| 00:03.14 | gandhijee | anyone know why the polycom's double post on submit? |
| 00:03.59 | De_Mon | is there a letters to digits function/application? |
| 00:04.13 | De_Mon | or macro |
| 00:07.36 | wasim | De_Mon: app_directory |
| 00:09.53 | CunningPike | Later, guys |
| 00:11.34 | *** join/#asterisk florz (i=nobody@2001:1a50:503c:0:0:0:0:1) |
| 00:13.24 | VoicePulse | Any VoicePulse users awake? |
| 00:13.49 | De_Mon | app_directory? doesn't that match user names to extensions? |
| 00:14.31 | *** join/#asterisk slashdotw (i=vsftpd_S@200.80.197.5) |
| 00:16.04 | slashdotw | hi there, I'm new to asterisk.. haven't red much about it, but I want to ask a simple question to see if this app could satisfy my needs.. can I do that ? |
| 00:16.40 | slashdotw | just asking to avoid the common RTFM... =) |
| 00:16.45 | Nugget | it would be faster and less annoying if you didn't ask for permission to ask a question. |
| 00:17.17 | slashdotw | ok, let's see |
| 00:17.21 | Nugget | the best way to avoid getting told to RTFM is to RTFM. |
| 00:17.29 | slashdotw | sure... |
| 00:18.37 | slashdotw | anyway, I'm setting up a wireless network with some friends, and I have a landline at home |
| 00:19.23 | slashdotw | my idea is to set up an asterisk server, connect this linux box to the land line |
| 00:19.42 | slashdotw | and share this line with my friends |
| 00:20.06 | slashdotw | of course I have to use some sw to link this server with my friend's PCs |
| 00:20.51 | slashdotw | basically, set a linux server to share the internet connection and the phone line |
| 00:21.03 | Nugget | that's quite feasible. about the only thing that might prove to be too much of a pain in the ass would be if you wanted to do data over it (like for a tivo or something) |
| 00:21.23 | slashdotw | sorry, what's a tivo |
| 00:21.27 | Nugget | modems and voip are brittle even under the best of circumstances. |
| 00:21.43 | Nugget | a pvr thing that needs to dial out to home base every few days |
| 00:22.32 | slashdotw | you think I could get this running |
| 00:23.00 | slashdotw | connecting the linux server to the landline using just a modem ? |
| 00:24.00 | slashdotw | then link to each PC with voIP (some software I don't know yet) using the wireless network |
| 00:25.50 | slashdotw | have to go, thanks nugget, later. |
| 00:26.04 | pcrook | I can't take this anymore. |
| 00:26.19 | *** part/#asterisk pcrook (n=pcrook@fw.latis.com) |
| 00:35.46 | Malthus | dlynes: I think the DTMF ignoring has stopped with the upgrade to 1.2.7.1 |
| 00:40.47 | *** join/#asterisk kio (n=kio@ool-4577ae5e.dyn.optonline.net) |
| 00:40.47 | *** join/#asterisk coppice (n=chatzill@153.192.17.210.dyn.pacific.net.hk) |
| 00:43.10 | *** join/#asterisk Gunnar (n=gunnar@34.80-203-209.nextgentel.com) |
| 00:45.47 | *** join/#asterisk riddlebox (n=james@24-171-10-102.dhcp.stls.mo.charter.com) |
| 00:54.51 | *** join/#asterisk shido6 (n=shido6@d38-45-81.commercial1.cgocable.net) |
| 00:56.24 | *** join/#asterisk jeebusmobile (n=jeebusmo@29palms-cuda1-68-170-33-192.losaca.adelphia.net) |
| 01:02.51 | Dr-Linux | whenever i reload chan_zap.so , i get on CLI, Ignoring signalling but everthing works fine |
| 01:03.03 | Dr-Linux | why this error occures? |
| 01:03.11 | hads|home | That's normal. |
| 01:03.14 | Dr-Linux | sorry warning |
| 01:03.41 | Dr-Linux | hads|home: but there should be a reason beyond this :S |
| 01:04.07 | hads|home | It's just telling you that it's not resetting the signalling. There are posts on -users about it somewhere. |
| 01:04.35 | russellb | Dr-Linux: the reason is that you can't change the signalling without restarting asterisk - that's all |
| 01:04.40 | russellb | not a problem |
| 01:05.23 | *** join/#asterisk joat (n=joat@ip70-160-147-169.hr.hr.cox.net) |
| 01:05.31 | Dr-Linux | it's ok, if it's not a problem |
| 01:06.36 | Dr-Linux | russellb: actually i never do entire reload, i just reload specfic edited application/channel module |
| 01:07.02 | russellb | yeah, it's the same thing |
| 01:07.16 | russellb | plain "reload" just runs reload on every module that is loaded |
| 01:08.32 | Dr-Linux | russellb: also Wiki says, for better performance give a restart to asterisk in off hours |
| 01:08.45 | russellb | heh ... silly wiki. |
| 01:08.47 | Dr-Linux | but here someone said, no need to restart asterisk |
| 01:09.12 | russellb | there should not be a need to restart asterisk |
| 01:09.22 | russellb | under normal operation |
| 01:09.32 | Dr-Linux | russellb: actually i didn't restart my asterisk since 136 days |
| 01:09.37 | russellb | nice! |
| 01:09.42 | russellb | that's awesome |
| 01:10.27 | Dr-Linux | even this was my first astrisk installation |
| 01:10.40 | russellb | what version is it |
| 01:10.43 | Dr-Linux | but have no issues, that's why i care about. |
| 01:10.56 | Dr-Linux | 1.2.0 |
| 01:11.25 | Dr-Linux | even i did some patch as well, and they patch was for 1.2.6 |
| 01:11.25 | russellb | well it's always nice to hear from the users that *aren't* having problems |
| 01:11.44 | Dr-Linux | but it worked on 1.2.0 |
| 01:11.46 | Dr-Linux | :) |
| 01:12.07 | Dr-Linux | even during patch i was going to restart asterisk and compile again |
| 01:12.25 | Dr-Linux | but anthm helped good, and done all without restart asterisk :) |
| 01:13.29 | Dr-Linux | today i just configured MultiTech gateways with asterisk, woww very good |
| 01:21.31 | *** join/#asterisk kainam (n=Jake@202.137.160.110) |
| 01:23.36 | *** join/#asterisk iceyp (n=icepick@firewall.unix.co.nz) |
| 01:24.15 | iceyp | Hey guys, is it possible to use Dial() to point to a context and use prefixs within there? I.e. I have a exten => s,2,Dial(SIP/${MYNUM}@akl.italk.co.nz,20,Ttr) at the moment, but i want it to do LCR based on my prefix list |
| 01:26.02 | iceyp | ahh wait Goto() might be what im agter |
| 01:26.08 | iceyp | nah maybe not |
| 01:30.46 | *** join/#asterisk BadPacket (n=BadPacke@unaffiliated/badpacket) |
| 01:30.52 | FuriousGeorge | im using sip phones w/ ulaw and an iax service, and dtmf is not that reliable. there's not really anything i can do about that, is there? |
| 01:33.07 | iceyp | is it possible to do something like exten => s,2,Dial(${MYNUM}@context,20,Ttr) and it will use the LCR dial from that @context? |
| 01:33.51 | iceyp | Id ont want to use a static provider for the Dial() it needs to be based on my cheapest-route context |
| 01:35.04 | *** join/#asterisk achandra (n=achandra@12.44.122.130) |
| 01:35.23 | BadPacket | anyone else notice that voicepulse is now sending caller id name for inbound calls? |
| 01:35.58 | achandra | okay...been eading up and searching for info on getting the res_snmp feature to work...and running into some stumbling blocks...can someone help out on getting this working? |
| 01:38.38 | *** join/#asterisk Dr-Linux (n=huh@202.59.73.131) |
| 01:39.09 | achandra | tried to find the res_snmp package but could find it...also couldnt find the sample mibs...whats required to get them? |
| 01:41.21 | Qwell | achandra: trunk |
| 01:41.47 | achandra | Qwell: in the sources or ?? |
| 01:41.48 | De_Mon | can I change the format of DATETIME? |
| 01:42.18 | tainted- | how do i remove '011' from ${EXTEN} |
| 01:42.30 | tainted- | using dialplan |
| 01:42.35 | tainted- | i'm total dialplan newb |
| 01:42.37 | russellb | ${EXTEN:3} |
| 01:43.03 | Qwell | russellb: home yet? |
| 01:43.09 | russellb | Qwell: yeah, just got here |
| 01:43.13 | achandra | Qwell: sorry..when you refer to trunk...what is path that I can find it? |
| 01:43.21 | Qwell | cool |
| 01:43.23 | russellb | svn co http://svn.digium.com/svn/asterisk/trunk |
| 01:43.29 | Qwell | achandra: svn trunk. if you don't know...you shouldn't run it |
| 01:43.48 | russellb | http://www.asterisk.org ... downloads |
| 01:43.50 | *** join/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 01:44.00 | achandra | even more reason too...its all experimental..thanks for the link... |
| 01:44.12 | paolob | Hi guys! How do I enable addresses translation? thank you! |
| 01:44.18 | Qwell | russellb: this is an overly broad question... Do you have any idea how many * users and/or * deployments there are? |
| 01:44.25 | russellb | Qwell: one trillion |
| 01:44.28 | Qwell | heh |
| 01:44.35 | russellb | no, i have absolutely no idea |
| 01:44.42 | russellb | many, many thousdands |
| 01:44.45 | De_Mon | 2 hunder thousand |
| 01:44.48 | russellb | we have absolutely no way to know |
| 01:44.52 | coppice | no |
| 01:44.53 | Qwell | russellb: yeah... |
| 01:45.02 | FuriousGeorge | put spyware in the server that makes it phone home |
| 01:45.03 | Qwell | I wonder how far off my estimates of "tens of thousands" was |
| 01:45.03 | De_Mon | well, you could go off of downloads like firefox does |
| 01:45.15 | De_Mon | no that wouldn't help nevermind |
| 01:45.16 | Qwell | meh, go off of iaxtel registrations :p |
| 01:45.24 | russellb | pfft, yeah right |
| 01:45.51 | coppice | digium should have some idea about the number of PSTN connected installs, through the cards they sell |
| 01:46.05 | Qwell | coppice: That's true, except, gotta figure resellers too |
| 01:46.06 | coppice | there is no way to know the number of pure IP installs |
| 01:46.20 | russellb | there are probably more pure IP users these days |
| 01:46.22 | paolob | Guys, how do I get asterisk work with hostnames instead of IP's? thank you! |
| 01:46.30 | russellb | paolob: it just works ... |
| 01:46.32 | Qwell | paolob: it "Just Works" |
| 01:46.35 | Qwell | jynx! |
| 01:46.38 | russellb | heh |
| 01:47.01 | coppice | its quite a lot, though. I stopped counting when there appeared to be a minimum of 10,000 spandsp FAX users |
| 01:47.14 | paolob | russellb, Qwell , no, it can't register: chan_sip.c:1966 create_addr: No such host: sips.technodelta.com |
| 01:47.14 | paolob | May 12 21:46:50 WARNING[4206]: chan_sip.c:5350 transmit_register: Probably a DNS error for registration to 18095687419@sips.technodelta.com |
| 01:47.15 | russellb | coppice: nice |
| 01:47.19 | Qwell | coppice: that many? damn |
| 01:47.31 | Qwell | paolob: Can you lookup that hostname from the same box? |
| 01:47.36 | russellb | paolob: that error looks pretty clear to me |
| 01:47.40 | paolob | Qwell, yes |
| 01:47.57 | coppice | that's only based on vague, woolly inputs I had, though |
| 01:48.01 | riddlebox | does anyone have any problem getting both lines on the sipura 2100 working? |
| 01:48.03 | paolob | russellb, ? |
| 01:48.14 | russellb | you can type "host sips.technodelta.com" and it resolves? |
| 01:48.31 | russellb | we should track ftp downlods when 1.4 is released |
| 01:48.58 | paolob | russellb, $ host sips.technodelta.com |
| 01:48.58 | paolob | sips.technodelta.com CNAME sipps.technodelta.com |
| 01:48.58 | paolob | sipps.technodelta.com A 83.138.130.145 |
| 01:49.18 | De_Mon | russellb it woln't help. all the distros will repackage |
| 01:49.20 | Qwell | Do it from asterisk |
| 01:49.22 | Qwell | !host sips.technodelta.com |
| 01:49.24 | coppice | dunno what that says. most serious installs are through resellers who will only download once |
| 01:49.28 | russellb | De_Mon: yeah, that's the problem |
| 01:49.32 | Qwell | De_Mon: anybody who uses distro packages, should be shot, and not included :p |
| 01:49.46 | paolob | *CLI> !host sips.technodelta.com |
| 01:49.47 | paolob | sips.technodelta.com CNAME sipps.technodelta.com |
| 01:49.47 | paolob | sipps.technodelta.com A 83.138.130.145 |
| 01:49.48 | De_Mon | fine. I'll keep my patches |
| 01:50.08 | De_Mon | is freepbx still based on asterisk core? |
| 01:50.30 | Qwell | De_Mon: it doesn't use any asterisk code...since it isn't a...pbx |
| 01:50.38 | russellb | despite the name |
| 01:50.45 | coppice | its * packaged with other stuff |
| 01:50.53 | De_Mon | yeah, it's packaged with * |
| 01:50.54 | Qwell | freepbx? It's JUST AMP |
| 01:51.01 | paolob | russellb, Qwell, I remember a *.conf command to tell asterisk to translate hostnames... |
| 01:51.12 | russellb | paolob: you're probably thinking of dnsmgr.conf |
| 01:51.46 | russellb | but it resolves hostnames by default, dnsmgr allows managed lookups refreshed over time - useful for dynamic dns |
| 01:52.26 | De_Mon | can I format ${DATETIME} to YYYYMMDD-HHMMSS? |
| 01:52.28 | paolob | russellb, and then, why doesn't my asterisk resolve? |
| 01:52.36 | russellb | i have no idea. |
| 01:53.04 | De_Mon | lots of variable magic appears to be the only way :/ |
| 01:53.30 | russellb | paolob: i have been travelling all day, so I can't really think right now, sorry ... |
| 01:53.38 | *** join/#asterisk Assid (n=assid@203.115.83.214) |
| 01:53.42 | Assid | heya |
| 01:54.26 | Qwell | russellb: all day...that's an understatement |
| 01:54.51 | Assid | heya Qwell |
| 01:54.56 | Assid | whats cookin |
| 01:56.29 | paolob | russellb, ... I found it: srvlookup=yes ! |
| 01:57.13 | Assid | wooohoo |
| 01:58.02 | *** join/#asterisk Eggplant (n=none@dsl-216-155-212-221.cascadeaccess.com) |
| 01:58.14 | *** join/#asterisk Damin_PDA (n=pocketir@99.sub-70-217-223.myvzw.com) |
| 01:58.20 | Damin_PDA | werd... |
| 01:59.17 | De_Mon | dawg |
| 02:04.34 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
| 02:12.21 | achandra | okay..so after reading up on trunk...which contains the res_snmp package...is there any way to build that into the current working asterisk or do you need to recompile..all of it? |
| 02:12.36 | Damin_PDA | wow |
| 02:14.10 | *** join/#asterisk llirk (i=krill@snorkel.rtfm.net.au) |
| 02:14.14 | russellb | achandra: it depends which version you are currently running |
| 02:14.25 | russellb | if it's a recent version of trunk, then yes |
| 02:14.28 | russellb | if it's 1.2, then no |
| 02:14.44 | llirk | 12:13:21:950 H.225 Facility message Received (incoming, ooh323c_1) |
| 02:14.44 | llirk | 12:13:21:950 Unsupported userInput message type received - ignoring.(incoming, ooh323c_1) |
| 02:15.06 | llirk | Does anyone know what settings are required to get asterisk to recognise the DTMF coming from an Avaya PBX? |
| 02:15.37 | *** join/#asterisk Agrajag- (n=filip@c211-30-4-5.artrmn1.nsw.optusnet.com.au) |
| 02:16.05 | achandra | russelb: currently running 1.2.6 |
| 02:16.12 | Agrajag- | gday. i have a basic SIP setup - but now i'd like users to be able to dial other SIP addresses on the net, like foo@ekiga.net or whatever - how do i do this? |
| 02:16.14 | russellb | ok, then you will have to rebuild all of it |
| 02:16.22 | russellb | also, at the moment, res_snmp is not enabled by default |
| 02:16.24 | coppice | llirk: its "ubble, bubble, toil and trouble, DTMF work and cauldrens bubble" |
| 02:16.29 | russellb | you will need to run "make menuselect" to enable it |
| 02:16.51 | llirk | coppice, thanks. seriously, any ideas? |
| 02:16.56 | achandra | russelb: of the "top" so to speak...on the asterisk install |
| 02:17.10 | achandra | russelb: correct? |
| 02:17.10 | coppice | the snag with SNMP in the current *, is the cards don't monitor most of the important information |
| 02:17.14 | russellb | right, in the root source directory |
| 02:18.17 | achandra | russelb: and id suspect the version in svn is not for production use correct? |
| 02:18.25 | coppice | or rather the card drivers. only the tormenta 2 card is properly monitored |
| 02:18.30 | russellb | achandra: correct |
| 02:18.47 | De_Mon | how can I see what digits are being pressed after a peer has connected to *? |
| 02:19.01 | achandra | russelb: darn...otherwise cacti would be able to monitor a hell of a lot.. |
| 02:19.21 | russellb | feel free to try it, just don't be surprised when it breaks |
| 02:19.34 | *** join/#asterisk papa_e (i=papa_e@ip68-4-40-21.pv.oc.cox.net) |
| 02:20.02 | papa_e | dtmf with sipphone/gizmo on did -- any ideas? * just ignores any digits |
| 02:20.11 | papa_e | even with dtmfmode=inband |
| 02:20.12 | achandra | russellb: just curious though...is there another way to get the information..other than res_ to say cacti or mrtg available....or just gotta wait for it... |
| 02:20.48 | russellb | wait, i guess |
| 02:21.01 | russellb | it will be released in about 2 months or so |
| 02:21.18 | achandra | what other tools are available to effectively monitor * ? |
| 02:21.40 | achandra | at least the 1.2.6 version?..I know its a little genreic question...but... |
| 02:24.52 | paolob | Guys, what's this: "channel.c:2323 set_format: Unable to find a codec translation path from g729 to slin" ? I get it calling through wengo |
| 02:26.03 | coppice | that message seems clear enough. what don't you understand? |
| 02:27.03 | *** join/#asterisk anthm (n=anthm@CPE-69-76-83-52.wi.res.rr.com) |
| 02:27.03 | *** mode/#asterisk [+o anthm] by ChanServ |
| 02:29.48 | achandra | russellb: found something here.....illhave to play with it and see if it works... |
| 02:29.55 | achandra | http://www.faino.it/en/asterisk.html |
| 02:30.00 | Agrajag- | gday. i have a basic SIP setup for users on my lan - but now i'd like users to be able to dial other SIP addresses on the net, like foo@ekiga.net or whatever - is this possible? how is it setup? |
| 02:31.48 | *** join/#asterisk Peter-Brazil (n=pierre@201.41.234.40) |
| 02:32.05 | Peter-Brazil | hello my friends |
| 02:34.04 | Peter-Brazil | already the asterisk with videophone da grandstream? the gxv-3000 |
| 02:36.40 | *** join/#asterisk r0d3nt (n=RatMan@foster.stonedcoder.org) |
| 02:36.55 | coppice | anyone here actually played with the grandstream videophone? |
| 02:37.43 | Qwell | coppice: I've seen one.. |
| 02:37.50 | Qwell | looked okay |
| 02:38.09 | Peter-Brazil | but it is using o h264 |
| 02:38.23 | Peter-Brazil | the asterisk support this codec? |
| 02:39.41 | coppice | in pictures it looks better than most video phones, but that isn't saying much. most make a brick look stylish |
| 02:41.21 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 02:41.21 | *** mode/#asterisk [+o denon] by ChanServ |
| 02:48.39 | *** join/#asterisk Mavvie (n=edwin@203.222.131.252) |
| 02:50.15 | paolob | Guys, anyone could tell me what's this error: "chan_sip.c:1966 create_addr: No such host: sips.technodelta.com |
| 02:50.15 | paolob | May 12 22:49:16 WARNING[4206]: chan_sip.c:5350 transmit_register: Probably a DNS error for registration to 18095687419@sips.technodelta.com" ? "!host sips.technodelta.com" works perfectly! |
| 02:50.28 | *** part/#asterisk a1fa (n=a1fa@207.210.210.202) |
| 02:51.17 | paolob | it's asterisk2pstn voip server |
| 03:01.37 | Qwell | paolob: what version of * is this? |
| 03:02.10 | paolob | Qwell, Asterisk 1.2.1 built by kk @ nyx on a x86_64 running Linux on 2006-01-02 00:19:54 UTC |
| 03:02.25 | Qwell | upgrade |
| 03:02.30 | Qwell | then complain |
| 03:04.06 | paolob | Qwell, is it a bug? |
| 03:04.27 | Qwell | maybe, upgrade first, then test it |
| 03:06.07 | paolob | Qwell, ok |
| 03:11.52 | *** join/#asterisk dusan2 (i=dusan@209-223-47-160-static.oplink.net) |
| 03:12.52 | paolob | Qwell, I upgraded to 1.2.7, all is the same :-( |
| 03:17.24 | dlynes | paolob: Try this, from a Linux command line: ping sips.technodelta.com |
| 03:18.01 | paolob | dlynes, it works! |
| 03:18.07 | dlynes | paolob: does ping tell you what the ip address is for sips.technodelta.com? |
| 03:18.26 | dlynes | paolob: what works? the ping, or asterisk? |
| 03:18.28 | paolob | dlynes, 83.138.130.145 |
| 03:18.35 | paolob | the ping |
| 03:18.42 | FuriousGeorge | is there anything that can be done about unreliable dtmf, im calling w/ sip phones on iax voip service |
| 03:19.03 | dlynes | Ok, now in your asterisk configuration, put in 83.138.130.145 instead of sips.technodelta.com and see if that works |
| 03:19.11 | FuriousGeorge | its not that it doesnt work, its that if i dont give it a long hard push, i might get number+/- 1 instead of number |
| 03:19.34 | FuriousGeorge | i use ulaw too, soo... |
| 03:21.15 | FuriousGeorge | i was reading on the bugtracker that jitterbuffer can mess with DTMF, but that report said it was "all or nothing" where my problem is more of a "sorta" |
| 03:21.58 | paolob | dlynes, it works with the IP |
| 03:22.24 | paolob | dlynes, but WHY it doesn't with the host name?!? |
| 03:22.45 | coppice | if jitterbuffering messes up DTMF it means you are sending the DTMF as audio. that's a bad thing to do, anyway |
| 03:23.09 | FuriousGeorge | coppice: i thought dtmf was always inband with ulaw, lemme look into that |
| 03:24.14 | coppice | ulaw over IP != ulaw over PSTN. people keep treating them like they are the same |
| 03:24.48 | coppice | then they will tell you faxing over IP with ulaw works, because they saw it work one day. same with DTMF |
| 03:25.31 | dlynes | paolob: maybe you had a typo in the hostname? |
| 03:25.54 | dlynes | paolob: maybe you changed dns servers while asterisk was running, and haven't restarted asterisk since, so it's still caching the old ip address? |
| 03:26.17 | dlynes | paolob: maybe you had a space in there with the hostname, that you don't have in there with the ip address? |
| 03:26.24 | dlynes | paolob: could be any number of reasons why |
| 03:26.50 | dlynes | paolob: replace the ip address with the hostname, make sure there's no spaces in there, save, restart asterisk and try again |
| 03:27.26 | FuriousGeorge | coppice: actually looking at the setting its set to rfc2833 |
| 03:28.16 | dlynes | coppice: i have never gotten faxing to work over ip with ulaw :) |
| 03:28.40 | Qwell | I fax over lpc10 |
| 03:28.51 | FuriousGeorge | i use voip faxing only i call it email |
| 03:29.02 | coppice | glynes: exactly. some people do it by luck, and then tell the world in glowing terms that is works just fine. |
| 03:29.28 | dlynes | coppice: heh...just like qwell's bs about it working in lpc10 :) the mr. roboto codec :) |
| 03:29.37 | FuriousGeorge | maybe sending the dtmf inband will actually work better, lemme try that |
| 03:31.44 | FuriousGeorge | eh |
| 03:31.52 | papa_e | aghh, fucking dtmf doesn't work for shit on did numbers from sipphone/gizmo |
| 03:31.55 | papa_e | what a waste of time/money |
| 03:32.45 | dlynes | papa_e: Did you find out what dtmfmode they're using, before giving up? |
| 03:32.50 | FuriousGeorge | papa_e: perhaps its not gizmo's fault. my dtmf works but dtmf over asterisk has never been as reliable as dtmf over pstn or switched pbxs for me |
| 03:32.51 | *** join/#asterisk chaoscon (n=ph33r@smartserv/ceo/chaoscon) |
| 03:32.55 | dlynes | papa_e: not everyone uses rfc2833 |
| 03:33.33 | papa_e | well, i've tried everything to no avail, switching codes, dtmfmodes, etc., doesn't work |
| 03:33.58 | papa_e | regarding relability, i've used a bunch of providers in the past, never had problems with dtmf detection, pstn has always been 100% with asterisk |
| 03:34.05 | papa_e | s/codes/codecs/ |
| 03:34.10 | dlynes | ah |
| 03:34.47 | FuriousGeorge | papa_e: now that u mention it, gizmo is just sipphone and it always just worked for me with default dtmf settings and every codec |
| 03:35.21 | papa_e | george, placing outbound calls, or receiving calls? |
| 03:35.42 | FuriousGeorge | outbound only |
| 03:35.48 | papa_e | i think their configurations/setups are different for virtual numbers and regular accounts |
| 03:37.35 | FuriousGeorge | you should contact their support ans ask them wtf is up with their dtmf. they dont force you to use their client |
| 03:39.46 | papa_e | heh, still waiting from a reply from their support department |
| 03:40.25 | papa_e | overall, it's a really decent service, nice voice quality, unbeatable price, etc., but damn, i'd hate to switch over something as lame as this |
| 03:41.15 | *** join/#asterisk llirk (n=majestic@210-84-10-4.dyn.iinet.net.au) |
| 03:46.32 | *** join/#asterisk Olobola (n=casper_s@216.218.221.166) |
| 03:46.58 | *** join/#asterisk Assid (n=assid@203.115.83.214) |
| 03:48.30 | *** join/#asterisk Cresl1n (n=matt@user-24-236-124-147.knology.net) |
| 03:49.47 | llirk | hi |
| 03:52.02 | *** join/#asterisk bmg505 (n=leon@c1-167-12.rndf.isadsl.co.za) |
| 03:54.30 | Olobola | this is interesting: pinging wimax from the outside in is rock solid, from the inside to the out side 20% loss. How is this possible? |
| 04:01.43 | *** join/#asterisk tuxd00d (n=tuxinato@netblock-68-183-112-100.dslextreme.com) |
| 04:07.11 | coppice | its murphy's law |
| 04:17.53 | Olobola | ha!, it is actually. |
| 04:19.45 | *** join/#asterisk NoRemorse (n=bah@203-214-54-170.dyn.iinet.net.au) |
| 04:19.53 | *** join/#asterisk Altair256 (n=Altair25@tn-greenback1a-30.rhmdky.adelphia.net) |
| 04:20.03 | NoRemorse | heya all, is there an asterisk to teamspeak gateway module? |
| 04:22.15 | Altair256 | not that I know of NoRemorse, but that would be an awesome idea! |
| 04:22.45 | NoRemorse | hehe yeah |
| 04:23.03 | Altair256 | have you google'd it yet? |
| 04:23.03 | Qwell | why not just...run teamspeak? |
| 04:23.09 | NoRemorse | well thats 2 asterisk projects that are looking good for me to start. other one is CALEA |
| 04:23.25 | Qwell | CALEA? Isn't that the underage thing? |
| 04:23.30 | NoRemorse | lolo |
| 04:23.30 | Altair256 | Qwell, with a TStoAsterisk bridge, you could have someone call in and be bridged to the Teamspeak server |
| 04:23.38 | Altair256 | no internet connection required for the participants |
| 04:23.41 | NoRemorse | qwell thats 2 stupid things u ahve said in a row |
| 04:23.45 | Qwell | oh, guess not, heh |
| 04:24.04 | Qwell | What's the thing I'm thinking of? |
| 04:24.58 | Altair256 | NoRemorse, as TeamSpeak is not OSS, you're going to be fighting an uphill battle |
| 04:25.14 | Qwell | the thing where kids have to get parents to fill out a form, before they can register for some sites...nevermind :p |
| 04:26.08 | NoRemorse | nah CALEA is a lawfull intercept API |
| 04:26.09 | Altair256 | NoRemorse, I'd approach it by using a single TeamSpeak client running on the Asterisk box... |
| 04:26.18 | Altair256 | and tie it as a client to a Conference |
| 04:26.31 | NoRemorse | yeah, if TS isnt open source probaly not worth pusuing, what about ventrillo? |
| 04:26.33 | Altair256 | then let asterisk users just dial in to the Conference |
| 04:26.47 | Altair256 | well NoRemorse, there is a Teamspeak linux client |
| 04:26.54 | Altair256 | so you could run the client on the linux box |
| 04:27.00 | Qwell | Altair256: There is also a skype client for linux |
| 04:27.06 | NoRemorse | cool. worth looking at anyways |
| 04:27.15 | NoRemorse | skype is dead end |
| 04:27.17 | Altair256 | then create the bridge from an Asterisk Conference to the TeamSpeak client |
| 04:27.24 | Altair256 | Skype is useless, imo |
| 04:27.27 | Altair256 | waste for everyone |
| 04:27.31 | NoRemorse | someone has done skype to sip |
| 04:27.43 | NoRemorse | its commercial tho |
| 04:28.22 | *** join/#asterisk jsaunders (i=jsaunder@S01060060971c5817.va.shawcable.net) |
| 04:28.42 | Altair256 | hrm... NoRemorse... |
| 04:28.57 | Altair256 | why not just let everyone use a SIP client? |
| 04:29.06 | Altair256 | and cut out Teamspeak/Ventrillo all together? |
| 04:29.38 | Altair256 | unless you create a separate bridge from each asterisk user to the TS server, you're going to lose the "user presence" functionallity that TS/Vent have |
| 04:29.39 | NoRemorse | so many times I have needed to tell my Warcraft Guild on TS that I cant make a raid or somesuch and rather that turn on PC and run TS would be good to just dial a confernece extension |
| 04:29.56 | Altair256 | hrm |
| 04:30.17 | NoRemorse | the people who use TS are happy with it and would not consider even a free Sip soft client |
| 04:30.55 | NoRemorse | what I could then offer is a DID into the TS confernece so everyone can use the concepy, not just those who are clients of my asterisk box |
| 04:31.50 | NoRemorse | is it easy to start a sourceforge project for something like this? do you need permission from Digium? |
| 04:31.54 | Altair256 | yes, I definitely see the idea |
| 04:32.03 | Altair256 | reading notes from the dev of TS right now |
| 04:32.25 | NoRemorse | cool. I have previous sofwtare project management experience i'd love to give something back to the asterisk community |
| 04:32.44 | NoRemorse | and I know alot of people have wanted CALEA for awhile now |
| 04:32.54 | Altair256 | NoRemorse, http://www.savvy.nl/blog/forum/viewtopic.php?id=62 might be a starting point. Post from last November by the TS devs |
| 04:33.06 | NoRemorse | ah thanks |
| 04:33.28 | Altair256 | Communications Assistance for Law Enforcement Act? |
| 04:33.31 | NoRemorse | the TS gateway is more of something to play with in the meantime, cut my teeth on so to speak b4 I attempt CALEA |
| 04:33.38 | NoRemorse | that be it :) |
| 04:34.27 | Altair256 | bleh... I hate Big Brother. But at least in the USA, we keep passing more and more l laws allowing (even requiring) it |
| 04:35.06 | NoRemorse | thats the thing, it is legislated, some people would give their left etsticle to have the business or product they pushh "legislated" as a requirement by law :) |
| 04:35.16 | Altair256 | http://www.voip-news.com/news/calea-voip-050406/ |
| 04:35.27 | *** join/#asterisk Mavvie (n=edwin@203.222.131.252) |
| 04:35.29 | Altair256 | for VoIP providers... doesn't mention CALEA for internal PBX's |
| 04:36.23 | NoRemorse | nope probably never will |
| 04:36.53 | NoRemorse | this is a big picture thing. |
| 04:37.34 | Altair256 | I tell you what we really want... call barge for SIP channels |
| 04:37.36 | Qwell | calea + voip == fucking retarded |
| 04:38.01 | NoRemorse | not really qwell, when most vocie services are on VOIP, what other option is there? |
| 04:38.07 | Altair256 | yes Qwell, but unfortunatley, the US government has decided that they are going to enforce it. But only to providers, not to end users |
| 04:38.09 | tainted- | Qwell tell us how u _really_ feel about calea + voip |
| 04:38.21 | tainted- | lol |
| 04:38.28 | Qwell | NoRemorse: let's see... |
| 04:38.33 | Qwell | NOT violating my privacy? |
| 04:38.35 | tainted- | are people really going to comply with that garbage |
| 04:38.36 | NoRemorse | longw ay away of course, but if they done have intercept capabilities for voip its just an avenue for crooks to do business thru that they know is safe |
| 04:38.38 | Qwell | NOT making me pay for it? |
| 04:38.49 | NoRemorse | fuck your privacy, if one less sept 11 happens its worth it |
| 04:39.05 | Qwell | I could give a shit less...let thousands of people die |
| 04:39.07 | tainted- | NoRemorse lol |
| 04:39.09 | NoRemorse | yes payment is a serious controversy atm, who should pay for all this stuff |
| 04:39.11 | Altair256 | tainted-, they have to or they will be shut down |
| 04:39.17 | tainted- | NoRemorse seriously dude.. GO FUCK YOURSELF |
| 04:39.26 | NoRemorse | lol yeah wahetver u say idiot |
| 04:39.28 | Altair256 | NoRemorse, have you read the news lately? |
| 04:39.32 | Altair256 | the NSA can kiss my butt |
| 04:39.43 | NoRemorse | re the NSA recording all the phone call cdr data? |
| 04:39.46 | Qwell | and no, it isn't "who is going to pay for it". WE are going to pay for it |
| 04:39.54 | Altair256 | I agree with Qwell in spirit, but not necessarily in his way of saying it |
| 04:39.56 | Qwell | http://www.eff.org/Privacy/Surveillance/CALEA/ |
| 04:40.16 | NoRemorse | well yeah in the long run, either the carriers will pay and add the cost to conenctivity or the govmnt will pay out of taxes |
| 04:40.22 | Qwell | wrong |
| 04:40.29 | Altair256 | we pay the taxes |
| 04:40.35 | Qwell | we will pay, from a MANDATED CALEA charge |
| 04:40.37 | Altair256 | we pay for the increased costs for the carriers |
| 04:40.43 | Altair256 | we pay it if they simply tack it on to our bill |
| 04:40.52 | NoRemorse | ah well, not much u can do about that qwell |
| 04:40.53 | Altair256 | at the end of the day. We pay it. End of story |
| 04:40.56 | Qwell | "The FCC explores a mechanism by which these costs will pass to their customers, including a Commission mandated flat monthly charge." |
| 04:41.01 | NoRemorse | except sell CALEA compliance services :) |
| 04:41.44 | Altair256 | well... on private networks, just use a VPN to encrypt your IAX2 tunnels... |
| 04:41.55 | Altair256 | and tell everyone to go away |
| 04:42.21 | NoRemorse | anyways sorry for starting an argument I didnt want to, and I didnt realise it was such a sensitive topic, in Australia there hasnt been any sugegstion of a user pays LI charge, at this stage the carriwer are looking at footing the bill which they are not too hapy about |
| 04:42.35 | jsaunders | Do you have any remorse? Heh. |
| 04:42.44 | NoRemorse | any.. what? |
| 04:42.46 | Altair256 | NoRemorse, even if the carrier "foots the bill" it still eventually passes to the consumer |
| 04:42.57 | NoRemorse | yes Altair256 tahts what I said above, agreed |
| 04:43.10 | Altair256 | and you will find that most of us hate the idea of anyone being able to tell us that they can listen in on our networks whenever they feel like it |
| 04:43.47 | Altair256 | giving up personal freedom in the hopes that it will increase group security is assinine. At least at the level that has happened since September 11th |
| 04:43.51 | NoRemorse | from an end user point of view I agree totally, but from an LI service provide rpoint of view..... well |
| 04:43.52 | jsaunders | You can listen to who's network? |
| 04:44.17 | jsaunders | Ah, nsa. |
| 04:45.12 | Altair256 | jsaunders, FCC made CALEA to force telephony "providers" to have a way for law enforcement groups to be able to tap individual lines |
| 04:45.21 | Altair256 | this law was passed like back in 1994 |
| 04:45.49 | Altair256 | now, 13 years latter, the FCC is demanding that all VoIP providers meet this requirement as well |
| 04:46.00 | Altair256 | they have 90 days (from May 4, 2006) to comply, or they will be shut down |
| 04:46.10 | NoRemorse | isnt it 2007? |
| 04:46.40 | Altair256 | er, I don't believe so |
| 04:46.41 | Altair256 | http://www.voip-news.com/news/calea-voip-050406/ |
| 04:46.50 | Altair256 | ah, you're right |
| 04:47.04 | Altair256 | ahh... they have to "begin compliance in 90 days" |
| 04:47.09 | NoRemorse | yep |
| 04:47.43 | NoRemorse | we went thru this in australia about 12 montgs ago, mainly because carrier here didn't even comply at the voice level. |
| 04:47.50 | Altair256 | "they must show progress towards this end before the 90 days is up. " |
| 04:48.03 | NoRemorse | so the authrorities cracked it and stated voip had to be sticthed up as well |
| 04:50.43 | Altair256 | in the USA, this list is useful for businesses that are using voice services to know what their state's specific laws say about surveillance |
| 04:50.43 | Altair256 | http://www.ncsl.org/programs/lis/CIP/surveillance.htm |
| 04:51.32 | Altair256 | here are the federal laws http://www.ncsl.org/programs/lis/CIP/surveillance.htm#Federal |
| 04:51.45 | NoRemorse | <PROTECTED> |
| 04:52.09 | NoRemorse | anyways thanks for the chat, cathca |
| 04:59.31 | *** part/#asterisk Agrajag- (n=filip@c211-30-4-5.artrmn1.nsw.optusnet.com.au) |
| 05:06.33 | *** join/#asterisk Olobola (n=casper_s@216.218.221.166) |
| 05:08.18 | Olobola | my linksys was dropping packets, not wimax. |
| 05:09.57 | brookshire | i've found most linksys routers have a limit at 3mbit |
| 05:10.11 | brookshire | :( |
| 05:10.13 | brookshire | at least the older ones |
| 05:20.39 | *** join/#asterisk SkramX (n=mark@admins.sentiensystems.net) |
| 05:20.43 | SkramX | any sccp-knowledgable people around? |
| 05:20.57 | SkramX | heh |
| 05:21.00 | Qwell | Never heard of it! |
| 05:21.24 | SkramX | My * server receives the data sent from my 12sp+, however i dont hear the server's output |
| 05:21.27 | SkramX | <PROTECTED> |
| 05:21.30 | SkramX | <PROTECTED> |
| 05:21.32 | SkramX | :|? |
| 05:21.36 | Qwell | bug Sergio. |
| 05:21.42 | SkramX | i have tos to 18, and silence off |
| 05:22.01 | jsaunders | Good thing I live in Canada. |
| 05:22.14 | Qwell | SkramX: or, use chan_skinny |
| 05:22.18 | SkramX | meh |
| 05:23.24 | SkramX | Qwell: know where sergio is ? |
| 05:23.30 | Qwell | his list |
| 05:23.43 | Qwell | well, sometimes. he's been quite...inactive lately |
| 05:24.18 | Qwell | bbl, tv |
| 05:25.05 | dlynes | jsaunders: yeah, no kidding :) |
| 05:25.18 | jsaunders | Heheh |
| 05:25.31 | dlynes | jsaunders: where's "va.shawcable.net"? |
| 05:25.50 | dlynes | Never seen that one before |
| 05:26.27 | CunningPike | I'm going to guess Victoria |
| 05:27.38 | dlynes | oh yeah...hadn't thought of that :) |
| 05:27.49 | dlynes | I knew it wasn't vancouver |
| 05:27.52 | dlynes | that's vc |
| 05:29.30 | CunningPike | jsaunders: Put us out of our misery |
| 05:30.18 | jsaunders | Chilliwack |
| 05:30.43 | CunningPike | :S |
| 05:30.44 | dlynes | va == chilliwack? |
| 05:30.45 | dlynes | lol |
| 05:30.47 | jsaunders | Heh |
| 05:32.14 | *** join/#asterisk Olobola (n=casper_s@216.218.221.166) |
| 05:32.34 | Olobola | wimax is actually pretty solid. |
| 05:32.49 | jsaunders | Good to hear. |
| 05:32.59 | Olobola | $100 per megabit. |
| 05:33.19 | dlynes | bleh.... |
| 05:33.23 | jsaunders | Krikey |
| 05:34.47 | jsaunders | I say just launch a hot air balloon w/ jets and a few auto-aiming dishes. |
| 05:35.47 | Olobola | well the roof of a 13 story building always helps |
| 05:36.06 | *** join/#asterisk kristalino (n=kristali@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr) |
| 05:39.13 | dlynes | yeah...stupid thing....video suddenly decided to stop working, but audio still works |
| 05:39.31 | dlynes | I guess that'll teach me to be cheap, by buying a dvd player on ebay :p |
| 05:41.25 | tainted- | a dvd player is like 30$ at walmart |
| 05:41.44 | tainted- | can it be much cheaper than that? |
| 05:42.11 | dlynes | When I bought this one, the cheapest one was about $65 |
| 05:42.16 | dlynes | I got this one for $25 |
| 05:45.24 | tainted- | wow that is cheap |
| 05:46.03 | brookshire | olobola: i've had other experiences |
| 05:46.57 | brookshire | the town I am in has over subscribed their service |
| 05:47.28 | brookshire | plus... anytime a virus breaks out, it eats all the bandwidth |
| 05:47.32 | brookshire | for everyone |
| 05:48.10 | *** join/#asterisk xxi (i=foobar@cpe-70-112-73-77.austin.res.rr.com) |
| 05:48.38 | tainted- | how do i match 5551115555 as well as 15551115555 |
| 05:48.48 | tainted- | w/ one dialplan regex |
| 05:49.06 | tainted- | _X5551115555 ? |
| 05:50.18 | brookshire | tainted: i would use a macro |
| 05:50.57 | tainted- | can u describe the logic u would use? |
| 05:51.28 | brookshire | well.. i would just pass 555115555 and 5551115555 to the macro :) |
| 05:51.49 | tainted- | oh |
| 05:51.52 | tainted- | lol |
| 05:51.53 | brookshire | ${MACRO_EXTEN:1} |
| 05:52.04 | brookshire | that removes the first digit |
| 05:52.49 | tainted- | are there asterisk best practice dialplan macro archives anywhere? |
| 05:52.55 | tainted- | wiki? |
| 05:53.11 | brookshire | not that i know of |
| 05:53.17 | brookshire | macro is somewhat like a function |
| 05:53.25 | tainted- | yea |
| 05:53.38 | tainted- | so i would imagine there would be common macros |
| 05:55.06 | *** join/#asterisk PakiPenguin (i=uppal@linuxpakistan/admin/pakipenguin) |
| 05:55.19 | PakiPenguin | morning |
| 05:56.48 | SkramX | Heya. |
| 05:58.34 | brookshire | tainted: here is what we use |
| 05:58.38 | brookshire | [outgoing] |
| 05:58.41 | brookshire | exten => _9NXXXXXX,1,Dial(${TRUNK}/${EXTEN}) |
| 05:58.41 | brookshire | exten => _91NXXXXXX,1,Dial(${TRUNK}/${EXTEN}) |
| 05:58.41 | brookshire | exten => _91NXXNXXXXXX,1,Dial(${TRUNK}/${EXTEN}) |
| 05:58.41 | brookshire | exten => _9011X.,1,Dial(${TRUNK}/${EXTEN}) |
| 05:58.47 | brookshire | i guess it's not really a macro |
| 05:59.18 | brookshire | but it will do the job i'm sure |
| 05:59.54 | tainted- | gotcha |
| 06:00.59 | PakiPenguin | or you could just do _9X.,1,Dial(${TRUNK}/${EXTEN}) |
| 06:01.23 | tainted- | or _X., |
| 06:01.32 | tainted- | but we don't want to do that :D |
| 06:01.50 | PakiPenguin | :) hehe |
| 06:02.23 | PakiPenguin | well tainted- not reallly _X :) that would give access to everything / every number |
| 06:03.17 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 06:06.05 | *** join/#asterisk anderiv (n=anderiv@207.67.87.34) |
| 06:10.55 | *** join/#asterisk angler- (n=angler@pdpc/sponsor/digium/angler) |
| 06:13.52 | file | meep |
| 06:14.06 | angler- | file, whats going on? |
| 06:14.23 | file | I'm in a hotel near a Paris airport... |
| 06:14.34 | file | leaving in 15 mins to catch a flight to Montreal, then to home |
| 06:14.49 | *** join/#asterisk Uzzi (n=Andrea@host229-236.pool872.interbusiness.it) |
| 06:14.49 | angler- | Ahrimanes, im in atlanta this weekend |
| 06:14.58 | angler- | woops wrong person |
| 06:16.27 | brookshire | angler: are you drunk yet? |
| 06:16.36 | *** part/#asterisk catch23 (n=catch23@c-67-191-252-3.hsd1.ga.comcast.net) |
| 06:17.13 | angler- | brookshire, should be but not... tomorrow hopefully |
| 06:21.42 | *** join/#asterisk Thazza (n=me@229.9.233.220.exetel.com.au) |
| 06:25.56 | brookshire | HI! |
| 06:28.46 | *** join/#asterisk obantaX (n=obanta@CPE-24-27-131-132.neb.res.rr.com) |
| 06:34.01 | *** join/#asterisk mrtwister|mobile (n=andrius@cable-10-68.cgates.lt) |
| 06:38.44 | *** join/#asterisk angler- (n=angler@pdpc/sponsor/digium/angler) |
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| 07:02.22 | Qwell | brookshire: go to bed :p |
| 07:02.41 | brookshire | NO!!O@I#O!@I#!@ |
| 07:02.44 | brookshire | NEVAH!@#!@ |
| 07:09.09 | *** join/#asterisk ghenry (n=ghenry@mail.suretecsystems.com) |
| 07:17.00 | *** join/#asterisk Tommmo (n=tps@203.62.181.52) |
| 07:20.51 | Tommmo | im trying to work out how i can permit/restrict certain call types on a per-phone basis in asterisk. at the moment i have several contexts with a bunch of phones in eahc |
| 07:21.18 | Tommmo | i know you can include outbound contexts in a phone;s context but i need to do it per phone. any ideas? |
| 07:23.52 | *** join/#asterisk julien[re] (n=julien[r@AStDenis-103-1-4-27.w81-248.abo.wanadoo.fr) |
| 07:23.57 | julien[re] | hi there |
| 07:24.06 | julien[re] | i've got a small question: |
| 07:24.08 | *** join/#asterisk TonyM (n=TonyM@80.168.135.167) |
| 07:24.21 | julien[re] | how can i make the user type a phone number, followed by the hash key? |
| 07:25.04 | *** join/#asterisk kimosabe (n=kimosabe@dsl-200-78-71-61.prod-infinitum.com.mx) |
| 07:34.36 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219) |
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| 07:50.55 | *** part/#asterisk jef (i=fischer@pdpc/supporter/active/Jef) |
| 07:57.10 | *** join/#asterisk gmaruz1 (n=gmaruzz@217-133-80-112.b2b.tiscali.it) |
| 07:58.29 | ghenry | anyone know if someone has done a dialplan or something using AGI to check for new tickets in Request Tracker (RT)? |
| 07:59.16 | ghenry | Also, can you get Asterisk to call you back if it finds something? i.e. internally event driven, not having to dial in? Cron? or soemthing else? |
| 08:00.50 | stoffell | ghenry, regarding RT, you could write it if it doesn't exist, don't know that. |
| 08:01.06 | stoffell | regarding the call back, it's possible by using asterisk manager, or .call files (check on voip-info.org how that works) |
| 08:01.34 | ghenry | I thought as much ;-) |
| 08:01.38 | file | my expenses on paid wireless this month = oh dear god |
| 08:01.43 | stoffell | file, have fun in france ;) |
| 08:01.50 | file | I'm leaving! |
| 08:01.53 | file | I'm flying back to Canada |
| 08:02.02 | ghenry | wil do the RT thing myself then ;-) THanks stoffell |
| 08:02.05 | stoffell | file, cool, long trip :) |
| 08:02.10 | file | dunno really |
| 08:02.22 | brookshire | ghenry: we have |
| 08:02.40 | ghenry | are queues hard to understand? Not sure what an agent is? Just a user who can get called? |
| 08:02.43 | ghenry | brookshire: yeah? |
| 08:02.43 | file | I'm expecting... based on some rough calculations I did while I was in line since their computer system wasn't working... about 7-8 hours |
| 08:02.57 | file | brookshire: did you get my SMS? |
| 08:02.59 | ghenry | brookshire: Did you put it in the RT wiki? |
| 08:03.11 | brookshire | umm.. no |
| 08:03.14 | ghenry | or voip-info? |
| 08:03.23 | brookshire | what are you try to do? |
| 08:03.33 | brookshire | we just check for valid ticket numbers |
| 08:03.37 | wasim | file: have a safe flight and thanks for visiting alaw land |
| 08:03.45 | brookshire | and put the user into the correct queue |
| 08:03.50 | file | wasim: :P |
| 08:03.51 | brookshire | sales, support, etc |
| 08:03.54 | ghenry | brookshire: use the rt command tool to call use or give us a total when dealing of new tickets, and maybe subject |
| 08:03.56 | stoffell | i'm wondering, is the RTP traffic when using * and sip hardphones, really limited to 16384-32767 ? (a report generator sees a lot more udp traffic on other ports) |
| 08:04.24 | ghenry | brookshire: why do you check for valid ticket numbers in rt? |
| 08:04.27 | ghenry | brb |
| 08:04.29 | dlynes | stoffell: you can set it to whatever ports you want, afaik |
| 08:04.33 | dlynes | stoffell: check rtp.conf |
| 08:04.47 | stoffell | dlynes, hm, okay, checking it now |
| 08:04.51 | brookshire | ghenry: it's how we track support customers |
| 08:05.01 | julien[re] | hi there, could i match an extension finishing by a given number? |
| 08:05.28 | julien[re] | eg. i want to match extensions finishing by "12345" |
| 08:05.29 | stoffell | damn dlynes, so simple... sorry to have missed that ;) |
| 08:06.11 | harlequin516 | Anyone know if those Vonage Phone 2port phone adapters that are at every retail outlet will work with a home custom asterisk setup? |
| 08:06.17 | *** join/#asterisk tainted- (n=identd@ppp-71-134-51-75.dsl.irvnca.pacbell.net) |
| 08:06.30 | harlequin516 | Or is it something that the firmware will only allow vonage to use the device? |
| 08:06.31 | ghenry | right, chat later |
| 08:06.47 | brookshire | ghenry: rt has an api |
| 08:06.49 | brookshire | it's in perl |
| 08:07.03 | brookshire | you can pull any info you want from it |
| 08:07.04 | stoffell | hm, any reason why * ships default with ports 10000-20000 instead of 16384-32767 ? |
| 08:07.21 | *** join/#asterisk cced1 (n=dev2003@222.33.36.205) |
| 08:07.24 | *** part/#asterisk cced1 (n=dev2003@222.33.36.205) |
| 08:08.29 | ghenry | brookshire: yup |
| 08:14.03 | *** join/#asterisk anderiv (n=anderiv@207.67.87.34) |
| 08:20.11 | kimosabe | lol |
| 08:23.01 | *** join/#asterisk RoyK (n=roy@10.80-203-106.nextgentel.com) |
| 08:29.17 | tainted- | are 481's normal part of asterisk? |
| 08:29.39 | tainted- | is that a result of qualify / OPTION requests? |
| 08:35.42 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219) |
| 08:41.51 | harlequin516 | What's 481's? |
| 08:43.50 | kimosabe | i want 2 unlock pap 2 any one help |
| 08:48.48 | papa_e | best sip aware router out there? what works well for a small network (handful of pcs)? |
| 08:57.29 | kimosabe | does any one have pap2 password |
| 09:07.11 | papa_e | the pap2 password is "wank4fun" |
| 09:10.20 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
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| 09:46.15 | nroej | hi |
| 09:49.23 | llirk | hi |
| 09:49.43 | *** join/#asterisk nroej (n=joern@heaven.cyphertext.de) |
| 09:49.44 | nroej | re |
| 09:50.06 | llirk | ay |
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| 11:26.59 | ghenry | what cool things can you do with asterisk and LDAP? |
| 11:43.05 | *** join/#asterisk Assid (n=assid@203.115.83.214) |
| 11:59.25 | Thazza | . |
| 12:01.59 | *** join/#asterisk z00dax (n=kbsingh@base.karan.org) |
| 12:02.20 | *** part/#asterisk z00dax (n=kbsingh@base.karan.org) |
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| 12:56.40 | pjo | hi all, I have an asterisk server at home to do the local PBX stuff. I would like to have someone terminate my (international) calls via an IAX account. Any recomended provdiers of this service? |
| 13:03.47 | *** join/#asterisk Dr-Linux (n=huh@202.59.73.131) |
| 13:04.38 | Dr-Linux | is there any web/link for asterisk features patches? |
| 13:05.17 | llirk | changelog? |
| 13:07.21 | Dr-Linux | llirk: i'm not sure if asterisk new versions have some new features, i mean after 1.2.0 |
| 13:19.44 | *** join/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.233.208.Dial1.SanJose1.Level3.net) |
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| 13:24.30 | *** join/#asterisk littleball (n=littleba@cm55.epsilon171.maxonline.com.sg) |
| 13:25.01 | littleball | hello, i did sip to PSTN call, the quality is bad. what is the reason? sip to sip is good and pstn to pstn also good. |
| 13:31.35 | kaldemar | http://www.voip-info.org/wiki/view/Asterisk+echo+avoidance |
| 13:32.59 | *** part/#asterisk Funkmaster (n=spider@s55932490.adsl.wanadoo.nl) |
| 13:33.58 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.cust.bezeqint.net) |
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| 13:46.05 | *** join/#asterisk doolph (n=doolph@HACKERMACHINE.cpe.cableonda.net) |
| 13:46.39 | doolph | I am trying to connect to a sip provider, I am registered, but when I try to call that DID, I get into the SIP provider voicemail |
| 13:47.42 | littleball | the SIP to PSTN quality is really bad. anybody can give suggestion? |
| 13:48.10 | doolph | get a better sip provider? |
| 13:48.24 | littleball | no, i am provider :-) |
| 13:48.53 | pjo | I'm personally looking for an IAX2-PSTN solution - of course that limits your clients since most will be using sip |
| 13:48.53 | littleball | i use xlite to connect to my asterisk system, which do PSTN termination over e1 |
| 13:49.16 | doolph | what codec |
| 13:49.30 | littleball | i tried gsm, ulaw and alaw |
| 13:49.36 | littleball | no much difference. |
| 13:49.42 | *** join/#asterisk robin_sz (n=robin@adsl.redpoint.org.uk) |
| 13:49.46 | robin_sz | meep? |
| 13:50.01 | littleball | sorry, what is meep? |
| 13:50.18 | robin_sz | ok, questions .... I have a bunch of Snom 320s connected to * and then to some local ISDN lines ... |
| 13:50.28 | robin_sz | and I keep getting user complaints of echo |
| 13:50.40 | Dr-Linux | littleball: what about your zapapta.conf? |
| 13:50.58 | robin_sz | I would have though for non-sip calls, echo would ne not a problem |
| 13:50.59 | littleball | Dr_Linux, which part? |
| 13:51.17 | robin_sz | presumably then, the echo is infact over-helpful echo cancellation, right? |
| 13:51.41 | littleball | can i paste here? |
| 13:51.44 | robin_sz | no |
| 13:51.58 | tzafrir_laptop | ~pb |
| 13:52.00 | jbot | hmm... pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
| 13:52.04 | Dr-Linux | littleball: you must have only one zapata.conf each asterisk server. |
| 13:52.30 | robin_sz | well, no ... you can hav as many zapat.comfs as you like |
| 13:52.33 | littleball | Dr-Linux, which parameter you refer to? |
| 13:52.35 | robin_sz | but only one will be used |
| 13:52.41 | SpaceBass | morning |
| 13:52.43 | littleball | yes |
| 13:52.48 | SpaceBass | trying to execute an BASH script via AGI |
| 13:52.50 | SpaceBass | I can execute it fine from the bash prompt, but the AGI call just returns 0 but does nothing |
| 13:53.03 | tzafrir_laptop | SpaceBass, is it executable? |
| 13:53.19 | SpaceBass | tzafrir_laptop, good question, yeah it is, and its in the agi-bin dir |
| 13:53.39 | robin_sz | have it print something to a file as it executes |
| 13:53.42 | littleball | Dr-Linux, yes. only one file. which parameter you think is important |
| 13:53.45 | tzafrir_laptop | replace it with a wrapper script: |
| 13:53.55 | SpaceBass | robin_sz, thats all it does now |
| 13:54.01 | tzafrir_laptop | #!/bin/sh |
| 13:54.09 | SpaceBass | yeah, I have the shebang |
| 13:54.25 | robin_sz | well, check what * debug is telling you |
| 13:54.29 | Dr-Linux | littleball: depends, what's wrong with the voice quality? |
| 13:54.33 | tzafrir_laptop | exec strace -o /tmp/log ./script.orig |
| 13:54.35 | SpaceBass | its just a bash script and right now all it does is SSH into another box (using key auth) and touch a file |
| 13:54.38 | robin_sz | I had the wrong filename, and it said something similar |
| 13:54.48 | SpaceBass | the * cli says that it returns 0 |
| 13:54.51 | littleball | Dr-Linux, latency and not clear |
| 13:54.58 | tzafrir_laptop | SpaceBass, Have you tried it with the asterisk user? |
| 13:55.10 | SpaceBass | tzafrir_laptop, yeah, this is all with the asterisk user |
| 13:55.11 | littleball | Dr-Linux, give me your phone number, i can call u over xlite |
| 13:55.21 | tzafrir_laptop | Chances are it doesn't have the host in the known host file or something similar |
| 13:55.21 | littleball | can? |
| 13:55.33 | SpaceBass | script is in /var/lib/asterisk/agi-bin and the owner and group are asterisk |
| 13:55.34 | Dr-Linux | littleball: awww, you can't call me, i'm in Pakistan :) |
| 13:55.48 | littleball | i see |
| 13:55.53 | littleball | it is expensive |
| 13:56.03 | Dr-Linux | littleball: is the voice is very slow? echo ? or noise ? |
| 13:56.11 | littleball | i think all |
| 13:56.12 | robin_sz | so ... I have 3 problems .. echo on PSTN routed outgoing calls, occasionaly users cannot hear the called party, they redial and its fine .. both are BAD |
| 13:56.17 | tzafrir_laptop | SpaceBass, then try the wrapper script approach. Look at the trace you get from strace and see what exactly happened. One mistake there: |
| 13:56.23 | tzafrir_laptop | exec strace -f -o /tmp/log ./script.orig |
| 13:56.44 | SpaceBass | tzafrir_laptop, what do you mean by a wrapper scritpt? |
| 13:57.05 | SpaceBass | and does the exec strace command go in the script or the asterisk cli? |
| 13:57.25 | Dr-Linux | littleball: if your hear slow, incease a bit rxgain , |
| 13:57.45 | Dr-Linux | littleball: if there is noise , then echotraining=yes |
| 13:57.53 | tzafrir_laptop | cd /var/lib/asterisk/agi-lib; mv script script.orig; #create 'script' with the two lines I gave above; chmod +x script |
| 13:58.03 | tzafrir_laptop | assuming the script is called "script" |
| 13:58.12 | SpaceBass | gotcha |
| 13:58.17 | SpaceBass | thanks a lot, I'll try that |
| 13:59.14 | robin_sz | and .. a voicemail problem .. how can I get rid of Alison at the end of the user recorded greeting ... she says "please speak after the tone" ... my french clients dont like it a bit |
| 14:00.03 | SpaceBass | alros, pour qui non? Elle est Allison, elle est increable! |
| 14:00.06 | littleball | Dr-Linux, i already with echotraining=yes |
| 14:00.58 | robin_sz | SpaceBass, possibly but she has to go. |
| 14:01.26 | SpaceBass | :) |
| 14:01.58 | Dr-Linux | littleball: your pstn line is crystel clear? |
| 14:02.21 | SpaceBass | tzafrir_laptop, strace is not found |
| 14:02.23 | robin_sz | I have a similar problem |
| 14:02.53 | robin_sz | Dr-Linux, I have a similar problem .. our ISDN lines were echo free on the old crappy PBX .. with * we now got ocasional sever echo |
| 14:03.31 | SpaceBass | i get a little echo with SIP sometimes....PSTN was a nightmare with my x100p cards....had to ditch em |
| 14:04.03 | Dr-Linux | robin_sz: what's your * rx/tx ? |
| 14:04.15 | tzafrir_laptop | SpaceBass, the install it |
| 14:04.17 | robin_sz | ? |
| 14:04.30 | tzafrir_laptop | apt-get install strace, or the equivalent with your distro |
| 14:04.31 | robin_sz | Dr-Linux, ? what dou you mean? |
| 14:04.38 | SpaceBass | ahh |
| 14:04.48 | SpaceBass | i'm way too hung over to have thought that far ahead of you :) |
| 14:04.52 | robin_sz | Dr-Linux, I have SNOM 320s on their own subnet on one side |
| 14:05.20 | robin_sz | Dr-Linux, and an Eicon Diva card to 4 ISDN lines on the other (8 B channels total) |
| 14:05.38 | robin_sz | Dr-Linux, using the CAPI interface |
| 14:06.45 | Dr-Linux | robin_sz: echo problem is only with Snom's ? |
| 14:07.05 | robin_sz | well, we only have snoms .. so ... |
| 14:07.12 | robin_sz | about 25 320s and 20 360s |
| 14:08.07 | Dr-Linux | robin_sz: did you try any soft phone? to verify |
| 14:08.23 | robin_sz | not yet ... |
| 14:08.36 | Dr-Linux | robin_sz: you just need to confirm, if this is a asterisk side issue, or Snom |
| 14:08.41 | robin_sz | hmmm |
| 14:08.45 | Dr-Linux | robin_sz: i'm sure it's server side issue |
| 14:08.56 | Dr-Linux | bcoz 20 snom can't be dump |
| 14:09.16 | robin_sz | dump? |
| 14:09.18 | doolph | anyone can help me with incoming calls? what's wrong with this setup, I always get into provider voicemail http://pastebin.ca/55553 |
| 14:09.40 | littleball | hello, how to reload zapata.conf file without restart asterisk? |
| 14:09.42 | Dr-Linux | dump hhm... like bad |
| 14:09.48 | robin_sz | right |
| 14:10.06 | littleball | DR-Linux, local PSTN is very good. it is E1 to operator. |
| 14:10.11 | SpaceBass | tzafrir_laptop, 2 questions...what am I looking for from the strace? my orignal script takes arguments, do I specify one in the strace command or pass it to the wrapper script? |
| 14:10.13 | Dr-Linux | littleball: reload chan_zap.so |
| 14:10.28 | SpaceBass | guess I just answered the 2nd question....will specify in the wrapper |
| 14:10.34 | littleball | within cli console or linux shell? |
| 14:10.50 | littleball | Dr-Linux |
| 14:10.50 | Dr-Linux | littleball: CLI consol |
| 14:11.22 | tzafrir_laptop | SpaceBass, oops, silly me |
| 14:11.29 | tzafrir_laptop | exec strace -f -o /tmp/log ./script.orig "$@" |
| 14:12.00 | SpaceBass | whats the $@ for? |
| 14:12.02 | doolph | http://pastebin.ca/55554 <<-- help |
| 14:12.20 | tzafrir_laptop | 'reload chan_zap.so' within the asterisk CLI |
| 14:12.44 | littleball | Dr-Linux, hanks |
| 14:12.58 | SpaceBass | someone check out doolph's SIP debug...interesting...like asterisk destroys the call before it even answers, despite being registered |
| 14:13.26 | littleball | Dr-Linux, the echo in pstn side is big |
| 14:13.41 | Dr-Linux | that's how sip debug works |
| 14:14.02 | littleball | any suggestion? the PSTN users feel big echo |
| 14:14.12 | tzafrir_laptop | SpaceBass, see '@' in http://www.gnu.org/software/bash/manual/bashref.html#SEC25 |
| 14:14.21 | Dr-Linux | littleball: show me your rxgain=? |
| 14:14.36 | littleball | rxgain=6.0 |
| 14:14.36 | littleball | txgain=0.0 |
| 14:14.36 | tzafrir_laptop | Basiclly: all the parameters of the command |
| 14:14.47 | SpaceBass | tzafrir_laptop, thanks....I'll resarch it...ok, i run the wrapper script and get ptrace not permitted |
| 14:15.00 | SpaceBass | or rather: strace: ptrace(PTRACE_TRACEME, ...): Operation not permitted |
| 14:15.01 | Dr-Linux | littleball: do it "rxgain=2.0" |
| 14:15.12 | Dr-Linux | and then try. |
| 14:15.39 | littleball | Dr-Linux, what is the principle? |
| 14:16.38 | Dr-Linux | littleball: just reduce rxgain from 6.0 to 2.0 and try :) |
| 14:16.46 | tzafrir_laptop | That's odd. setuid problems? NM. The second option for tracing a shell script is by setting the '-x' option . I just don't know where the standard error goes to with shell scripts |
| 14:17.01 | littleball | ok. Dr-Linux |
| 14:17.16 | tzafrir_laptop | try adding ' -x' in the end of first line (the #! line) |
| 14:17.28 | SpaceBass | tzafrir_laptop, think I can specifiy the stderr right? with redirection |
| 14:17.45 | tzafrir_laptop | redirecting standard error to a file: |
| 14:17.57 | tzafrir_laptop | exec 2>/tmp/log |
| 14:18.03 | SpaceBass | bingo |
| 14:18.04 | SpaceBass | thanks |
| 14:19.03 | SpaceBass | ahhh with out the rediection it just displays on the terminal |
| 14:19.39 | *** join/#asterisk Altair256 (n=Altair25@tn-greenback1a-30.rhmdky.adelphia.net) |
| 14:19.39 | SpaceBass | note to self, don't start blog entery about AGI stuff if you dont know how to make it work |
| 14:20.47 | littleball | Dr-Linux, no changes |
| 14:21.10 | Dr-Linux | littleball: do you reload after changing the rxgain values ? |
| 14:21.43 | littleball | yes |
| 14:21.51 | littleball | i am using sip to pstn |
| 14:22.09 | littleball | pstn user feel big echo. sip user no echo |
| 14:22.25 | SpaceBass | tzafrir_laptop, or anyone... if the BASH script takes ~3 seconds to execute, would that cause problems with the AGI in the dialplan? |
| 14:22.54 | Dr-Linux | littleball: try rxgain=0.0 |
| 14:22.56 | robin_sz | SpaceBass, youd be better off forking |
| 14:23.46 | littleball | Dr-Linux, i already tried 0 |
| 14:23.55 | littleball | no difference |
| 14:24.32 | Dr-Linux | littleball: pastebin your zapata.conf |
| 14:25.02 | SpaceBass | ok tzafrir_laptop the debug looks fine, in other words its doing what it should |
| 14:26.17 | *** join/#asterisk camelon (n=chiardon@206.106.252.232) |
| 14:26.51 | SpaceBass | and the wrapper script you suggested is working (typo on my part) but there is noting in /tmp/log its writing everything to the screen |
| 14:26.54 | SpaceBass | even if i redirect |
| 14:27.06 | camelon | Morning guys! |
| 14:27.39 | robin_sz | is there a set of french voicemail soundfiles ? so at least "please speak after the tone" could be in French? |
| 14:27.45 | littleball | Dr-Linux,http://pastebin.com/715307 |
| 14:27.59 | *** join/#asterisk Uzzi (n=Andrea@host135-170.pool8260.interbusiness.it) |
| 14:28.14 | littleball | Dr-Linux, i try to increase rxgain now |
| 14:28.18 | SpaceBass | robin_sz, c'est ici : http://www.voip-info.org/wiki/view/Asterisk+sound+files+international |
| 14:28.25 | camelon | robin_sz go to voip there is a complete set |
| 14:28.41 | robin_sz | right .. |
| 14:28.43 | robin_sz | thanks .. |
| 14:28.46 | robin_sz | oh wait ... |
| 14:28.59 | robin_sz | the s "silent" option on voicemail ... |
| 14:29.11 | robin_sz | that might do it ... |
| 14:29.18 | robin_sz | :wq |
| 14:30.29 | Dr-Linux | littleball: rxgain is your recieving audio, if you will increase it more then 2.0 you will get echo problems |
| 14:30.30 | Dr-Linux | # |
| 14:30.31 | Dr-Linux | rxgain=8.2 |
| 14:30.31 | Dr-Linux | # |
| 14:30.31 | Dr-Linux | txgain=1.0 |
| 14:30.40 | Dr-Linux | that's bad |
| 14:33.01 | *** join/#asterisk eKo1 (n=bernd@190.4.7.90) |
| 14:33.14 | Dr-Linux | brb |
| 14:33.47 | eKo1 | Quick ?: Is there a way for Asterisk to return custom SIP messages like 503 You are in trouble or something like that? |
| 14:34.03 | eKo1 | Would I have to modify chan_sip.c? |
| 14:36.25 | SpaceBass | here is my dialplan for my AGI stuff ... could this be the problem? http://pastebin.ca/55561 |
| 14:36.32 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219) |
| 14:40.42 | robin_sz | coo. sorted. the 's' flag to the voicmail command rmoves Allison. |
| 14:42.18 | tzafrir_laptop | SpaceBass, if "its doing everything right" then what is wrong? |
| 14:45.01 | *** join/#asterisk ManxPower (n=ewieling@dpc67142183150.direcpc.com) |
| 14:45.02 | littleball | Dr-Linux, does asterisk support g723 and g729? |
| 14:45.29 | *** join/#asterisk keyhack (n=keyhack@c-24-60-209-35.hsd1.ma.comcast.net) |
| 14:46.20 | *** join/#asterisk Assid (n=assid@203.115.83.214) |
| 14:46.47 | Assid | heya |
| 14:47.08 | SpaceBass | d'oh |
| 14:47.23 | *** join/#asterisk spunz_ (n=spunz@h081217096096.dyn.cm.kabsi.at) |
| 14:47.29 | Assid | okay what if you have an external box hosting asterisk.. and 2 sip users within the same network.. will it use internet bandwith to make the calls to each other ? even tho the calls are from the same network? |
| 14:48.12 | SpaceBass | Assid, if you do canreinvite=yes |
| 14:48.45 | *** join/#asterisk zepmantra (i=waaa@203.76.216.123) |
| 14:48.49 | Assid | under what conditions would canreinvite cause a problem otherwise? and why would someone want to put it off? |
| 14:49.15 | Assid | SpaceBass: so it will use internal bandwith if i use canreinvite ? and nat=route ? |
| 14:49.40 | SpaceBass | not 100% sure, but i think so |
| 14:51.05 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219) |
| 14:52.25 | stoffell | Assid, if you use the same codecs on the sip users and they support it, you can use canreinvite=yes |
| 14:52.27 | *** join/#asterisk heart (n=zippetto@lugbari/people/heart) |
| 14:52.50 | stoffell | Assid, best is to check with a network sniffer (ethereal) to make sure :) |
| 14:54.12 | *** join/#asterisk Dr-Linux (n=huh@202.59.73.131) |
| 14:55.19 | *** join/#asterisk Led_Zeppelin (n=dummy@cpe-24-210-79-197.columbus.res.rr.com) |
| 14:55.50 | Led_Zeppelin | hey is it possible to route my cell phone calls to some service? I got a hefty $200 bill last month, and I need an alternative |
| 14:56.16 | Led_Zeppelin | or actually, can someone recommend me a good service to place calls? |
| 14:57.55 | keyhack | Led_Zeppelin: VoIP? |
| 14:58.01 | *** join/#asterisk marv (n=marv@12-219-145-181.client.mchsi.com) |
| 14:58.44 | Led_Zeppelin | keyhack: well, anything I guess. I am just frustrated with my cell provider. I mostly make 1800, and 1877 calls too, I need a cheap service for that |
| 14:58.51 | Led_Zeppelin | currenty I have no landline in my apt |
| 15:01.46 | *** join/#asterisk DrPete (n=Pete@host-84-9-255-194.bulldogdsl.com) |
| 15:02.19 | *** join/#asterisk wunderkin (n=kev@ip68-226-113-228.ph.ph.cox.net) |
| 15:04.43 | *** join/#asterisk Ariel_ (n=Ariel@70.46.87.158) |
| 15:08.01 | SpaceBass | Led_Zeppelin, which provider? |
| 15:09.02 | keyhack | SpaceBass: He's looking for one |
| 15:09.10 | SpaceBass | which cell provider? |
| 15:10.40 | SpaceBass | well, I'm off to Lowes...my lease favorite place in the world... |
| 15:11.31 | SpaceBass | some cell providers allow you to make unlimited (free) calls to your home number....set up a * box, get a Broadvoice Lite BYOD plan and say that is your home number, then set up DISA on the * server...call you house and route calls through the BV account |
| 15:14.21 | keyhack | SpaceBass: BroadVoice is decent, the quality is not the best, who do you see people recommending for business quality lines? |
| 15:15.04 | SpaceBass | I use BV for business and find the quality to be excelent...better than POTS at least... Talasip is another big one that I hear a lot about |
| 15:15.14 | SpaceBass | I use nufone for some outgoing stuff...they are uncompressed ulaw which is nice |
| 15:16.19 | SpaceBass | Led_Zeppelin, another solution... if your provider doesnt offer free calls to a 'home' number... get a smartphone and load a SIP client...then get unlimited data from your provider....lower your minutes to next to nothing and use VoIP |
| 15:17.36 | keyhack | SpaceBass: how much is nufone? |
| 15:17.41 | SpaceBass | its metered |
| 15:17.47 | SpaceBass | you load up your account via pay pal |
| 15:17.53 | SpaceBass | not sure what the rates are |
| 15:18.02 | SpaceBass | like .02 cents / min or something....pretty cheap I think |
| 15:18.03 | keyhack | SpaceBass: yeah they don't put any rates on there |
| 15:18.43 | SpaceBass | i use it for sending faxes since its uncompressed |
| 15:18.43 | SpaceBass | and for changing my caller id to prank friends :) |
| 15:19.27 | SpaceBass | talasip is a good bet |
| 15:19.32 | keyhack | I can't find talasip |
| 15:19.38 | SpaceBass | www.telasip.com |
| 15:19.45 | keyhack | oh, tela not tala |
| 15:20.01 | SpaceBass | oops...typo |
| 15:20.59 | littleball | hello, what codec is used by e1/pri? |
| 15:21.21 | keyhack | SpaceBass: Residential seems decent pricing, $14.95 for unlimited US |
| 15:21.42 | keyhack | SpaceBass: The business ones are good, but none are unlimited |
| 15:21.58 | SpaceBass | yeah |
| 15:22.08 | SpaceBass | i use broadvoice for my business line |
| 15:22.16 | SpaceBass | but I'm a more of a telecomuter |
| 15:22.27 | keyhack | SpaceBass: The $29.95 right? |
| 15:22.51 | SpaceBass | I use the BYOD lite plan |
| 15:23.01 | keyhack | oh, so your using residential for business haha |
| 15:23.04 | SpaceBass | I always go over 100 mins, but rarely pay more than $20 / month |
| 15:23.18 | SpaceBass | yeah |
| 15:23.53 | SpaceBass | ok, gotta run |
| 15:24.10 | keyhack | ok thanks |
| 15:26.16 | *** join/#asterisk gmaruzz (n=gmaruzz@217-133-80-112.b2b.tiscali.it) |
| 15:27.00 | *** join/#asterisk bjohnson_ (n=bjohnson@jecinc.tor.istop.com) |
| 15:27.41 | *** join/#asterisk Kokey (n=jramirez@dsl-200-78-65-27.prod-infinitum.com.mx) |
| 15:30.43 | *** join/#asterisk Kokey (n=jramirez@dsl-200-78-65-27.prod-infinitum.com.mx) |
| 15:34.34 | *** join/#asterisk gmaruz1 (n=gmaruzz@217-133-80-112.b2b.tiscali.it) |
| 15:35.27 | DrPete | whats a typcal partition scheme for an asterisk server, just a few ip phones, a x100p etc, nothing major. anything speacial i need? storage space etc |
| 15:35.47 | *** join/#asterisk coppice (n=chatzill@153.192.17.210.dyn.pacific.net.hk) |
| 15:37.30 | SpaceBass | ahhh just get asterisk@home and let it do it :) |
| 15:38.39 | DrPete | lol, i could, hehe. I was have never used asterisk@home, only ever used asterisk in the past. I have not used it in ages, and I want to make a dedicated server for it |
| 15:39.04 | SpaceBass | I'm quite a fan of it...guess I'm biased |
| 15:39.43 | *** part/#asterisk eKo1 (n=bernd@190.4.7.90) |
| 15:40.45 | SpaceBass | www.archatechs.com had some info on A@H and www.nerdvittles.com had a startup guide |
| 15:43.44 | Led_Zeppelin | SpaceBass: wait.. |
| 15:43.59 | Led_Zeppelin | how woud would you do that? I was just googling that idea... |
| 15:44.22 | DrPete | thanks |
| 15:44.36 | Led_Zeppelin | man, you guys have too many cool ideas. Do they actually work though? :-) |
| 15:46.43 | DrPete | control itunes with my phone, lol nice |
| 15:47.10 | *** join/#asterisk jtodd (n=jtodd@garthim.fox-den.com) |
| 15:48.46 | Led_Zeppelin | is SpaceBass ever here again? |
| 15:50.08 | *** join/#asterisk ManxPower (n=ewieling@dpc67142183150.direcpc.com) |
| 15:50.17 | SpaceBass | here now |
| 15:50.23 | SpaceBass | thought I was going...but didnt make it |
| 15:50.33 | SpaceBass | the itunes thing is pretty slick |
| 15:50.35 | *** join/#asterisk litage (n=nick@203.220.55.70) |
| 15:50.37 | *** join/#asterisk Snake-Eyes (n=blog@203.220.55.70) |
| 15:50.55 | Led_Zeppelin | SpaceBass: cool! you are back! I like your idea about getting a smartphone (a new gadget) |
| 15:51.04 | Led_Zeppelin | What would I need for that? |
| 15:51.22 | SpaceBass | Led_Zeppelin, I have the sprint ppc-6700 ...its VERY cool..but HUGE and bulky compaired to a regular phone |
| 15:51.29 | SpaceBass | cingular has teh 2125 which would work well |
| 15:51.55 | Led_Zeppelin | SpaceBass: currently I only have cell phone. and 90% of the calls are outgoing, and I want to save my minutes |
| 15:52.18 | Led_Zeppelin | SpaceBass: my phone bill is crazy high now! I want to totally reduce the phn bill, and I have broadband internet connection |
| 15:52.42 | coppice | if I seriously wanted to save my minutes, I wouldn't waste them on IRC :-) |
| 15:52.57 | SpaceBass | Led_Zeppelin, ahh if thats the case, then you should just build a Asterisk@home server, get a cheap plan and make all of your calls over voip |
| 15:53.13 | SpaceBass | of course that doesnt help when you are away from the house |
| 15:53.30 | SpaceBass | but there are some things you can do...like what I suggested about 'free calls to a home number'....I know sprint does that |
| 15:53.37 | Led_Zeppelin | SpaceBass: well, I work from home most of the times, and these calls are from home |
| 15:53.56 | dlynes | Just go on the all-you-can-eat cell phone plan, and ditch your home phone |
| 15:54.03 | Led_Zeppelin | you have too many kick ass ideas SpaceBass :-) |
| 15:54.23 | Led_Zeppelin | dlynes: i looked for those, they are like $80/month!! |
| 15:54.34 | dlynes | Heh...mine's $45/mo |
| 15:54.44 | Led_Zeppelin | who do you have? |
| 15:54.47 | dlynes | $45Cdn/mo :) |
| 15:54.58 | Led_Zeppelin | who is it? the provider |
| 15:55.00 | dlynes | Fido (Microcell Solutions) |
| 15:55.31 | Led_Zeppelin | SpaceBass: any good services (cheap) for asterisk? like 9.99 per month |
| 15:55.32 | dlynes | Not that you can get Fido in the US, but... |
| 15:55.39 | Led_Zeppelin | heh yah :-) |
| 15:55.48 | Led_Zeppelin | thanks for getting my hopes up, dyn |
| 15:55.51 | Led_Zeppelin | thanks for getting my hopes up, dlynes |
| 15:55.55 | dlynes | lol |
| 15:56.14 | Led_Zeppelin | i was reading up on *, you can do some crazy things on it! |
| 15:56.22 | dlynes | Yeah...you guys pay too much for cable tv, too much for cell phone, too much for a lot of stuff |
| 15:56.29 | SpaceBass | nah...not too many ideas...just dont have a life :) |
| 15:56.44 | SpaceBass | Led_Zeppelin, I like broadvoice.com which has a $5.00/month BYOD lite plan |
| 15:56.53 | Led_Zeppelin | what is BOYD? |
| 15:56.59 | SpaceBass | Led_Zeppelin, also if you only want outgoing, check out www.nufone.net |
| 15:57.02 | dlynes | Bring Your Own Drugs |
| 15:57.06 | SpaceBass | BYOD = bring you own device |
| 15:57.08 | Led_Zeppelin | hahaha |
| 15:57.20 | Led_Zeppelin | SpaceBass: BOYD = * in your case? |
| 15:57.22 | SpaceBass | dlynes wow...that reminds me of some parties! |
| 15:57.36 | SpaceBass | Led_Zeppelin, exactly...the Device ends up being * |
| 15:58.22 | dlynes | There's a company here that does $10Cdn/mo for all you can eat voip plan |
| 15:58.27 | *** join/#asterisk angler- (n=angler@pdpc/sponsor/digium/angler) |
| 15:58.40 | SpaceBass | Telasip.com has cheap plans too...incoming and outgoing |
| 15:58.49 | SpaceBass | OHHH and www.freedigits.com <--- free incoming calls |
| 15:58.54 | Dr-Linux | woww dlynes wokeup! :) |
| 15:59.07 | SpaceBass | OHHHH and if you are doing toll free... check out free world dialup and goiax.com |
| 15:59.10 | dlynes | Dr-Linux: ummmm....it's only 9am |
| 15:59.26 | Led_Zeppelin | 14.95 isn't that bad either |
| 15:59.33 | dlynes | Dr-Linux: Waiting for my buddy to wake up...going to look at some new vw's and audi's this morning |
| 15:59.37 | Led_Zeppelin | okay, can I hookup a IP phone with asterisk? |
| 16:00.06 | dlynes | no, absolutely not...asterisk is not an ip pbx...where on earth did you ever get that idea from? |
| 16:00.08 | Led_Zeppelin | instead of me speaking on a mic, and headphone, I rather speak on a phone (IP, preferabbly) |
| 16:00.18 | littleball | hello. what is rxgain and txgain? |
| 16:00.27 | SpaceBass | Led_Zeppelin, so once you get Asterisk@home up and running, then you get a FREE account with goiax.com for your 800 calls, then get a Broadvoice BYOD lite plan for incoming calls and non-800 outgoing and you are set |
| 16:00.35 | *** join/#asterisk iq|mobile (n=iq@71-38-73-211.omah.qwest.net) |
| 16:00.42 | dlynes | littleball: They're settings in your zapata.conf file, and they adjust the gain on incoming and outgoing calls |
| 16:00.47 | SpaceBass | Led_Zeppelin, then go on ebay and get an IP phone |
| 16:01.00 | Dr-Linux | dlynes: coool, here is 9 PM though, i'm off from work and feeling bore |
| 16:01.10 | camelon | little ball . .you are needing lot of readings . . googling now |
| 16:01.18 | dlynes | Dr-Linux: so go rent a porn flick |
| 16:01.24 | SpaceBass | Led_Zeppelin, or you can get a Linksys PAP2 box...$40 at most stores (best buy, etc) and google for hacking it...then you can plug a regular phone into it |
| 16:01.24 | littleball | dlynes, foor rx and tx, it is from asterisk point view ? or phone user point of view? |
| 16:01.36 | Led_Zeppelin | WOW1 |
| 16:01.40 | Led_Zeppelin | you are the best Supaplex :-) |
| 16:01.44 | SpaceBass | does anyone acutally buy porn from a store anymore dlynes ? |
| 16:01.45 | Led_Zeppelin | SpaceBass! |
| 16:01.46 | Led_Zeppelin | you are the best |
| 16:01.51 | SpaceBass | thought that is what bandwidth is for |
| 16:01.52 | dlynes | SpaceBass: lol |
| 16:02.06 | Led_Zeppelin | and funny too |
| 16:02.07 | Dr-Linux | dlynes: lol pron flick? mean a chick? or what? :P |
| 16:02.08 | SpaceBass | Led_Zeppelin, I'm just a newbie too....albeit a zealot newbie |
| 16:02.08 | dlynes | SpaceBass: i dunno...the porn shops here are still pretty busy |
| 16:02.16 | SpaceBass | its those back rooms! |
| 16:02.19 | Led_Zeppelin | SpaceBass: you know more than me :-) |
| 16:02.28 | coppice | SpaceBass: there are still plenty of magazines, so i guess the answer is yes |
| 16:02.32 | dlynes | Dr-Linux: nah...you said you had a girlfriend...i figured a pron flick for when you can't be with her :) |
| 16:02.33 | SpaceBass | good point |
| 16:03.53 | Dr-Linux | yeah |
| 16:03.58 | wunderkin | great, my bt101 is semi-unoperational |
| 16:04.01 | Dr-Linux | but wtf is "flick" ? |
| 16:04.12 | Led_Zeppelin | who else are good providers? |
| 16:04.21 | Led_Zeppelin | looking at broadboice + telesip so far |
| 16:04.27 | Led_Zeppelin | anyone else you recommend for my situation? |
| 16:04.32 | Dr-Linux | Vonics |
| 16:05.13 | wunderkin | oh i see why |
| 16:05.17 | dlynes | littleball: txgain is for outbound; rxgain is for inbound |
| 16:05.58 | dlynes | Dr-Linux: movie |
| 16:06.01 | Led_Zeppelin | personally, I don't even need a phone number |
| 16:06.06 | Led_Zeppelin | I just need a place to make calls... |
| 16:06.10 | SpaceBass | Led_Zeppelin, www.goiax.com has FREE accounts for toll-free calls |
| 16:06.18 | Led_Zeppelin | SpaceBass: yep. |
| 16:06.30 | SpaceBass | Led_Zeppelin, www.nufone.com has metered accounts if you don't want to pay by the month...just charge it via pay pall |
| 16:06.36 | Led_Zeppelin | I have t-mobile as my cell, can I have those calls be forwared to my * box |
| 16:06.57 | SpaceBass | yes |
| 16:06.59 | dlynes | Led_Zeppelin: if you have a did, yes |
| 16:07.09 | SpaceBass | but t-mobile may charge a forwarding fee...i know sprint does |
| 16:07.26 | dlynes | I think all cell phones do |
| 16:07.41 | dlynes | Cell phone companies up here charge you by the minute for forwarding |
| 16:07.42 | Led_Zeppelin | yah, now I think about it...all cells probally do |
| 16:08.27 | Led_Zeppelin | how is sunrocket? |
| 16:08.30 | Led_Zeppelin | is that any good? |
| 16:08.38 | SpaceBass | BUT, like I said, some cell providers have a add-on where you can call your home for free ...so for $5 you get unlimited calls to one number...just tell them that your broadvoice (or what ever) is your home number |
| 16:08.50 | SpaceBass | I hear good things about sunrocket over at flyertalk |
| 16:09.08 | *** join/#asterisk XeSolutions (n=Steve@68.225.80.195) |
| 16:09.08 | littleball | dlynes, just to confirm out/in is from asterisk system point of view, right? |
| 16:09.34 | *** join/#asterisk asterboy (n=kevin@S010600485480f4be.ed.shawcable.net) |
| 16:09.34 | Led_Zeppelin | does SR work with * ? |
| 16:09.37 | XeSolutions | Anyone able to point me to some documentation or something on how to install Asterick? |
| 16:09.43 | dlynes | littleball: define "asterisk system point of view" |
| 16:09.49 | SpaceBass | XeSolutions, www.archatechs.com www.nerdvittles.com |
| 16:09.49 | asterboy | ~docs |
| 16:09.50 | jbot | [docs] probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 16:09.57 | littleball | dlynes, thanks. |
| 16:10.04 | SpaceBass | Led_Zeppelin, now that I dont know |
| 16:10.06 | SpaceBass | never used sr |
| 16:10.17 | dlynes | littleball: for? I asked you a question...you answered it with thanks :) |
| 16:10.27 | littleball | dlynes, asterisk -->out -->phone user |
| 16:10.38 | littleball | phone user -->in asterisk |
| 16:10.44 | littleball | phone user -->in --> asterisk |
| 16:11.05 | dlynes | littleball: Yes, outbound from asterisk == txgain; inbound into asterisk dialplan == rxgain |
| 16:11.55 | littleball | this is my question. i still don't understand how these two parameter affect echo. can u explain. especially, sip <-->asterisk<-->pstn phone |
| 16:12.26 | littleball | in my testing, pstn phone user hear loud echo |
| 16:12.47 | littleball | i try to change txgain and rxgain. but what is the principle |
| 16:12.47 | littleball | ? |
| 16:12.50 | dlynes | littleball: they don't really affect echo; mainly whether dtmf will be recognized or not, and how much distortion you'll have in the voice, or whether you'll even be able to hear anyone |
| 16:13.12 | dlynes | littleball: You've got three echo options |
| 16:13.42 | littleball | how to remove the echo? i already echo cancel to yes in zapdata.conf file |
| 16:13.48 | *** join/#asterisk darkskiez (n=mhb@bb-87-81-62-203.ukonline.co.uk) |
| 16:13.59 | asterboy | littleballs, Try Mark2 |
| 16:14.03 | dlynes | littleball: echotraining=yes/xxx, where xxx is a value (800 is default, I think)...it's the number of milliseconds of echo training it does at the start of a call (during that time when it's echo training, you won't hear anything) |
| 16:14.18 | dlynes | littleball: echocancel=yes/no for enabling it |
| 16:14.31 | littleball | dlynes, i already set to echotraining=yes |
| 16:14.41 | dlynes | littleball: echocancelwhenbridged=yes/no if you've got an fxo that's making a call to an fxs, or the reverse |
| 16:14.54 | dlynes | littleball: is the echo really bad on every call? |
| 16:15.02 | dlynes | littleball: or is it slight on every call? |
| 16:15.08 | dlynes | littleball: or is it only on some calls? |
| 16:15.09 | asterboy | littleballs, make sure your hardware pci is not sharing. |
| 16:15.13 | littleball | asterboy, what is Mark2? dlynes, yes, everycall |
| 16:15.28 | littleball | sharing? asterboy |
| 16:15.30 | asterboy | It's a compile time option for EC |
| 16:15.40 | asterboy | cat /proc/interrupts |
| 16:15.41 | dlynes | littleball: Try adjusting your echotraining to echotraining=1200, and see if that fixes it |
| 16:16.15 | asterboy | make sure your hardware is not sharing with anything on the pci slot |
| 16:16.52 | littleball | asterboy, i don't think so. this server only for digium card ussage. no other hardware.. |
| 16:17.10 | asterboy | littleballs, make sure by typing cat /proc/interrupts |
| 16:17.22 | asterboy | or use a utility like lspci |
| 16:17.23 | dlynes | littleball: if it doesn't, you might be sharing an interrupt or you'll have to try a different echo canceller, a hardware echo canceller, or if you're using an x100p, ditch it, and replace it with a tdm400 |
| 16:17.36 | dlynes | asterboy: cat /proc/interrupts isn't reliable |
| 16:17.45 | asterboy | no but a good quickie |
| 16:17.50 | dlynes | littleball: try lspci -v | grep IRQ |
| 16:18.01 | dlynes | littleball: if you have more than one item with the same IRQ, you're sharing interrupts |
| 16:18.07 | ManxPower | lspci does not show the interrupts after APIC has been enabled by the OS |
| 16:18.20 | dlynes | ManxPower: ah |
| 16:18.44 | dlynes | ManxPower: but I've also heard that apic can affect your timing, too? |
| 16:18.51 | dlynes | ManxPower: i.e. for the worse |
| 16:18.57 | asterboy | lspci is not reliable either |
| 16:19.10 | asterboy | it is a good idea to disable apci |
| 16:19.10 | ManxPower | dlynes, FAR less than sharing IRQs |
| 16:19.20 | dlynes | ah |
| 16:19.34 | dlynes | but it's better to just get rid of everything that can affect it, non? |
| 16:19.52 | asterboy | at boot type, noacpi or acpi=off |
| 16:20.11 | dlynes | asterboy: he was talking about apic, not acpi, but yeah...acpi should be disabled, too |
| 16:20.21 | asterboy | ah |
| 16:21.28 | dlynes | asterboy: apic's an option you can enable for a lot of smp architectures, and very rarely on uniprocessor systems; I've only ever seen it on rack mount systems |
| 16:21.46 | ManxPower | ACPI = sleep modes. APIC = 255 IRQs for devices to be on |
| 16:22.21 | dlynes | ManxPower: normally you only have 32 irq's? |
| 16:22.30 | asterboy | ~apic |
| 16:22.31 | jbot | apic is, like, Advanced Programmable Interrupt Controller, a new PIC, and about time too... |
| 16:22.40 | ManxPower | dlynes, no, normally 16 IRQa |
| 16:22.45 | asterboy | jbot apic is also http://en.wikipedia.org/wiki/IO-APIC |
| 16:22.47 | jbot | asterboy: okay |
| 16:22.53 | asterboy | ~apic |
| 16:22.54 | jbot | from memory, apic is Advanced Programmable Interrupt Controller, a new PIC, and about time too... http://en.wikipedia.org/wiki/IO-APIC |
| 16:22.54 | dlynes | ManxPower: ah...thought that was raised with pci |
| 16:23.39 | dlynes | 16 irq's is kinda idiotic |
| 16:23.45 | ManxPower | APIC is supported by almost all modern motherboards. |
| 16:24.04 | dlynes | ManxPower: how modern are you talking? |
| 16:24.18 | dlynes | ManxPower: Only one of the machines I've bought in the last 2 years has it |
| 16:24.19 | ManxPower | dlynes, my 2001 motherboard had it. |
| 16:24.36 | ManxPower | dlynes, Install an SMP kernel on it, you'll see APIC |
| 16:24.47 | ManxPower | and you can install SMP kernels on uniprocesser machines |
| 16:25.01 | dlynes | ManxPower: yeah, but afaik, you have to have apic support on the machine, too |
| 16:25.10 | dlynes | ManxPower: only one of my bioses has an apic option |
| 16:25.21 | *** join/#asterisk alephcom (n=alephcom@host75.net14.mcsnet.ca) |
| 16:25.33 | ManxPower | dlynes, Oh! just because the bios won't let you configure it, does not mean it's not there. |
| 16:25.39 | dlynes | ah |
| 16:26.12 | dlynes | Yeah...i've never enabled in the kernel because i thought none of my machines supported it |
| 16:26.52 | dlynes | I guess that's what the bioses refer to as a pnp os, then? |
| 16:27.38 | ManxPower | dlynes, that's something different. |
| 16:27.48 | ManxPower | dlynes, try enabling it on a machine |
| 16:27.48 | dlynes | That might actually help me out on a couple of systems I've got where there's only one pci slot, and whatever device is in the slot insists on sharing the same irq as the network card |
| 16:27.51 | *** join/#asterisk abatista (n=Ariel@70.46.87.158) |
| 16:27.56 | ManxPower | enable APIC in the Linux Kernel, that is |
| 16:28.13 | dlynes | or will apic not fix that problem? |
| 16:31.19 | asterboy | littleballs, I've had the greatest success with EC issues using this link: |
| 16:31.20 | asterboy | http://www.asteriskdocs.org/modules/tinycontent/content/docbook/current/docs-html/x1695.html#AEN1718 |
| 16:31.51 | asterboy | put on mark2 and agressive...voila...not echo |
| 16:31.59 | *** join/#asterisk af_ (n=af@ip-143-220.sn1.eutelia.it) |
| 16:32.11 | asterboy | ~echo |
| 16:32.12 | jbot | i heard echo is normal and a natural part of Voice over IP. What? You don't like it? Nah. It's great! You really don't want it? Well, read http://lists.digium.com/pipermail/asterisk-users/2005-February/088794.html then. echo, echo ..echo... cho..o... .. It also helps ADHD people keep track of their conversations. It's all good! |
| 16:32.56 | wunderkin | yes you won't get echo with agressive, since it is half duplex |
| 16:33.02 | asterboy | jbot remove echo |
| 16:33.05 | *** join/#asterisk Defraz (n=t0tal@tim.mychoice.cc) |
| 16:33.10 | wunderkin | nothing like a one-way conversation |
| 16:33.34 | asterboy | jbot no, echo is http://lists.digium.com/pipermail/asterisk-users/2005-February/088794.html |
| 16:33.37 | jbot | asterboy: okay |
| 16:33.42 | asterboy | ~echo |
| 16:33.43 | jbot | rumour has it, echo is http://lists.digium.com/pipermail/asterisk-users/2005-February/088794.html |
| 16:33.49 | ManxPower | dlynes, the only way to know is to try. |
| 16:34.01 | dlynes | Yeah...I was going to do that anyways |
| 16:34.07 | dlynes | Was just kinda hoping for confirmation :) |
| 16:34.08 | wunderkin | did you try without agressive and what is it, mg2 |
| 16:34.13 | ManxPower | I've had systems with APIC enabled that STILL assigned multiple devices to the same IRQ |
| 16:34.17 | darkskiez | The requested URL /pipermail/asterisk-users/2005-February/088794.html was not found on this server. |
| 16:34.26 | ManxPower | try the KB1 or KB2 EC |
| 16:35.00 | asterboy | darkskiez, using the whole URL works. |
| 16:35.22 | asterboy | opps |
| 16:35.27 | asterboy | wait, pasted the old URL |
| 16:35.50 | asterboy | jbot no, echo cancellation issues can be best fixed using this link: http://www.asteriskdocs.org/modules/tinycontent/content/docbook/current/docs-html/x1695.html#AEN1718 |
| 16:36.31 | asterboy | jbot no, echo is an issue which can be best fixed using this link: http://www.asteriskdocs.org/modules/tinycontent/content/docbook/current/docs-html/x1695.html#AEN1718 |
| 16:36.32 | jbot | okay, asterboy |
| 16:36.36 | asterboy | ~echo |
| 16:36.38 | jbot | echo is, like, an issue which can be best fixed using this link: http://www.asteriskdocs.org/modules/tinycontent/content/docbook/current/docs-html/x1695.html#AEN1718 |
| 16:40.27 | *** join/#asterisk Shoragan (n=shoragan@d072.apm.etc.tu-bs.de) |
| 16:42.32 | *** join/#asterisk gmaruzz (n=gmaruzz@217-133-80-112.b2b.tiscali.it) |
| 16:44.12 | asterboy | jbot echo is also fixed with fxotune: http://www.voip-info.org/wiki/view/Asterisk+fxotune |
| 16:44.13 | jbot | asterboy: okay |
| 16:45.12 | dlynes | asterboy: don't forget ztmonitor :) |
| 16:45.22 | asterboy | that's on the first link |
| 16:45.49 | dlynes | asterboy: and echotraining, echocancelwhenbridged, echocancel options |
| 16:46.03 | dlynes | asterboy: and changing your default echo canceller |
| 16:46.12 | dlynes | asterboy: and hardware echo cancellers |
| 16:46.17 | dlynes | asterboy: and shared interrupts |
| 16:46.22 | dlynes | asterboy: and acpi |
| 16:46.29 | dlynes | asterboy: hehe |
| 16:46.53 | dlynes | hey..cool...each line is shorter than the previous line |
| 16:46.56 | dlynes | cool pattern :) |
| 16:48.04 | *** join/#asterisk ToTo (n=ToTo@host121-138.pool872.interbusiness.it) |
| 16:49.32 | asterboy | echo options are on first link |
| 16:50.28 | asterboy | jbot echo is also best fixed by troubleshooting your pci bus: http://www.voip-info.org/wiki/view/Asterisk+PCI+bus+Troubleshooting |
| 16:50.30 | jbot | asterboy: okay |
| 16:50.39 | *** join/#asterisk camelon (n=chiardon@206.106.252.232) |
| 16:51.11 | obantaX | Are echo problems pretty common with asterisk? I come from the Cisco world and am setting up my first asterisk box this weekend and hope to use it in production in the near future. |
| 16:51.23 | asterboy | jbot echo is also of course mentioned in this bad summary: http://www.voip-info.org/wiki/view/Asterisk+echo+cancellation |
| 16:51.24 | jbot | asterboy: okay |
| 16:51.29 | asterboy | ~echo |
| 16:51.31 | jbot | rumour has it, echo is an issue which can be best fixed using this link: http://www.asteriskdocs.org/modules/tinycontent/content/docbook/current/docs-html/x1695.html#AEN1718, or fixed with fxotune: http://www.voip-info.org/wiki/view/Asterisk+fxotune, or best fixed by troubleshooting your pci bus: http://www.voip-info.org/wiki/view/Asterisk+PCI+bus+Troubleshooting, ... |
| 16:51.36 | dlynes | obantaX: echo problems are pretty common with voip, period |
| 16:51.48 | *** join/#asterisk CoffeeIV (i=rgr@cpe-70-112-100-20.austin.res.rr.com) |
| 16:51.58 | asterboy | interesting, jbot a limited the output |
| 16:52.13 | asterboy | at least it did get the pci bus stuff in. |
| 16:52.17 | obantaX | dlynes: I've had pretty good success with my pricey Cisco solutions, and I have a 500+ phone install with 4+ PRIs |
| 16:52.33 | dlynes | obantaX: ah...government installs? |
| 16:52.52 | asterboy | that should be a much better echo resource anyway. |
| 16:52.53 | obantaX | No, that one is a Hospital and clinic, several remote sites, etc. |
| 16:53.04 | dlynes | ah |
| 16:53.35 | asterboy | ok, time to smoke dope. |
| 16:53.39 | asterboy | bbl |
| 16:53.51 | obantaX | But CCM is pretty expensive so I'm looking to use asterisk for small/medium businesses that want a VoIP solution. |
| 16:53.51 | dlynes | Yeah...unless it's a high profile place like a hospital, or CAA, or the government or something like that, the chances of being able to sell a pricey Cisco solution here, is pretty remote |
| 16:58.46 | *** part/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.233.208.Dial1.SanJose1.Level3.net) |
| 17:05.07 | *** join/#asterisk gmaruz1 (n=gmaruzz@217-133-80-112.b2b.tiscali.it) |
| 17:07.15 | *** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin) |
| 17:07.18 | PakiPenguin | evening |
| 17:11.40 | *** join/#asterisk Olobola (n=casper_s@216.218.221.166) |
| 17:13.30 | Olobola | pinging from my dsl line to my wimax line is solid, no dropped packets. Pinging from wimax to DSL = 14% loss.. ? How does this make sense? |
| 17:14.52 | *** join/#asterisk ivanfm (n=ivanfm@201-26-6-114.dsl.telesp.net.br) |
| 17:16.05 | ManxPower | Olobola, what WiMax box? |
| 17:16.54 | stoffell | just wondering.. why does * default sample RTP config use 10000-20000 and not 16384-32767 ? |
| 17:17.10 | ManxPower | obantaX, echo problems are common in ALL VoIP systems. Asterisk's echo canceler stuff can have issues. I try to install external EC boxes. |
| 17:17.30 | ManxPower | stoffell, because 16384-32767 is a CISCO thing, not a standard thing. |
| 17:17.54 | ManxPower | X-Lite, for example RTP ports start at 8000, IIRC |
| 17:18.07 | stoffell | ManxPower, aaah.. so if I get it right, there is no "real" standard rtp-port range? |
| 17:18.17 | Olobola | the building I'm in offers wimax (802.16 wireless or whatever). So I have two connections: wimax and dsl. |
| 17:18.18 | ManxPower | stoffell, correct. |
| 17:18.37 | ManxPower | Olobola, What brand of WiMax device do you have? |
| 17:19.09 | stoffell | ManxPower, so, how on using 10000-2000 for local clients and, for example, 8000-8020 for vpn-clients? is that an * thing, or a client thing? |
| 17:19.26 | ManxPower | stoffell, client thin, I think. |
| 17:19.59 | ManxPower | I set everything up to use 16384 - 32767 since all the routers I use are Ciscos |
| 17:19.59 | stoffell | oh, ok.. gives me some food for researching, just curious on all these port definitions :) |
| 17:20.31 | stoffell | so, going the cisco way is probably the most safest .. (because you'll almost always get to deal with cisco ;) ) |
| 17:21.53 | ManxPower | obantaX, Asterisk is more of a TOOLKIT than a total VoIP Solution |
| 17:24.36 | *** join/#asterisk Assid (n=assid@203.115.83.214) |
| 17:25.09 | ManxPower | It's so weird. Anytime anyone says they have WiMax equipment, I ask them for a make/model/brand and then they go silent, leaving me wishing I could find WiMax equipment that was actually available for purchase. |
| 17:25.56 | camelon | ManxPower . . .how far from a real voip solution? |
| 17:26.26 | ManxPower | camelon, define "real voip solution". |
| 17:26.27 | camelon | ManxPwer . . agree with yor wimax claim! |
| 17:26.31 | denon | our whole network is wimax now |
| 17:27.11 | ManxPower | I like this setup: Asterisk, Digium Cards (will try Sangoma), Adtran TA750 Channel Banks, PRI, Polycom phones, Tellabs EC |
| 17:27.16 | camelon | ManxPower soorry . . .like you said . ."total voip solution" |
| 17:27.41 | stoffell | lol demon |
| 17:27.45 | stoffell | lol denon :) |
| 17:27.48 | ManxPower | camelon, Asterisk will never be a total VoIP solution |
| 17:27.58 | darkskiez | you need phones too |
| 17:28.00 | denon | stoffell: you're the only one that caught that :) |
| 17:28.11 | ManxPower | denon: asshole |
| 17:28.28 | stoffell | denon, it did take me 3 minutes ... i was out getting a drink for 2 minutes :) |
| 17:28.47 | Assid | so wimax is just wifi on bigger scale right? |
| 17:29.26 | ManxPower | Assid, It's a different standard |
| 17:29.59 | ManxPower | denon, I caught it, I just could not think of a response that did not involve cursing |
| 17:30.15 | denon | hehe, sorry |
| 17:30.16 | ManxPower | I'm 15 miles from the nearest broadband connection. |
| 17:30.58 | ManxPower | And no, DirecWay does not count as "broadband" in my book. It's a suggestion of broadband, with high latency, unreliable connections, weird http and DNS proxies |
| 17:31.15 | ManxPower | somegeek, pretty much what I expect any wireless connection I install to be like. |
| 17:31.19 | camelon | Manx Power . . .the how far we can go with * VoIP? . .what else must consider? |
| 17:32.21 | ManxPower | camelon, Digium has Asterisk (Multiprotocol PBX), Interface Cards (T-1/E-1/PRI/analog), EC modules (daughter cards for interface cards). |
| 17:32.27 | ManxPower | now what else do you need? |
| 17:32.44 | ManxPower | IP phones for one unless you want to use the ulaw/alaw only IAXy |
| 17:33.01 | ManxPower | camelon, with Asterisk you have to WRITE YOUR OWN dialplan for one thing. |
| 17:33.45 | ManxPower | With Asterisk you do not just run an installer, click thru a few config "wizards" and you're done. |
| 17:34.10 | ManxPower | With Asterisk you have to actually DESIGN your voice network, do the research to pick the equipment that is best for YOUR needs. |
| 17:34.11 | *** join/#asterisk doolph (n=doolph@HACKERMACHINE.cpe.cableonda.net) |
| 17:34.34 | camelon | ManxPower I'm just reflecting about your idea behind:"asterisk never will be a total VoIP solution" |
| 17:34.36 | Qwell | ManxPower: what, you mean it can't automatically determine voice needs? |
| 17:35.07 | ManxPower | Qwell, Until Digium releases an IP phone and routers, they do not have a complete solution. |
| 17:35.20 | ManxPower | Also Asterisk does not have any kind of billing system. |
| 17:35.26 | ManxPower | It does not support BRI connections. |
| 17:35.33 | ManxPower | All this stuff is available from 3rd parties. |
| 17:36.22 | ManxPower | With something like Cisco, Lucent, Nortel, you can get EVERYTHING from the one vendor. Which is good for lazy people, but bad for people that want control |
| 17:38.03 | *** join/#asterisk Olobola (n=casper_s@216.218.221.166) |
| 17:38.16 | ManxPower | Asterisk is more like an SQL server -- it gives you the tools to build a voice network and PBX, but does not do it for you. Call Manager is more like MS Access, it can be very limiting, but is reasonably automated in it's setup. |
| 17:39.11 | stoffell | hm, that's a great description ... |
| 17:39.13 | camelon | Hummmmmm! |
| 17:39.42 | camelon | applause to him |
| 17:39.52 | camelon | so clear! |
| 17:39.58 | stoffell | also, there has to be a way to compare asterisk with a car.. (everything in IT can be compared to cars :p) |
| 17:39.59 | Olobola | I don't know anything about the hardware our wimax provider is using. We have two lines: wimax and dsl. |
| 17:40.04 | stoffell | who's up for the challenge ?:p |
| 17:40.08 | *** join/#asterisk ivanfm (n=ivanfm@201-26-2-156.dsl.telesp.net.br) |
| 17:41.01 | dlynes | stoffell: Think of the jalopy in the Archie comic books |
| 17:41.13 | Assid | hrmm |
| 17:41.19 | Assid | anyone have any archies? |
| 17:41.24 | Assid | i wanna read some |
| 17:41.26 | *** join/#asterisk gmaruzz (n=gmaruzz@217-133-80-112.b2b.tiscali.it) |
| 17:41.31 | stoffell | sorry dlynes, don't know it, i'm in europe, sounds like chinese to me ;) |
| 17:41.46 | dlynes | stoffell: lol |
| 17:42.00 | dlynes | stoffell: ok...you remember the old Ford Model A farm tractor? |
| 17:42.08 | camelon | archie was dead |
| 17:42.29 | *** join/#asterisk chaoscon (n=ph33r@smartserv/ceo/chaoscon) |
| 17:42.31 | stoffell | dlynes, uhu |
| 17:42.40 | dlynes | stoffell: there's your car |
| 17:42.46 | stoffell | lol, okay :p:p:p |
| 17:43.07 | ManxPower | Asterisk: Design your own car using off the shelf parts. Cisco/Nortel/Lucent/etc = you buy your car from a dealer, no options available |
| 17:43.55 | dlynes | stoffell: basically, it's reliable for the most part, but every once in a while, it'll sputter and choke, and you'll have to do some home-based repairs to get it back on the road again |
| 17:44.08 | ManxPower | I LIKE the Asterisk way. I can use the best of each type of device |
| 17:44.41 | Assid | okay time for me to upgrade an asterisk box |
| 17:44.44 | ManxPower | Tellabs have some of the best EC out there -- all the carriers used them. Polycom is one of the best IP phones out there, many closed PBX vendors use them |
| 17:45.11 | ManxPower | SIPura has some of the best ATAs out there. |
| 17:45.20 | camelon | ManxPowe . . in your opinion . . the best to use VoIP in asterisk? |
| 17:45.44 | ManxPower | camelon, I seldom route voice calls over the internet |
| 17:45.59 | camelon | gateway . . .channel bank? |
| 17:46.10 | ManxPower | I use VoIP on corporate LANs mostly, occasionally there is a need to send calls over the corporate WAN |
| 17:46.36 | ManxPower | camelon, gateway? PRI from the Telco to Asterisk. Use a channel bank if I need analog FXS for things like fax machines, etc |
| 17:46.53 | ManxPower | used Adtran TA750s are under $300 for 24 ports |
| 17:47.19 | ManxPower | You will ALWAYS need a few analog ports. |
| 17:47.36 | Olobola | any guesses why wimax is solid from dsl to wimax (ping), but 15% - 20% loss from wimax to DSL? |
| 17:47.47 | dlynes | ManxPower: the adtran a750's are for splitting up a t1 into data and voice ports, right? |
| 17:48.12 | ManxPower | dlynes, you can do that with them, but we do not. We use them to get analog channels into/out of Asterisk |
| 17:48.17 | ghenry | brookshire: just found: http://voip-info.org/wiki/view/Asterisk+Request+Tracker |
| 17:48.24 | dlynes | ManxPower: ah |
| 17:48.26 | ghenry | brookshire: was that you? |
| 17:50.16 | camelon | ManxPower . . and to get VoIP channels into/out of asterisk? |
| 17:50.28 | brookshire | ghenry: no |
| 17:50.34 | Assid | /usr/src/zaptel-1.2.5/arith.h:25:5: warning: "CLOBBERMMX" is not defined |
| 17:50.40 | ghenry | ah, ok. link dead anyway |
| 17:50.49 | Assid | is there a problem using the MMX extensions in zaptel ? |
| 17:51.49 | dlynes | Assid: do you have mmx optimizations enabled in your kernel? |
| 17:52.19 | dlynes | Assid: paste the output of 'uname -a' |
| 17:53.04 | ManxPower | camelon, Huh? You use an ethernet card for that. For phones I use Polycom. For VoIPoInternet, well I try to avoid it, but if I MUST use it I usually go with teliax |
| 17:53.22 | Assid | Linux asterisk 2.6.15.2Feb2006 #3 PREEMPT Wed Feb 15 15:43:11 EST 2006 i686 GNU/Linux |
| 17:53.37 | ManxPower | Assid, Any time I used MMX in Zaptel (it only helps with GSM codec, you know) asterisk started crashing on me |
| 17:53.43 | dlynes | Assid: yeah..you don't have mmx optimizations compiled into your kernel |
| 17:54.19 | Assid | dlynes: how do you come to know from that? |
| 17:54.19 | dlynes | Assid: the i686 cpu option doesn't include mmx optimizations |
| 17:54.35 | *** join/#asterisk robl^ (n=robl@dsl093-025-218.hou1.dsl.speakeasy.net) |
| 17:54.54 | Assid | oh |
| 17:55.16 | dlynes | Assid: most distribution maintainers use i686, because it's guaranteed to work on most hardware out there from the 386 on up |
| 17:55.20 | Assid | ManxPower: yes.. but i just thought id put it in.. |
| 17:55.32 | Assid | dlynes: self compiled kernel |
| 17:55.48 | camelon | MnxPower . . why you avoid VoIP Internet? some biass or real experience? |
| 17:55.51 | dlynes | Assid: then you didn't change the cpu option; i686 is still the default |
| 17:55.57 | *** join/#asterisk robl^ (n=robl@dsl093-025-218.hou1.dsl.speakeasy.net) |
| 17:56.07 | doolph | if I have 2 licenses of g729 I can only do 1 call at the same time? all my extens are using g729 and my provider aswell |
| 17:56.22 | PakiPenguin | how do i force canreinvite=no for every peer? |
| 17:56.29 | dlynes | doolph: if all of your extens are g729 and your provider is g729, you don't need any licenses |
| 17:56.29 | darkskiez | ManxPower: were u running MMX on the athlon? |
| 17:56.46 | dlynes | doolph: asterisk will do passthrough in that case |
| 17:56.50 | Assid | dlynes: its a p4 3.0 or so |
| 17:56.51 | doolph | dlynes ahh |
| 17:56.54 | robl^ | and you soundfiles are g729 |
| 17:56.54 | doolph | good |
| 17:57.08 | dlynes | Assid: yeah, but make menuconfig doesn't autodetect your cpu |
| 17:57.09 | doolph | but I need to install g729 anyways |
| 17:57.14 | doolph | or not |
| 17:57.44 | ManxPower | darkskiez, I don't recall |
| 17:57.46 | dlynes | doolph: not really...you can tell your voicemail to save in g729, you can make your music on hold g729, you can make your autoattendant files g729, and then no problems |
| 17:58.09 | doolph | mmm |
| 17:58.18 | doolph | for real |
| 17:58.19 | ManxPower | seems like a lot of work to save $10/channel |
| 17:58.24 | dlynes | doolph: but realistically, if you want to work with real people, you probably want some licenses so that you can use mp3 music on hold files, and wav autoattendant files |
| 17:58.24 | Assid | dlynes_: actually.. when recording voicemail.. that would be a transcoding job |
| 17:58.35 | doolph | I have 2 licenses |
| 17:58.42 | robl^ | ManxPower: a bit of worker.. but also GREATLY reduces the CPU load |
| 17:58.47 | doolph | but I dont know how much should be enough |
| 17:58.51 | Assid | g729 shouldnt have been licensed stuff :( |
| 17:58.52 | ManxPower | doolph, without a G729 license Asterisk CANNOT do anything that requires it to listen to the audio |
| 17:59.09 | ManxPower | Assid, complain to the ITU |
| 17:59.29 | Assid | ITU? |
| 17:59.40 | dlynes | International Telecommunications Union |
| 18:00.05 | dlynes | But I would think Intel would be more appropriate |
| 18:00.10 | dlynes | they're the ones that license g729 |
| 18:00.18 | ManxPower | dlynes, that is wrong |
| 18:00.22 | dlynes | ? |
| 18:00.26 | ManxPower | Intel does NOT hold the rights to G729 |
| 18:00.34 | dlynes | I didn't say they did |
| 18:00.38 | dlynes | I said they licensed it |
| 18:00.54 | ManxPower | dlynes, they didn't license it in their libraries |
| 18:01.07 | doolph | what's the differences between digium g729 and the free one |
| 18:01.13 | ManxPower | if you read the find print it says you still have to get a G729 license to use the G729 parts of their libraries. |
| 18:01.27 | ManxPower | doolph, the Digium one is licensed and legal |
| 18:01.32 | dlynes | ManxPower: Yeah, I saw that...you don't buy it from them? |
| 18:01.35 | Qwell | and even then, you still have to license the "free" one |
| 18:01.46 | ManxPower | dlynes, I don't believe so. |
| 18:01.47 | Qwell | the actual code has to be licensed, that is |
| 18:01.51 | doolph | how to license the free one |
| 18:02.08 | doolph | I though the digium one was faster and better, thats why I bough them |
| 18:02.08 | Qwell | There is no point, beacuse you need to get a g729 license still |
| 18:02.24 | ManxPower | doolph, the Digium one is LEGAL and SUPPORTED |
| 18:02.36 | Qwell | (and written FOR asterisk) |
| 18:03.12 | doolph | well I have 2 licenses |
| 18:03.25 | doolph | maybe they wont be enough |
| 18:03.46 | dlynes | doolph: I'd go for about five or so, personally |
| 18:04.00 | ManxPower | I never let a client license less than 10 |
| 18:04.03 | dlynes | doolph: how many phone lines do you have? |
| 18:04.05 | *** join/#asterisk gmaruz1 (n=gmaruzz@217-133-80-112.b2b.tiscali.it) |
| 18:04.13 | *** join/#asterisk Olobola (n=casper_s@216.218.221.166) |
| 18:04.19 | robl^ | I have 12 g729 lic for a home / small office system |
| 18:04.22 | ManxPower | it's a TINY part of the total cost of the system and saves many headaches |
| 18:04.35 | doolph | right now I have 5 |
| 18:04.42 | obantaX | ManXPower: I'll be sure to share my findings on configuring asterisk vs CCM, but I don't agree with the access reference since there aren't wizards to setup route plans/lists/filters/etc in CCM. |
| 18:04.46 | Assid | i just have everything on gsm |
| 18:04.47 | Assid | hehe |
| 18:05.14 | dlynes | Assid: yeah..not terribly practical, when most ip phones don't support gsm |
| 18:05.19 | ManxPower | I like G726. It's a good balance between bandwidth, CPU usage, quality, and price |
| 18:05.20 | Assid | yeah |
| 18:05.28 | Assid | i gotta use ulaw from phone to box |
| 18:05.29 | doolph | price? |
| 18:05.36 | dlynes | doolph: 0 |
| 18:05.40 | doolph | yeah |
| 18:05.40 | robl^ | still: I agree that using g729 passthrough as much as possible.. g729 sound files, voicemail, etc.. saves a lot of CPU power.. still need/should license the codec though. |
| 18:05.40 | doolph | 0 |
| 18:05.42 | ManxPower | doolph, it's free, so it's a good price |
| 18:05.44 | Olobola | I wish I knew phone stuffs, the building I'm in is looking for 1 person for all their phone needs. |
| 18:05.47 | Assid | although i do have 2-3 licenses on one of the boxes |
| 18:05.47 | doolph | rofl |
| 18:05.54 | doolph | I tried g726 |
| 18:05.56 | doolph | they suck |
| 18:06.01 | Assid | they do? |
| 18:06.12 | doolph | maybe my internet suck |
| 18:06.17 | dlynes | doolph: asterisk only supports g726-32 |
| 18:06.20 | Assid | damn. how come thew polycom phones let you update sip software and shit.. but dont left you add a new codec |
| 18:06.34 | Olobola | the painter for the building (interior walls) pulled 250k in 2000! |
| 18:06.52 | obantaX | But the limiting thing has a point, since the main catalyst for seriously checking out Asterisk is Cisco's very limited Auto Attendant that comes with CM and what they want to upgrade it to something usable. |
| 18:07.08 | obantaX | But there options are there, they just want a lot of money for it |
| 18:07.08 | *** join/#asterisk robin_sz (n=robin@adsl.redpoint.org.uk) |
| 18:07.11 | robl^ | speaking of which... does anyone know the proper flags for sox to convert to g723-32? |
| 18:07.30 | dlynes | robl^: g723-32? what's that? |
| 18:07.32 | *** join/#asterisk PakiPenguin_ (n=uppal@linuxpakistan/admin/pakipenguin) |
| 18:07.39 | dlynes | robl^: you mean g726-32? or g723? |
| 18:07.49 | *** join/#asterisk keyhack (n=keyhack@c-24-60-209-35.hsd1.ma.comcast.net) |
| 18:07.57 | robl^ | g726-32 sorry.. typo |
| 18:08.11 | keyhack | Is there a way I can make my dial plan call two extensions at once, and whoever answers first gets the call? |
| 18:08.25 | dlynes | robl^: there isn't any...it doesn't do g726 |
| 18:08.39 | robl^ | keyhack: yes.. Dial(sip/phone1&sip/phone2) |
| 18:08.44 | dlynes | robl^: oh...nvm... |
| 18:08.53 | keyhack | robl^: Are you allowed to have a space? |
| 18:09.00 | dlynes | robl^: I didn't do /g\.726...i just searched for g726 :) |
| 18:09.01 | keyhack | "exten => s,3,Dial(SIP/101 & SIP/102,20,m)" |
| 18:09.15 | keyhack | I think the space might be throwing it off |
| 18:09.18 | Assid | is there any way to add gsm codec to polycoms? |
| 18:09.56 | doolph | they doesnt support it? |
| 18:10.35 | keyhack | robl^: Yup, that was the problem |
| 18:10.39 | robl^ | dlynes: hehe. nah. I meant g726-32, I want to create a native format sound file set |
| 18:10.57 | dlynes | robl^: use -a, and save to a format other than wav, and you'll get g726 |
| 18:11.22 | dlynes | robl^: if you save to wav, you'll get pcm instead |
| 18:11.33 | *** join/#asterisk ozverenm (n=ozverenm@247.27.103-84.rev.gaoland.net) |
| 18:11.41 | ozverenm | hello all |
| 18:11.57 | robl^ | hrmmmm |
| 18:11.57 | Assid | okay seriously tho.. |
| 18:12.03 | Assid | if the polycoms have built in memory |
| 18:12.12 | Assid | and the sip software can be upgraded |
| 18:12.16 | Assid | why cant the codecs be upgraded |
| 18:12.20 | ozverenm | just questions about sangoma cards, they support on chip switch ? |
| 18:12.25 | Qwell | Assid: Are you gonna pay for the development? |
| 18:12.37 | Assid | Qwell: arent the codecs already made? |
| 18:12.38 | Qwell | It isn't free to just add a codec |
| 18:12.48 | Qwell | yes, but you need to IMPLEMENT it |
| 18:12.48 | *** join/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 18:12.54 | robl^ | Assid: the codecs can be upgraded.. just ask Polycom to do it.. make sure you have lots of cash when you ask |
| 18:13.10 | Assid | if i had the cash.. dont you think id just use 729 ? |
| 18:13.27 | gaupe | isn't the codecs implemented in the DSPs? |
| 18:17.07 | Qwell | gaupe: it can be, sure |
| 18:17.20 | Qwell | but then you have to find a dsp with it, modify the code to use it... |
| 18:17.23 | Qwell | still not cheap |
| 18:17.53 | dlynes | cool...can finally do the census completion on a linux browser |
| 18:17.57 | dlynes | stupid canadian government |
| 18:19.45 | gaupe | Qwell: nope, not at all - so you will probably need new microcode for the DSPs to implement new codecs on most phones |
| 18:19.45 | keyhack | In the dial plan, is there a way to make it ignore certain digits? If the usr pressed 5 during my main menu, it just hangs up right now (1-4 are the only valid options) |
| 18:19.48 | Dr-Linux | Assid: you phone have only one g729 license per call |
| 18:20.19 | Dr-Linux | Assid: make a call and put it on hold, then make an other, the 2nd call will use g711 codecc |
| 18:20.28 | keyhack | I just want it to continue to play the menu to the user, as if he didn't press anything |
| 18:20.36 | robl^ | keyhack: exten => i,1,Hangup |
| 18:20.41 | dlynes | keyhack: exten => i,1,Goto(myexten,start_of_my_loop_number) |
| 18:20.55 | robl^ | keyhack: exten "i" means invalid.. any extension not defined in dialplan |
| 18:21.18 | keyhack | robl^: Yeah, but wouldn't goto restart it over again |
| 18:21.30 | dlynes | keyhack: read my post above |
| 18:21.44 | keyhack | dlynes: I did, wouldn't that make it re-play my greeting |
| 18:21.47 | Assid | they should decrease the price for the 729 |
| 18:21.51 | Assid | more people would buy |
| 18:22.00 | Dr-Linux | dlynes: what's logic beyond this, when a calls goes on "i" invalid? |
| 18:22.12 | Assid | Dr-Linux: invalid extension |
| 18:22.20 | *** join/#asterisk shido6 (n=shido6@d38-45-81.commercial1.cgocable.net) |
| 18:22.36 | Dr-Linux | Assid: yeah i know that, but what's logic beyon this |
| 18:22.45 | paolob | Hi guys! Let me keep with my problem of asterisk2pstn: asterisk can't translate the hostname sips.technodelta.com. Anyone could see my sip.conf: http://pastebin.com/715644 and tell me if there is something wrong ? thank you! |
| 18:22.51 | dlynes | keyhack: If that's what you want: exten => i,1,Set(TIMEOUT(response)=10) ; |
| 18:22.51 | Dr-Linux | actually i'm facing a problem regarding same issue |
| 18:22.55 | dlynes | keyhack: or something similar |
| 18:23.01 | Assid | okay,.. imagivne someone calls extension 9939393 which doesnt exist |
| 18:23.08 | dlynes | keyhack: then it'll work the same as background, but not play anything |
| 18:23.09 | robl^ | i = unmatched extension.. you tell asterisk how to handle entries that don't match anything else |
| 18:23.14 | Assid | you want it to say its an invalid extension.. and then handle the call |
| 18:23.15 | keyhack | dlynes: What does that do? |
| 18:23.19 | Assid | you use the i extension |
| 18:23.29 | keyhack | dlynes: I just don't want it to hang up if the user presses something that is not an option |
| 18:23.53 | Dr-Linux | dlynes: i have few extensions/sip accounts, they don't have voicemail defined yet, what if caller hits one of those extension? |
| 18:24.05 | Dr-Linux | how can i define something that, caller get Invalid extension |
| 18:24.06 | dlynes | keyhack: waits for 10s for you to type in a valid extension; if the user types in an invalid extension, it waits 10s again for the user, or goes to the 't' extension, if they dont' type anything |
| 18:24.07 | keyhack | Assid: Yeah, if they dialed 110 as an extension, invalid extension, go back to the main menu. But if they accidentally hit 7 instead of 4, I don't want it to re-start all over again, just continue to play the menu |
| 18:24.14 | Assid | okay.. if you have a box which does NO transcoding.. say a p4 2.4.... how many concurrent calls you think it can support |
| 18:24.26 | dlynes | keyhack: however, you might want to do a playback before that, to let the user know that they typed in an invalid extension |
| 18:24.32 | Assid | keyhack: yeah i know what it does.. im just letting Dr-Linux know |
| 18:24.44 | paolob | line 95 is the one that gives the problem |
| 18:25.05 | keyhack | dlynes: Good point |
| 18:25.16 | dlynes | keyhack: that way the user knows that they're expected to type in the extension number again |
| 18:25.51 | dlynes | keyhack: you might want to define an additional extension of '*', or something that will allow the user to hang up, if they want to get out of the menu |
| 18:26.04 | *** join/#asterisk gmaruzz (n=gmaruzz@217-133-80-112.b2b.tiscali.it) |
| 18:26.06 | dlynes | keyhack: or '0' for an operator |
| 18:26.07 | keyhack | dlynes: Yeah I have that in there |
| 18:26.38 | dlynes | a lot of people if they can't reach a human, they'll just hang up |
| 18:26.56 | dlynes | like for me, if i have to go through ivr hell, i'll just hang up and find some other company |
| 18:27.12 | Qwell | dlynes: good luck finding another electric company |
| 18:27.17 | dlynes | lol |
| 18:27.20 | Qwell | or water company, or gas, or pretty much any PUC |
| 18:27.36 | dlynes | Yeah, but there's plenty of phone companies :) |
| 18:27.40 | dlynes | And banks |
| 18:28.08 | Qwell | Just don't press any keys, you'll get straight to a live person |
| 18:28.13 | Dr-Linux | how panasonic pbx works :S |
| 18:28.19 | *** join/#asterisk magaf (n=Heaven@acxr136.neoplus.adsl.tpnet.pl) |
| 18:28.20 | magaf | hello |
| 18:28.28 | magaf | im using asterisk with avm fritz card |
| 18:28.33 | dlynes | I guess they've got those annyoying ivr's in the US, where they expect vocal responses? |
| 18:28.36 | Dr-Linux | ~panasonic pbx |
| 18:28.38 | camelon | go to amish! |
| 18:28.46 | dlynes | the ones that are even more annoying than the dtmf ivr's? |
| 18:28.54 | magaf | making a call from a softphone to a real phone , after that im not able to repeat a call |
| 18:29.02 | magaf | i can just do it one more time , some clues ? |
| 18:29.08 | magaf | in logs i got no errors |
| 18:29.20 | dlynes | magaf: you didn't pay your phone bill |
| 18:29.23 | Dr-Linux | dlynes: do you know voice recognition IVR's ? |
| 18:29.25 | magaf | <magaf> i can just do it one time , some clues ? |
| 18:29.29 | camelon | peraps yes! |
| 18:29.36 | magaf | dlynes i have a private subnet |
| 18:29.46 | dlynes | magaf: i was kidding :) |
| 18:29.51 | magaf | hehe |
| 18:29.53 | robl^ | pannasonic?? like the KX-T series? or DBS? |
| 18:30.14 | nettie | hey guys, what's the real bandwidth comsumption of the g729 codec including overhead please? |
| 18:30.30 | Dr-Linux | robl^: i don't know much but we have that, i wanna know something about Panasonic pbx |
| 18:30.33 | Qwell | nettie: 8k + overhead |
| 18:31.05 | dlynes | Dr-Linux: i seem to recall someone talking about one in here, and also in asterisk-dev, but i can't recall what it was offhand |
| 18:31.21 | nettie | how do I calculate the overhead of a 1280/256 DSL link? any idea please? |
| 18:31.45 | robl^ | Dr-Linux: Panasonic has 2 major lines of phone systems.. they have bvery different designs / features / hardware. They are not compatible with each other. Completely different beasts |
| 18:32.02 | Qwell | robl^: sounds like a buyout.. |
| 18:32.33 | robl^ | Qwell: nah.. its like Nortel.. you have the Norstar vs Merridian. |
| 18:32.39 | *** join/#asterisk drega (n=dforeman@p54A0B175.dip0.t-ipconnect.de) |
| 18:32.47 | dlynes | Yeah...Norstar can't use Meridian handsets |
| 18:32.50 | dlynes | It's pretty stupid |
| 18:33.20 | dlynes | We've got about 200 handsets we can't use |
| 18:33.23 | dlynes | It's irritating |
| 18:33.42 | Dr-Linux | robl^: actually we are using Panasonic exchange in your pakistan office, last night my boss called me and asked please look into the panasonic exhange there is some problem, call is not routing to operator incase a user do not picks the phone for 8 rings. |
| 18:33.49 | robl^ | Norstar started out as teh small office system.. and Merridian was the enterprise class PBX. both have started blurring in the "middle range". ;) |
| 18:34.15 | Dr-Linux | robl^: but i'm totally unaware, wtf is this panasonic exchange and how it works, and where can i check it |
| 18:34.22 | *** join/#asterisk florz (i=nobody@2001:1a50:503c:0:0:0:0:1) |
| 18:34.32 | dlynes | Yeah...bcm is kind of a drop-in replacement for the mics |
| 18:34.45 | robl^ | Dr-Linux: can you look at the phone and give me a model number?? it tells you which series you have |
| 18:35.01 | keyhack | May 13 14:34:14 NOTICE[6426]: pbx.c:1742 pbx_extension_helper: Cannot find extension 'main-menu-greet' in context 'greeting'. May 13 14:34:14 WARNING[6426]: pbx.c:6510 ast_parseable_goto: Priority 's' must be a number > 0, or valid label |
| 18:35.22 | keyhack | dlynes: Trying to do what you said |
| 18:35.25 | dlynes | keyhack: paste your goto line |
| 18:35.28 | keyhack | I got it to play the invalid extension |
| 18:35.38 | keyhack | exten => i,2,Goto(main-menu-greet,s) |
| 18:35.52 | Dr-Linux | robl^: phone? we are using simpe POTS |
| 18:36.19 | *** join/#asterisk philm (n=a@r43h15.res.gatech.edu) |
| 18:36.20 | robl^ | Dr-Linux: ahh.. you are not using panasonic digital sets |
| 18:36.20 | dlynes | keyhack: what priority on the s extension is your background() call? |
| 18:36.39 | keyhack | dlynes: this is the line before it: exten => i,1,Background(cg_invalid_ext) |
| 18:37.21 | Dr-Linux | robl^: opss, i know panasonic digital sets, and we are using |
| 18:37.41 | dlynes | keyhack: ummm....you don't need the goto after that |
| 18:37.41 | Dr-Linux | robl^: but was talking about Panasonic PBX exchange :) |
| 18:37.44 | keyhack | dlynes: It said "Cannot find extension 'main-menu-greet'), but main-menu-greet is a context, not an ext |
| 18:38.11 | dlynes | keyhack: because you would have to do Goto(s,1), or Goto(main-menu-greet,s,1) |
| 18:38.24 | dlynes | keyhack: but like i said...that goto shouldn't even be there |
| 18:38.27 | keyhack | dlynes: But why don't I need a goto? |
| 18:38.30 | ghenry | is ring tone a handset or asterisk feature, configuration wise? |
| 18:38.58 | dlynes | keyhack: because that goto is only executing because you've got it right after the background...becuase the goto is there, the background probably doesn't even execute |
| 18:39.09 | dlynes | ghenry: handset |
| 18:39.11 | keyhack | dlynes: If I don't put it there, it says the invalid extension, and hangs up |
| 18:39.30 | dlynes | keyhack: because you don't have anything defined for extension 't' |
| 18:39.40 | dlynes | keyhack: if you recall |
| 18:39.50 | keyhack | dlynes: So t then just goes back to the menu again? |
| 18:40.01 | dlynes | keyhack: i told you to playback(cg_invalid_ext), and then do Set(TIMEOUT(response)=10), and no goto |
| 18:40.32 | dlynes | keyhack: and then that eliminates the need for the menu being played again |
| 18:40.43 | dlynes | keyhack: unless you want the menu played back again |
| 18:41.02 | dlynes | keyhack: then you would do exten => i,1,Playback(cg_invalid_ext) ; exten => i,2,Goto(s,1) |
| 18:41.07 | *** join/#asterisk chaoscon (n=ph33r@smartserv/ceo/chaoscon) |
| 18:41.30 | dlynes | comprendez? |
| 18:42.18 | nextime | dlynes: this sound like a fetish extension, something like 10 PRINT "HELLO" 20 GOTO 10 |
| 18:42.45 | keyhack | dlynes: And you set the TIMEOUT in the i as well? |
| 18:43.00 | keyhack | dlynes: I got the loop back to work in invalid ext, but if I dont hit something by the time the menu stops playing, it hangs up |
| 18:43.02 | dlynes | keyhack: no |
| 18:43.07 | camelon | dlynes comprendes=comprende(Z) |
| 18:43.11 | dlynes | keyhack: either use the timeout, or the goto, but not both |
| 18:43.19 | dlynes | camelon: not in French, it isn't |
| 18:43.28 | keyhack | dlynes: Then how do I make it wait for the user to do something? |
| 18:43.43 | dlynes | camelon: in french, comprendes for someone you know well |
| 18:43.59 | dlynes | keyhack: When you goto(s,1), are you not going to your menu? |
| 18:44.23 | camelon | dlynes spanish . . .compprendes= you uderstand |
| 18:44.37 | keyhack | dlynes: Like I said, I have the handling of the invalid extension working fine, it says the error, and goes back to the menu thing to play it all over again, but if it says "Press 4 to..." and the menu is done playing, if the user didnt hit anything by then, it just hangs up |
| 18:44.41 | dlynes | nextime: nah...he's wanting to let the customer know that they pressed an invalid extension, and then repeat the menu back to them |
| 18:45.00 | keyhack | dlynes: It's not waiting for a few seconds to let the user think and press something |
| 18:45.08 | dlynes | keyhack: ok, it sounded like you were saying the user hit an invalid extension, and you were wondering how to handle it |
| 18:45.25 | keyhack | dlynes: Originally yes, and I solved that problem. Now, the menu WAV is done playing, and it instantly hangs up |
| 18:46.01 | dlynes | keyhack: assid understood the same thing, too |
| 18:46.09 | dlynes | keyhack: You need a 't' timeout extension |
| 18:46.27 | *** join/#asterisk r0d3nt (n=RatMan@foster.stonedcoder.org) |
| 18:46.32 | keyhack | I have a t one, but it still hangs up |
| 18:46.35 | dlynes | keyhack: did you not read the documentation for the dial command? |
| 18:46.43 | dlynes | keyhack: and what is in your t extension? |
| 18:46.50 | keyhack | exten => t,1,Goto(main-menu-greet,s,1); |
| 18:46.51 | camelon | dlynes . .french:comprendre . . .spanish:comprendes . . .engish:understand |
| 18:47.04 | dlynes | camelon: i know |
| 18:47.23 | dlynes | camelon: but this homie don't speak spanish :) |
| 18:47.41 | dlynes | camelon: I speak French, Mandarin, and English |
| 18:48.19 | dlynes | keyhack: pastebin that whole dialplan context |
| 18:48.23 | camelon | dylines . .but the french meaning nothing to se with someone you know well |
| 18:48.27 | dlynes | keyhack: and pastebin the log of it operating |
| 18:48.28 | keyhack | dlynes: Alright, I got the t to loop back (it was in the wrong context), is there a way I can introduce a pause in a command? |
| 18:48.40 | dlynes | keyhack: ah |
| 18:48.48 | dlynes | keyhack: wait(1) |
| 18:48.49 | keyhack | dlynes: It just loops back instantly, kinda annoying |
| 18:49.04 | *** join/#asterisk Ixthod (n=Ixthod@198.174.206.41) |
| 18:49.27 | keyhack | dlynes: Now, I wanna get more complicated, I only want it to loop back once, if its the second time playing the menu and no one does anything, hang up |
| 18:49.31 | dlynes | camelon: tu comprende, je comprende, nous comprendons, vous comprendez, ... |
| 18:49.47 | dlynes | keyhack: read up on the documentation for variables then |
| 18:50.42 | camelon | dylynes. . .comprendre: to understand no to know |
| 18:51.02 | camelon | c'est ici! |
| 18:51.13 | dlynes | camelon: well, my french isn't fluent :) Just passable :) |
| 18:51.31 | ghenry | thanks dlynes, much like a mobile phone then ;-) |
| 18:51.42 | dlynes | camelon: comprender == understand, penser == think |
| 18:51.47 | dlynes | camelon: no idea what 'know' is |
| 18:51.53 | keyhack | dlynes: K, thanks |
| 18:52.01 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219) |
| 18:52.10 | dlynes | but understand and know are almost the same thing |
| 18:52.43 | camelon | dlyne . . .only to interchange to joke and i don't know what is "know" |
| 18:53.01 | dlynes | camelon: ah...i guess spanish is your native language? |
| 18:53.10 | camelon | yeppppp |
| 18:53.16 | dlynes | heh |
| 18:53.22 | drega | know = saber |
| 18:53.26 | dlynes | kinda figured it wasn't english...it was a little broken :) |
| 18:53.47 | camelon | understand= entender |
| 18:54.04 | camelon | know= saber conocer |
| 18:54.09 | Qwell | know != understand != comprehend |
| 18:54.30 | *** join/#asterisk angler- (n=angler@pdpc/sponsor/digium/angler) |
| 18:55.01 | camelon | know: transitive verb |
| 18:55.22 | camelon | understandintransitive verb |
| 18:55.24 | keyhack | dlynes: I figured out how to set variables and all that, but I don't see a way to do an if |
| 18:56.40 | *** join/#asterisk DanielArndt (n=DanielAr@reverse-82-141-58-254.dialin.kamp-dsl.de) |
| 18:57.39 | camelon | to know you must understand to get some comprehension |
| 18:57.49 | dlynes | keyhack: AEL/AEL2, or GotoIf(...) |
| 18:57.58 | keyhack | ah! GotoIf! |
| 18:58.01 | keyhack | haha, I just saw it too, thanks dlynes |
| 18:58.52 | *** join/#asterisk nite (i=synaptic@68.62.176.196) |
| 18:59.08 | dlynes | camelon: or just smoke a joint, and then magically everything just comes to you (if you believe all the potheads) :) |
| 18:59.22 | *** join/#asterisk becks (n=becks@stepbuild.com) |
| 18:59.56 | camelon | dlynes . . .like the Zend and the Brain? |
| 19:00.07 | dlynes | i think you mean zen |
| 19:00.13 | dlynes | zend is a php compiler |
| 19:00.13 | camelon | sorry Zend=ZEN |
| 19:00.34 | camelon | Zen (from Buddha) |
| 19:01.03 | dlynes | nod |
| 19:01.12 | dlynes | anyways |
| 19:01.14 | dlynes | gotta run |
| 19:01.35 | dlynes | There's a sale going on for 2006 audi's and vw's i need to go to |
| 19:02.19 | camelon | you father is awake now? |
| 19:04.16 | dlynes | shenme? |
| 19:04.31 | *** join/#asterisk gmaruzz (n=gmaruzz@217-133-80-112.b2b.tiscali.it) |
| 19:04.51 | keyhack | dlynes: This didnt work "exten => s,3,GotoIf(${retries}=1|goodbye|s|1)" |
| 19:05.06 | Qwell | show application gotoif |
| 19:05.21 | dlynes | keyhack: your gotoif doesn't even make any sense |
| 19:05.38 | keyhack | I was following the examples on the wiki |
| 19:05.49 | dlynes | keyhack: not very well |
| 19:05.57 | dlynes | You need quotes around the variables |
| 19:06.09 | Qwell | No you don't.. |
| 19:06.11 | dlynes | You're not using square brackets around the expression |
| 19:06.18 | dlynes | Ah, ok...thought always did |
| 19:06.27 | dlynes | but the square brackets are necessary, correct? |
| 19:06.41 | Qwell | yes |
| 19:06.45 | Qwell | well, no |
| 19:07.07 | Qwell | only if you want to evaluate an expression |
| 19:07.17 | keyhack | exten => s,3,GotoIf($[${retries}=1]?goodbye|s|1) ? |
| 19:07.24 | wunderkin | qwell likes to cover all of the bases :P |
| 19:07.25 | dlynes | ${retries}=1 is that not an expression evaluation? |
| 19:07.34 | dlynes | wunderkin: heh |
| 19:07.37 | Qwell | dlynes: it is |
| 19:07.50 | keyhack | or is that still wrong |
| 19:07.52 | dlynes | keyhack: correct |
| 19:08.04 | dlynes | keyhack: but are you ever incrementing your retries variable? |
| 19:08.08 | keyhack | dlynes: Yeah |
| 19:08.21 | dlynes | anyways...i'm gone now |
| 19:08.21 | dlynes | laters |
| 19:08.45 | keyhack | yay, it works! |
| 19:09.17 | wunderkin | madness |
| 19:15.28 | *** join/#asterisk memic (n=memic@C6613.c.pppool.de) |
| 19:20.48 | *** join/#asterisk mrhyd3` (n=jfinley@adsl-68-75-197-196.dsl.akrnoh.ameritech.net) |
| 19:21.52 | mrhyd3` | anyone have a resource on getting a Cisco 7971 using SIP or skinny with Asterisk besides voip-info? |
| 19:22.36 | *** join/#asterisk nortex (n=nortex@adsl-69-153-190-164.dsl.lbcktx.swbell.net) |
| 19:22.54 | *** join/#asterisk gcarrillog (n=gcarrill@201.152.19.192) |
| 19:22.56 | gcarrillog | hi |
| 19:23.08 | gcarrillog | im mexica, my english isnt good |
| 19:23.24 | gcarrillog | i have some questions |
| 19:23.36 | nortex | I need some advice on H323 connections and trunking them to Nortel BCM. Can some one help? |
| 19:24.00 | gcarrillog | someone know if exist a program like "destar" for FreeBSD? |
| 19:33.01 | mrhyd3` | loader.c:258 ast_load_resource: /usr/lib/asterisk/modules/chan_sccp.so: undefined symbol: ast_moh_stop I'm running * 1.9, any ideas? |
| 19:33.17 | Qwell | 1.9? |
| 19:33.35 | mrhyd3` | Yes |
| 19:33.40 | Qwell | No such thing |
| 19:34.17 | mrhyd3` | sorry, 1.09 |
| 19:34.35 | Qwell | either upgrade, fix chan_sccp, or don't run it |
| 19:34.58 | Qwell | or bitch to the sccp-users mailing list |
| 19:35.57 | memic | hi, is there a good open source config gui for asterisk? |
| 19:36.09 | Qwell | memic: vi |
| 19:36.15 | memic | A%) |
| 19:36.27 | macTijn | :> |
| 19:36.49 | gcarrillog | lol |
| 19:37.17 | gcarrillog | memic destar |
| 19:37.49 | memic | gcarrillog is it good? |
| 19:38.16 | Qwell | no such thing as a good config gui |
| 19:38.26 | gcarrillog | ok |
| 19:38.33 | Qwell | mutually exclusive |
| 19:39.21 | memic | yea yea |
| 19:39.42 | *** join/#asterisk Egonis (n=Egonis@CPE0004ac23d6ea-CM00407b87dc7b.cpe.net.cable.rogers.com) |
| 19:40.25 | Egonis | I am just starting to build a third Asterisk PBX, but I want to try /dev/dsp based paging, i.e. call the extension, and it plays over /dev/dsp to a set of warehouse speakers. Is this hard to do? |
| 19:41.04 | *** part/#asterisk Egonis (n=Egonis@CPE0004ac23d6ea-CM00407b87dc7b.cpe.net.cable.rogers.com) |
| 19:41.26 | keyhack | Egonis: Wouldn't you use EAGI? |
| 19:41.27 | keyhack | oh |
| 19:41.28 | keyhack | hes gone |
| 19:46.31 | *** join/#asterisk key2 (n=key2@gob75-2-81-56-64-17.fbx.proxad.net) |
| 19:50.34 | *** join/#asterisk adorah (n=FreePBX6@84.94.122.50.cable.012.net.il) |
| 19:51.10 | *** join/#asterisk gmaruz1 (n=gmaruzz@217-133-80-112.b2b.tiscali.it) |
| 19:52.22 | *** part/#asterisk file (i=jcolp@216.237.114.82) |
| 19:55.07 | *** join/#asterisk file (i=jcolp@216.237.114.82) |
| 19:55.59 | xachen | hrm |
| 19:56.14 | file | home = good |
| 19:56.20 | SkramX | :) |
| 19:56.41 | xachen | lamer. you use a bouncer with no vhost??~?!?! |
| 19:56.50 | xachen | :p |
| 19:57.07 | *** join/#asterisk batphone (n=bugz@cpe-70-123-126-15.houston.res.rr.com) |
| 19:57.35 | SkramX | xachen: ptr == vhost, no/ |
| 19:57.36 | SkramX | *? |
| 19:57.49 | batphone | so now we are trying to have a way for users to delete voicemail to email from the pbx |
| 19:57.54 | SkramX | I just dont see why people call it different things. its a ptr record which causes reverse dns |
| 19:57.57 | xachen | SKramX: you use a PTR to make a vhost |
| 19:58.08 | SkramX | okay, then vhost == rDNS |
| 19:58.11 | xachen | indeed |
| 19:58.11 | SkramX | My bad. |
| 19:58.18 | batphone | im thinking of a reply the user can send to the voicemail to email message with DELETE as the first line of the msg |
| 19:58.21 | xachen | I like PTR'd records :) |
| 19:58.26 | SkramX | :) |
| 19:58.30 | batphone | but then we'd have to create a unique id for the message in the subject |
| 19:58.37 | SkramX | batphone: doable. Just make a script and pipe the emails to it. |
| 19:58.46 | SkramX | I have never done such, but would be useful, indeed. |
| 19:59.02 | batphone | where would this script fit into the asterisk configs? |
| 19:59.08 | batphone | my question is about that exactly |
| 19:59.12 | tzafrir_laptop | batphone, you'll have to add some token to that message so you'll have some confidence that it refers to a specific message |
| 19:59.38 | batphone | tzafrir_laptop: i was thinking of a hash of its time and wav file chksum |
| 19:59.40 | tzafrir_laptop | So a random joe attacker won't easily delete all of your voicemail |
| 19:59.43 | batphone | storable in a database |
| 19:59.56 | batphone | 2 fields, one is the has, one is the path to the message |
| 20:00.07 | batphone | this could be useful with a web client as well |
| 20:00.18 | batphone | which is something we are all constantly working on anyway |
| 20:00.30 | batphone | since asterisk does not store messages in a very useful format to either the admin or the user |
| 20:00.48 | batphone | only for its purposes are those msg0003.wav files any good |
| 20:00.59 | batphone | especially when you have a thousand phones |
| 20:01.34 | wunderkin | get rid of 999 |
| 20:02.51 | doolph | what do you guys suggest for a callshop |
| 20:03.49 | batphone | <PROTECTED> |
| 20:03.53 | batphone | app_voicemail.c |
| 20:03.54 | batphone | yay! |
| 20:04.40 | *** join/#asterisk sevard (i=sev@merrill-49-29.resnet.ucsc.edu) |
| 20:06.17 | *** join/#asterisk SexyKen (n=Ken@c-24-5-129-114.hsd1.ca.comcast.net) |
| 20:06.20 | SexyKen | Hello. |
| 20:06.24 | SexyKen | Anyone here use Teliax? |
| 20:07.18 | sevard | I do |
| 20:07.25 | batphone | yeah. |
| 20:07.29 | SexyKen | Do you by any chance CoLo with them too? |
| 20:07.34 | sevard | no |
| 20:07.37 | batphone | haha no way |
| 20:07.42 | sevard | I didn't think they offered colo |
| 20:07.44 | batphone | i cant even get them to port a number correctly |
| 20:07.47 | SexyKen | I've got a box over there and it's been offline for 5 hours |
| 20:07.55 | SexyKen | I've called David about 20 times. |
| 20:07.59 | sevard | wtf |
| 20:08.02 | SexyKen | They have no support available right now. |
| 20:08.07 | sevard | dave is really good about that stuff |
| 20:08.10 | *** join/#asterisk xai (n=pasta@about/networking/0.0.0.0/xai) |
| 20:08.17 | SexyKen | He isn't answering his phone. |
| 20:08.17 | sevard | that sucks, ken |
| 20:08.20 | SexyKen | I can't access my KVM |
| 20:08.30 | sevard | is he supposed to be on call 24/7? |
| 20:08.33 | sevard | it is saturday |
| 20:08.41 | SexyKen | I entered a support ticket that's been unanswered for hours. |
| 20:08.45 | dlynes | sevard: you would think for a colo |
| 20:08.50 | xai | anyone aware of a * design diagram, uml, or dia ? |
| 20:08.51 | dlynes | sevard: i wouldn't colo anywhere, without 24/7 |
| 20:09.03 | sevard | dlynes: sure, but I didn't know of dave himself was oncall |
| 20:09.12 | SexyKen | I called RockyNet -- they say that TelIAX has their stuff unmanaged and do it all themselves |
| 20:09.14 | sevard | SexyKen: you hvae one of those awesome KVM/IP boxne? |
| 20:09.16 | SexyKen | This is wack. |
| 20:09.37 | SexyKen | My server over there is attached to a hp kvm, yes. |
| 20:09.48 | SexyKen | But what good does it do me now...not a darn thing. |
| 20:10.03 | SexyKen | Dave himself told me that if I ever have a problem, to call him, yes. |
| 20:10.08 | sevard | Maybe their service provider is experiencing a router issue |
| 20:10.16 | SexyKen | Nope, RockyNet is the provider |
| 20:10.20 | SexyKen | And they said they're having no issues. |
| 20:10.21 | sevard | really |
| 20:10.26 | *** join/#asterisk Egonis (n=Egonis@CPE0004ac23d6ea-CM00407b87dc7b.cpe.net.cable.rogers.com) |
| 20:10.38 | SexyKen | And the server just became inaccessible this morning....makes no sense. |
| 20:10.41 | sevard | that sucks bro, if i worked for teliax i'd be that shit like penis on your mom |
| 20:11.06 | SexyKen | sevard -- what? |
| 20:11.09 | sevard | what? |
| 20:11.27 | SexyKen | Rephrase? |
| 20:11.34 | sevard | I'd be there for you, baby. |
| 20:12.13 | sevard | SexyKen: you should colo with me I'll host it in _space_ |
| 20:12.30 | batphone | haha |
| 20:13.04 | sevard | by space i mean off of stolen wireless connections.. but we can pretend it's in spacccccce |
| 20:13.19 | batphone | nice |
| 20:13.33 | batphone | i take it you have an uptime guarantee then |
| 20:13.53 | sevard | not responsible for solar flares |
| 20:14.47 | sevard | asteroids/space mutants/people finding their wifi is being stolen/space mishaps |
| 20:15.21 | sevard | other than that 9.99% guarenteed uptime |
| 20:15.45 | Qwell | 5 9's? |
| 20:15.48 | Qwell | 9.9999% |
| 20:15.56 | batphone | im not looking forward to writing something for courier to parse message replys |
| 20:16.02 | sevard | 9.999999999999999% YOU'RE TAKEN CARE OF |
| 20:16.20 | Qwell | I wonder... |
| 20:16.27 | batphone | i installed linux on my neighbors wrt54g |
| 20:16.35 | Qwell | If a company guarantees 5 9's...can you sue them if they get 6 9's? |
| 20:16.37 | batphone | they moved out |
| 20:16.37 | sevard | batphone: hahaha |
| 20:16.57 | file | Qwelllllllllllllllllll |
| 20:17.02 | Qwell | filllleeeee |
| 20:17.05 | batphone | Qwell: hel0p |
| 20:17.05 | Qwell | hello |
| 20:17.05 | file | hi1 |
| 20:17.12 | Qwell | typo city |
| 20:17.20 | batphone | this whole unified messaging thing is killing me |
| 20:17.27 | *** join/#asterisk swytch (n=ezcall@d80-170-39-248.cust.tele2.fr) |
| 20:17.35 | jsaunders | 9.99% gauranteed uptime? Heheh. What's terrible. |
| 20:17.41 | jsaunders | s/What's/That's/ |
| 20:17.42 | sevard | batphone: you should have built a fake linksys page that doesn't upgrade and pretends for forward ports |
| 20:17.42 | batphone | im in need of a way for users to delete voicemails from the pbx once they have recieved the voicemail to email message |
| 20:18.01 | batphone | sevard: now that would be fun |
| 20:18.16 | sevard | batphone: rewrite app_voicemail.c, dlynes will help you |
| 20:18.19 | sevard | <PROTECTED> |
| 20:18.34 | batphone | right now there are 2 options |
| 20:18.47 | batphone | write a mailcmd wrapper and hack courier |
| 20:18.59 | batphone | or write a damn web client |
| 20:19.14 | batphone | or both |
| 20:19.20 | dlynes | sevard: lol...actually i am working on app_voicemail2.c |
| 20:19.33 | sevard | dlynes: do you have a project page |
| 20:19.38 | batphone | dlynes: whats it do different? |
| 20:19.41 | dlynes | sevard: nope |
| 20:19.47 | sevard | get a project page, son |
| 20:19.57 | dlynes | batphone: Mostly cleaning up the code, but I'm going to add native smtp support, too |
| 20:20.06 | dlynes | batphone: not to mention a lot more commenting of the code |
| 20:20.07 | batphone | sevard will host it for you on my neighbors linksys |
| 20:20.27 | sevard | you have guarenteed uptime |
| 20:20.30 | batphone | dlynes that native smtp support is going to kick major ass |
| 20:20.44 | sevard | (subeject to tos) |
| 20:20.49 | Qwell | If you break odbc_storage, I'm kicking your ass :P |
| 20:20.57 | dlynes | batphone: and someone's already written a patch to move the adsi support out of app_voicemail |
| 20:21.00 | dlynes | Qwell: lol |
| 20:21.04 | batphone | but it wont be able to recieve replys if there is another mailer running |
| 20:21.25 | dlynes | Qwell: Kevin asked me to write it as app_voicemail2, so it'll be a complete rewrite |
| 20:21.35 | Qwell | I stand by what I said :P |
| 20:21.38 | sevard | dlynes: project page, son1 |
| 20:21.44 | dlynes | heh |
| 20:22.00 | batphone | dlynes: host it here: 192.169.0.1 username: admin password: admin |
| 20:22.01 | dlynes | Anyways...I will be installing odbc on this system that i'm going to be testing it on |
| 20:22.08 | dlynes | it'll be conversing with postgresql |
| 20:22.40 | dlynes | batphone: since when does asterisk need to receive connections on port 25? |
| 20:23.07 | dlynes | batphone: i'm only going to implement a native smtp client in asterisk, not a whole damned mail server |
| 20:23.10 | [Airwolf] | Can someone tell me if I only need to install asterisk-addons for h323 support or do I need something else to get it running ? |
| 20:23.29 | dlynes | batphone: if you want a mail server, try postfix, qmail, or sendmail |
| 20:23.46 | [Airwolf] | Because the documentation isn't quite clear about this |
| 20:23.51 | dlynes | [Airwolf]: you need to install pwlib, openldap, gdl, and openh323 |
| 20:23.52 | ManxPower | [Airwolf], the H232 stuff in asterisk-addons SHOULD have a README file. |
| 20:23.52 | batphone | dlynes: i have to use courier >( |
| 20:24.01 | ManxPower | dlynes, no, he does not. |
| 20:24.15 | dlynes | ManxPower: openldap and gdl are prerequisites for pwlib |
| 20:24.16 | ManxPower | the H323 stuff in Asterisk Addons uses it's own H323 stuff, IIRC |
| 20:24.21 | batphone | dlynes: im just looking for a way for users to delete voicemails after they got the email if they want |
| 20:24.27 | dlynes | ManxPower: oh...no more need for openh323? |
| 20:24.30 | batphone | without having to work very hard ;P |
| 20:24.59 | ManxPower | dlynes, there are at least FOUR H323 channel drivers for Asterisk. The one in asterisk-addons never needed openH323 |
| 20:25.16 | *** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239) |
| 20:25.32 | [Airwolf] | ManxPower, I was reading the wiki. |
| 20:25.55 | dlynes | ManxPower: ah...i just know about chan_oh323(?) (requires it), chan_ooh323 (requires it), chan_h323 (wasn't sure...written by jerjer), and woomera (not sure) |
| 20:25.58 | ManxPower | [Airwolf], you know the wiki is frequently wrong and always outdated. |
| 20:26.17 | ManxPower | which is why the source code has readmes |
| 20:26.25 | dlynes | And the most recent posting I'd seen about JerJer's code was in 2003 or something |
| 20:26.28 | [Airwolf] | ManxPower, I noticed, but it gets me started anyway. |
| 20:26.32 | dlynes | so i figured it'd gone dormant |
| 20:26.56 | [Airwolf] | So basicly I should just install asterisk addons and I should be able to user h323. |
| 20:27.05 | [Airwolf] | Without installing openh323 |
| 20:27.25 | [Airwolf] | Because all the channel drivers confuses me. :) |
| 20:29.23 | ManxPower | [Airwolf], Asterisk addons is a COLLECTION of various software for Asterisk that cannot be part of the main asterisk source tree because of licensing reasons. |
| 20:29.37 | ManxPower | you don't "install asterisk addons" |
| 20:29.47 | dlynes | ManxPower: Will that openh323 channel driver connect to a gatekeeper, or will it only connect to openh323 phones? |
| 20:30.07 | dlynes | erm h323 phones i mean |
| 20:30.34 | ManxPower | i don't know. try it and see |
| 20:30.50 | batphone | [blocks B ] mail-mta/courier (is blocking mail-filter/maildrop-2.0.1) |
| 20:30.51 | ManxPower | only people with NO OTHER CHOICE use H323 |
| 20:30.57 | batphone | wtf man... |
| 20:30.57 | Qwell | file: Right Said Fred - I'm too sexy? |
| 20:31.04 | file | Qwell: good idea |
| 20:31.05 | batphone | so you cant use courier with maildrop on gentoo |
| 20:31.05 | [TK]D-Fender | file " Basket Case" |
| 20:31.07 | Qwell | heh |
| 20:31.11 | sevard | file: you and me baby we ain't nothing but mammels |
| 20:31.20 | [Airwolf] | ManxPower, the pbx already there supports only h323 |
| 20:31.27 | file | sevard: that's bloodhoung gang, partial lyrics... not the actual song title there :P |
| 20:31.27 | [Airwolf] | So, I have no choice |
| 20:31.27 | [Airwolf] | :P |
| 20:31.35 | sevard | file: duhhhhhh\ |
| 20:31.48 | sevard | file: i was just singing to you |
| 20:31.52 | sevard | or suggesting... |
| 20:31.55 | Qwell | wtf |
| 20:31.55 | file | bills bills bills, bills to pay |
| 20:32.11 | [TK]D-Fender | file : "Money For Nothing" ? |
| 20:32.34 | file | you people are gr8t |
| 20:32.40 | ManxPower | "Tainted Love"? |
| 20:33.14 | file | I suppose I should update this debian unstable install |
| 20:33.45 | jsaunders | Anyone know how to decrease dial delay in dial plan for Sipura ata's? I see S0 to dial immediately... but I'm wondering if there is a default delay value for the dial plan. |
| 20:34.02 | jsaunders | SPA-1001 & SPA-2002 |
| 20:35.23 | *** join/#asterisk Johnnie (n=jdlewis@pdpc/supporter/active/Johnnie) |
| 20:36.07 | *** join/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 20:39.21 | ManxPower | jsaunders, There is no delay in Asterisk for SIP dialed calls. Your problem is in the SIPura dialplan |
| 20:39.48 | file | dejavu |
| 20:40.22 | jsaunders | What makes you think I was refering to asterisk? I said "in dial plan for Spira ata's" did I not? |
| 20:40.33 | jsaunders | s/Spira/Sipura/ |
| 20:40.50 | sevard | that he did |
| 20:41.04 | ManxPower | jsaunders, none of my SIPuras have delays when dialing. |
| 20:41.18 | sevard | jsaunders: are you talking about live-dial? |
| 20:41.24 | jsaunders | I'll show ya the dial plan that came stock. |
| 20:41.32 | *** join/#asterisk linuxtuxie (n=kku@21.165-201-80.adsl.skynet.be) |
| 20:41.33 | ManxPower | but I hand crafted my SIPura dialplan one bit at a time, using bits I grew and harvested myself |
| 20:41.34 | jsaunders | (*xx|[3469]11|0|00|[2-9]xxxxxx|1xxx[2-9]xxxxxxS0|xxxxxxxxxxxx.) |
| 20:41.42 | sevard | like the amount of time it takes to request a call after you've finished dialing? |
| 20:41.43 | jsaunders | I should prolly do the same. |
| 20:41.55 | ManxPower | jsaunders, the dialplan that comes stock of course has delays since they have no idea what you need to disl |
| 20:42.09 | jsaunders | Makes sense, I'll build my own. Thanks for the suggestion. :D |
| 20:42.19 | ManxPower | I assumed you built your own dialplan |
| 20:42.22 | sevard | sipura documentation is the crap. |
| 20:42.27 | jsaunders | Negatory, just got these things yesterday. |
| 20:42.41 | jsaunders | Documentation may be crap but I certainly enjoy the product. |
| 20:44.04 | ManxPower | (9,1[2-9]xx[2-9]xxxxxx|9,[2-9]xxxxxx|[2-8]xxx) |
| 20:44.17 | *** join/#asterisk salviadud (n=dude@dsl-201-129-86-188.prod-infinitum.com.mx) |
| 20:44.23 | salviadud | !seen _paulo_ |
| 20:45.26 | linuxtuxie | Hello all, I have a problem connecting to sip.voipstunt.org. |
| 20:45.26 | linuxtuxie | I am running asterisk 1.2.7.1 on a debian behind a debian firewall which is running siproxd |
| 20:45.26 | linuxtuxie | If I enable siproxd debugging I see "SIP/2.0 401 Unauthorized". I suspect something is wrong in my sip.conf....but I cannot figure out what it could be..can anybody help ? |
| 20:47.43 | Egonis | linuxtuxie: there is no NXDOMAIN for that address |
| 20:48.50 | linuxtuxie | oops..sorry..my mistake, a amde a type during my explanation..it is sip.voipstunt.com |
| 20:50.06 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219) |
| 20:50.46 | linuxtuxie | in asterisk I get: |
| 20:50.46 | linuxtuxie | homeserver*CLI> sip show registry |
| 20:50.46 | linuxtuxie | Host Username Refresh State |
| 20:50.46 | linuxtuxie | sip.voipstunt.com:5060 myusername 120 Request Sent |
| 20:50.47 | linuxtuxie | May 13 22:50:16 NOTICE[1600]: chan_sip.c:5289 sip_reg_timeout: -- Registration for 'myusername@sip.voipstunt.com' timed out, trying again (Attempt #37) |
| 20:51.52 | linuxtuxie | in the siproxd debug info I see the registration request leaving towards sip.voipstunt.com...and I get a 401 back |
| 20:52.05 | linuxtuxie | the username/password however are correct |
| 20:52.18 | linuxtuxie | because I can use the m$ client |
| 20:53.10 | *** join/#asterisk XeSolutions (n=Steve@68.225.80.195) |
| 20:53.14 | *** join/#asterisk Johnnie (n=jdlewis@pdpc/supporter/active/Johnnie) |
| 20:53.53 | XeSolutions | Can someone help me? I just installed Asterisk per the instructions yet when I go to the web address the GUI wont come up I just get a page not found error. |
| 20:54.24 | jsaunders | asterisk has built in gui admin? news to me. |
| 20:54.37 | XeSolutions | well its asterisk@home |
| 20:54.40 | jsaunders | ;) |
| 20:54.46 | [TK]D-Fender | XeSolutions : Please read the channel topic... |
| 20:54.49 | Qwell | "per the instructions"...HA |
| 20:54.53 | jsaunders | Clarity = good |
| 20:54.53 | Qwell | Step 1: Insert CD. |
| 20:54.56 | Qwell | Step 2: Sit there |
| 20:55.27 | Qwell | s/Sit there/Sit there and watch it blow away anything on your drive(s) without asking./ |
| 20:55.36 | jsaunders | heheh |
| 20:56.54 | [TK]D-Fender | Actually in a sense * does have a built-in gui.. just not pixel-based or in HTTP :) "asterisk -r" ... the ultimate GUI! |
| 20:57.09 | Qwell | load mod_gtk.so |
| 20:57.10 | Qwell | :D |
| 20:57.18 | jsaunders | Umm... do I need to give you a link to the acronym gui? |
| 20:57.19 | jsaunders | :) |
| 20:57.44 | Qwell | phpshell |
| 20:58.17 | [TK]D-Fender | jsaunders : With all the ascii pr0n going around channel these days, I'd like to expand on that definition a bit :) |
| 20:58.31 | jsaunders | Ranks right up there w/ pernouncing it 'sequel'. *shudders* |
| 20:58.39 | jsaunders | Mmm, ascii pr0n. |
| 20:58.44 | *** join/#asterisk gunsch (n=linux@pD954064B.dip0.t-ipconnect.de) |
| 20:58.48 | jsaunders | Some rather talented individuals out there. |
| 20:58.50 | Qwell | jpeg's are better :P |
| 20:58.57 | Qwell | ...or mpeg |
| 20:59.20 | linuxtuxie | has anybody got an idea what could be checked/verified/tested concerning with my sip problem ? |
| 20:59.45 | Qwell | linuxtuxie: check your firewall |
| 20:59.48 | Qwell | 5060 udp |
| 20:59.56 | linuxtuxie | already done that.. |
| 21:00.03 | linuxtuxie | iptable rules are correct |
| 21:00.28 | linuxtuxie | already dropped my complete fw and only used the 3 bare fw rules over |
| 21:01.21 | salviadud | guys, i'm trying to setup a .call file using voipbuster, but it gives me a call failed to go through, reason 0 |
| 21:01.28 | salviadud | here's my code http://pastebin.com/715884 |
| 21:02.04 | salviadud | i'm wondering if it's even possible to do that |
| 21:02.05 | linuxtuxie | meaning: |
| 21:02.06 | linuxtuxie | $IPTABLES -A PREROUTING -t nat -i $INT_ETH -p udp --dport 5060 -j REDIRECT |
| 21:02.06 | linuxtuxie | $IPTABLES -A INPUT -p UDP -s 0/0 --dport 5060 -j ACCEPT |
| 21:02.06 | linuxtuxie | $IPTABLES -A INPUT -p UDP -s 0/0 --dport 7070:7080 -j ACCEPT |
| 21:02.37 | *** join/#asterisk lzhang (n=rjrae@67.95.13.46) |
| 21:03.25 | salviadud | would it have anything to do with authentication, as in username and password? |
| 21:04.41 | linuxtuxie | according to me not...because if I use the same username/password in the windows client provided by voipstunt.com it works |
| 21:04.42 | robl^ | what about your RTP streams? nothing allowing those? |
| 21:05.20 | linuxtuxie | which ports are those ? |
| 21:05.28 | salviadud | somebody, throw me frriggin' bone here! |
| 21:05.40 | *** part/#asterisk gunsch (n=linux@pD954064B.dip0.t-ipconnect.de) |
| 21:07.23 | linuxtuxie | In fact I used the recommendated instructions from siproxd which can be found at http://siproxd.sourceforge.net/siproxd_guide/siproxd_guide_c6s5.html |
| 21:08.19 | lzhang | hey, I just recompiled asterisk, and when I do an 'asterisk -vvvc' it hangs at 'Asterisk Event Logger Started', right before it should be beginning to load the modules. Is it possible that someone could help me find why this is happening? |
| 21:10.00 | *** join/#asterisk r0d3nt (n=RatMan@foster.stonedcoder.org) |
| 21:12.36 | doolph | anyone can recomend a callshop software |
| 21:14.03 | jsaunders | Aha! Sipura dial plans... "The Interdigit_Long_Timer specifies the default maximum time (in seconds) allowed between dialed digits, when no candidate digit sequence is as yet complete." & "The Interdigit_Short_Timer specifies the default maximum time (in seconds) allowed between dialed digits, when at least one candidate digit sequence is complete as dialed". |
| 21:14.29 | jsaunders | default 10 & 3 respectively. |
| 21:20.32 | *** join/#asterisk CoffeeIV (i=rgr@cpe-70-112-100-20.austin.res.rr.com) |
| 21:22.28 | *** join/#asterisk Johnsie (n=jdlewis@pdpc/supporter/active/Johnnie) |
| 21:26.28 | ManxPower | If anyone is good at shell scripts and wants help me, take a look at this: http://pastebin.ca/55586 |
| 21:27.01 | ManxPower | It TOTALLY fails if there are any spaces in the file names. |
| 21:27.07 | TheBigSpark | I am having trouble getting BLF to work on a grandstream gxp-2000, can anyone help |
| 21:28.28 | linuxtuxie | ManxPower: You can use the -exec option with find |
| 21:28.42 | salviadud | i hate the fxo port on my sipura |
| 21:28.54 | ManxPower | linuxtuxie, Yes, but this will be part of a MUCH larger script. |
| 21:28.55 | salviadud | i can't crank the volume up |
| 21:29.03 | ManxPower | salviadud, OF COURSE YOU CAN |
| 21:30.09 | linuxtuxie | bash does not like spaces in a for |
| 21:30.19 | linuxtuxie | it will split everyting on a space |
| 21:30.24 | ManxPower | linuxtuxie, I know, which is why I'm escaping them using sed |
| 21:30.58 | ManxPower | Maybe I can tr all " " to ^, then undo that later |
| 21:31.04 | Qwell | nah |
| 21:31.08 | Qwell | ManxPower: change $IFS |
| 21:31.15 | Qwell | field seperator |
| 21:31.28 | linuxtuxie | indeed..a space needs to be avoided |
| 21:31.36 | salviadud | really? |
| 21:31.47 | salviadud | is it the pstn to spa gain option? |
| 21:31.50 | linuxtuxie | you need to use a substitute for the space like | |
| 21:32.12 | linuxtuxie | and then later replace the | again by a space |
| 21:32.14 | salviadud | ManxPower, if I dial out with my sipura, the other party can't hear me very well |
| 21:32.19 | ManxPower | IFS=\n |
| 21:32.29 | ManxPower | salviadud, well, do into the SIPura config and up the gain on the FXO |
| 21:32.31 | Qwell | ManxPower: just empty string works, I think |
| 21:32.59 | ManxPower | Qwell, if you were closer I'd kiss you |
| 21:33.01 | salviadud | ManxPower, will do |
| 21:33.28 | swytch | ManxPower: it fails because your SPACE character separates the elements in the list to "for .. in .." |
| 21:33.31 | Qwell | ManxPower: Just the thought disturbs me... |
| 21:33.45 | ManxPower | Qwell, 8-) |
| 21:34.07 | ManxPower | swytch, and yet escaping spaces with \ does no good at all |
| 21:35.21 | swytch | ManxPower: check what this does: files='"a b.c"';for file in $files;do echo $file;done |
| 21:36.40 | ManxPower | swytch, files="Happy\ Days Fred\ Wilma" |
| 21:36.44 | ManxPower | that does no good at all. |
| 21:36.52 | ManxPower | Qwells suggestion works just great. |
| 21:37.13 | robin_sz | so .. echos |
| 21:37.16 | linuxtuxie | I have pasted my sip.conf at http://pastebin.ca/55588 maybe someone has an idea related to the voipstunt registration problem |
| 21:37.35 | robin_sz | I think what I have os not echo, but infact, echo cancellation being applied worngly |
| 21:37.35 | swytch | ManxPower: i guess the "for" in the shell just takes the backslash as a normal char. the IFS (default to space) separates the other chars (including the backslash. |
| 21:37.49 | robin_sz | so what generated the echo cancellation? asterisk? |
| 21:37.53 | robin_sz | or ther phone? |
| 21:38.25 | ManxPower | I wonder if files="Happy\\ Days Fred\\ Wilma" would have worked |
| 21:40.10 | robin_sz | the other problem I have is ocassional "one sided" calls ... we cant hear them, they can hear us |
| 21:42.14 | ManxPower | robin_sz, echo is caused by 2-wire circuits |
| 21:42.41 | ManxPower | If you use SIP phones, PRI with Asterisk, you still can get echo when you call analog phones |
| 21:42.48 | *** join/#asterisk faberk (n=faberk@host54-228.pool80181.interbusiness.it) |
| 21:43.17 | *** join/#asterisk catch23 (n=catch23@c-67-191-252-3.hsd1.ga.comcast.net) |
| 21:44.14 | robin_sz | ManxPower, so, we have ISDN lines ... we get lots of echo when dialling out to local numbers. using an "normal" phone onthe isdn port, we dont get echos .. just with asterisk |
| 21:44.46 | *** join/#asterisk PMantis (n=pmantis@cpe-66-66-115-197.rochester.res.rr.com) |
| 21:44.48 | ManxPower | robin_sz, that is not correct. You get echo, you just can't hear it because the latency is so low, you hear the echo as "sidetone" |
| 21:45.01 | robin_sz | right .. that true enough |
| 21:45.08 | ManxPower | if you could add 100ms of latency to those calls you would hear it. |
| 21:45.21 | robin_sz | so where is the long latnecy suddenly appearing from then? |
| 21:45.29 | robin_sz | some local calls are very pooor |
| 21:45.39 | ManxPower | so the problem, of course, is because VoIP's latency is high enough for you to hear any echo. |
| 21:46.12 | ManxPower | That's just the way VoIP works. You need good echo cancelation in your PSTN/VoIP gateway, and as you can hear, Asterisk's EC is not all that great. |
| 21:46.26 | robin_sz | but we dont use a VOIP provider |
| 21:46.47 | robin_sz | just snoms->asterisk->isdn_to_pstn |
| 21:46.52 | ManxPower | I never said you did. |
| 21:47.07 | PMantis | Can someple please help me to rewrite this line, to avoid the error? |
| 21:47.11 | PMantis | exten => s,n,GotoIf($[${CALLERIDNUM}z = 1${CALLERIDNAME}z]?dosetcid:) |
| 21:47.11 | PMantis | syntax error, unexpected TOKEN, expecting $end; Input: 15851111111z = 1Cell Phone NYz |
| 21:47.12 | ManxPower | yes, the SNOM->Asterisk is SIP and that is VoIP and that has latency. |
| 21:47.21 | robin_sz | OK, |
| 21:47.30 | robin_sz | so mostly it is fine .. |
| 21:47.48 | robin_sz | b ut say, 10% of local calls have huge echo (high latnecy?) |
| 21:47.56 | PMantis | eww... s/someple/someone/ |
| 21:48.10 | ManxPower | robin_sz, prolly that %10 of calls are what Asterisk's EC cannot deal with |
| 21:48.24 | robin_sz | the CEO is talking about dumping the whole thing and going back to a "normal" PBX ... :( |
| 21:48.32 | ManxPower | robin_sz, set echocancel=off sometime and you'll hear that most calls to POTS lines have echo |
| 21:48.40 | [TK]D-Fender | PMantis : you on * 1.0.x? |
| 21:49.06 | robin_sz | right ... |
| 21:49.11 | ManxPower | robin_sz, if you were on a PRI or CT1 CE1 I would say "get tellabs" |
| 21:49.13 | *** join/#asterisk chaoscon (n=ph33r@smartserv/ceo/chaoscon) |
| 21:49.16 | PMantis | [TK]D-Fender, No, 1.2.7.1 |
| 21:49.21 | ManxPower | robin_sz, try the KB2 EC |
| 21:49.27 | robin_sz | KB2? |
| 21:49.43 | PMantis | [TK]D-Fender, It's cacking on the begining of "Phone" |
| 21:49.53 | [TK]D-Fender | PMantis : exten => s,n,GotoIf($["${CALLERID(num)}" = "1${CALLERID(name)}"]?dosetcid) |
| 21:50.12 | PMantis | [TK]D-Fender, Quotes? That simple?? Cool! |
| 21:50.40 | [TK]D-Fender | PMantis : You really wanted that "1" prefix in the comparison? |
| 21:51.04 | PMantis | yes |
| 21:51.17 | ManxPower | Correct. KB2 is only available in zaptel trunk, the version of zaptel trunk dated around Feb 19 works with Asterisk 1.2.x |
| 21:51.22 | [TK]D-Fender | PMantis : well there you have it then. |
| 21:51.37 | PMantis | [TK]D-Fender, That's only one of the comparison lines. I compare both with and without the 1, and a another combination. |
| 21:51.41 | PMantis | [TK]D-Fender, Thanks! |
| 21:51.57 | [TK]D-Fender | PMantis : np, and use as a templet for your others. |
| 21:52.01 | [TK]D-Fender | template* |
| 21:52.05 | linuxtuxie | has someone got any ideas concerning my sip proxy/registration problem...or does someone have a better solution for siproxd ? |
| 21:52.49 | robin_sz | ManxPower, im using the CAPI interface to my interface card, so I dont think zap comes into it |
| 21:53.05 | ManxPower | robin_sz, it sucks to be you |
| 21:53.16 | robin_sz | tell that to my wife |
| 21:53.19 | ManxPower | robin_sz, I cannot help you at all with PRI |
| 21:53.20 | camelon | where I can find the last "how to" that give me the fresh asterisk grammar? |
| 21:53.22 | ManxPower | ..er..BRI |
| 21:53.34 | robin_sz | I have: echocancelold=yes |
| 21:54.12 | PMantis | May 13 17:51:59 WARNING[21181]: ast_expr2.y:843 op_div: non-numeric argument |
| 21:54.17 | robin_sz | erm, I also seem to have a ; commented out echcancel=yes line |
| 21:54.18 | PMantis | Hmmm, what'st hat all about? :) |
| 21:55.02 | [TK]D-Fender | PMantis : paste the source line that caused it |
| 21:55.34 | PMantis | [TK]D-Fender, Ahh, I think I need quoted here, too.. |
| 21:55.38 | robin_sz | hmmm |
| 21:55.41 | *** join/#asterisk Shoragan (n=shoragan@134.169.175.72) |
| 21:55.43 | PMantis | [TK]D-Fender, CLI shows: extn-ARG2=Zap/2&IAX2/SteveMonsoon&IAX2/sbsiaxy01&SIP/linksys-pap-l2 |
| 21:55.46 | PMantis | [TK]D-Fender, from a setvar |
| 21:56.32 | batphone | telechoice rocks! |
| 21:56.33 | [TK]D-Fender | PMantis : ate both in full |
| 21:57.10 | PMantis | [TK]D-Fender, exten => s,5,SetVar(extn-ARG2=${ARG2}z) |
| 21:57.52 | PMantis | Hmm, replaced by set |
| 21:58.05 | [TK]D-Fender | PMantis : Setvar is deprecated, and I'm pretty sure dashes in var names is a bad idea... |
| 22:01.30 | hads|home | KB2? I just grepped zaptel trunk and didn't see anything. |
| 22:01.37 | camelon | TK D-Fender . . . where i can find some document to up to date the * code syntaxis? . .no deprecated things |
| 22:02.26 | [TK]D-Fender | camelon : THEBOOK is largely up to date. though use "show applications" from CLI with give you the function list and "show application [x]" its details |
| 22:03.12 | [TK]D-Fender | ~thebook |
| 22:03.13 | jbot | i guess thebook is Asterisk: The Future of Telephony, released under a Creative Commons license and available at http://www.asteriskdocs.org << Read the book online! |
| 22:03.14 | xachen | wierd |
| 22:03.16 | xachen | 1.2.7.1 |
| 22:03.18 | camelon | TK D-Fender . . . thanks |
| 22:03.28 | xachen | I am able to cause MOH play faster by the second |
| 22:03.38 | xachen | despite the fact its playing a class I told not to |
| 22:03.48 | linuxtuxie | Can anybody point me in the right direction connection regarding "SIP/2.0 401 Unauthorized" ? |
| 22:04.09 | PMantis | [TK]D-Fender, Cool.. got rid of the error in my dialplan... thanks |
| 22:05.47 | *** join/#asterisk Cresl1n (n=matt@user-24-236-124-147.knology.net) |
| 22:06.29 | *** join/#asterisk gmaruzz (n=gmaruzz@217-133-80-112.b2b.tiscali.it) |
| 22:08.13 | [TK]D-Fender | PMantis : glad to hear |
| 22:08.36 | robin_sz | OK, I explicitly set echo-cancel to yes, I'll see if that helps |
| 22:09.14 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 22:09.40 | robin_sz | what do the force, g164 and g165 options do? |
| 22:09.50 | [TK]D-Fender | linuxtuxie : bad user/pass |
| 22:10.35 | linuxtuxie | <PROTECTED> |
| 22:10.54 | linuxtuxie | <PROTECTED> |
| 22:11.32 | [TK]D-Fender | linuxtuxie : then its the specific fields that you are filling that don't match. |
| 22:11.50 | linuxtuxie | <PROTECTED> |
| 22:12.22 | [TK]D-Fender | linuxtuxie : Irrelevent. The error means what it says. bad user/pass. You aren't filling out the right field with the right values. |
| 22:13.09 | *** join/#asterisk linuxtuxie (n=kku@21.165-201-80.adsl.skynet.be) |
| 22:13.24 | robin_sz | so .. it seems the Eicon Diva has an inbuilt echo cancellor in the DSP ... |
| 22:13.26 | linuxtuxie | damn gaim crashed |
| 22:13.35 | robin_sz | so * should not be trying to do it, right? |
| 22:13.48 | robin_sz | what does echotail do? |
| 22:13.57 | [TK]D-Fender | [18:12] <[TK]D-Fender> linuxtuxie : Irrelevent. The error means what it says. bad user/pass. You aren't filling out the right field with the right values. |
| 22:13.58 | linuxtuxie | [TK]D-Fender: I have posted my sip.conf at http://pastebin.ca/55588 |
| 22:14.15 | robin_sz | linuxtuxie, great, are your passwords in it? |
| 22:14.23 | linuxtuxie | nope |
| 22:14.52 | [TK]D-Fender | linuxtuxie : Funny I don't see a "sectet" line in your peer entry AT ALL... |
| 22:15.03 | Qwell | [TK]D-Fender: I would hope not |
| 22:15.14 | [TK]D-Fender | Secret* |
| 22:15.19 | [TK]D-Fender | Qwell : Why is that? |
| 22:15.24 | linuxtuxie | see line 18: secret=mypwd |
| 22:15.33 | [TK]D-Fender | I m blind... |
| 22:15.34 | Qwell | because sectet=blah wouldn't work :p |
| 22:15.37 | robin_sz | I have txgain=0.8 and rxgain=0.8 .. spound "reasoanble" ?? |
| 22:16.25 | hads|home | robin_sz: Could be reasonable, have you used ztmonitor? |
| 22:16.34 | robin_sz | nope .. |
| 22:16.37 | linuxtuxie | [TK]D-Fender: Asterisk is printing: |
| 22:16.37 | linuxtuxie | <PROTECTED> |
| 22:16.50 | robin_sz | hads|home, what does that do? |
| 22:17.11 | hads|home | Show's you what levels you are running at. |
| 22:17.18 | robin_sz | hads|home, and will it work on my CAPI devices? |
| 22:17.28 | hads|home | ah... that I do not know sorry. |
| 22:17.33 | linuxtuxie | and if I enable sip debug I see: |
| 22:17.44 | linuxtuxie | Retransmitting #5 (no NAT) to 80.239.235.201:5060: |
| 22:17.44 | linuxtuxie | REGISTER sip:sip.voipstunt.com SIP/2.0 |
| 22:17.44 | linuxtuxie | Via: SIP/2.0/UDP 192.168.200.1:5060;branch=z9hG4bK29c2ec25;rport |
| 22:17.44 | linuxtuxie | From: <sip:linuxtuxie@sip.voipstunt.com>;tag=as55a32b8d |
| 22:17.44 | linuxtuxie | To: <sip:linuxtuxie@sip.voipstunt.com> |
| 22:17.45 | linuxtuxie | Call-ID: 568317f543360eec574524b86ab5912d@192.168.200.1 |
| 22:17.45 | hads|home | I only used zap devices. |
| 22:17.47 | linuxtuxie | CSeq: 400 REGISTER |
| 22:17.49 | linuxtuxie | User-Agent: Asterisk PBX |
| 22:17.51 | linuxtuxie | Max-Forwards: 70 |
| 22:17.53 | linuxtuxie | Expires: 120 |
| 22:17.55 | robin_sz | hads|home, sounds like a zaptel thing to me |
| 22:17.55 | linuxtuxie | Contact: <sip:linuxtuxie@192.168.200.1> |
| 22:17.57 | linuxtuxie | Event: registration |
| 22:17.59 | linuxtuxie | Content-Length: 0 |
| 22:18.27 | [TK]D-Fender | linuxtuxie : You aren't accounting for NAT.... |
| 22:18.36 | [TK]D-Fender | linuxtuxie : look at the contact. |
| 22:18.36 | hads|home | Yeah apologies, too hungover to realise you were using capi. |
| 22:18.43 | robin_sz | np |
| 22:19.04 | linuxtuxie | <PROTECTED> |
| 22:19.22 | robl^ | mmmm... hungover |
| 22:19.37 | linuxtuxie | 192.168.200.1 is indeed my internal ip where asterisk is running on |
| 22:20.26 | linuxtuxie | but this sip packet should be rewritten on the fw by siproxd |
| 22:20.47 | [TK]D-Fender | linuxtuxie : Don't bet on it. |
| 22:21.04 | [TK]D-Fender | linuxtuxie : And sorry I can't help you with siproxd. |
| 22:22.07 | *** join/#asterisk kimosabe (n=kimosabe@dsl-200-78-71-61.prod-infinitum.com.mx) |
| 22:22.11 | *** join/#asterisk adorah (n=FreePBX1@84.94.134.7.cable.012.net.il) |
| 22:22.25 | *** join/#asterisk brockj49464_home (n=chatzill@63.87.56.153) |
| 22:22.40 | linuxtuxie | [TK]D-Fender: Here you can find the debug output from siproxd http://pastebin.ca/55591 |
| 22:22.51 | kimosabe | has any one unlocke the pap2 3.1.9 LSc |
| 22:23.15 | *** join/#asterisk CrummyGummy (n=wayne@dsl-145-79-206.telkomadsl.co.za) |
| 22:23.23 | linuxtuxie | [TK]D-Fender: you see the register packet leave thw fw...and you see the answer 401 coming back |
| 22:25.08 | kimosabe | does anyone know how i can obtain the GPP_K for my pap2 |
| 22:25.49 | [TK]D-Fender | linuxtuxie : Only rael guess is your pass is bad. |
| 22:27.31 | camelon | rael could do that? really? |
| 22:27.36 | linuxtuxie | I'm afraid not :( A colleague of mine is also using voipstunt...but he is running it on his fw, and this seems to be working fine |
| 22:27.55 | linuxtuxie | I mean he is running asterisk on his fw |
| 22:28.52 | linuxtuxie | So I am afraid it has got something to do with the sip proxy...in the beginning asterisk would not even play with siproxd because rvlookup was set to yes... |
| 22:28.55 | [TK]D-Fender | real* |
| 22:29.18 | adorah | voipstunt charges for unanswered calls that's why I stopped using their service |
| 22:29.57 | linuxtuxie | currently I would be happy to get any voip service running ;) |
| 22:30.27 | linuxtuxie | I have just taken voipstunt because a colleague got it runnning on his setup |
| 22:31.15 | adorah | there r a few French providers that u can connect to Asterisk.. |
| 22:31.26 | *** join/#asterisk jujutsuka (n=jujutsuk@68-65-33-193.chvlva.adelphia.net) |
| 22:31.37 | jujutsuka | hello |
| 22:31.57 | linuxtuxie | and do they work through a sip proxy ? |
| 22:32.45 | adorah | some offer sip some support IAX. I sold some of them GSM gateways so I know they r resonable service |
| 22:33.15 | linuxtuxie | iax would indeed be even better :) |
| 22:33.41 | jujutsuka | is it possible to configure asterisk and fwdout to work with a cell phone? |
| 22:35.02 | adorah | <jujutsuka> in what sense? u can connect a GSM gateway to an analog port of an asterisk..it is transparent |
| 22:35.21 | jujutsuka | i have verizon, so i think it's CDMA |
| 22:35.39 | Qwell | verizon and sprint are cdma, yes |
| 22:35.51 | adorah | CDMA all the same..just the device is more expensive |
| 22:36.01 | jujutsuka | i want to convert my free nights and minutes to usable peak minutes for my cell phone, if that's possible |
| 22:36.11 | ManxPower | well transparent if the cellular gateway provides far end disconnect supervision, and transparent still means you have to wait for DTMF to be sent while dialing |
| 22:36.58 | adorah | analog to GSM works much the same as analog to PSTN |
| 22:38.04 | jujutsuka | is what i'm describing possible? i can set up asterisk on my home pc to connect to my PSTN line if necessary. i'm just kind of in over my head and wondering exactly how to go about this |
| 22:38.07 | *** part/#asterisk PMantis (n=pmantis@cpe-66-66-115-197.rochester.res.rr.com) |
| 22:38.24 | *** join/#asterisk RoyK (n=roy@10.80-203-106.nextgentel.com) |
| 22:39.07 | adorah | the gateway at least the ones I used to work with has polarity reversal or battery reversal so they disconnect |
| 22:41.42 | jujutsuka | is it possible to put asterisk on a smartphone, or do i need it running on a PC server? |
| 22:42.06 | kimosabe | can some one help me wth vuxk vonage |
| 22:45.24 | *** join/#asterisk darkskiez (n=mhb@bb-87-81-62-203.ukonline.co.uk) |
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| 22:49.42 | kimosabe | can some one help me with gpp?k |
| 22:49.51 | kimosabe | can some one help me with gpp_k please |
| 22:50.18 | *** join/#asterisk obanta (n=obanta@CPE-24-27-129-176.neb.res.rr.com) |
| 22:57.43 | ManxPower | never hear od gpp_k |
| 23:01.31 | kimosabe | manxpower im tring to conext to a pap2 device with 3.1.9 LSc firmwear i see on line that some people have been able to obtain the passwords with that method |
| 23:01.58 | ManxPower | kimosabe, only if the box was never connected to the internet. |
| 23:02.32 | VoicePulse | kimosabe: There is a HUGE thread about it on dslreports.com -- if there is any useful info on the Internet regarding your question, it will be in that "Unlocking PAP2" thread. |
| 23:02.56 | kimosabe | yes it did conect 2 the internet |
| 23:02.57 | *** join/#asterisk wm4k (n=wm4k@dsl213-218-233-192.as15444.net) |
| 23:04.23 | kimosabe | voicepulse what option at dslreports |
| 23:04.59 | *** join/#asterisk Johnnie (n=jdlewis@pdpc/supporter/active/Johnnie) |
| 23:05.01 | robl^ | once it connects to the interenet it is forever locked to Vonage |
| 23:05.07 | robl^ | (just about) |
| 23:05.26 | kimosabe | robl thats teribble |
| 23:05.40 | VoicePulse | kimosabe: http://www.broadbandreports.com/forum/remark,14450684 Start at the end, it's 100 pages long :b |
| 23:06.06 | kimosabe | i have read there is a way 2 doble reset the box and get it to tftp to ure machine where it will upload a password firle wich will then allow u 2 log in |
| 23:06.15 | kimosabe | thanks |
| 23:07.07 | robl^ | kimosabe: there is.. but only if it is BEFORE it talks to Vonage's servers.. after that, its locked solid |
| 23:07.23 | *** join/#asterisk gmaruz1 (n=gmaruzz@217-133-80-112.b2b.tiscali.it) |
| 23:07.25 | kimosabe | ok |
| 23:07.44 | kimosabe | that means im stcuk technvalley becuase any time i ant 2 flsh it asks for passwd |
| 23:07.59 | kimosabe | ok thanks this is a brick then |
| 23:08.26 | *** part/#asterisk justinu (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
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| 23:25.14 | *** part/#asterisk swytch (n=ezcall@d80-170-39-248.cust.tele2.fr) |
| 23:25.37 | [TK]D-Fender | :O |
| 23:26.48 | camelon | ZZZZZZZZZZZZZ |
| 23:27.39 | camelon | TK D-fender . . .may I ask you a question? |
| 23:27.45 | kimosabe | does anyone have sipura200 firmear |
| 23:27.53 | Qwell | kimosabe: sipura does |
| 23:27.53 | kimosabe | does anyone have sipura2000 firmwear |
| 23:28.05 | kimosabe | qwell question is it in bin format ? |
| 23:28.16 | kimosabe | or does it just like try 2 reflash the device |
| 23:28.19 | [TK]D-Fender | camelon : Well thats one.. I may even grant you a second! |
| 23:28.49 | [TK]D-Fender | kimosabe : Go to Sipura's website. Its all right there. |
| 23:29.36 | kimosabe | ok |
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| 23:30.27 | *** join/#asterisk peanuter (n=peanuter@216.176.177.138) |
| 23:30.58 | [TK]D-Fender | camelon: .... well go ahead and ask already! |
| 23:30.59 | peanuter | i'm looking for an 800 forward service like nufone offered |
| 23:31.07 | peanuter | anyone know of one |
| 23:31.34 | camelon | from your experience . . loking for a * expansion . . . (add > 50 extensions) . . . and balancing . . .investmen . . . confidence and minimum troubles . . . wich technology to choose: voip or channel banks? |
| 23:32.36 | drray | do you already own phones? |
| 23:32.44 | camelon | TK D-Fender . . .the latter . . my questio |
| 23:32.46 | *** join/#asterisk Led_Zeppelin (n=dummy@cpe-24-210-79-197.columbus.res.rr.com) |
| 23:32.53 | Led_Zeppelin | hello. |
| 23:33.00 | drray | I prefer channel banks for stations and VOIP for trunking |
| 23:33.08 | *** join/#asterisk chaoscon_ (n=ph33r@smartserv/ceo/chaoscon) |
| 23:33.18 | [TK]D-Fender | camelon : VoIP. Channel banks = T1 which needs cards in your PC and there is a cap on that. VoIP in passthrough/non transcode is nearly irrelevent. |
| 23:33.27 | Led_Zeppelin | I am looking for a good service that will work with asterisk, and is cheap for a college student. I prefer getting a IP phone too, any advice? |
| 23:33.49 | [TK]D-Fender | camelon : If you're still looking for analog phones, get a mass gateway like AudioCodes, Mediatrix, etc. |
| 23:35.05 | camelon | drray . . .no qwn phones yet |
| 23:38.22 | camelon | TK D-Fender . . . if you were starting from 0? |
| 23:39.18 | russellb | Led_Zeppelin: go for a prepaid service, that's the way to go if you're not doing a whole lot of calls. where are you in school? |
| 23:39.24 | [TK]D-Fender | camelon : Depends on budget, call volume, wiring, etc. a lot of factors. Describe the company and its phone usage. |
| 23:40.14 | Led_Zeppelin | russellb: OSU (Ohio) |
| 23:40.16 | *** join/#asterisk delmar (n=delmar@203-114-178-231.static.inspire.net.nz) |
| 23:40.29 | russellb | Led_Zeppelin: cool. engineering student? |
| 23:41.10 | Led_Zeppelin | yah, ofcourse :-) |
| 23:41.21 | camelon | TK D-Fender . .. 2000 calls a day (schedulling) . . . 2E1 . . . 4.000 USD budget . . . 6 Cat network |
| 23:41.23 | russellb | Led_Zeppelin: heh, nice. I'm a computer engineering student at Clemson Univ. |
| 23:41.26 | Led_Zeppelin | only enginnering students do this kinda stuff |
| 23:41.32 | Led_Zeppelin | russellb: cool. |
| 23:41.39 | Led_Zeppelin | russellb: what yr? |
| 23:41.42 | russellb | seniro |
| 23:41.45 | russellb | er, senior. |
| 23:41.54 | Led_Zeppelin | cool. |
| 23:42.02 | Led_Zeppelin | Applying for any jobs so far? |
| 23:42.08 | Led_Zeppelin | or going to grad school? |
| 23:42.14 | russellb | Led_Zeppelin: well, I already have a job with Digium, actually |
| 23:42.22 | Led_Zeppelin | oh nice! |
| 23:42.29 | russellb | I have been working on Asterisk for a couple years now ... |
| 23:42.29 | Led_Zeppelin | where are they based out of? |
| 23:42.33 | russellb | Huntsville, AL |
| 23:42.49 | *** join/#asterisk kbitter (i=kbitter@201.29.133.114) |
| 23:43.02 | Led_Zeppelin | cool, you must be good with this stuff then :-) |
| 23:43.11 | Led_Zeppelin | I know a lot of companies use * |
| 23:43.40 | russellb | let me know if you want to get into development :) |
| 23:43.46 | Led_Zeppelin | My cousins friend's friend's friend's company uses it for their company. |
| 23:43.57 | [TK]D-Fender | can : is that 4000 USD just for phones? |
| 23:44.06 | Led_Zeppelin | russellb: i would like to do some :-). Volunteering ofcourse. |
| 23:44.29 | [TK]D-Fender | camelon : rather... |
| 23:44.43 | russellb | Led_Zeppelin: cool, well there are always things to work on. We usually have a list of "janitor projects" that are good to get you going |
| 23:44.56 | camelon | noooop . . . for all the devices required . . is a SOHO company |
| 23:44.56 | Led_Zeppelin | russellb: any cool projects? I am faily versed in Unix, and DBMS |
| 23:45.03 | Led_Zeppelin | russellb: don |
| 23:45.04 | [TK]D-Fender | camelon : And sorry, there is no solution I would suggest that would even fit that budget. |
| 23:45.23 | Led_Zeppelin | don't really do a whole lot lowlevel stuff, russellb. Mostly data warehousing techniques and stuff like that. |
| 23:45.41 | Led_Zeppelin | and mostly RDBMS stuff :-/ |
| 23:46.01 | russellb | Led_Zeppelin: well, there are even some larger projects in that area ... |
| 23:46.15 | Qwell | odbc_storage! |
| 23:46.31 | Led_Zeppelin | russellb: we should get together and work on some of this stuff. |
| 23:46.44 | russellb | Led_Zeppelin: drop me an email at russell@digium.com, and we can find a project for you |
| 23:46.48 | Led_Zeppelin | see how my skillz go up against a Tiger :-0 |
| 23:46.54 | [TK]D-Fender | camelon : not for 50 phones. |
| 23:46.54 | camelon | TK -DFender . . . what the smallest budget to start? |
| 23:47.55 | camelon | TK D-Fender . . . and wich way (VoIP or analogs)? |
| 23:48.07 | [TK]D-Fender | camelon : if you want basic Cat5 sip hardphones without the need for speakerphone I'd suggest Polycom IP 301's @ 115$ ea. 50 * 115$ = 5750 |
| 23:48.58 | camelon | Hummmm . . .ans what else? both ChBs? |
| 23:49.32 | [TK]D-Fender | camelon : No those are SIP hardphones, no need for CB. |
| 23:49.46 | camelon | TK D-Fender . . .sorry . . .SIP phones . .damnn! |
| 23:50.13 | Led_Zeppelin | Okay, if i have an asterisk server running, can I have regular computers (with mics and speakers), connect to the * server to use as a phone? |
| 23:50.29 | russellb | yup, using "soft phones" |
| 23:50.41 | Led_Zeppelin | oh, is that what its called :-) |
| 23:52.39 | camelon | T D-Fender. . . to manage 2000 calls daily (incoming and outgoing) . . . what could secure me a reliable system? |
| 23:53.39 | [TK]D-Fender | camelon : Good hardware Echo Cancelling PRI card, basic P4 3.2ghz + processor is just fine. |
| 23:53.58 | camelon | TK D-Fender . . .all supprted with polycom phones . .no additional equipment . . .and what about the boadband? |
| 23:54.00 | Led_Zeppelin | is there a list of * compatable providers? |
| 23:55.46 | [TK]D-Fender | camelon : Broadband? this is all LAN to your T/E 1's |
| 23:57.20 | camelon | TK D-Fender . . . yeeeeppp . . . outgoing through the PRI . . . but my LAN wouldn't be improved? |
| 23:58.30 | [TK]D-Fender | camelon : Generally I'd say no unless you are running it in parallel with heavy PC's (inline with the phone). Switches typically avoid the usual pitfalls. |