyn | reminds me... any work been done on remote MWI yet? |
| 14:04.56 | *** join/#asterisk newmedian (~crowlther@Quebec-HSE-ppp230300.qc.sympatico.ca) |
| 14:05.02 | RoyK | bkw_: don't know yet. I had to cancel going to the VON after I hurt my arm at a party last weekend :( |
| 14:05.03 | exonic | The wiki only has informatin pretaining to ${CALLINGPRES} variable, which states what it is, great! and then the applications setCallingPres() and CallingPres() which are of no help. |
| 14:05.06 | bkw_ | vaewyn, you can use sipsak and externotify |
| 14:05.12 | anthm | bkw, here's one for you, what's the difference between #asterisk amd a gay bar..... the bar has beer |
| 14:05.14 | darwin35 | athm where is that patch |
| 14:05.29 | bkw_ | anthm, HAHAHAHAHAHAHAH |
| 14:05.44 | bkw_ | the bar has jello shots too |
| 14:06.04 | bkw_ | wait let me look around.... |
| 14:06.09 | anthm | patch is in mantis decaying into dust |
| 14:06.11 | bkw_ | nope I see now drag queens here. |
| 14:06.12 | anthm | somewhere |
| 14:06.19 | vaewyn | bkw_: talking like... machine 1 is voicemail server... machine 2,3,4 land SIP phones... 2,3,4 register with 1 to receive notification when their active phones receive vm |
| 14:06.34 | exonic | I'm not gay nor stupid, I woudl really like some advice. |
| 14:06.44 | bkw_ | vaewyn, you just have to send a sip message.... via extern notify |
| 14:06.50 | bkw_ | ;) |
| 14:06.58 | bkw_ | exonic, well you can do the fugly dial hack |
| 14:07.12 | bkw_ | or get anthms patch |
| 14:07.15 | bkw_ | its on the bug tracker |
| 14:07.19 | exonic | bkw_, I know, that's my only solution, It seems like quite the nasty hack. |
| 14:07.21 | bkw_ | you can send everything over IAX then |
| 14:07.22 | *** join/#asterisk sandnigg0r (~niggerplz@66-55-197-254.gwi.net) |
| 14:07.25 | exonic | bkw_, ahh I see |
| 14:07.29 | vaewyn | bkw_: any mail or wiki or such I could look at for more info off the top of your head? or should I hit google? :} |
| 14:07.30 | bkw_ | gotta dig for it |
| 14:07.44 | exonic | bkw_, alright, thank you |
| 14:07.49 | bkw_ | vaewyn, get one of them there sip packets for MWI... |
| 14:07.54 | bkw_ | and template it |
| 14:08.04 | bkw_ | then fire it off to each phone from the main voicemail server :P |
| 14:08.09 | vaewyn | been there... done that... got the tee-shirt |
| 14:08.12 | bkw_ | or better yet |
| 14:08.17 | bkw_ | I just had an evil evil idea |
| 14:08.33 | bkw_ | you can point the voicemail spool to an NFS share on all the servers |
| 14:08.38 | vaewyn | send message via registered IAX2 links |
| 14:08.39 | bkw_ | but only record voicmail on the main server |
| 14:08.50 | *** join/#asterisk MikeJ[Laptop] (~ircatjerr@mi.origenfinancial.com) |
| 14:08.53 | vaewyn | ohh geeze... that's evil |
| 14:08.53 | bkw_ | then when it sees the voicemail it will generate the MWI |
| 14:08.59 | bkw_ | but it would work |
| 14:09.09 | bkw_ | am I wrong? |
| 14:09.12 | vaewyn | yeah... also a ton of polling |
| 14:09.16 | robin_sz | bkw_: sure it would. NEXT! |
| 14:09.20 | bkw_ | go hack it to slow down pooling |
| 14:09.23 | bkw_ | er polling |
| 14:09.34 | mountie | vaewyn: who cares if it's an evil hack. This *is* asterisk, remeber? |
| 14:09.53 | anthm | waht are you trying to do? |
| 14:09.59 | vaewyn | I'm thinking message via IAX2 registered connections much better idea |
| 14:10.22 | vaewyn | much more programming though... *sigh* |
| 14:10.57 | vaewyn | kindof like switch => ... but backwards |
| 14:10.59 | darwin35 | hey bkw where is the iso we talked about |
| 14:11.32 | *** join/#asterisk file[class] (~jcolp@66.199.241.90) |
| 14:11.37 | file[class] | good... morning... I think |
| 14:12.28 | vaewyn | I would just put the vm boxes on the landing machines... but if you have 2 phones with same mailbox they might not be landing on the same server.... :{ |
| 14:12.47 | syle | if you specify a dial plan and you hangup, does the plan execute to finish or just terminate |
| 14:13.11 | vaewyn | terminates and jumps to exten => h,1,.... if it exists |
| 14:13.40 | mountie | vaewyn: What you really want is the "event framework" kpflemming has been talking about for months, and then a way to "distribute" those events over IAX2 trunking to certain servers. |
| 14:13.46 | mountie | But that doesn't exists. |
| 14:14.02 | robin_sz | OK, so ... tell me, I have an IAX user defined with a name and password |
| 14:14.06 | vaewyn | mountie: exactly :} in fact.. talked to him about it at VON |
| 14:14.22 | robin_sz | what will happen if two clients authineticate with the same credentials? |
| 14:14.23 | syle | you guys might find this useful |
| 14:14.27 | file[class] | everyone should come to Cluecon! |
| 14:14.29 | syle | http://www.thenetbrain.com/files/app_machinedetect.c |
| 14:14.51 | syle | i knew their was a way to detect answering machines lol |
| 14:15.09 | file[class] | but is it accurate? |
| 14:15.36 | vaewyn | wonder if kpfleming is going to cluecon... cause that is close... with lots of hackers... :} we could get the event framework going in no time |
| 14:15.59 | file[class] | :) |
| 14:16.03 | file[class] | another reason to go! |
| 14:16.19 | vaewyn | maybe it would be enough time for me to depot my dialplan based voicemail *evil grin* |
| 14:16.27 | vaewyn | debut even |
| 14:16.37 | vaewyn | egads my brain |
| 14:16.38 | *** join/#asterisk morris (~morris@pcworkshop.plus.com) |
| 14:16.59 | morris | Is it possible to specify 2 different signalling methods (zaptel) depending on the call type ? |
| 14:17.05 | syle | well it waits for silence so i don;t know about accurate |
| 14:17.07 | morris | for example, incomming be ls and outgoing ks |
| 14:17.15 | syle | accurate if it was an answering machine |
| 14:17.21 | file[class] | morris: no, try again |
| 14:17.22 | syle | not so much if a real person picked up |
| 14:18.52 | *** join/#asterisk Zeeek (~Zeeek@Zeeek.sustaining.supporter.pdpc) |
| 14:19.23 | blitzrage | does the Answer() command have any arguments such as a channel name? |
| 14:19.34 | file[class] | blitzrage: no... |
| 14:19.37 | blitzrage | or does it just answer any ringing channel? |
| 14:19.37 | Zeeek | I refuse to answer that |
| 14:19.48 | syle | crap |
| 14:19.53 | *** join/#asterisk hellop (~hellop@cpe-70-93-40-171.hawaii.res.rr.com) |
| 14:19.54 | syle | what yum command you use to install pthreads |
| 14:19.54 | file[class] | blitzrage: it answers the channel that calls it |
| 14:20.09 | blitzrage | I guess it just answers any ringing channel coming in on the console channel? |
| 14:20.16 | syle | header files |
| 14:20.18 | blitzrage | sorry, I'm talking about the CLI command |
| 14:20.24 | exonic | So now i'm looking to the source for solutions. Seems in asterisk 1.0.7 doesn't even have IAX_IE_CALLINGPRES in the source code, but CVS does. Asterisk doesn't really support the full IAX standard? |
| 14:20.26 | blitzrage | not the app :) |
| 14:20.29 | file[class] | blitzrage: ah the CLI command :P Answer() is dialplan silly |
| 14:20.37 | file[class] | it answers whatever call is coming in |
| 14:20.39 | blitzrage | file[class]: yah, I'm not quite THAT dumb :) |
| 14:20.43 | blitzrage | file[class]: thanks! :) |
| 14:21.19 | blitzrage | how goes today file? |
| 14:21.31 | file[class] | oh I'm okay... got up at 5:30AM to go to a grad breakfast |
| 14:21.33 | file[class] | so I'm rather tired |
| 14:21.39 | file[class] | and it took an hour and a half to get my pancakes :( |
| 14:21.48 | morris | Zeeek, It seems.. when using BT line, i am able to make an outbound call via SIP and when they pick up.. it stays connected.. BUT Centrex on NTL does not. |
| 14:22.03 | *** join/#asterisk newl (~newlook@203.59.241.42) |
| 14:22.07 | rg1 | zaptel.conf question - i need to provision a port on my te110p to use "emi" (e&m immediate) signalling - anyone know the syntax for that? |
| 14:22.22 | Zeeek | morris, post that immediately to the mailing list and put all those BT users out of their misery! |
| 14:22.26 | newmedian | are you making a flap, jack? |
| 14:22.54 | blitzrage | Mr. Joshua |
| 14:23.01 | file[class] | Mr. Leif |
| 14:23.31 | blitzrage | mmm... I could go for some breakfast |
| 14:23.38 | blitzrage | trying to get a CLI reference done first though |
| 14:23.42 | file[class] | it was good, somewhat |
| 14:23.46 | file[class] | I'm not feeling all that good now |
| 14:23.50 | anthm | maybe you should have lunch |
| 14:23.58 | anthm | cluecon has lunch |
| 14:24.03 | file[class] | indeed |
| 14:24.11 | Zeeek | blitzrage when that happens there won't be anymore questions here |
| 14:24.12 | vaewyn | hahahaha |
| 14:24.13 | newmedian | you need Quebec real maple syrup to go with pancakes. Not generic "table syrup" water glue. |
| 14:24.13 | file[class] | the $650 conference fee INCLUDES all your lunches |
| 14:24.20 | file[class] | as well as hotel, and the normal conference access |
| 14:24.24 | anthm | dine with the most insane developers of our time |
| 14:24.36 | Zeeek | the lunhces must be great at $130/ea |
| 14:24.40 | file[class] | newmedian: the pancakes soaked the syrup up |
| 14:24.41 | vaewyn | newmedian: Nonono... Michigan Maple Syrup.... not that Canadian drool |
| 14:24.43 | vaewyn | :} |
| 14:24.50 | file[class] | added no taste... rather evil |
| 14:25.04 | file[class] | anthm: did David boy get the opteron into the dc? |
| 14:25.12 | anthm | gotta ask him still |
| 14:25.12 | darwin35 | anyone here running head with inbound audio isses |
| 14:25.19 | *** join/#asterisk santiago (~santiago@63.245.86.178) |
| 14:25.26 | file[class] | or did Alan break down and go insane |
| 14:25.50 | blitzrage | vaewyn: *gasp* |
| 14:26.11 | blitzrage | darwin35: HEAD from CVS-HEAD-05/16/05-13:59:55 with no issues |
| 14:26.26 | file[class] | anthm: and I think why he was going on about the money and stuff was because I told him to get my plane ticket soon, so he might be wondering what money to use for that |
| 14:26.37 | blitzrage | Zeeek: lol... do you really think that? |
| 14:26.57 | Zeeek | blitzrage only between 0800-0822 GMT |
| 14:27.14 | Zeeek | before that it"'s all H323 stuff and after |
| 14:27.23 | Zeeek | it's NAT |
| 14:27.38 | darwin35 | ok not current head |
| 14:27.45 | darwin35 | as of this am |
| 14:27.53 | darwin35 | or yesterday |
| 14:28.05 | vaewyn | GIMME HEAD! errrr... funny... but still just not right ;P |
| 14:28.25 | newmedian | well, you can have some drool |
| 14:28.38 | Zeeek | blitzrage in fact I am wondering how to say "please take a look at http://asteriskdocs.org where you'll find that info and more" in some easy to understand way |
| 14:28.39 | bkw_ | Connected to Asterisk CVS-HEAD-05/19/05-19:23:54 currently running on daffy (pid = 29973) |
| 14:28.41 | bkw_ | works fine here |
| 14:28.57 | *** join/#asterisk pointer (pointer@aj.catt.com) |
| 14:29.46 | file[class] | bkw_: never leave me! NEVERRRRRRRR |
| 14:29.47 | *** join/#asterisk tafazzi (~Dario@eporroj0-1.customer.nettuno.it) |
| 14:29.51 | darwin35 | dialing out works fine |
| 14:30.01 | syle | omg compile errors woohoo |
| 14:31.06 | syle | cc -D_GNU_SOURCE test.c i beleive is way to compile in posix recursive threads am i not right? |
| 14:31.21 | file[class] | -lpthread |
| 14:31.40 | syle | naw i already have #include <pthread.h> not that |
| 14:32.00 | file[class] | uh that's just a header file |
| 14:32.06 | file[class] | you actually have to link against the library that provides everything |
| 14:33.44 | syle | problem i am having is with PTHREAD_MUTEX_RECURSIVE |
| 14:34.02 | file[class] | ugh slowly falling asleep |
| 14:34.43 | *** join/#asterisk rube2 (~rube@83-131-18-141.adsl.net.t-com.hr) |
| 14:35.44 | *** join/#asterisk cjk (~cjk@80.92.64.103) |
| 14:36.03 | syle | tmp/ccUNtZLK.o(.text+0x3f): In function `do_waiting': |
| 14:36.03 | cjk | hi, is it possible to get g729 working with oh323? |
| 14:36.09 | blitzrage | Zeeek: thats ok... soon you'll have something to reference for all the newbs |
| 14:36.12 | syle | undefined reference to `ast_set_read_format' |
| 14:36.53 | file[class] | syle: yeekz you're not compiling it right |
| 14:37.15 | file[class] | syle: go look around... |
| 14:37.17 | syle | cc -I../asterisk -D_GNU_SOURCE kk.c even linking in real asterisk dir here |
| 14:37.32 | file[class] | that's wrong though |
| 14:37.59 | file[class] | the easiest way is to put your source in the correct directory, and modify the Makefile... or use astxs from the contrib directory |
| 14:38.19 | Nuxi | I have two telephones on my desk. One goes to a Motorola MTA and the other goes to the wall. Is there some way I can combine these so that I can receive calls from both but on one telephone? |
| 14:38.19 | *** join/#asterisk Dishwasha (~chatzilla@208.251.32.70) |
| 14:38.34 | syle | well it seems that PTHREAD_MUTEX_RECURSIVE problem was a result of not defining _GNU_SOURCE , i;ve done that now new problems lol |
| 14:38.49 | file[class] | syle: just do it the right way like I said |
| 14:38.53 | blitzrage | Dishwasha: where you from geographically? |
| 14:39.01 | Dishwasha | Austin, TX |
| 14:39.11 | Nugget | <-- Austin, TX |
| 14:39.12 | blitzrage | Dishwasha: hahahahaha |
| 14:39.29 | Dishwasha | waso funny? |
| 14:39.48 | Dishwasha | cool, nug |
| 14:39.54 | *** join/#asterisk truescot (~truescot2@213.201.171.186) |
| 14:40.46 | truescot | can anyone tell me what the asterisk command line equivelant of |more is please? |
| 14:40.46 | tzafrir_laptop | asterisk -rx 'command' |less |
| 14:40.46 | Connor- | <PROTECTED> |
| 14:40.56 | Nugget | or just use the scrollback in your xterm/terminal window |
| 14:41.16 | truescot | nope |
| 14:41.17 | tzafrir_laptop | truescot, is you want to code one, take a look at the "pager" feature of the mysql shell |
| 14:41.25 | truescot | running from linux command line |
| 14:42.23 | tzafrir_laptop | truescot, alternatively, run the whole session inside a script(1) session |
| 14:42.27 | PandemiK | is it possible to call a skype number from asterisk ? |
| 14:42.41 | tzafrir_laptop | PandemiK, you can always call. |
| 14:43.03 | PandemiK | tzafrir_laptop: directly online ? even if the user don't have a PSTN number ? |
| 14:43.03 | tzafrir_laptop | But it won't get you anything useful |
| 14:43.23 | truescot | tis cool, i will just connect to it thru a windows machine and scroll back that way, much easier :) |
| 14:43.31 | file[class] | PandemiK: no. |
| 14:43.33 | Nugget | yes, that is the easiest way. |
| 14:43.54 | rube2 | Pandemik: there where some attempts to get the linux Skype API working with *, but nothing usefull yet |
| 14:44.01 | tzafrir_laptop | truescot, you can also record your session when you connect through screen(1) |
| 14:44.04 | Nuxi | truescot, what? you're not running linux on your desktop too? |
| 14:44.12 | Dishwasha | "According to this SFGate article, federal regulators have given VoIP providers 120 days to provide 911 service to their customers." |
| 14:44.12 | PandemiK | rube2: ok, I'm looking on the wiki. but this would be a cool feature |
| 14:44.21 | truescot | i am on one of my desktops but the other is windows |
| 14:44.33 | truescot | plus a mac and a couple of sun machines |
| 14:44.54 | file[class] | PandemiK: Skype is proprietary and closed, all we have is their API and some reverse engineered notes.. nothing concrete to work with |
| 14:45.13 | tzafrir_laptop | PandemiK, skype is a propritary(sp?) protocol, you can't just connect to it |
| 14:45.18 | PandemiK | file[class]: ok, it seems to be a good bounty for this |
| 14:45.20 | truescot | but i am not exactly a linux guru :) can find my way around use vi etc |
| 14:45.30 | file[class] | yeah just because there's a bounty doesn't mean it's possible |
| 14:45.32 | blitzrage | what does 'iax2 provision' do from the CLI? |
| 14:45.45 | fenlander | The Skype API just lets you control their app - I don;t think you could write chan_skype |
| 14:45.50 | file[class] | blitzrage: provisions an IAX2 device of course |
| 14:46.00 | blitzrage | like an iaxy? |
| 14:46.02 | file[class] | yes |
| 14:46.12 | file[class] | you specify the host, and if you want, a template... |
| 14:46.20 | blitzrage | where do I get the template? |
| 14:46.21 | tzafrir_laptop | actually their license has some interesting practical consequeses regading using it for a chan_skype |
| 14:46.24 | blitzrage | and what does it consist of? |
| 14:46.29 | file[class] | lemme get it... |
| 14:46.32 | blitzrage | file[class]: thanks! |
| 14:46.40 | tzafrir_laptop | I'm not sure it would be legal without an interactive user |
| 14:46.57 | file[class] | blitzrage: iaxprov.conf |
| 14:47.04 | blitzrage | haha... thanks! |
| 14:47.50 | file[class] | I can't even get to anything Skype related from here |
| 14:48.17 | tzafrir_laptop | "here" being? |
| 14:48.24 | file[class] | school |
| 14:48.32 | rg1 | has anyone attached an "fxo" TRUNK line to the Asterisk system? |
| 14:48.36 | *** join/#asterisk dos000 (~dos000@66.11.173.123) |
| 14:48.38 | file[class] | Access to this web page is restricted. |
| 14:48.38 | file[class] | Il n'est pas possible d'avoir accès à cette page Web. |
| 14:48.53 | tzafrir_laptop | rg1, me, for instance |
| 14:48.59 | fenlander | Are they trying to cut down on bandwidth? |
| 14:49.11 | file[class] | no they block stuff |
| 14:49.15 | file[class] | like Skype is able to get through their insane firewall ;) |
| 14:49.16 | tzafrir_laptop | unless I misread your Q. |
| 14:49.21 | file[class] | since it uses HTTPS |
| 14:49.25 | fenlander | I've heard that some univsities block sip |
| 14:49.40 | file[class] | all external UDP, ICMP, and almost all TCP is blocked here |
| 14:49.46 | file[class] | only 3 or 4 ports are not blocked |
| 14:49.48 | rg1 | tzafrir-alaptop - what card did you connect that runk to? |
| 14:49.51 | rg1 | trunk |
| 14:50.08 | tzafrir_laptop | A guy did a lecture in our local LUG recently about firewall traversing |
| 14:50.30 | tzafrir_laptop | rgl, so I did miss-read your q. |
| 14:50.32 | file[class] | and I don't have access to do anything to this system, so it's hard to get past it |
| 14:50.49 | rg1 | tzafrir - when I say TRUNK, i mean a DID-TRUNK as opposed to a regular old POTS line |
| 14:51.12 | dos000 | anyone oy you sip expert can take a look at this asterisk sip debug logs (http://pastebin.ca/12259) and give hints as to what is going on ? |
| 14:51.19 | tzafrir_laptop | rgl, if you have a question, better ask it |
| 14:51.32 | file[class] | rg1: the zap hardware/drivers don't support analog DID trunks |
| 14:51.51 | dos000 | the regitration is getting rejected for i do not know why ! |
| 14:52.05 | PandemiK | I'll go SIP, and if Skype doesn't, I simply won't be using Skype. : good conclusion ;-) |
| 14:52.12 | PandemiK | taken from the skype forum |
| 14:52.19 | rg1 | file - right, but what i have is the DID-TRUCK line going into an ADIT 600 |
| 14:52.59 | rube2 | Pandemonik: Skype is using Level 3 for termination anyway |
| 14:53.16 | fenlander | So do skype need to provide 911? |
| 14:53.31 | gambolputty | can iax dialed calls be unauthenticated at all? |
| 14:53.46 | file[class] | rg1: lemme just ask this, what's your question? |
| 14:53.51 | Dishwasha | dos000: well, it's giving you an unauthorized and then a forbidden so I'd imagine you a) don't have the username/auth username correct, b) you don't have the password correct, c) they are not allowing registrations from your IP subnet, d) any of the above |
| 14:53.54 | file[class] | gambolputty: yes they'll go to the guest user in iax.conf |
| 14:54.23 | gambolputty | and then from there to a desired extension? |
| 14:54.33 | file[class] | whatever extension they dialed. |
| 14:54.51 | Dishwasha | fenlander: Probably only for their paid service when they allow you to call POTS |
| 14:55.42 | rg1 | file - what kind of signalling do i need to use and what kind of line do I need from SBC to make this work with Asterisk (i.e. ground start/ e&m/loopstart, etc.) |
| 14:56.08 | file[class] | rg1: well if it's going into the Adit, don't you think you should check the specs on that instead? |
| 14:56.15 | *** join/#asterisk jsolares (~jsolares@200.30.141.86) |
| 14:56.22 | dos000 | Dishwasha, do you know how the response field is made ? i want to compute it my self so i can verify it. |
| 14:56.41 | Dishwasha | dos000: No |
| 14:56.58 | *** join/#asterisk morris (~morris@pcworkshop.plus.com) |
| 14:58.24 | morris | Hmm.. ok Got calls working back and forth reasonably well (give or take a few minor issues) 2 problems faced with atm which are quite bad. 1: The sip phone is able to hear everything said with a .24 second delay at high volume, obviosly that is extremly distracting. 2: volume appears to be quite low generaly on the pstn side. |
| 14:58.29 | morris | any suggestions ? |
| 14:58.29 | morris | ;/ |
| 14:58.45 | morris | .25 * |
| 14:59.11 | dos000 | Dishwasha, can you tell from the logs what the sip device is using as username/auth ? |
| 14:59.19 | fenlander | morris: what was your Zap hangup problem earlier? |
| 14:59.28 | dos000 | Dishwasha, this is the first time i am reading * logs |
| 15:00.18 | Dishwasha | nope, no username listed |
| 15:00.23 | morris | fenlander, hmm, when making connection outbound.. as soon as the destination answered zap would hang up |
| 15:00.39 | fenlander | morris: how did you fix it? |
| 15:00.44 | Dishwasha | Digest username="19990002000" |
| 15:00.48 | Dishwasha | that might be it |
| 15:01.35 | dos000 | Dibbler, what about auth ? |
| 15:02.07 | Dibbler_ | ? |
| 15:02.39 | morris | i think, setting to signalling=fxs_ls and turning off busy detect, hanguponpolarityswitch.. having those off seems ok.. altho.. it doesnt detect hangups lol |
| 15:02.56 | fenlander | oh |
| 15:02.58 | morris | not particularly important |
| 15:03.01 | morris | in this instance |
| 15:03.12 | morris | brb reboot |
| 15:04.21 | dos000 | Dishwasha, what is the caller id for and how does asterisk use/expect it to be ? |
| 15:05.49 | dos000 | Dishwasha, i have an xten device on the same network and it is able to get registered no swet. i see that it sends an encrypted field as caller id tho. |
| 15:07.06 | Dishwasha | dos000: better to pastebin your sip.conf |
| 15:07.11 | *** join/#asterisk cmaj (~chris@65-37-6-42.nrp2.roc.ny.frontiernet.net) |
| 15:08.41 | blitzrage | iax2 show cache command is related to the switch => statement for remote dialplans right? |
| 15:09.09 | file[class] | yes |
| 15:09.13 | blitzrage | w00t |
| 15:09.20 | file[class] | it keeps an internal cache of extensions for an amount of time... |
| 15:09.52 | file[class] | then they expire... and it requeries if the extension is used again |
| 15:10.21 | blitzrage | do you know the expiration time? |
| 15:10.55 | file[class] | 10 minutes by default |
| 15:11.00 | file[class] | 600 seconds... |
| 15:11.06 | blitzrage | where is that configured, iax.conf? |
| 15:11.11 | file[class] | chan_iax2.c |
| 15:11.15 | blitzrage | ahhhh, thanks! |
| 15:11.24 | file[class] | bbl |
| 15:11.29 | rube2 | does anyone know of some useful resource about connecting asterisk to legacy PBX? the wiki section is not very informative.. |
| 15:12.05 | dos000 | Dishwasha, i pasted the relevant section at http://pastebin.ca/12262 |
| 15:13.11 | Dishwasha | dos000: Try putting in an authuser=19990002000 |
| 15:15.48 | dos000 | Dishwasha, still failed .. forbiden same message :-( . yes i reloaded asterisk ! |
| 15:15.59 | Dishwasha | i don't see your register=> line |
| 15:16.15 | dos000 | Dishwasha, in where ? |
| 15:16.57 | Dishwasha | your pastebin |
| 15:18.06 | dos000 | Dishwasha, thats all i get .. asterisk first returns a trying a message then it does .. forbiden |
| 15:18.53 | *** join/#asterisk jeremywhiting (~jeremy@70-56-99-134.slkc.qwest.net) |
| 15:19.00 | Dishwasha | dos000: do you know what a register=> line is? |
| 15:19.01 | dos000 | Dishwasha, at the very end tho there is a log about registration failing |
| 15:19.09 | Dishwasha | it's in your sip.conf |
| 15:20.16 | dos000 | Dishwasha, ok .. i see what you mean now. but no i do not know what register => would do ! forgive my ignorance |
| 15:20.30 | truescot | can anyone tell me sinply how to set up 2 different cards (dev kit and te110p) in the zapata.conf file so that i can have seperate values for things like busydetect and callprogress? |
| 15:20.48 | Dishwasha | dos000: That's what actually initiates the registration process |
| 15:20.48 | Dishwasha | http://www.voip-info.org/wiki-Asterisk+config+sip.conf |
| 15:21.11 | *** join/#asterisk enirob (~enirob@ool-18bdeca2.dyn.optonline.net) |
| 15:21.43 | dos000 | Dishwasha, i have an xten softphone on the network .. and it is able to register with asterisk. btw i am using the sip.conf that was generated by make samples. do you want to see the whole sip.conf ? |
| 15:21.50 | *** join/#asterisk jsolares (~jsolares@200.30.141.86) |
| 15:21.52 | enirob | one of the phone lines plugged into my asterisk pbx went down this morning, and i couldn't make outbound calls even though i have 6 lines plugged in |
| 15:22.11 | enirob | is there a way that asterisk can detect that the line has no dial tone and choose a different line/ |
| 15:22.12 | enirob | ? |
| 15:22.27 | enirob | automatically (without me having to change the config whenever a line goes down) |
| 15:22.35 | *** join/#asterisk jas_williams (~Jason@host217-43-100-176.range217-43.btcentralplus.com) |
| 15:22.44 | Dishwasha | dos000: who set up the xten softphone? |
| 15:22.46 | syle | chanisavail i think |
| 15:23.05 | truescot | einrob, i am no asterisk genious but can you not jsut group together the outbount lines and choose first available? |
| 15:23.08 | dos000 | Dishwasha, from the looks of it looks like you misunderstood what i am trying to do. |
| 15:23.42 | dos000 | Dishwasha, asterisk is the server .. i dont want it to be a client . which the register line would do. |
| 15:23.54 | enirob | truescot, that's what i'm trying to figure out how to do |
| 15:24.20 | Dishwasha | dos000: client/server potato/potaato |
| 15:24.39 | jas_williams | Dishwasha/Sink |
| 15:24.39 | dos000 | Dishwasha, asterisk needs to act as the sip proxy here |
| 15:24.43 | syle | http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20ChanIsAvail |
| 15:24.58 | Dishwasha | dos000: Then you have the wrong product, * does not do sip proxy, SER does |
| 15:25.23 | dos000 | Dishwasha, tow ! |
| 15:26.01 | enirob | thanks, syle |
| 15:26.15 | dos000 | Dishwasha, do you care to explain how i am able to register to asterisk with my xten softphone and i am able to make extension calls ? |
| 15:26.29 | *** part/#asterisk enirob (~enirob@ool-18bdeca2.dyn.optonline.net) |
| 15:26.49 | dos000 | Dishwasha, what the hell does asterisk do as far as sip is concerned ? |
| 15:26.53 | Dishwasha | dos000: I'm not sure what kind of explanation you're looking for. |
| 15:27.18 | *** join/#asterisk PTG123 (~PTG123@ip68-106-24-139.ph.ph.cox.net) |
| 15:27.33 | Dishwasha | dos000: it's a SIP capable PBX, it will register local SIP clients and forward SIP requests to an external peer is one possible scenario |
| 15:27.45 | dos000 | Dishwasha, i have a sip device on the network and i am trying to make asterisk use it as an end point |
| 15:27.45 | *** part/#asterisk truescot (~truescot2@213.201.171.186) |
| 15:28.01 | jas_williams | dos000: that will work |
| 15:28.02 | *** join/#asterisk af_ (~af@ip-181-7.sn1.eutelia.it) |
| 15:28.32 | jas_williams | dos000: but all rtp traffic may travel through asterisk |
| 15:29.14 | dos000 | jas_williams, what does it take ? i did the apropriate section in sip.conf. the xten software has no problem with it. however this sip device (azacall200) just gets a forbiden message back from asterisk at register time |
| 15:30.04 | jas_williams | dos000: post sip.conf and a sip debug for me to look at |
| 15:30.05 | Dishwasha | dos000: ah, my misunderstand, I figured azacall200 was a VoIP provider not a local phone |
| 15:30.08 | *** part/#asterisk km- (~pgrace@brdgw1.rttx.com) |
| 15:30.12 | dos000 | jas_williams, tow ! ... but i'll worry about it later .. once i pass demo stage ! |
| 15:31.16 | dos000 | jas_williams, Dishwasha i'll paste the whole sip.conf for your reading pleasure ... i apreciate your patience ! |
| 15:31.29 | jas_williams | dos000: if it is on a static IP address it does not need to register to asterisk |
| 15:31.50 | *** join/#asterisk naula (~nschmidt@67.154.228.132) |
| 15:32.04 | *** part/#asterisk PandemiK (~PandemiK@62.2.255.42) |
| 15:32.08 | *** join/#asterisk twilson (~terry@63.77.68.11) |
| 15:32.14 | jas_williams | dos000: but it can |
| 15:32.33 | jas_williams | dos000: don't forget to blank out passwords |
| 15:32.55 | Dishwasha | dos000: http://www.voip-info.org/wiki-AzaTel |
| 15:34.12 | twilson | Anyone using realtime sip want to try out the patch at http://bugs.digium.com/view.php?id=4138 It fixes a registration issue that was preventing sharing a realtime SIP db backend with multiple * boxes. Just need to make sure that it doesn't break anything for anyone. |
| 15:34.36 | *** join/#asterisk jsolares (~jsolares@200.30.141.86) |
| 15:35.00 | *** join/#asterisk iCEBrkr (icebrkr@chrome.cyberdyne.org) |
| 15:35.02 | dos000 | jas_williams, http://pastebin.ca/12264 http://pastebin.ca/12259 no worries about passwd .. this demo passwd ! |
| 15:37.01 | Dishwasha | dos000: I think they problem is going to be in your azatel device configuration |
| 15:37.24 | *** join/#asterisk Blackthorn (blackthorn@ws-10.smyth.net) |
| 15:37.32 | darwin35 | ok its work when you dial from a cell phone or a normal home phone but when you dial asterisk to asterisk sip ro sip it looses the ring and audio |
| 15:37.57 | Blackthorn | Hello. I was just reading the recent fcc order about e911. Whats the take on this? And what can we do (if possoble) to comply with this? |
| 15:38.02 | dos000 | Dishwasha, ooh nooo ! |
| 15:38.57 | Dishwasha | Blackthorn: Are you a VoIP provider? |
| 15:39.29 | Blackthorn | I am providing bussiness and home telephone service to several customers via wireless network in our town. |
| 15:40.06 | Dishwasha | Blackthorn: Do you use a VoIP provider to bridge to the PSTN or do you resell another service that connects to the PSTN? |
| 15:40.10 | Blackthorn | sending the LD to nuphone, and local did (and transfered numbers from ilec) is attached to a local voice pri. |
| 15:40.11 | *** join/#asterisk funxion (~chatzilla@mtnuser.icgws.com) |
| 15:41.13 | Blackthorn | I send local calls out our voice pri. |
| 15:41.34 | Dishwasha | Blackthorn: I would get a single dedicated phone line per location, make sure that line is tied to the address of your customer in the 911 database, and route 911 calls to those lines per customer |
| 15:42.00 | dos000 | jas_williams, any idea what is going on ? |
| 15:42.06 | Dishwasha | Blackthorn: You could just dial 911 to your local PRI, but it would give your address and not your customers' |
| 15:42.40 | Blackthorn | adding a pots line to each location defeats the reason to have voip services delivered to the customer. |
| 15:43.59 | dos000 | Dishwasha, i have not seen anywhere ppl making azatel boxes configured with asterisk .. maybe there is a catch somewhere |
| 15:44.26 | newl | Most businesses have fax lines..there's your extra PSTN service. |
| 15:44.38 | astoria | Blackthorn: if you're just a CPE installer, IANAL but I don't think you have to comply with this order. |
| 15:44.52 | astoria | this was only for service providers placing calls between VoIP and the PSTN. |
| 15:45.14 | astoria | people like NuFone. |
| 15:45.48 | astoria | However, if you're installing a PBX for a business you ought to provide 911 capabilities, ethically.. |
| 15:46.35 | Blackthorn | I have one voice pri. any local number that is dialed is sent out that pri. Right now if somone hits 911 it just forwards them to the 911 office phone. |
| 15:47.01 | astoria | If you have a voice PRI then 911 should go to the right PSAP. |
| 15:47.04 | astoria | Thats not VoIP |
| 15:47.22 | astoria | Unless you have some kind of virtual PRI |
| 15:48.02 | Blackthorn | when voip customer dials 911, i could send it out the pri. i think the callder-id will come up correctly. |
| 15:48.34 | *** join/#asterisk morris (~morris@pcworkshop.plus.com) |
| 15:48.36 | astoria | Yeah. You could do that. If your customer is not on your premises, the PSAP would have your address though. |
| 15:49.11 | Dishwasha | astoria: correct, which means you get sued when the firedepartment shows up on your doorstep and your customer burns to a crisp |
| 15:49.36 | astoria | Yep. |
| 15:50.03 | newl | The solution is to correctly disclose the limitations of the service to avoid misguided expectations. :) |
| 15:50.22 | Dishwasha | newl: Nope, not gonna cut it anymore |
| 15:50.48 | newl | Maybe not back home..works a treat here in Oz though. :) |
| 15:51.26 | *** join/#asterisk [Outcast] (~astlivehe@c-24-218-94-11.hsd1.ma.comcast.net) |
| 15:51.26 | Dishwasha | newl: You have to information your customers about the limitations for the next 120 days, but after 120 days you have to provide the service |
| 15:51.40 | Blackthorn | A few weeks ago i talked to the 911 cordinator in our county. basicly they said they were expecting ani information when a 911 call arrives. |
| 15:52.00 | Dishwasha | "The vote came after testimony from people including a Florida woman who had her infant die after being unable to call 911 from her internet phone. . VoIP providers are also required to notify their customers of the deadline and of the limitations of VoIP 911 service." |
| 15:52.27 | Blackthorn | and if you don't provide it in 120 days you must shutdown. |
| 15:53.05 | Dishwasha | blackthorn: so if you can map callerid and registery the callerid with 911 with the customer's address you should be fine then |
| 15:53.44 | funxion | can anyone explain tha admin feature for meetme? |
| 15:53.56 | Blackthorn | thats what i'm thinking too dishwasha... I'm thinking i'm going to have to update the address information with my pri provider of each telephone number that is assigned to my pri. |
| 15:54.10 | Dishwasha | blackthorn: sounds like a plan |
| 15:54.52 | funxion | it doesnt seem to work |
| 15:54.56 | Zeeek | guys, what about if you're in Mexico or Europe and call 911 on nufone? |
| 15:55.04 | funxion | lol |
| 15:55.36 | Zeeek | isn't all this 911 just a foot in the door to begin regulating, then TAXing voip? |
| 15:56.10 | Zeeek | then a secret "back door to monitor" will be required |
| 15:56.26 | Zeeek | pretty soon, people will start disappearing... |
| 15:56.54 | Dishwasha | Zeeek: yes, saving babies from dying is just a smokescreen to give more money to the efforts in IRAQ so we can keep the economy pumping |
| 15:57.30 | Zeeek | I understand that part, there are many examples of this with cellphones as well |
| 15:57.44 | cjk | is there a way to store registration information in the databasse |
| 15:57.51 | cjk | i mean in mysql for example |
| 15:57.55 | Blackthorn | well, it could be zeeek. But the law states comply or they are goign to shut you down. I doubt the fcc will spend the tme to do that. but you havea lot of 911 cordinator that will report it just cuz they want there $$$$ |
| 15:59.06 | darwin35 | ok its work when you dial from a cell phone or a normal home phone but when you dial asterisk to asterisk sip ro sip it looses the ring and audio |
| 15:59.13 | Zeeek | Honest to God, if someone's voip phone was connected to say, BV and there was an outage, dialing 911 won't help. I guess I still don't see voip as rock solid. |
| 15:59.35 | darwin35 | also http://pastebin.ca/12266 all the lines register but the main ones |
| 16:00.24 | Nuxi | Most peoples phones don't work when the power goes out anyway. Cordless, built in answering machine. Doodads. You have to plug most phones into the outlet and the wall these days. |
| 16:00.48 | Zeeek | Thats why I still use corded phones |
| 16:00.58 | Zeeek | (at least one in each location) |
| 16:01.18 | Dishwasha | Nuxi: I guess you've never heard of generators |
| 16:01.28 | Dishwasha | or ups's |
| 16:01.40 | Nuxi | sure, I everybody has those now days in their home. NOT. |
| 16:01.41 | darwin35 | for 911 |
| 16:01.49 | *** join/#asterisk jarrod (jarrod@t06700.humbled.org) |
| 16:02.02 | newl | Dishwasha: I don't live in the US any more so those regulations do not apply. :) |
| 16:02.03 | Zeeek | Let me give an anecdotal bit of info here, YMMV... I'm very old. I've lived in the USA and in Europe. I can only remember one time when I didn't have dialtone on the PSTN. Now, let's talk about voip... different story :) |
| 16:02.09 | darwin35 | what if the powersupply dies in the pc |
| 16:02.14 | darwin35 | or the mb fries |
| 16:02.15 | jarrod | so is asterisk not able to translate a sip ulaw session to xxx g729 session? |
| 16:02.25 | darwin35 | or the raid bites the dust |
| 16:02.29 | Dishwasha | Nuxi: Sorry, I don't think of home I think of businesses when you say everybody |
| 16:02.45 | Nuxi | It's the homes that are suing 'cause kids are dying. |
| 16:02.52 | darwin35 | you have to buy a codec licence from digum |
| 16:03.03 | Zeeek | my own opinion is still have a fallback PSTN line where possible, and if you want a cellphone as well |
| 16:03.12 | newl | they're too lazy to use their cell phones ;) |
| 16:03.37 | darwin35 | then you have to put a asterisk box in each location with a tdm in it |
| 16:04.01 | darwin35 | and make the 911 calls go to the tdm |
| 16:04.07 | Zeeek | in fact I just stuck a hw manual switch on our main office phone that connects it directly to the incoming phone line in case asterisk is off for any reason |
| 16:04.40 | darwin35 | then your stuck with 1 phone |
| 16:06.00 | iCEBrkr | Fucking 2.6 kernel |
| 16:06.00 | newl | So? Most offices are stuck with zero phones when the power goes out because they're too cheap to purchase failover cards for their respective PBX. |
| 16:06.02 | iCEBrkr | oops |
| 16:06.04 | iCEBrkr | wrong channel |
| 16:06.05 | Nugget | Linux is poo. |
| 16:06.06 | iCEBrkr | :X |
| 16:06.07 | jarrod | darwin: i have |
| 16:07.22 | funxion | does anyone know if the admin function of meetme works or not? |
| 16:07.30 | *** join/#asterisk crimd (~crimd@168-215-110-210.gen.twtelecom.net) |
| 16:08.12 | *** join/#asterisk jas_williams (~Jason@host217-43-100-176.range217-43.btcentralplus.com) |
| 16:08.36 | Nuxi | That's why I run * on Windows. lol |
| 16:09.16 | *** join/#asterisk vandien (~stephan@p5090552F.dip.t-dialin.net) |
| 16:09.26 | Qwell | Nuxi: Haven't your parents ever told you not to play with broken glass? |
| 16:10.21 | *** part/#asterisk vandien (~stephan@p5090552F.dip.t-dialin.net) |
| 16:10.38 | newl | That's classic. :) |
| 16:11.24 | cjk | is there a way to store registration information in mysql an not in the astdb file ? |
| 16:11.33 | Qwell | newl: yeah, I thought it was good :p |
| 16:11.42 | darwin35 | res_mysql |
| 16:11.48 | darwin35 | or odbc |
| 16:11.53 | dos000 | Dishwasha, jas_williams, i finally figured what was wrong ... thanks a MILLION ! |
| 16:12.00 | Dishwasha | dos000: Great! |
| 16:14.26 | cjk | darwin35: ok odbc sounds good. what is the module called? |
| 16:14.47 | *** join/#asterisk mcmay (~mcmay@c-24-18-171-179.hsd1.wa.comcast.net) |
| 16:16.42 | darwin35 | look in you conf files |
| 16:16.45 | darwin35 | its there |
| 16:17.35 | *** join/#asterisk cpatry (~grepmoo@65.39.228.5) |
| 16:17.49 | cjk | darwin35: i am using odbc for cdr etc..... but couldnot found it for the astdb |
| 16:20.09 | jas_williams | dos000: I missed the solution what was the problem for reference ? |
| 16:20.32 | *** join/#asterisk Aze` (~aze@host229-162.pool80105.interbusiness.it) |
| 16:22.10 | *** join/#asterisk Corydon-w (forest@vcchgate.vcch01.springfield.tn.us.vcch.net) |
| 16:22.27 | *** join/#asterisk Patrick^ (~patrickm@dhcp-0-e-2e-1f-77-1a.cpe.mountaincable.net) |
| 16:23.29 | *** join/#asterisk yxa (empty@cm121.gamma228.maxonline.com.sg) |
| 16:23.48 | jarrod | did someone mention an available g729 'free' codec used for 'testing' purposes? |
| 16:24.08 | Chuji | NO, don't ask that here |
| 16:24.17 | jarrod | it would be for 'testing' only |
| 16:24.17 | Chuji | It's 10 flippen bux |
| 16:24.18 | jarrod | duh |
| 16:24.30 | jarrod | yea but it takes 24 hours |
| 16:24.32 | jarrod | i want it now |
| 16:24.33 | jarrod | heh |
| 16:24.44 | Qwell | So call yesterday |
| 16:24.57 | jarrod | my time machine's flux capacitor is broke |
| 16:24.58 | jarrod | :( |
| 16:25.07 | Chuji | Order it right now, I'll bet it's to you in a couple hours |
| 16:25.11 | Chuji | Mine always come fast |
| 16:25.55 | Chuji | By the time you j0rk with the illegal one, you could have it |
| 16:26.08 | jarrod | hey can i buy multiples |
| 16:26.16 | Chuji | Yup |
| 16:26.22 | jarrod | er i mean |
| 16:26.34 | jas_williams | Of course one is not going to be much use |
| 16:26.58 | Jedirl | outside the US, it's still illegal to use the ''''free'''' G729? |
| 16:27.14 | Jedirl | as software patents don't apply (yet) in europe... |
| 16:27.37 | jarrod | can i buy 23 now |
| 16:27.39 | jarrod | and then later buy 23 more |
| 16:27.43 | jarrod | and have 46 on my box? |
| 16:27.43 | jas_williams | it's illegal in any country with copyright laws Europe included |
| 16:27.44 | Blackthorn | I am using the g.726-40 byte codec across my wireless network. any reason to go to g729 will it sould just as good or better? what would be the benifit? |
| 16:28.08 | Jedirl | jas_williams: but because of the license of the free one? or because of patent grants? |
| 16:28.24 | jas_williams | jarrod: yes of course but will only beable to use 23 now |
| 16:28.40 | jarrod | but when i add 23 later |
| 16:28.43 | jarrod | ill have 46 available |
| 16:28.45 | jarrod | yes? |
| 16:28.46 | Jedirl | I'm not going to use g729 anyway, just curious, I just use a-law |
| 16:28.50 | jas_williams | yes |
| 16:32.16 | darkskiez | is it better to run asterisk on a P4 with or without hyperthreading |
| 16:32.17 | *** join/#asterisk smash- (~smash@198.107.16.189) |
| 16:33.01 | smash- | hey i got a question, for a sangoma a104 card. On my PRI order form, they are asking me. Coding, Framing format, and Signaling. |
| 16:33.18 | funxion | Im trying to setup synamic meetme rooms where only an authenticated user can create but users with pin for meetme can join can anyone help me? |
| 16:33.29 | smash- | does anyone know what kinda frame format it should use? or that ur zaptel cards use >? |
| 16:33.32 | *** join/#asterisk loick (~loick@APuteaux-151-1-26-179.w82-124.abo.wanadoo.fr) |
| 16:33.48 | darkskiez | smash-, its probably for their statistics rather than anything to do with the order. |
| 16:33.54 | smash- | do pri's just use standard esf , ansi and d8bz |
| 16:34.02 | darkskiez | but i could be very wrong. |
| 16:34.02 | *** join/#asterisk newmedian (~crowlther@Quebec-HSE-ppp230300.qc.sympatico.ca) |
| 16:34.04 | funxion | b8zs esf |
| 16:34.59 | funxion | is that noone uses the meetme feature or noone listening |
| 16:35.30 | darkskiez | maybe nobody uses dynamic ones. |
| 16:35.32 | funxion | framing format would be like NI2 or dms100 |
| 16:35.48 | darwin35 | this sucks |
| 16:36.15 | darkskiez | if it makes you feel any better nobody answered my question either, and i'm sure people run * on P4's |
| 16:36.16 | darwin35 | the pbx works but has auth isses and asterisk to asterisk issues |
| 16:36.33 | funxion | darkskiez: heres the thing I'm trying to make a meetme room where noone can join unless the admin is already in the conf room but the admin can leave the conf and the users can stay |
| 16:36.50 | funxion | what was ur ? darkskiez |
| 16:36.56 | darwin35 | read the meet me wiki |
| 16:37.00 | funxion | o |
| 16:37.01 | darwin35 | it explains how to do that |
| 16:37.02 | newmedian | patience is key to #asterisk. People may be afk, bored, out watching Star Wars, unable to respond, unwilling to respond |
| 16:37.04 | funxion | hyperthreading |
| 16:37.05 | darkskiez | funxion, hyperthreading - on or off, |
| 16:37.16 | darkskiez | (with a te110p card too) |
| 16:37.19 | funxion | I have it on |
| 16:37.23 | funxion | I have the sam card |
| 16:37.38 | smash- | funxion do you have sangoma card? |
| 16:37.39 | funxion | sam=same |
| 16:37.42 | funxion | no |
| 16:37.47 | funxion | I have te110p |
| 16:37.55 | darkskiez | currently CPU0 is getting the timer interrupts and CPU1 sis getting the te110p interrupts. |
| 16:38.07 | *** join/#asterisk paulc (~paulc@S010600062586a0b4.vc.shawcable.net) |
| 16:38.08 | funxion | darwin35:I have read but what they suggest does not work |
| 16:38.16 | darkskiez | i've not plugged it into a line yet, its just doing that cool led strobe. |
| 16:38.23 | *** join/#asterisk buddah (~hnic@208.179.86.5) |
| 16:38.26 | funxion | lol |
| 16:39.03 | darkskiez | i cant figure out what makes the pri card channels 1-31 and the tdm400 channels 32-35 |
| 16:39.22 | darkskiez | why could the tdm card not be channels 1-4 and the pri 5-35 |
| 16:39.28 | buddah | is there a place other than in the sip entry i set codecs that get sent? |
| 16:39.34 | smash- | funxion is ezf a valid format? |
| 16:39.54 | buddah | i have disallow=all, and allow ulaw and alaw for this one quintum gateway, and it seems its getting sent g729 still |
| 16:39.54 | funxion | smash no |
| 16:39.57 | funxion | esf |
| 16:40.02 | funxion | extended super frame |
| 16:40.05 | smash- | yeah |
| 16:40.09 | funxion | its the framing of the circuit |
| 16:40.10 | smash- | guy at sangoma told me ezf |
| 16:40.19 | smash- | i dun know y |
| 16:40.23 | darkskiez | its german, extended zuper frame. |
| 16:40.26 | funxion | lol |
| 16:40.27 | smash- | oo |
| 16:40.32 | jarrod | hey is there a term associated with having an asterisk box forwarding a call from say a sip proxy to another asterisk box so that the original sip session negotiates with the asterisk box that will terminate the call |
| 16:40.57 | jas_williams | darkskiez: the module load order sets the channel numbers so the PRI must have loaded first |
| 16:41.04 | smash- | and signaling is Ansi? |
| 16:41.40 | darkskiez | jas_williams, right, cheers, currently my modules seem to be loaded by hotplug, so are they going to stay in the same order.. |
| 16:41.45 | jas_williams | jarrod: a reinvite ? |
| 16:42.28 | jarrod | yea |
| 16:42.47 | funxion | smash |
| 16:42.47 | jarrod | hwo would i tell an asterisk gateway to send a reinvite to use another gateway |
| 16:42.51 | newmedian | darkskiez/funxion, I thought Hyperthreading was an ungood in general, e.g. http://lists.digium.com/pipermail/asterisk-users/2004-June/049705.html |
| 16:42.58 | smash- | yeah? |
| 16:43.15 | funxion | one sec |
| 16:43.22 | smash- | ~docs |
| 16:43.23 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 16:43.34 | funxion | framing and coding is B8zs ESF |
| 16:43.50 | darkskiez | i did manage to crash my machine |
| 16:43.59 | darkskiez | i ran ztcfg whilst asterisk was running |
| 16:44.17 | darkskiez | locked hard. |
| 16:44.20 | smash- | fun |
| 16:44.37 | newmedian | It's all fun and games until someone loses a Production Asterisk. ;) |
| 16:44.39 | funxion | I use dms100 for format |
| 16:44.45 | funxion | or switchtype |
| 16:44.48 | darkskiez | its not production yet. |
| 16:44.53 | darkskiez | will be in a month |
| 16:44.58 | funxion | and u need to specify who is net and who is user |
| 16:45.01 | smash- | whats the diff? |
| 16:45.04 | darkskiez | im soooooooo tempted to run cvs head. |
| 16:45.22 | funxion | well with pri one side has to be net and one side has to be user |
| 16:45.27 | funxion | asterisk can do both |
| 16:45.32 | jas_williams | jarrod: canreinvite=yes in sip.conf for the peer but it may not work |
| 16:45.41 | funxion | some switches/pbx's can only be user |
| 16:45.50 | *** join/#asterisk m0f0x (m0f0x@m0f0x.user) |
| 16:45.59 | funxion | darkskiez that sholdnt happen |
| 16:46.04 | bkw_ | I run cvs-head |
| 16:46.08 | bkw_ | it works great |
| 16:46.46 | jas_williams | I run widows it works great *!KA~..... |
| 16:47.20 | jas_williams | CVS-Head is fine just some options keep changing between versions |
| 16:47.20 | bkw_ | widows? |
| 16:47.33 | bkw_ | yep and i'm in the middle of all those changes |
| 16:47.34 | bkw_ | haha |
| 16:47.37 | *** join/#asterisk [Outcast] (~knoppix@c-24-218-94-11.hsd1.ma.comcast.net) |
| 16:48.38 | jas_williams | bkw_: I'm not sure I like the set application. |
| 16:48.47 | [Outcast] | does anyone know of any tools with tell me what dtmf button was pushed from a audio file. |
| 16:48.59 | jas_williams | Toooo many possibilities |
| 16:50.05 | jas_williams | [Outcast]: http://www.audio-software.com/dtmf-tone.htm |
| 16:51.15 | Nuxi | just have * call itself and have it process the dtmf. |
| 16:51.32 | [Outcast] | jas_williams : thanks |
| 16:56.12 | *** part/#asterisk darkskiez (~darkskiez@usergc137.dsl.pipex.com) |
| 16:57.20 | *** join/#asterisk dos000 (~dos000@66.11.173.123) |
| 16:57.31 | *** join/#asterisk Marlow (~martin@cerberus.bluetree.ie) |
| 16:57.50 | dos000 | anyone know how much (roughly) is the g729 license from digium per channel ? |
| 16:57.58 | Marlow | 10$ |
| 16:58.02 | Zeeek | Nuxi LOL |
| 16:58.20 | dos000 | marcus5, one time ee ? |
| 16:58.46 | dos000 | Marlow, i mean one time licence fee ? |
| 16:59.10 | smash- | 1o dollars |
| 16:59.15 | smash- | 1 time fee |
| 16:59.42 | *** join/#asterisk Koshatul (~evangelio@CPE-138-217-191-111.qld.bigpond.net.au) |
| 17:00.04 | Marlow | dos000 |
| 17:00.11 | Marlow | dos000: once off, yep |
| 17:00.22 | smash- | once per channel |
| 17:00.50 | jarrod | ok i ordered g.729 codec licenses |
| 17:00.55 | jarrod | lets see how long it takes them to approve |
| 17:01.24 | Marlow | jarrod: usually 1 day .. max .. |
| 17:01.38 | jarrod | thats too long |
| 17:01.39 | jarrod | i want them now |
| 17:01.46 | jarrod | heh |
| 17:01.46 | Marlow | jarrod: even though .. you might get lucky (today) or not (monday) |
| 17:02.01 | jarrod | im betting monday |
| 17:02.01 | smash- | jarrod u can do transcoding too |
| 17:02.04 | Marlow | jarrod: it's still mid day over there .. |
| 17:02.16 | smash- | its 10am here. |
| 17:02.18 | jarrod | smash: you can? |
| 17:02.21 | smash- | yeah |
| 17:02.27 | jarrod | between 711 and 729? |
| 17:02.37 | jarrod | why doesnt it do it :( |
| 17:02.41 | smash- | i transcode * > g729 |
| 17:02.44 | *** part/#asterisk cpatry (~grepmoo@65.39.228.5) |
| 17:03.00 | jarrod | is there something you need to configure? |
| 17:03.04 | smash- | well my pbx uplinks |
| 17:03.06 | smash- | to another one |
| 17:03.09 | smash- | um |
| 17:03.14 | smash- | yeah u need a special file |
| 17:03.18 | smash- | u got the codec yet? |
| 17:03.20 | darwin35 | buy the codec license from digium and dl the module |
| 17:03.27 | bewest | do any of the variables have just the number that is calling? ${CALLERID} and its friends all have text in it don't they? |
| 17:03.34 | jarrod | well i have one on another server i bought long ago |
| 17:03.42 | jarrod | but i send to it ulaw and it sends to my upstream carrier 729 |
| 17:03.48 | jarrod | then cant translate between them |
| 17:04.06 | bewest | also, how do you print something to the console from extensions.conf? I thought it was NoOp(${VAR}), but that doesn't work |
| 17:04.06 | jarrod | ${CALLERIDNUM} |
| 17:04.06 | jarrod | ? |
| 17:04.08 | smash- | hrm |
| 17:04.19 | bewest | the wiki says ${CALLERIDNUM} commonly has text |
| 17:04.36 | jarrod | agi script to parse it heh |
| 17:04.36 | Qwell | bewest: somebody said DNID the other day, but I'm not sure when that applies |
| 17:05.09 | bewest | ok, so there's no way to get it without some parsing |
| 17:05.09 | jarrod | smash the other caveat is h323 is in the loop heh |
| 17:05.17 | jarrod | sip phone -> iax2 connect -> h323 carrier |
| 17:05.37 | jarrod | ulaw all the way then g729 to the carrier |
| 17:06.00 | smash- | hrm |
| 17:06.03 | smash- | i dun kow |
| 17:06.05 | bewest | isn't DNID the number they entered to reach you? not the number they are calling from? |
| 17:06.25 | smash- | dailnumber id? |
| 17:06.27 | jas_williams | ${CALLERIDNUM} is the number ${CALLERID} is name and number if availiable |
| 17:06.44 | bewest | oh I see |
| 17:07.30 | darwin35 | or cidname and cidnum |
| 17:07.32 | bewest | jas_williams, so the "text" the wiki mentions in CALLERIDNUM would only be spaces in the groupings of the number? |
| 17:08.09 | bewest | how do you print something outon the CLI from the dialplan? I thought NoOp(${foo}) would print to cli? |
| 17:08.20 | bewest | assuming ${foo} was a variable |
| 17:09.36 | jas_williams | exten=> xxx,1,NoOP($CALLERIDNUM}) works |
| 17:10.24 | jas_williams | where xxx is an extension in your extensions.conf |
| 17:10.59 | jas_williams | missed { bracket exten=> xxx,1,NoOP(${CALLERIDNUM}) works |
| 17:11.03 | *** join/#asterisk morris (~turntabli@ACD54FBF.ipt.aol.com) |
| 17:11.05 | [Outcast] | exit |
| 17:11.08 | morris | evening |
| 17:11.41 | *** join/#asterisk dalabera (~Dalabera@mail.pmrtechnologies.com) |
| 17:11.53 | bewest | ok thanks jas_williams |
| 17:12.08 | dalabera | hi, is there any load configuration command for the voicemail.conf on the cli? |
| 17:13.08 | jas_williams | dalabera: No it's read every time voicemail is accessed |
| 17:13.58 | cjk | is there a way to store registration information in mysql an not in the astdb file, any hint and link to a doc... |
| 17:14.20 | Marlow | cjk: dbodbc |
| 17:14.23 | morris | jas_williams: good evening |
| 17:14.26 | Marlow | cjk: does the job .. |
| 17:14.34 | jas_williams | evening |
| 17:14.50 | cjk | Marlow: ok but really where can i find some informaiton on how to set it up. i cant really find anything |
| 17:15.02 | Marlow | cjk: it's on voip-info.org .. |
| 17:15.22 | cjk | Marlow: of course, thats where i am searching now for an hour. do you have any more information? |
| 17:15.51 | *** part/#asterisk Zeeek (~Zeeek@Zeeek.sustaining.supporter.pdpc) |
| 17:22.44 | *** join/#asterisk bofh42 (~bofh42@p54820381.dip0.t-ipconnect.de) |
| 17:24.01 | morris | argh |
| 17:24.06 | Nuxi | cjk, agi is the answer to all problems. |
| 17:24.32 | *** join/#asterisk heison (~heison@gw-yyz1.somanetworks.com) |
| 17:24.33 | cjk | Nugget: i use a lot of agi really, but how can this solve my problem |
| 17:25.01 | morris | anyone that cares to hear about my stressful life.. i wanted to ssh into asterisk box at work.. (this happened last night too) I am unable to connect. I have the port rules setup.. when i got into work lastnight it seemed we had a power outage! guess its happened again argh |
| 17:25.46 | morris | bleh |
| 17:26.04 | jas_williams | morris: or the IP address has changed :) |
| 17:26.36 | morris | jas.. thankfuly its static!!! |
| 17:26.44 | *** join/#asterisk dros7 (~simon@laotan.openflows.org) |
| 17:26.45 | morris | however.. that hasnt proven to be much use to me lol |
| 17:27.01 | morris | argh its so anoying |
| 17:27.12 | morris | u know, with a BT line its perfect |
| 17:27.18 | morris | got the ntl line kinda working |
| 17:27.22 | morris | but its not really acceptable |
| 17:27.23 | morris | :( |
| 17:27.51 | morris | sympathy guys ;p |
| 17:30.15 | bewest | anyone know any reason why exten => xxx, y, NoOp(${var}) would not print anything on the CLI? |
| 17:30.37 | newmedian | is it me, or is the sponsored by digium logo now have to * above the i rather than previously having one? Do I just need more coffee? |
| 17:30.43 | newmedian | two |
| 17:30.44 | jas_williams | morris: You need to find out what signalling NTL are using Ground start Loop Start or E&M |
| 17:31.21 | morris | definatly not kewlstart or wahtever it is |
| 17:31.24 | morris | as that works with bt |
| 17:31.26 | newmedian | http://freenode.net/digium-logo.png |
| 17:31.42 | morris | that g looks like its waving pom poms |
| 17:32.00 | jas_williams | bewest: It's not in the right context/extension do you have any output in the CLI |
| 17:32.23 | bewest | I get output in CLI when there's an error |
| 17:32.31 | newmedian | On the digium site itself there is only one * above the second i. |
| 17:32.34 | morris | jas_williams, do u think maybe a call to ntl technical might help? |
| 17:32.39 | bewest | for example, NoOP(${foo) is missing a brace, I would get that error on CLI |
| 17:33.19 | bewest | does some debug need to be on? |
| 17:33.31 | jas_williams | set verbose 9 ? |
| 17:33.57 | bewest | ah it was 0 |
| 17:34.30 | bewest | ok |
| 17:34.35 | bewest | now I see lots of stuff |
| 17:34.38 | bewest | thank you |
| 17:34.51 | *** join/#asterisk Alric (~nbowyer@masq.hyperusa.com) |
| 17:36.52 | iCEBrkr | LILO needs to STFU already |
| 17:37.38 | *** join/#asterisk drumkilla (~russell@user-24-214-77-225.knology.net) |
| 17:37.38 | *** mode/#asterisk [+o drumkilla] by ChanServ |
| 17:43.41 | *** join/#asterisk L|NUX (~linux@202.5.145.54) |
| 17:44.12 | *** join/#asterisk hekuran (~heka@80.80.174.140) |
| 17:45.21 | *** join/#asterisk Exstatica (Exstatica@jumping.on.the.bed.are.not.umpteenmonkeys.com) |
| 17:53.16 | jarrod | grr |
| 17:53.30 | jarrod | someone from asterisk expedite my order heh |
| 17:53.36 | jarrod | show me some lub |
| 17:54.42 | iCEBrkr | Show them da money! |
| 17:54.47 | [TK]D-Fender | I'm sure bkw could show you some lube ;) |
| 17:54.53 | iCEBrkr | [TK]D-Fender: hahah |
| 17:55.08 | jarrod | heh |
| 17:55.12 | jarrod | i gave them zee $$$$ |
| 17:55.33 | *** join/#asterisk jooray (~jooray@smtp.rebelium.net) |
| 17:57.18 | *** join/#asterisk Grooby (~Grooby@12.22.232.212) |
| 17:57.48 | *** join/#asterisk ciphergroup (~cagate@216.90.111.178) |
| 17:58.09 | jooray | hello, i am working on a little voip installation, that will use sip and softphones (primarily) over larger LAN. the problem is, I want it all encrypted. Is it possible, that all traffic goes through the machine, that runs asterisk (as a proxy)? I can establish an ipsec connection to one machine and let all the communication (and data) never go direct, but through that machine. is this possible or any other ideas how to acomplish asterisk+sip+encryption |
| 17:59.16 | ciphergroup | hello all |
| 18:00.28 | *** join/#asterisk opa__ (~opa__@80.66.95.148) |
| 18:00.29 | ciphergroup | has nay one every used a single line analog phone with voltage message waiting, not FSK, with asterisk |
| 18:02.17 | *** join/#asterisk SarahEmm (~sarahemm_@Toronto-HSE-ppp3704598.sympatico.ca) |
| 18:03.50 | Alric | Anyone ever see a Polycom suddenly lose its configuration? I've been statically configuring them from their web interfaces. This one worked fine for months, and suddenly I come in one day and it reverted to an old dialplan, and seems to have lost its SIP proxy settings. Happened to another one I just got too. |
| 18:04.03 | nestAr | hrmm |
| 18:04.14 | nestAr | no, sorry.. i use a boot server for most of mine |
| 18:04.26 | nestAr | the ones at home haven't puked so far so good |
| 18:04.58 | iCEBrkr | jooray: Do you really think someone's gonna 'hack' your VoIP data? |
| 18:05.17 | iCEBrkr | Cuz you know, there's all those top secret conversations going on. |
| 18:05.42 | *** join/#asterisk RoyK (~roy@host-81-191-147-248.bluecom.no) |
| 18:05.57 | *** join/#asterisk Blissex (~Blissex@82-69-39-138.dsl.in-addr.zen.co.uk) |
| 18:05.59 | opa__ | HI everybody! Who do understand new JB at IAX2/sip? |
| 18:06.35 | opa__ | i tring to port it to h323... :( |
| 18:06.40 | iCEBrkr | Who do understand? |
| 18:07.03 | jooray | iCEBrkr: i don't care, that's what the boss wants |
| 18:07.27 | jooray | iCEBrkr: but yes, it is possible to listen to the voip data, I have seen various applications, that actually do it |
| 18:07.39 | jooray | iCEBrkr: via the arp attack, it's quite easy to do so |
| 18:07.44 | *** join/#asterisk Cheetah (~Snak@62.217.48.111) |
| 18:07.58 | Cheetah | 'evening |
| 18:08.00 | jooray | iCEBrkr: you can't risk this in some environments, when you have >500 people working in company and even most confidential data is spoken over voip |
| 18:11.33 | *** join/#asterisk jayk_ (jayk@vapid.reprehensible.net) |
| 18:11.38 | jayk_ | is anybody here familar with asttapi? |
| 18:11.53 | jayk_ | when i select a contact to dial, it dials my asterisk phone, and once i pick up it dials the number.. |
| 18:12.06 | jayk_ | is there a way to set it up so that my cisco 7960 phone automatically picks up when using asttapi? |
| 18:13.17 | Cheetah | can someone help me with this: |
| 18:13.17 | Cheetah | modem_request: Requested device 'g1' does not exist |
| 18:13.28 | Cheetah | I'm using isdn4linux ;) |
| 18:13.32 | Cheetah | it works if I am using this: |
| 18:14.07 | iCEBrkr | jooray: Fanatic. |
| 18:14.13 | Cheetah | exten => _.,1,Dial(Modem/ttyI1:${EXTEN}) |
| 18:14.23 | Cheetah | but not with: |
| 18:14.23 | Cheetah | exten => _.,1,Dial(Modem/g1:${EXTEN}) |
| 18:14.26 | Cheetah | any ideas? |
| 18:14.55 | jayk_ | try g0? |
| 18:15.01 | jooray | btw. any good free/open source SIP softphones for linux and mac os x? i have seen lots of them, but none with source code (excpet for linphone, which is linux-only). i have played with x-lite, but i really need some open-source alternative.... |
| 18:15.46 | dos000 | jooray, there is the sipx stuff if i remember correctly |
| 18:15.53 | Cheetah | jayk_, lemme check that. but in my modems.conf i added it to group 1 |
| 18:16.14 | *** join/#asterisk oej (~oej@213.204.186.40) |
| 18:16.19 | Cheetah | nope, doesnt work |
| 18:16.34 | jayk_ | dunno |
| 18:16.43 | opa__ | Cheetah, did you look to src of modem_req? |
| 18:17.43 | Cheetah | opa__, source code? i'm not really up to that right now :D |
| 18:18.08 | dos000 | jooray, you tried http://www.sipfoundry.org/sipXphone/ yet ? |
| 18:18.56 | smash- | hey |
| 18:20.13 | Cheetah | Requested device 'g0' does not exist |
| 18:21.30 | jooray | dos000: no, I'll try |
| 18:21.30 | jooray | :) |
| 18:21.33 | jooray | dos000: thanks |
| 18:22.34 | opa__ | Cheetah, try add to 3 groups: 4,1,3 |
| 18:22.51 | Cheetah | opa__, one sec |
| 18:23.18 | *** join/#asterisk zotz (~zotz@208.196.247.140) |
| 18:25.26 | opa__ | Cheetah, else ad debug: cur_group = get_group(v->value); ast_log(LOG_ERROR,"CurGroup %d\n",cur_group); in load_module() |
| 18:25.54 | Cheetah | i'm using precompiled debian packages |
| 18:26.09 | opa__ | Cheetah, :) |
| 18:26.49 | jooray | and one last question. anyone played with some kind of jabber integration? so it would be possible to call a jabber contact through asterisk? |
| 18:26.50 | Cheetah | already fixed it :) |
| 18:27.27 | *** join/#asterisk Keizer (~keizer@pcp02851954pcs.ctftmy01.fl.comcast.net) |
| 18:27.48 | opa__ | Cheetah, what was? |
| 18:28.13 | Cheetah | i forgot to add group=1 BEFORE the "device =>" lines |
| 18:28.16 | jas_williams | jooray: http://www.voip-info.org/wiki-Asterisk+Jabber |
| 18:28.29 | SuPrSluG | i'm getting breaks (sound drops) on calls. i'm using dundi thru openvpn. other than bandwidth. what are the likely suspects |
| 18:28.31 | jooray | thanks |
| 18:28.31 | jooray | :) |
| 18:28.31 | iCEBrkr | Anyone got the wiki url to show me how to get 2 asterisk box connected to each other? |
| 18:28.58 | *** join/#asterisk LoRez_ (lorez@lorez.staff.freenode) |
| 18:29.23 | SuPrSluG | ~connecting 2 asterisk boxes |
| 18:29.57 | jas_williams | ~google connecting 2 asterisk boxes |
| 18:30.07 | iCEBrkr | hehe |
| 18:32.45 | Keizer | Is there any data on how to make VOIP cost effective solution? |
| 18:33.03 | Keizer | I've already got a T1 for voice but not data. I use cable modem for data |
| 18:33.35 | iCEBrkr | Keizer: Um, take the VoIP cost for a long distance call.. and compare it to what you're paying now |
| 18:33.38 | iCEBrkr | :P |
| 18:33.46 | jooray | thanks all, goodbye :) |
| 18:33.48 | *** part/#asterisk jooray (~jooray@smtp.rebelium.net) |
| 18:33.53 | Keizer | I don't know how to get the cost comparison |
| 18:33.57 | Keizer | I don't want to use Vonage |
| 18:34.00 | Keizer | Is there an alternative? |
| 18:34.25 | Keizer | The Asterisk PBX would be neat to use. I have a Dual Opteron server I could put it on that runs Linux |
| 18:34.25 | iCEBrkr | Who pays the phone bill? Get the cost of the phone blill, and the number of minutes you've used. |
| 18:34.26 | jas_williams | many |
| 18:34.40 | iCEBrkr | Then multiply that by what some VoIP provider is gonna charge you . haha |
| 18:35.46 | iCEBrkr | Keizer: Tho, unfortunately, there's costs in getting phones and beefier network gear |
| 18:36.01 | Keizer | I wouldn't mind getting the network gear. |
| 18:36.07 | *** join/#asterisk Klaus (~klaus@sconk.dk) |
| 18:36.41 | Keizer | I just wish there was a spec sheet on how to setup Asterisk with the total solution. Like that new DUNDI technology. How would that save money? |
| 18:37.22 | iCEBrkr | Dundi would save you money cuz it'd route calls to registered PSTN gateways which every VoIP service charges for. |
| 18:37.30 | jas_williams | Keizer: Engage a consultant to do a business case for you |
| 18:37.40 | iCEBrkr | ...or at least that's what I gather from DUNDi |
| 18:38.18 | Keizer | jas_williams, I could consult with my friend who is a CCIE |
| 18:39.54 | Keizer | Who is a VOIP carrier? |
| 18:40.25 | iCEBrkr | Broadcom, VoicePulse, FWD, iConnect, |
| 18:40.30 | iCEBrkr | there's a list on the Wiki |
| 18:40.30 | iCEBrkr | err |
| 18:40.33 | iCEBrkr | Broadvoice even. |
| 18:40.46 | shmaltz | does tftp run on tcp or udp? |
| 18:41.03 | jas_williams | shmaltz: TCP |
| 18:41.16 | Nivex | udp |
| 18:41.22 | Nivex | ftp is tcp, tftp is udp |
| 18:41.23 | darwin35 | Astralinks |
| 18:41.23 | SarahEmm | tftp is udp. |
| 18:41.34 | shmaltz | my inet is set just for udp |
| 18:41.36 | shmaltz | so this OK |
| 18:41.52 | shmaltz | oh I got it |
| 18:41.57 | shmaltz | stupid me |
| 18:42.05 | *** join/#asterisk Tili (Tili@202-133-65-134-dialup.sat.net.pk) |
| 18:42.57 | *** join/#asterisk tld (~tld@45.80-202-167.nextgentel.com) |
| 18:43.28 | Cheetah | hm |
| 18:44.06 | Cheetah | opa__, when someone hangs up, i see that the modem hung up in the asterisk console but the SIP client doesnt notice it |
| 18:45.16 | opa__ | Cheetah, try add exten => xxxx,2,hangup |
| 18:45.51 | Cheetah | thanks.. lemme try that |
| 18:46.40 | ciphergroup | has any one worked with the Citel Gateways |
| 18:46.47 | Cheetah | hm |
| 18:46.51 | Cheetah | opa__, its already there |
| 18:46.53 | RoyK | jas_williams: tftp is udp, normally port 69....... rtfm! |
| 18:46.54 | Cheetah | exten => _.,1,Dial(Modem/g1:${EXTEN}|60|r) |
| 18:46.54 | Cheetah | exten => _.,2,Hangup |
| 18:47.08 | *** join/#asterisk kdayn (~kdayn@codeine.svnets.lv) |
| 18:47.45 | Keizer | Meh |
| 18:47.50 | Keizer | I'll research this later |
| 18:47.58 | Flav | any thoughts on the 911 order by the fcc? |
| 18:48.41 | morris | its a sensible idea.. altho i wouldnt wanna trust an internet based phone for an emergency ;p |
| 18:49.32 | denon | you know, that opinion's going to change as the 'net gets more robut |
| 18:49.33 | opa__ | Cheetah, i thinked ast wait till t extention. try call in both direction : from sip to modem and reverse. let set verbose 3 |
| 18:49.35 | denon | robust |
| 18:49.55 | denon | I mean, look at the pstn .. it's interconnected by many carriers, over microwave, satellite, fiber, copper, over oceans etc |
| 18:50.01 | shmaltz | anybody ever heard of this? |
| 18:50.03 | shmaltz | I have an internet connection that is triple natted, like this: |
| 18:50.05 | shmaltz | INTERNET < NAT > 192.168.0.0/24 < NAT > 192.168.1.0/24 < NAT > 192.168.2.0/24 and I can't receive any tftp (UDP) packets on 192.168.2.0 but I could on 192.168.0.0 (didn't try 192.168.1.0) |
| 18:50.29 | *** join/#asterisk jsolares (~jsolares@200.30.141.86) |
| 18:50.32 | denon | no reason the Internet should be any different, especially with ip6 |
| 18:50.44 | [TK]D-Fender | shmaltz : Why the cascading NAT? |
| 18:51.13 | shmaltz | [TK]D-Fender, to make a long story short, because 3 office are sharing the same cable connection |
| 18:51.13 | opa__ | Cheetah, turn on debug/dump on SIP and look for disconnect command |
| 18:51.55 | shmaltz | D-Fender, nope I don't plan on making SIP work on this, but I wouldn't mind to try IAX ;/ |
| 18:51.59 | Cheetah | opa__, http://pastebin.com/287089 |
| 18:52.09 | *** join/#asterisk onkeltimm (~chatzilla@dsl-082-083-015-215.arcor-ip.net) |
| 18:52.21 | *** join/#asterisk Gerrath (Gerrath@shanev.lifecor.com) |
| 18:52.22 | [TK]D-Fender | shmaltz : Should just have them route the subnet towards the central gateway.... anyways, UDP is connectionless and you'd have to forward ports AFAIK to get it to work... |
| 18:52.35 | [TK]D-Fender | But that would restrict you to a single host... |
| 18:52.46 | Cheetah | opa__, the last SIP message is 404 not found |
| 18:52.54 | shmaltz | ~afaik |
| 18:52.55 | jbot | somebody said afaik was some kind of acronym...but nobody knows what it means, as far as I know. |
| 18:53.19 | shmaltz | D-Fender, it makes and it doesn't make sense |
| 18:53.33 | shmaltz | the NAT servers should take care of it on the way |
| 18:53.54 | shmaltz | I mean they should know who is the one that sent the request so they can turn it back |
| 18:55.10 | [TK]D-Fender | shmaltz : the problem is UDP is connectionless and you are trying to target anyting at all behind a NAT. Only work on the the deeper levels accessing the inner ones. |
| 18:56.21 | opa__ | Cheetah, what is last "Spawn Ext"? 3-d call? or and of second? check dialplan more. |
| 18:56.59 | iCEBrkr | ok, sweet, I got both my asterisk servers talking to each other.. Now what? :D |
| 18:57.00 | shmaltz | I don't understand this, it shouldn't matter if it's connectionless or not, although it does for the fact that none of the sides will verify that the packet actualy arrived, however in my case one of the routers (at least thats what I understand is happening) are not natting any UDP traffic at all |
| 18:57.13 | [TK]D-Fender | iCEBrkr : WORLD DOMINATION |
| 18:57.27 | Cheetah | opa__, after some time the console says something like this: (after the real phone hung up) |
| 18:57.38 | iCEBrkr | [TK]D-Fender: Yea, I was kinda thinking that myself. |
| 18:57.39 | shmaltz | iCEBrkr, try make it talk to your toaster oven |
| 18:57.53 | iCEBrkr | shmaltz: Yeah, gotta start small. ;P |
| 18:57.56 | shmaltz | nah, you way to far from world domination |
| 18:58.03 | iCEBrkr | :( |
| 18:58.27 | shmaltz | I just finnished an install for someone, where they can roll up their rolldown gate, as well as cloes it or stop it |
| 18:58.34 | shmaltz | all using asterisk |
| 18:58.42 | shmaltz | they can call in and do it |
| 18:59.08 | iCEBrkr | DTMF menu? |
| 18:59.08 | shmaltz | they can even have the bell ring to the cell phone and answer the phone like this |
| 18:59.10 | iCEBrkr | Press 1, to close your gate, or press 2 to close the gate on the FedEx guy. |
| 18:59.19 | shmaltz | yep using DTMF, but not an IVR for secuiryt reasons |
| 18:59.48 | iCEBrkr | Some how someone gets into that extension and the gates going up and down cuz they're randoming jamming buttons. |
| 18:59.54 | shmaltz | you just type exten *501 to open the gate, *502, to close it, and *503 to stop it |
| 18:59.57 | [TK]D-Fender | shmaltz : I have already created an * interface to my toaster oven...... |
| 19:00.00 | iCEBrkr | Ah |
| 19:00.24 | iCEBrkr | I wanna get Sphinx working darn it. |
| 19:00.25 | opa__ | Cheetah, it it 1 call? or 2? |
| 19:00.34 | iCEBrkr | I got it working just have to figure out the eagi() stuff. |
| 19:00.42 | shmaltz | what that doesn is call the relay that opens the gate, and on *501 sends DTMF to open and so on |
| 19:00.46 | [TK]D-Fender | I'd pay for solid instructions on getting sphinx integrated with *...... |
| 19:00.52 | Cheetah | opa__, what? |
| 19:01.01 | iCEBrkr | [TK]D-Fender: ouch. |
| 19:01.02 | *** join/#asterisk bsd3 (~bsd@203.145.128.6) |
| 19:01.07 | shmaltz | I'm using an RC-3 from http://vikingelectronics.com |
| 19:01.22 | iCEBrkr | [TK]D-Fender: well if someone would document how to use the sound capture stuff of eagi's |
| 19:01.27 | bsd3 | hi, friends! |
| 19:02.08 | *** part/#asterisk Marlow (~martin@cerberus.bluetree.ie) |
| 19:02.33 | bsd3 | i have written 2 tiny scripts to view CSV/PGSQL CDR data via web, now where do i post these? |
| 19:03.23 | shmaltz | the RC-3 is interfaced to this: |
| 19:03.25 | shmaltz | http://www.manaras.com/book046.pdf |
| 19:03.26 | shmaltz | page 20, and C2 wiring on page 23 |
| 19:03.40 | iCEBrkr | bsd3: on your web page in the section where you have a list of accomplishments :P |
| 19:03.46 | iCEBrkr | and contributions |
| 19:04.27 | bsd3 | iCEBrkr: i don'thanks have any web page as yet |
| 19:04.41 | iCEBrkr | booooooooo |
| 19:04.50 | iCEBrkr | Yeah, homepages are so 1995. |
| 19:04.51 | bsd3 | :) |
| 19:04.58 | jas_williams | post them on wiki |
| 19:05.11 | bsd3 | ok, thanks |
| 19:05.16 | Cheetah | opa__, could it be the cause that the Dial command did not return zero? |
| 19:07.06 | *** join/#asterisk JerJer[mobile] (~jj@ip68-103-26-140.ks.ok.cox.net) |
| 19:07.25 | jas_williams | Cheetah: the reason is you have a catchall extension such as exten => _.,1,Dial This is bad change it to _X. |
| 19:07.49 | Cheetah | jas_williams, will this fix any problems with this? ;) |
| 19:07.53 | jas_williams | yes |
| 19:08.19 | jas_williams | it will stop the second dial |
| 19:08.24 | jas_williams | after hangup |
| 19:09.00 | Cheetah | hmm |
| 19:09.04 | Cheetah | not I can't dial anymore |
| 19:09.38 | Cheetah | err my fault |
| 19:09.39 | jas_williams | what do you have as the dial line now ? |
| 19:09.58 | Cheetah | i fixed it ;) |
| 19:10.05 | Cheetah | i had a _X without the . |
| 19:10.13 | jas_williams | k |
| 19:10.28 | Cheetah | jas_williams, lets say I want to play a "thank you" sound after the modem hung up, is that possible? |
| 19:10.57 | Cheetah | right now I do this: |
| 19:10.58 | Cheetah | exten => _X.,1,Dial(Modem/g1:${EXTEN}|60|r) |
| 19:11.22 | jas_williams | yes but who would it play to they have hung up:) use an exten=> h,1,Background(Thankyou) |
| 19:12.02 | Cheetah | well, its just a thought :D |
| 19:12.31 | jas_williams | the h extension is called after hangup |
| 19:12.33 | opa__ | Cheetah, oh... is exten => i,1,hangup ? |
| 19:12.37 | Cheetah | ahh nice :D |
| 19:12.49 | Cheetah | I'm still trying to understand the way extensions.conf works |
| 19:13.06 | jas_williams | opa__: i means invalid number |
| 19:13.41 | Cheetah | where is that documented? |
| 19:14.09 | opa__ | Cheetah: in pbx.c :) |
| 19:14.32 | jas_williams | http://www.voip-info.org/wiki-Asterisk+config+extensions.conf |
| 19:14.47 | Cheetah | opa__, you little sourcesniffer ;D |
| 19:15.41 | Cheetah | and, stupid question, is it possible to change the "dialing..." sound that appears before the remote user has picked up? |
| 19:15.56 | Cheetah | i want to change that because the people in germany get confused with the american dial tone :D |
| 19:16.02 | opa__ | Cheetah, i programmer of some private channels for embeded system... and has no debuger better then logs |
| 19:16.23 | Cheetah | opa__, ouchies ;) |
| 19:16.40 | *** join/#asterisk Patrick^ (~patrickm@birch4.mountaincable.net) |
| 19:17.21 | znoG | you guys reckon a P2 300MHz box with 64MB ram is enough for Asterisk to run well on a maximum of 2 simultaneous calls? (both calls with transcoding) |
| 19:17.37 | jas_williams | Cheetah: depends on where the ring tone is being generated, by a sip device or by asterisk |
| 19:18.07 | Cheetah | i see, well. it seems that the ring tone switches to the one generated by asterisk if I add a "|60|r" to the dial command |
| 19:18.10 | Cheetah | what does that do? |
| 19:18.36 | RaYmAn-Bx | znoG: the ram seems a bit low tbh, but depending on codec (and whether it's VoIP Only) it should work fine |
| 19:19.08 | RaYmAn-Bx | I used to run something similar with VoIP only on a PII 333mhz with 384mb ram just fine..while the box was doing a bunch of other things at the same time |
| 19:19.10 | znoG | yeah those are my thoughts too. I was going to go 128MB anyway but I thought i'd check to see if it would work OK with 64 |
| 19:19.16 | opa__ | Cheetah, try exten => h,1,hangup according wiki |
| 19:19.24 | RaYmAn-Bx | znoG: only trying it can tell :) |
| 19:19.57 | Cheetah | opa__, i already fixed that problem by using "_X." instead of "_." |
| 19:20.41 | jas_williams | Cheetah: look in indications.conf for asterisk tones |
| 19:21.32 | Cheetah | ah :) |
| 19:21.51 | *** join/#asterisk robf (~robf@208.188.247.3) |
| 19:23.18 | *** join/#asterisk anthm (~anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 19:23.18 | *** mode/#asterisk [+o anthm] by ChanServ |
| 19:23.27 | znoG | RaYmAn-Bx: :) will do, cheers. |
| 19:23.54 | *** join/#asterisk Micc (~dotirc@c-24-18-35-120.hsd1.wa.comcast.net) |
| 19:24.24 | Micc | Is realtime in stable yet? |
| 19:24.35 | jas_williams | Micc: No |
| 19:25.47 | Micc | jas_williams, when will it be? |
| 19:26.25 | jas_williams | No idea you need to ask a developer |
| 19:27.07 | jas_williams | I run CVS Head |
| 19:27.15 | Micc | I've been running cvs head too. |
| 19:27.31 | Micc | But I am about to use it with a larger production system. |
| 19:27.36 | Micc | Not sure I want to do that. |
| 19:27.39 | Cheetah | can I set the language to "de" by default in extensions.conf? |
| 19:27.43 | Micc | But I need the realtime stuff. |
| 19:29.30 | *** join/#asterisk moonwick (~moonwick@core.dump.net) |
| 19:29.43 | jas_williams | Cheetah: Cheetah you need the tones first follow the details on http://www.voip-info.org/tiki-index.php?page=Asterisk%20indications%20Germany |
| 19:29.57 | Cheetah | jas, I've been there; done that :) |
| 19:29.58 | *** join/#asterisk vphirke (~vphirke@66.9.62.66) |
| 19:30.03 | Cheetah | however, the tone sounds like the us one still |
| 19:30.28 | jas_williams | Cheetah: have you restarted |
| 19:30.55 | Cheetah | uhm... of course |
| 19:31.01 | *** join/#asterisk odie_flocon (~Odie@ptr-64-201-182-211.ptr.terago.ca) |
| 19:32.48 | jas_williams | Cheetah: you did change this as well ? [general] |
| 19:32.48 | jas_williams | country=de |
| 19:33.06 | *** part/#asterisk bsd3 (~bsd@203.145.128.6) |
| 19:33.39 | jas_williams | any ringing generated by asterisk (with the ,r) should now be German tones |
| 19:34.38 | *** join/#asterisk wrmem (~wrmem@monnin-win.cso.uiuc.edu) |
| 19:36.07 | *** join/#asterisk meppl (mephisto@p54AAF807.dip.t-dialin.net) |
| 19:41.34 | Cheetah | odd |
| 19:41.37 | Cheetah | hm |
| 19:41.41 | Cheetah | back in a moment |
| 19:42.28 | *** join/#asterisk kl_pl_01 (kl_pl_01@82.160.252.17) |
| 19:42.36 | *** join/#asterisk hypa7ia (~leigh@655a47ec531361ac.session.tor) |
| 19:43.54 | shmaltz | anthm, you around? |
| 19:44.06 | anthm | yes? |
| 19:44.32 | shmaltz | someone on the list wants the app_valetparking for HEAD, whats the url? |
| 19:44.41 | *** join/#asterisk Gerrath (~Gerrath@shanev.lifecor.com) |
| 19:44.56 | anthm | www.cluecon.com |
| 19:46.23 | shmaltz | cluecon.com/what??????????? |
| 19:46.25 | *** join/#asterisk bah (048830696@AC865B6D.ipt.aol.com) |
| 19:50.05 | *** part/#asterisk bkw_ (~bkw_@bkw.developer.and.friend.of.asterisk) |
| 19:50.15 | anthm | http://www.pbxclue.com/asterisk_apps/ what does he want it for so he can call ppl and ask them to go to cluecon? |
| 19:50.20 | *** join/#asterisk Blackvel (~blackvel@dsl-084-057-125-030.arcor-ip.net) |
| 19:50.59 | *** join/#asterisk Cresl1n (~matt@216.207.245.23) |
| 19:51.02 | iCEBrkr | How much hacking would I have to do to get Asterisk and NEAX to co-exist? |
| 19:51.06 | *** join/#asterisk bkw_ (~brian@bkw.developer.and.friend.of.asterisk) |
| 19:51.06 | *** mode/#asterisk [+o bkw_] by ChanServ |
| 19:52.27 | shmaltz | I have no clue what he wants it for: |
| 19:52.29 | shmaltz | http://lists.digium.com/pipermail/asterisk-users/2005-May/108298.html |
| 19:54.11 | anthm | hmm the guy sounds like a fan, he wants valet parking so he can use it with chanspy and native moh |
| 19:54.33 | anthm | so he wants a trifecta of my inventions, tell him to come to cluecon and pay hommage |
| 19:55.02 | iCEBrkr | haha |
| 19:55.15 | *** join/#asterisk harryvv (~none@S010600a0c93f6f7e.vs.shawcable.net) |
| 19:55.16 | iCEBrkr | ~make asterisk work with neax |
| 19:55.17 | jbot | make: *** No rule to make target `asterisk work with neax'. Stop. |
| 19:55.18 | hypa7ia | mwahahah |
| 19:55.32 | iCEBrkr | grr |
| 19:55.35 | hypa7ia | yay! |
| 19:55.44 | harryvv | So any discussion on the 911 requirments today? |
| 19:57.50 | shmaltz | anthm, everybody on the list is a fan of you |
| 19:58.06 | anthm | kewl , tell them all to come to cluecon then ! |
| 19:58.21 | file | yeah, we'd all love to see them at Cluecon |
| 19:58.22 | *** join/#asterisk bkw_ (~bkw_@bkw.developer.and.friend.of.asterisk) |
| 19:58.22 | *** mode/#asterisk [+o bkw_] by ChanServ |
| 20:01.45 | *** join/#asterisk n4y (~tmalkut@fw.orasoft.net.pl) |
| 20:07.11 | Cresl1n | Howdy!!! |
| 20:07.12 | Cresl1n | :-) |
| 20:10.04 | *** join/#asterisk santiago (~santiago@63.245.86.248) |
| 20:12.09 | *** join/#asterisk Cheetah (~Snak@main-gw.bense.de) |
| 20:12.22 | Cheetah | hey again |
| 20:16.13 | harryvv | I wonder what kind of enforcment the fcc would exert on lec voip carriers if thay do not comply with e911 service in 120 days. |
| 20:16.30 | Cheetah | how can I dial to a connected SIP softphone with the Dial() command? |
| 20:16.31 | hypa7ia | C&D's and fines |
| 20:17.56 | iCEBrkr | Cheetah: Well, put the softphone in your extensions.conf and sip.conf |
| 20:18.48 | iCEBrkr | dial(sip/1234,tr) |
| 20:19.03 | Cheetah | what does the ,tr do? |
| 20:19.21 | iCEBrkr | CLI> show application dial |
| 20:19.29 | iCEBrkr | t = allow callee to transfer |
| 20:19.32 | iCEBrkr | r = ring |
| 20:19.41 | Cheetah | ah thanks :) |
| 20:20.37 | Cheetah | i get "invalid timeout specified: 'tr'" |
| 20:20.45 | iCEBrkr | oops |
| 20:20.48 | *** part/#asterisk SarahEmm (~sarahemm_@Toronto-HSE-ppp3704598.sympatico.ca) |
| 20:20.51 | iCEBrkr | dial(sip/123,20,tr) |
| 20:20.57 | Cheetah | ah thanks :D |
| 20:21.04 | harryvv | ice, I dont have t in my dialplan but then I really dont need to. does that enable say a secretaty to press another extention on the behalf of a caller asking to be transfered? |
| 20:21.13 | iCEBrkr | 20 will make it ring 20 seonds and then head to VMail if no answer |
| 20:21.21 | Cheetah | ok:9 |
| 20:21.23 | Cheetah | :) |
| 20:21.24 | Cheetah | thanks |
| 20:21.25 | Cheetah | brb |
| 20:21.32 | iCEBrkr | harryvv: Yea. |
| 20:21.37 | harryvv | i see |
| 20:21.40 | iCEBrkr | harryvv: tho, it's a blind transfer |
| 20:21.48 | harryvv | what do you mean |
| 20:22.00 | iCEBrkr | So it's like "Please hold.. *CLICK*" ( dial extension ) *HANG UP* |
| 20:22.15 | harryvv | i see |
| 20:22.21 | iCEBrkr | The secretary can't consult before passing the call through |
| 20:22.28 | harryvv | ahh |
| 20:22.36 | harryvv | or filter out calls. |
| 20:22.42 | iCEBrkr | Meaning Joe calls in asking for Bob.. Secretary can't call Bob and say "Here's Joe for ya.." and then hang up. |
| 20:23.02 | iCEBrkr | The call is imediately transfered ot Bob's extension. |
| 20:23.22 | harryvv | yea |
| 20:24.17 | iCEBrkr | Which I hate. |
| 20:25.00 | iCEBrkr | Tho, I'm sure you could use 'park' to some how make a consult and transfer. |
| 20:25.08 | iCEBrkr | I've tried, but didn't get very far. |
| 20:28.32 | harryvv | What about commercial pbx systems? I can see this is important for large coporations but not small ones ;) |
| 20:28.58 | iCEBrkr | Most commerical PBX's do that. |
| 20:30.46 | harryvv | I see |
| 20:30.58 | harryvv | Any wikis on how to make this happen? |
| 20:31.08 | iCEBrkr | Consult & Transfer? |
| 20:31.41 | shmaltz | harryvv, what you trying to accomplish? |
| 20:33.32 | *** join/#asterisk Sedorox (brandon@sedorox.staff.smartserv) |
| 20:34.55 | harryvv | shmaltz, nothing out of the ordinary but if a customer asked for the feature iCEBrkr was describing then its a good idea to accomplish it. |
| 20:36.57 | *** join/#asterisk smash- (~smash@198.107.16.189) |
| 20:37.08 | *** join/#asterisk ApEtc (apetc@ip68-99-136-197.ph.ph.cox.net) |
| 20:37.26 | *** join/#asterisk gpearson (~Graham@c-67-177-182-16.hsd1.in.comcast.net) |
| 20:38.36 | gpearson | Anyway under AMP to configure the *NN Commands. Just installed 1.0 and on the phone unable to setup digital receptionist |
| 20:42.41 | *** join/#asterisk Defraz (~t0tal@216.163.110.2) |
| 20:46.56 | *** join/#asterisk smash- (~smash@198.107.16.189) |
| 20:48.00 | *** join/#asterisk cmk (~cmk_@p54A3C973.dip.t-dialin.net) |
| 20:50.47 | bewest | does SIP use udp or tcp? |
| 20:50.53 | smash- | whats the diff between dms100 and esf? |
| 20:51.09 | Sedorox | udp |
| 20:51.20 | Sedorox | but if your firewalling.. do both |
| 20:51.34 | jeremywhiting | hi all, trying to debug soft-phone connection, can call from one to the other both ways, but either way I try one end can't hear the other |
| 20:51.46 | jeremywhiting | it's always computer a that can't hear computer b speaking no matter who calls who |
| 20:52.21 | harryvv | using sip? open up those firewall ports |
| 20:52.33 | harryvv | then communication both ways will work |
| 20:52.43 | smash- | whats the diff between dms100 and esf? |
| 20:52.52 | jeremywhiting | oh, ok thanks |
| 20:57.47 | *** join/#asterisk Moc[Train] (~mochouina@png1.pointshotwireless.com) |
| 20:58.24 | jeremywhiting | any other ideas anyone? |
| 20:59.51 | jeremywhiting | is one of my soundcards not full duplex or something maybe? |
| 21:00.08 | *** join/#asterisk Sedorox (brandon@sedorox.staff.smartserv) |
| 21:00.33 | *** join/#asterisk zSACREDz (~zSACREDz@adsl-69-227-27-92.dsl.irvnca.pacbell.net) |
| 21:01.19 | *** join/#asterisk mmlj4 (~looseduk@ip68-14-124-25.no.no.cox.net) |
| 21:02.56 | Qwell | jeremywhiting: its a nat issue |
| 21:04.02 | jeremywhiting | no, they're both on the same lan and so is the asterix box |
| 21:04.42 | jeremywhiting | but thanks for the idea |
| 21:05.02 | *** part/#asterisk Nuxi (~nuxi@cust-sdsl-204-250-82-202.bzn-co-i1000-01.bridgeband.net) |
| 21:11.33 | doughecka | ok, I need help with a T1 |
| 21:11.37 | doughecka | anyone here that can help me? |
| 21:11.50 | doughecka | ZT_SPANCONFIG failed on span 1: Invalid argument (22) |
| 21:11.57 | dalabera | how well does asterisk work with H323?? |
| 21:12.09 | Corydon-w | doughecka: probably a typo somewhere |
| 21:12.09 | doughecka | first T1 I have ever setup, so this is new for me |
| 21:12.10 | dalabera | better thatn sip or worst?? |
| 21:12.13 | ariel_ | doughecka, what is the issue? |
| 21:12.24 | doughecka | I get that error with ztcfg -vv |
| 21:12.26 | Corydon-w | doughecka: post your zaptel.conf on http://pastebin.ca/ |
| 21:13.05 | doughecka | http://pastebin.ca/12292 |
| 21:13.35 | doughecka | http://pastebin.ca/12293 |
| 21:13.42 | doughecka | that one has the loadzone, if thats important |
| 21:14.19 | Corydon-w | doughecka: t100p or te110p? |
| 21:14.43 | doughecka | te110p |
| 21:14.45 | Corydon-w | Is the associated module listed in the output of 'lsmod'? |
| 21:15.02 | Corydon-w | If not, run 'dmesg' and note the errors |
| 21:15.05 | doughecka | wcte11xp 20256 0 (unused) |
| 21:15.05 | doughecka | wcfxo 9216 0 |
| 21:15.05 | doughecka | zaptel 178560 0 [wcte11xp wcfxo] |
| 21:15.13 | Corydon-w | AHA |
| 21:15.28 | Corydon-w | You have an X100P in the same machine |
| 21:15.32 | doughecka | correct |
| 21:15.36 | doughecka | that bad? |
| 21:15.47 | doughecka | can I turn it off somehow? like disable it? |
| 21:15.47 | Corydon-w | Well, since you probed wcfxo first, that module claimed channel 1 |
| 21:15.53 | doughecka | ah |
| 21:15.54 | Corydon-w | rmmod wcfxo |
| 21:16.01 | doughecka | so I need 2-12 then |
| 21:16.09 | Corydon-w | and dchan=25 |
| 21:17.03 | doughecka | ZT_SPANCONFIG failed on span 1: No such device or address (6) |
| 21:17.03 | doughecka | span =2 then? |
| 21:17.03 | Corydon-w | Nope |
| 21:17.11 | doughecka | woot, that worked though :) |
| 21:17.19 | Corydon-w | Only 1 X100P? |
| 21:17.23 | doughecka | yea |
| 21:20.04 | doughecka | asterisk still doesnt start though, but ztcfg says it configured it |
| 21:20.21 | Corydon-w | Well, if ztcfg gives an error, then it didn't |
| 21:20.39 | MikeJ[Laptop] | hey Corydon-w |
| 21:20.40 | Corydon-w | The info columns are printed before it does the ioctl that actually does the configuration |
| 21:21.11 | *** join/#asterisk enzo123 (~jason@cpe-024-163-075-146.nc.res.rr.com) |
| 21:21.26 | doughecka | ztcfg has no error |
| 21:21.45 | Corydon-w | Okay, did you make the channel change to /etc/asterisk/zapata.conf ? |
| 21:21.46 | enzo123 | so everone got E911 figured out yet => |
| 21:21.51 | Corydon-w | to 2-12 ? |
| 21:22.02 | doughecka | yes |
| 21:22.20 | Corydon-w | Okay, start asterisk manually. What's the error? |
| 21:22.24 | blitzrage | darwin35: you around? |
| 21:22.39 | doughecka | Ouch ... error while writing audio data: : Broken pipe |
| 21:22.46 | Corydon-w | Before that |
| 21:22.55 | doughecka | <PROTECTED> |
| 21:23.01 | Corydon-w | Keep on going |
| 21:23.09 | Corydon-w | Should be an ERROR or WARNING |
| 21:23.39 | *** join/#asterisk dml337ira (~ira@206.127.39.5) |
| 21:23.50 | doughecka | I grepped it, no error or warning except that last one |
| 21:24.25 | Corydon-w | Do you have console => notice,error,warning in /etc/asterisk/logger.conf ? |
| 21:25.14 | doughecka | full => notice,warning,error,debug,verbose |
| 21:25.31 | doughecka | aha |
| 21:25.39 | Corydon-w | Add a line for console, so you can see it there |
| 21:25.43 | doughecka | Unknown signalling method 'pri-cpe' |
| 21:25.55 | Corydon-w | change that - to an underscore |
| 21:25.59 | doughecka | ah |
| 21:26.28 | doughecka | coool |
| 21:28.31 | doughecka | yay |
| 21:29.00 | doughecka | should outgoing caller id be 10 or 11 digits? |
| 21:29.52 | shmaltz | doughecka, 10 |
| 21:30.05 | lichen | so how is e911 implemented on voip systems? does the voip user just have to maintain a database of their current locations and the e911 crew can access that? |
| 21:30.40 | shmaltz | lichen, yeah, and when you are bored you set your address to 1600 Penn ave. in DC and call 911 |
| 21:30.46 | JerJer[mobile] | the provider has to have some method to update the ALI info |
| 21:30.53 | kFuQ | cool... * is in the OpenBSD 3.7 ports |
| 21:31.03 | JerJer[mobile] | shmaltz: there is going to have to be some sort of verification process, i'm sure |
| 21:31.27 | *** part/#asterisk wrmem (~wrmem@monnin-win.cso.uiuc.edu) |
| 21:31.28 | shmaltz | yeah, JerJer, how? |
| 21:31.30 | shmaltz | I live in NJ, but my IP shows up as from Canada |
| 21:31.31 | dpryo | Is it possible to configure asterisk as a client to cisco call manager? or a cisco VG200? (I don't have any other options from getting to PSTN) |
| 21:31.52 | JerJer[mobile] | dpryo: not without development |
| 21:32.23 | dpryo | JerJer[mobile]: Like a channel module for asterisk or something? |
| 21:32.28 | Corydon-w | doughecka: btw, if you're interested... http://nlug.org/canoe.php |
| 21:32.43 | Corydon-w | doughecka: it'll be up near the KY-TN line |
| 21:33.07 | doughecka | oh, that was last year as well |
| 21:33.26 | doughecka | cool |
| 21:33.29 | doughecka | hmm |
| 21:33.36 | doughecka | callerid isnt being passed out |
| 21:34.11 | Corydon-w | It won't be unless your telco allows you to set it |
| 21:34.34 | doughecka | oh? |
| 21:34.37 | doughecka | that sucks |
| 21:34.50 | doughecka | co direction = 2way |
| 21:34.54 | doughecka | whats that |
| 21:35.09 | Corydon-w | That means that you can receive calls AND make calls |
| 21:35.37 | doughecka | oh |
| 21:35.53 | Corydon-w | Some PRIs are set up for unidirectional only |
| 21:35.57 | doughecka | ah |
| 21:36.06 | doughecka | what would the set caller id thing be called? |
| 21:36.26 | Corydon-w | Probably 'restrict callerid' |
| 21:36.46 | Corydon-w | which means that they don't allow you to set it for other than a specific range |
| 21:36.50 | doughecka | hmm, I wonder if anyone is still there |
| 21:36.51 | Corydon-w | or a specific number |
| 21:36.54 | doughecka | ah, ok |
| 21:37.25 | Corydon-w | Tell them you want to be able to forward calls with the original callerid and to unrestrict your callerid settings |
| 21:37.37 | Corydon-w | That's a common legitimate usage |
| 21:37.39 | doughecka | sweet, got rid of the echo |
| 21:37.49 | doughecka | ok |
| 21:37.56 | Corydon-w | Be careful, though |
| 21:38.01 | doughecka | hmm, cant recieve calls either |
| 21:38.22 | Corydon-w | There's nothing inherent in the phone system that prevents you from sending your internal extension across the PRI |
| 21:38.41 | doughecka | yea |
| 21:38.46 | Corydon-w | I've made call from TN to CA which came across as callerid "103", which is my internal extension |
| 21:39.10 | doughecka | ah, it needed a did to recieve calls |
| 21:39.19 | doughecka | yea, I once made a call from FWD when they had free long distance |
| 21:39.19 | Corydon-w | How many digits did you get? |
| 21:39.30 | doughecka | and my caller id was like 70203 |
| 21:39.42 | Corydon-w | How many DID digits, rather |
| 21:39.45 | doughecka | I got 20 dids for 3 bucks a month |
| 21:39.50 | jarrod | werd to digium processing my g729 order |
| 21:40.07 | Corydon-w | 4 digit? 7 digit? 10 digit? |
| 21:40.07 | Jedirl | jarrod: already got your licenses? |
| 21:40.31 | *** join/#asterisk shido6 (~greg@d57-87-253.home.cgocable.net) |
| 21:40.51 | Corydon-w | I know one company that gets a 17 digit DID |
| 21:41.16 | doughecka | oh, I got a 3 digit |
| 21:41.30 | doughecka | they can change it to 4 if I need it |
| 21:42.00 | jarrod | jedirl: yessum |
| 21:42.14 | Corydon-w | We tend to tell our customers to get 10... so no confusion, even if they port other area codes to their system... |
| 21:42.26 | Jedirl | jarrod: how much time did it take ? |
| 21:43.05 | *** join/#asterisk Vco (~Vco@S0106080020aa7650.wp.shawcable.net) |
| 21:44.10 | doughecka | hmm, still not setting callerid right |
| 21:44.16 | doughecka | do I need to set something in my config first? |
| 21:44.43 | dsfr | Jedirl, he received them really quick. |
| 21:44.52 | Jedirl | great |
| 21:45.51 | doughecka | May 20 17:44:39 NOTICE[10963]: chan_zap.c:7395 pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 2 |
| 21:46.31 | *** join/#asterisk r0d3nt (~RatMan@wsip-24-234-241-84.lv.lv.cox.net) |
| 21:46.33 | Corydon-w | doughecka: are you doing a SetCIDNum in your dialplan? |
| 21:47.00 | doughecka | yea |
| 21:47.05 | doughecka | I think, let me check |
| 21:47.10 | Corydon-w | Or Set(CALLERID(num)=1234567890), if you're running on CVS HEAD |
| 21:47.35 | doughecka | <PROTECTED> |
| 21:48.04 | Corydon-w | Yep... just before you Dial... that should do it |
| 21:48.15 | Corydon-w | Unless your telco still isn't letting you change the CID |
| 21:48.22 | Corydon-w | Oh! |
| 21:48.37 | Corydon-w | One other thing. Are you setting usecallerid=no in zapata.conf ? |
| 21:48.52 | Corydon-w | You might have turned that off when using the X100P |
| 21:48.53 | doughecka | no |
| 21:48.59 | Corydon-w | also, hidecallerid=yes |
| 21:48.59 | doughecka | its not defined |
| 21:49.19 | doughecka | I get callerid just fine |
| 21:49.34 | Corydon-w | hidecallerid prevents you from setting it on the PRI |
| 21:49.51 | Corydon-w | i.e. it hides it from the remote end |
| 21:50.22 | doughecka | I just turned those off, still nothing |
| 21:50.30 | doughecka | usecallerid=yes |
| 21:50.30 | doughecka | hidecallerid=no |
| 21:50.39 | Corydon-w | You did a restart after changing them? |
| 21:50.42 | doughecka | yea |
| 21:50.59 | Corydon-w | Hmmm, I'd say telco problem, then |
| 21:51.15 | Jedirl | I have a "hardware-based" SIP VoIP gateway for which I want to set up a SIP proxy so that I can auth people trying to use the gateway... should I go asterisk? ser? partysip? what's the best option for this? |
| 21:51.31 | Corydon-w | Asterisk is not a sip proxy |
| 21:51.58 | Jedirl | just registrar then? |
| 21:52.00 | *** join/#asterisk PBXtech (~nik@64.122.26.53) |
| 21:52.10 | Corydon-w | Asterisk is a gateway |
| 21:52.13 | *** join/#asterisk freat[laptop] (~freat[lap@65.170.62.117) |
| 21:52.22 | Corydon-w | Yes, it can also be a registrar |
| 21:52.24 | PBXtech | what causes cpu irq load? typically hard drive? |
| 21:52.25 | freat[laptop] | hello |
| 21:52.56 | Jedirl | Corydon-w, what I want to do has to be done with a SIP proxy? |
| 21:52.56 | Corydon-w | PBXtech: anything that demands access to the PCI bus |
| 21:53.05 | rg1 | i used safe_asterisk to start asterisk - how do i shut asterisk down? |
| 21:53.09 | freat[laptop] | hey I downloaded HEAD 2 days ago... Im not able to transfer calls correctly (incoming from another IAX, hit transfer, dial outside line, transfer fails) |
| 21:53.11 | Corydon-w | Jedirl: I dunno, what do you want to do? |
| 21:53.14 | *** part/#asterisk Keizer (~keizer@pcp02851954pcs.ctftmy01.fl.comcast.net) |
| 21:53.21 | PBXtech | so its normal? i get pegs up in the 90% |
| 21:53.25 | freat[laptop] | anyone hear of a patch? |
| 21:53.36 | Jedirl | Corydon-w: have a place in which SIP devices can "log-in" and dial-out trought my SIP gateway |
| 21:53.41 | freat[laptop] | I searched Mantis but am kind of unfamiliar with it |
| 21:53.42 | Jedirl | through |
| 21:54.12 | Corydon-w | freat[laptop]: I wasn't aware Asterisk had a transfer button |
| 21:54.26 | Jedirl | Corydon-w: What I want is called a SIP proxy? a SIP registrar? both? |
| 21:54.44 | Corydon-w | Jedirl: Hate to say it, but "depends" |
| 21:54.49 | Jedirl | hehehe |
| 21:55.01 | Jedirl | could you help me to figure out please... |
| 21:55.11 | Corydon-w | Are you talking about majorly dynamic SIP clients? |
| 21:55.16 | Corydon-w | Or majorly static? |
| 21:55.25 | freat[laptop] | statis sip clients in the office |
| 21:55.27 | freat[laptop] | stic |
| 21:55.31 | freat[laptop] | dammit... static |
| 21:55.37 | doughecka | woooot |
| 21:55.39 | Jedirl | with "dynamic" you mean clients over the internet with unknown IPs? |
| 21:55.47 | doughecka | recieving/emailing the fax as a PDF works |
| 21:55.47 | sudhir492 | Can someone help me with priority of extensions? |
| 21:55.51 | doughecka | this is COOL |
| 21:56.04 | Jedirl | this is the case, "dynamic" clients in internet with unknown ip's |
| 21:56.13 | Corydon-w | No, I mean clients over the Internet that you may see once and never again? |
| 21:56.16 | *** join/#asterisk SarahEmm (~sarahemm_@Toronto-HSE-ppp3704598.sympatico.ca) |
| 21:56.16 | SarahEmm | clear |
| 21:56.19 | Jedirl | oh |
| 21:56.24 | Jedirl | nope |
| 21:56.39 | Corydon-w | Jedirl: what kind of scale? |
| 21:56.47 | Jedirl | just "authenticated" clients which can dial-out and receive calls |
| 21:56.51 | sudhir492 | In [default], I have include => context1, and in [context1] I have include => context2 |
| 21:57.03 | Jedirl | scale... well, my gateway has 6 E1s available for dialing out |
| 21:57.06 | Jedirl | and receiving calls |
| 21:57.08 | Corydon-w | Jedirl: how many, how fast are they signing up? |
| 21:57.08 | enzo123 | anyone get MSN to work with * ? |
| 21:57.12 | enzo123 | esp video |
| 21:57.21 | sudhir492 | Still something in context2 takes precedence over exten in context1 |
| 21:57.33 | Jedirl | how many... don't know, at the moment 150-200 |
| 21:57.35 | Corydon-w | Jedirl: it's still up to you whether you use a proxy or not |
| 21:57.38 | sudhir492 | Jedirl: what do you |
| 21:57.51 | sudhir492 | use your gateway for? Where |
| 21:57.55 | Corydon-w | Jedirl: are the majority calling each other (not through your gateway)? |
| 21:58.18 | Corydon-w | If the majority of calls are calling through the gateway, then it may be kind of pointless to use a proxy |
| 21:58.22 | Jedirl | no, the majority are dialing to numeration which has to be terminated over the E1 interfaces |
| 21:58.49 | *** join/#asterisk syslod (~yurplsl@65.114.15.70) |
| 21:58.51 | Corydon-w | Proxy only gains you something when you've got calls going SIP to SIP, not SIP to PSTN |
| 21:58.58 | Jedirl | Corydon-w: but my SIP gateway isn't a "registrar"... it registrates itself in a registrar |
| 21:59.04 | syslod | Helo |
| 21:59.16 | Corydon-w | "registrates"? |
| 21:59.20 | Jedirl | sorry |
| 21:59.20 | Jedirl | hehehe |
| 21:59.21 | *** join/#asterisk dimas (~ds@vbc.elcom.ru) |
| 21:59.22 | Jedirl | registers |
| 21:59.38 | Jedirl | having concurrent conversations in three languages right now :) |
| 21:59.53 | syslod | anyone know what PBX max exceeded is? |
| 21:59.57 | Corydon-w | Don't forget "nucular" |
| 22:00.06 | Jedirl | nucular? |
| 22:00.19 | *** join/#asterisk asteriskDOTbz (~logger@telux.net) |
| 22:00.21 | asteriskDOTbz | <PROTECTED> |
| 22:00.25 | Corydon-w | It's the way the American president says "nuclear" |
| 22:00.33 | Jedirl | hehehehehhehe |
| 22:02.04 | Jedirl | Corydon-w: the problem is my SIP gateway is kind of "dumb" device... if you use it in H.323, you have to use it with an external gatekeeper. With SIP, I thought I had to use a SIP proxy to achieve the same as with H.323... |
| 22:02.05 | syslod | The error is pbx.c:792 pbx_find_extension: Maximum PBX stack exceeded |
| 22:02.29 | Corydon-w | Jedirl: you don't have to, no |
| 22:02.50 | Corydon-w | You don't even necessarily have to with H.323 either |
| 22:02.55 | Jedirl | then how can I have some kind of database in which my clients register ? |
| 22:03.21 | Jedirl | and only when they're succesfully register, can dial-in and dial-out calls? |
| 22:03.58 | Jedirl | heh my english goes worse each day |
| 22:04.36 | doughecka | gets worse* |
| 22:04.36 | Corydon-w | You can do that with an Asterisk gateway now |
| 22:05.07 | Jedirl | Corydon-w: but my gateway isn't asterisk... |
| 22:05.21 | Corydon-w | Actually, SIP clients don't generally have to register with Asterisk to make calls... only to receive them |
| 22:05.25 | smash- | hey |
| 22:05.50 | Corydon-w | Jedirl: ah. Well, I don't know the answer to your question, then. You might ask the vendor of your gateway |
| 22:06.09 | Jedirl | ok thanks |
| 22:06.25 | smash- | Hey does anyone use ANSI , Q933 or LMI frame relay signaling |
| 22:06.47 | Corydon-w | we do frame relay here |
| 22:07.57 | Corydon-w | LMI=ansi |
| 22:11.27 | jarrod | hmm |
| 22:11.35 | smash- | does anyone use a sangoma card |
| 22:11.54 | jarrod | im goin sip->iax2->h323 and im only getting one-way communication. my iax2 trunk is reporting 'IAX2/ast-pri1/1 stopped sounds' |
| 22:12.13 | doughecka | this is rather cool stuff |
| 22:12.22 | jarrod | what could cause that :( |
| 22:13.41 | darwin35 | yes spandsp |
| 22:13.45 | darwin35 | go read |
| 22:13.48 | jarrod | ive done quite a few pages at high error rate |
| 22:14.01 | doughecka | it recieves them great |
| 22:14.13 | darwin35 | I send at 2400/9600 no issue |
| 22:14.23 | darwin35 | 14400 seems to have issues |
| 22:14.32 | jarrod | how are you sending? via print method ? |
| 22:14.40 | smash- | dude my timer doesnt work |
| 22:14.41 | jarrod | you dont send the fax over a voip codec to the asterisk box |
| 22:14.52 | darwin35 | no I jjust dop the file into a send dir |
| 22:14.53 | doughecka | I can recieve them fine, on a T1 |
| 22:15.03 | doughecka | would be cool to email the asteriskbox and have it fax it |
| 22:15.11 | jarrod | dop? |
| 22:15.18 | darwin35 | drop |
| 22:15.28 | jarrod | hmm |
| 22:17.16 | doughecka | bkw_: I have an asterisk box with T1 finally |
| 22:17.32 | doughecka | and its cool |
| 22:18.37 | syslod | Is there a route tool in * yet? |
| 22:21.22 | SarahEmm | cooooool! chan_zap supports TDD MODE out of the box :) |
| 22:22.52 | *** join/#asterisk jwitte (~jwitte_@firefly.alpha-lab.net) |
| 22:23.13 | Qwell | SarahEmm: out of the box, or in the box? |
| 22:23.21 | Qwell | I assume it needs to be in your * server to work. ;] |
| 22:23.46 | *** join/#asterisk JunK-Y (~junky@modemcable174.107-81-70.mc.videotron.ca) |
| 22:23.58 | SarahEmm | well, in the box ;) |
| 22:24.09 | SarahEmm | i'd expected to have to write support for TTY/TDDs into asterisk :) |
| 22:24.17 | Qwell | well, you need to think outsi...nevermind. ;) |
| 22:24.22 | SarahEmm | but chan_zap has it built in :) |
| 22:24.28 | SarahEmm | sending, anyway. haven't tried receiving yet. |
| 22:24.33 | SarahEmm | nor do i know how to :) |
| 22:25.25 | jarrod | why am i getting one-way audio!!! |
| 22:25.26 | jarrod | grrr |
| 22:25.39 | doughecka | omg I havent heard that in sooo long |
| 22:26.02 | jarrod | its gotta be a link in muh chain |
| 22:26.24 | jarrod | polycom -- sip -- SER -- sip -- AST1 -- IAX2 -- AST2 -- h323 -- carrier |
| 22:26.25 | jarrod | heh |
| 22:26.31 | SarahEmm | yikes |
| 22:26.43 | doughecka | holy |
| 22:26.44 | doughecka | crap |
| 22:26.49 | jarrod | no doubt hah |
| 22:26.56 | jarrod | :) |
| 22:27.05 | jarrod | atleast i get it one way :) |
| 22:27.51 | jarrod | haha |
| 22:28.12 | jarrod | i have one asterisk box that connects to upstream carriers for ld/international/1800 |
| 22:28.18 | jarrod | and another for PSTN |
| 22:28.27 | jarrod | thats why its messy like that :) |
| 22:28.33 | jarrod | and this one carrier REQUIRES h323/g729 |
| 22:28.34 | jarrod | eww |
| 22:29.47 | jarrod | i need to debug! |
| 22:37.07 | jarrod | GRRR |
| 22:38.06 | SarahEmm | meow? |
| 22:38.12 | jarrod | woof |
| 22:39.41 | jarrod | hmmmmmmmmmmmmmmm |
| 22:39.51 | jarrod | another thing that sucks about having a g729 call leg |
| 22:40.04 | jarrod | is if my phones dont register g729 it gives me error about translating between two codecs |
| 22:40.49 | *** join/#asterisk Manipura (~chatzilla@dsl-hunt-66-18-224-38-cgy.nucleus.com) |
| 22:40.49 | Manipura | Has anyone used Linked2Voip for termination? |
| 22:40.49 | jarrod | no u can use me tho :) |
| 22:53.37 | SarahEmm | is there any easyish way to debug AGI scripts? i'm writing in perl, and i'm having issues.. |
| 23:00.05 | JerJer[mobile] | you can run the agi script in a shell and see if complains about compiling |
| 23:00.11 | JerJer[mobile] | ie you screwed up |
| 23:00.22 | JerJer[mobile] | then just have lots of debug in your agi |
| 23:00.24 | SarahEmm | it didn't |
| 23:00.33 | SarahEmm | yeah, right after i asked that i found out about VERBOSE :) |
| 23:00.38 | SarahEmm | seems to work well, found the issue |
| 23:00.39 | SarahEmm | :) |
| 23:01.39 | *** join/#asterisk sedwards50 (~chatzilla@adsl-64-163-31-213.dsl.sndg02.pacbell.net) |
| 23:02.46 | sedwards50 | Any Digium hardware type guys around? |
| 23:03.45 | *** join/#asterisk prh (~paul@wacka.mjr.org) |
| 23:04.13 | *** join/#asterisk Zaw (zaw@zaw.subneural.net) |
| 23:04.32 | sedwards50 | Any Digium hardware type guys around? |
| 23:04.55 | SarahEmm | you don't need to repeat :) |
| 23:04.56 | sedwards50 | Anybody have experience with NFAS on PRI? |
| 23:05.30 | sedwards50 | Sarah -- thanks, I didn't think anybody was actually here -- no "traffic." |
| 23:06.50 | SarahEmm | i'm here |
| 23:07.03 | SarahEmm | i'm working on some AGI scripting tho |
| 23:07.09 | SarahEmm | and have no experience with digium PRI cards |
| 23:07.22 | sedwards50 | This is the "quietest" I've ever seen this channel. |
| 23:07.44 | sedwards50 | What language to you use to write AGI's? |
| 23:08.46 | SarahEmm | i've only done a few simple ones, but perl |
| 23:08.53 | SarahEmm | i'm just playing with TDD support right now |
| 23:09.07 | SarahEmm | sending works, i can't make receiving work, may not be support for it in chan_zap i think :( |
| 23:09.48 | sedwards50 | I use C myself. What are you trying to send/receive? |
| 23:10.17 | *** join/#asterisk zSACREDz (~zSACREDz@69.111.188.16) |
| 23:10.21 | SarahEmm | text on the channel |
| 23:10.25 | SarahEmm | send_text is fine |
| 23:10.32 | SarahEmm | receive_char is returning 0 (unsupported or timeout) |
| 23:12.52 | sedwards50 | No experience with sendtext. Sorry. |
| 23:15.18 | SarahEmm | s'ok :) |
| 23:15.20 | SarahEmm | sendtext is fine! :) |
| 23:15.25 | SarahEmm | receivechar isn't... |
| 23:15.43 | SarahEmm | seems silly to be able to send but not receive on a channel... |
| 23:16.24 | *** join/#asterisk Inv_arp (junya@adsl-8-230-54.mia.bellsouth.net) |
| 23:16.41 | sedwards50 | Any clue how to put an agent in a conference? |
| 23:21.42 | SarahEmm | hrm. changelog says it *Should* work in chan_zap.... |
| 23:26.23 | *** join/#asterisk Pete_Largo (~Pete_Larg@adsl-65-71-225-121.dsl.rcsntx.swbell.net) |
| 23:28.15 | SarahEmm | hmm. it's detecing a character was sent, but not returning it to me.. |
| 23:39.13 | enzo123 | anyone do video with * ? |
| 23:47.59 | *** join/#asterisk doolph (doolph@200.46.148.35) |
| 23:48.01 | doolph | hello |
| 23:48.09 | SarahEmm | hihi |
| 23:48.21 | doolph | anyone can help me? I am logged in FWD, I can make calls, but no receive |
| 23:48.30 | SarahEmm | ahh. looks like a perl issue, not an AGI issue :o) silly me. too rusty on perl... |
| 23:48.38 | SarahEmm | doolph: does anything at all happen when someone calls you? |
| 23:49.18 | doolph | it seems that is not registered |
| 23:49.30 | Pete_Largo | anyone know anything about libnet? I am trying to install libnet on Fedora Core 3 from source (not rpm) and am getting no-where. http://pastebin.ca/12306 |
| 23:52.02 | doolph | USER Context: iaxfwd (Pay attention here. Don't change it. or it won't work) |
| 23:52.04 | doolph | mmm |
| 23:52.15 | SarahEmm | doolph: okay, did you set it up to register? |
| 23:52.23 | SarahEmm | oh, are you connecting via IAX? |
| 23:52.56 | doolph | yes |
| 23:53.10 | SarahEmm | ahh.. i have no experience with FWD via IAX... |
| 23:54.11 | *** join/#asterisk bkw__ (~brian@adsl-69-154-144-52.dsl.tulsok.swbell.net) |
| 23:54.42 | doolph | then |
| 23:55.40 | doolph | do you have fwd on? |
| 23:55.48 | SarahEmm | me? |
| 23:55.49 | SarahEmm | not right now |
| 23:55.57 | SarahEmm | my asterisk box is up and down while i fight with this AGI script |
| 23:56.17 | JunK-Y | SarahEmm: whats wrong in ur agi? |
| 23:56.17 | doolph | argh |
| 23:56.42 | doolph | Name/Username Host Mask Port Status |
| 23:56.42 | doolph | fwd/414197 65.39.205.121 (S) 255.255.255.255 4569 Unmonitored |
| 23:56.49 | doolph | it suppose that is connected |
| 23:57.12 | *** join/#asterisk ariel_ (~Ariel@dsl-20-177.cofs.net) |
| 23:57.14 | JunK-Y | doolph: add qualify=yes |
| 23:57.29 | SarahEmm | JunK-Y: having trouble in perl AGI getting receive_char working to receive text on a channel |
| 23:57.33 | JunK-Y | you'll see something better then "Unmonitored" |
| 23:58.13 | doolph | for what is it |
| 23:58.38 | JunK-Y | doolph: i wont explain what ya can easily find in wikis/mailing list. |
| 23:59.13 | SarahEmm | JunK-Y: experienced in perl AGI at all? |
| 23:59.18 | JunK-Y | SarahEmm: and u sure your channel can receive char? |
| 23:59.27 | SarahEmm | JunK-Y: the changelog says that chan_zap can |
| 23:59.30 | JunK-Y | not in the perl-agi object, ive my own object. |
| 23:59.33 | SarahEmm | ahh okay. |
| 23:59.44 | SarahEmm | the changelog says that a long time ago receive_char support was added for zap channels |
| 23:59.49 | SarahEmm | send_char also works great, i've been using that |
| 23:59.53 | SarahEmm | err, send_text rather |
