| 00:00.03 | Silik0n | why is it people think that we really wanna hear them bitch and complain all the time, instead of seeing them submit patches? |
| 00:00.46 | greg_work | dca[laptop]: you have to pay $45/mo to get an 800# |
| 00:01.34 | harryvv | Silk, mabey thay had high hopes for asterisk and are not developers? |
| 00:02.06 | Veto | How much to custom prompts cost (intro,menu,directory)? |
| 00:02.15 | dca[laptop] | greg_work: email or call them. they'll work something out with you |
| 00:02.47 | *** part/#asterisk dca[laptop] (~trillian@sta-207-174-139-178.rockynet.com) |
| 00:03.21 | Veto | and does the lady at digium who does the samples do them commercially? |
| 00:04.17 | czero | veto I belive your problly talkin g about alison and yes she does |
| 00:04.31 | file[laptop] | hell is hell. |
| 00:06.04 | Veto | czero, thx...any idea what they charge for them (alison) to do prompts? |
| 00:06.09 | greg_work | Veto: http://www.digium.com/index.php?menu=thevoice |
| 00:06.29 | Veto | nm, thx greg_work |
| 00:08.25 | clh | anyone know the official name of T-1 cabling? |
| 00:08.27 | clh | and the connectors? |
| 00:08.28 | Moc_ | fucking metro... hehe |
| 00:08.36 | blitzrage | Moc_: I hear that |
| 00:08.36 | Moc_ | I got screwed today hehe |
| 00:08.51 | stevekstevek | clh: Category 2 and RJ45? |
| 00:08.54 | fearnor | rj-48. |
| 00:09.02 | Moc_ | I wanna be a good boy and take the metro/bus today hehe |
| 00:09.22 | stevekstevek | ouch. |
| 00:09.30 | stevekstevek | EIA-568B? |
| 00:09.32 | clh | so it is Cat-2 with RJ48 connectors right? |
| 00:09.51 | fearnor | 568b is pin color standard |
| 00:10.07 | clh | so if i have an adtran 750, then the cabling between it and a X100T card would be RJ48? |
| 00:10.09 | greg_work | i signed up for an account at iax.cc and got an 800 number.. outbound calls work fine, but i can't call the 800 from either a real pstn or routed over iax.cc .. from pstn it says "due to technical difficulties we cannot complete this call" |
| 00:10.13 | stevekstevek | hmm, I thought they usually came in shiny yellow :) |
| 00:10.44 | fearnor | clh: errrrrr |
| 00:10.50 | fearnor | x100p is single-port analog |
| 00:10.57 | fearnor | you mean T100P probably. |
| 00:11.13 | hermie | chl wiring isn't RJ anything, just the ends |
| 00:11.15 | fearnor | and yes, it'll be a crossover (i think) cable |
| 00:11.42 | hermie | yup, T1 xover |
| 00:12.07 | greg_work | hrm, apparently you can't call any 800 number from iax.cc? |
| 00:12.13 | clh | fearnor : yah that is correct |
| 00:12.37 | clh | but the cabling is Cat-2 and the connectors are RJ48 |
| 00:12.39 | clh | ? |
| 00:12.43 | clh | correct that time? |
| 00:12.46 | Veto | greg_work: they do work, but they got a bad batch of numbers. send email to help or call them. I got a new one that works fine DID. |
| 00:12.47 | hermie | chl: yup |
| 00:12.56 | stevekstevek | of course, cat-5 will work just as well. |
| 00:13.03 | stevekstevek | and it's easier to find. |
| 00:13.09 | hermie | chl: but you might wanna know that RJ48C == RJ45C |
| 00:13.24 | fearnor | hermie, technically, it isn't :) practically, it is ;) |
| 00:13.26 | Koshatul | is there a way to trouble shoot a iax peer coming up as unreachable ? |
| 00:13.40 | Koshatul | even when debug shows communication between the two machines |
| 00:13.54 | MrClean | How do I make an agent "unavailable" without logging them out of the system or dynamically removing them from the queue(s)? Is it even possible? |
| 00:13.58 | redder86 | Koshatul: is the peer there? ping responses? |
| 00:14.23 | hermie | fearnor: honestly, I think people selling 48c ends are probably just repackaging rj45 ends |
| 00:14.27 | Koshatul | redder86: yup, ping fine, and tcpdump shows traffic, i just consitantly get unreachable |
| 00:14.38 | redder86 | Koshatul: if the peer is defined with a name (rather than an IP), is there a DNS problem, possibly? |
| 00:14.47 | clh | so i really dont need cat2 for the link between the adtran and the T100P ? |
| 00:15.02 | Koshatul | redder86: i don't think so, if i change the password in the iax.conf i get aauth failure |
| 00:15.25 | redder86 | Koshatul: what version of Asterisk are you running? |
| 00:15.50 | Koshatul | redder86: Asterisk CVS-HEAD-01/12/05-09:37:09 built by root@magi on a i686 running Linux |
| 00:16.16 | Koshatul | should only be a few days old |
| 00:16.26 | redder86 | is the other IAX endpoint also an Asterisk box? |
| 00:16.33 | greg_work | anyone in the US want to try calling me? |
| 00:16.38 | Koshatul | not sure, it's a iax phone provider |
| 00:16.57 | Koshatul | www.austechpartnerships.com |
| 00:17.06 | Koshatul | i think it is |
| 00:17.13 | redder86 | Well, they should be able to troubleshoot with you more effectively. |
| 00:17.20 | Koshatul | emailed them 2 days ago |
| 00:17.27 | redder86 | It's hard to get a good idea of what's going on by just one endpoint. |
| 00:17.47 | redder86 | well, I don't know what to tell you. call them, switch, ? |
| 00:17.48 | Koshatul | no reply yet other than a autoattendant email saying that my email has been received and has a number, and i will be answered as soon as possible |
| 00:18.10 | Koshatul | redder86: yeah, i'm getting fustrated with it, can't call support, only email |
| 00:19.31 | stevekstevek | clh: No, Cat 5 or Cat 3 will work fine. The higher the "category", the more resistance to crosstalk, etc.. |
| 00:20.25 | Koshatul | redder86: one thing i haven't tried, i'm going to cvs update asterisk |
| 00:21.16 | redder86 | Koshatul: that might help. I don't use HEAD, though. That's too brave for me. I use v1_0 |
| 00:21.19 | redder86 | cvs |
| 00:22.00 | *** join/#asterisk jdb1968 (~jdb1968@S010600045af29653.cg.shawcable.net) |
| 00:22.39 | Koshatul | what is the param for cvs to grab a certain tag ? |
| 00:22.45 | redder86 | -r |
| 00:23.20 | jdb1968 | hey all, bit of a prob with zaptel on debian... hope someone can help |
| 00:23.38 | jdb1968 | make install gives: depmod: *** Unresolved symbols in /lib/modules/2.4.27-1-386/misc/wct4xxp.o |
| 00:23.43 | jdb1968 | for each module |
| 00:24.04 | jdb1968 | <PROTECTED> |
| 00:24.04 | jdb1968 | 2.4.27-1-386 |
| 00:24.54 | jdb1968 | have kernel-headers-2.4.27-1-386 symlinked to linux & linux-2.4 in /usr/src |
| 00:25.02 | jdb1968 | it actually builds and installs the modules /lib/modules/`uname -r`/misc |
| 00:25.30 | jdb1968 | but modproving them gives same depmod unresolved symbols error as make install time |
| 00:25.35 | jdb1968 | ideas? |
| 00:25.37 | *** join/#asterisk Umaro (~umaro@c-24-22-76-14.client.comcast.net) |
| 00:25.38 | *** join/#asterisk Zaw (hidden-use@zaw.subneural.net) |
| 00:25.57 | jdb1968 | looks like the kernel interface is out of sync with the headers/module |
| 00:26.06 | jdb1968 | but looks copesetic to me... |
| 00:26.18 | jdb1968 | k, i'll stop talking now... ideas? |
| 00:26.58 | jdb1968 | this is with -r v1-0 of zaptel |
| 00:29.25 | *** join/#asterisk eschvoca (~eschvoca@dailyglen.istop.com) |
| 00:34.03 | *** join/#asterisk harryvv (~noyb@S010600055d210201.vs.shawcable.net) |
| 00:34.08 | *** part/#asterisk ropeguru (~ropeguru@64.83.16.36) |
| 00:34.27 | harryvv | Can anyone recomend the grandstream phones? |
| 00:35.23 | *** join/#asterisk buddah (~hnic@208.179.86.5) |
| 00:35.45 | buddah | anyone know how to get polycom ip500 phones to grab the config off the central boot server rather than going to phone config? |
| 00:35.53 | jdb1968 | harryvv: cheap look and feel, im my experience the firmware is pretty stable |
| 00:36.15 | buddah | i got the cbs config working, so i junked up the phone config via web to see if it would grab the cbs config, and it didnt, it kept the phone config |
| 00:36.38 | redder86 | buddah: it should work like any other Polycom does |
| 00:36.46 | buddah | well |
| 00:36.50 | buddah | how is that? |
| 00:36.53 | buddah | only polycom i've used |
| 00:37.10 | harryvv | jdb, im not to concerned with the look and feel but the stability and does it work and you say it does. Thay will be test phones and demo phones but wont deploy them to bizz only if thay ask. |
| 00:37.25 | redder86 | buddah: ftp ... http://www.voip-info.org/wiki-Polycom+Phones |
| 00:37.50 | jdb1968 | others have complained about high %age of defects, but i haven't found that |
| 00:37.52 | redder86 | harryvv: I like the grandstreams, except that they don't have an alphanumeric display (only numbers) |
| 00:38.10 | harryvv | okay |
| 00:38.33 | redder86 | so I run X-Lite on the PC here just to get the Caller*ID Name |
| 00:38.35 | buddah | thanks |
| 00:38.42 | harryvv | again, I can explain that to a customer and thay can request a ip300 or 500 |
| 00:38.53 | harryvv | for now, keep the cost down :) |
| 00:39.12 | harryvv | grandsteam 101 budgetphone for 63 bucks from voipsupply |
| 00:39.21 | BoRiS | I recommend for the cheapest phone..Get a SPA-841 |
| 00:39.21 | redder86 | buddah: the biggest trick is getting your ftp server to accept the username |
| 00:39.22 | Nugget | ew, I'd hate only having caller id number. |
| 00:39.34 | *** join/#asterisk Connor_ (~billy@198-144-165-65.knx.tn.nxs.net) |
| 00:39.45 | Connor_ | Is there a way I can see if Asterisk is setting the ToS bit? |
| 00:39.56 | Umaro | yes |
| 00:39.59 | Umaro | er |
| 00:40.14 | stevekstevek | Connor_: tcpdump or ethereal. |
| 00:40.35 | redder86 | Nugget: yup, that's why I run X-Lite and Asterisk Dial()s both clients together |
| 00:41.06 | *** join/#asterisk jayden (~jayden@pcp02795302pcs.roylok01.mi.comcast.net) |
| 00:41.13 | Connor_ | stevekstevek: Knew that, just not sure what flags to throw to see the bits |
| 00:42.10 | RoyK | ~kill \* |
| 00:42.12 | jbot | ACTION shoots a super-inverse neutron gun at \* |
| 00:42.40 | buddah | damn i put a time server in the phone |
| 00:42.48 | *** join/#asterisk BuddaBoy (~budda@81-86-85-233.dsl.pipex.com) |
| 00:42.51 | buddah | and my boss looked at it, then looked at the clock on my computer |
| 00:42.57 | buddah | and realized mine was set ahead 15 min |
| 00:43.00 | buddah | so i can go home earlier |
| 00:43.02 | buddah | :( |
| 00:43.45 | Umaro | buddah: that's why you make the company's time server 15 minutes fast, and sync both your computer and your phone off it :P |
| 00:44.28 | RoyK | buddah, you need to find a dumber boss |
| 00:44.33 | buddah | lol |
| 00:44.34 | buddah | yeah |
| 00:44.35 | buddah | i do |
| 00:45.36 | jayden | you need a script that has it set 15 min slow in the morning, and over then advances it 3.75 seconds per minute, and then resets overnight |
| 00:45.43 | redder86 | but then you'd just show up to work 15 minutes late every day |
| 00:45.54 | jayden | see ^ |
| 00:45.55 | jayden | :) |
| 00:46.20 | BuddaBoy | hi. i'm wanting to experiment with writing a reporting system for asterisk based on its CDR data. however, i do not yet have asterisk installed and wanted some real-ish data to work off. |
| 00:46.26 | BuddaBoy | where could i find such sample data? |
| 00:48.30 | Connor_ | what is tos 0x60 and tos 0x10 ? |
| 00:48.31 | Koshatul | eheh Moc_ wants gameboy music on hold :) |
| 00:48.45 | Connor_ | Looks like asterisk is sending 0x10 and my ata is coming back with 0x60 |
| 00:48.49 | Moc_ | ;) |
| 00:48.51 | BuddaBoy | maybe somebody has a moderate sized /var/log/asterisk//cdr-csv//Master.csv i could borrow? :) |
| 00:49.06 | Moc_ | I want Tetris and Super Mario Midi for MoH ;) |
| 00:49.29 | Koshatul | 0x60 is realtime i think |
| 00:49.33 | Koshatul | ack |
| 00:49.36 | Koshatul | ignore me |
| 00:49.40 | implicit | 0x60 is not real time |
| 00:49.48 | Koshatul | Moc_: grab an old gameboy and plug it in line in |
| 00:50.00 | Koshatul | implicit: which is why i said ignore me |
| 00:50.13 | BuddaBoy | Moc_: use a GB emulator, and record the windows audio to an mp3? |
| 00:50.20 | implicit | Koshatul: yep |
| 00:50.28 | Moc_ | BuddaBoy, it not as fun ;) |
| 00:50.28 | implicit | delayed response :-p |
| 00:50.48 | implicit | BuddaBoy: that is also assuming that he is using windows |
| 00:51.27 | Connor_ | I'm also seeing something that says use 0xb8 for RTP and 0x68 for SIP |
| 00:51.43 | Koshatul | eheh, run a dedeicated machine with a gameboy emulator on screen piping it's music to line in on the same box and streaming it to the asteris kbox |
| 00:51.49 | Koshatul | and just to make it useful |
| 00:51.54 | Koshatul | play the games |
| 00:51.58 | Moc_ | Anyone know the Analog DID Trunk technology ? |
| 00:51.59 | BuddaBoy | lol |
| 00:52.07 | Moc_ | and how I can use it with * ? |
| 00:52.08 | Koshatul | so the MoH has *blip* *blip* *boing* *blip* |
| 00:52.33 | Moc_ | hehe |
| 00:52.34 | jayden | when I had DID trunks, they were all inband signalling before answer |
| 00:52.35 | DrPete | man, howcome the 7960 are so expenisve, are they that good? |
| 00:52.46 | Moc_ | DrPete, nope |
| 00:52.50 | Koshatul | i've got a 7960 |
| 00:52.54 | Moc_ | DrPete, check the Polycom IP 500 and IP 600 |
| 00:52.58 | sivana | is G723 supported in *? |
| 00:53.00 | Koshatul | if it wasn't cisco i would have thrown it out |
| 00:53.12 | file[laptop] | sivana: passthrough is supported, but there is no encoder/decoder available |
| 00:53.17 | Koshatul | well that and it has lots of little functions i can play with |
| 00:53.17 | Moc_ | g723 no but could be. G723.1 is not I think |
| 00:53.17 | sivana | is it worth it? |
| 00:53.21 | *** join/#asterisk _Sam-- (sam@ns2.kneedraggers.com) |
| 00:53.22 | RoyK | sivana, there's a patch for testing, but none for commercial use |
| 00:53.24 | sivana | ok |
| 00:53.27 | jayden | i've got a 7960, it is great for free.. .but I'm cheap |
| 00:53.43 | Koshatul | jayden: problem i have is it isn't backlit |
| 00:53.45 | RoyK | sivana, asterisk out of the box only supports passthrough |
| 00:53.49 | DrPete | Moc_: ok, are they any good? expensive? |
| 00:53.54 | Moc_ | Let say cisco need to put alot more option on their phones ... |
| 00:53.56 | sivana | spa-2000 only does 711, 726, 729A I guess then |
| 00:54.07 | jayden | I would be happy to test any free phone now that you mention... |
| 00:54.09 | Koshatul | and it's in my room at home, where there isn't much light (as any true tech room shouldn't, the light sources must all be blinking with network data) |
| 00:54.11 | Moc_ | DrPete, VERY GOOD... and hell better than the cisco 7960 |
| 00:54.13 | _Sam-- | hi...are the dialogic boards still supported? |
| 00:54.19 | Moc_ | for the price, the IP 500 is about 180$ US |
| 00:54.32 | _Sam-- | specifically one listed on the site...D/120JCT-LS 12-port analog + voice |
| 00:54.39 | RoyK | sivana, 729 is supported. there's a 'free' one for testing and you can buy licenses from digium for $10 a piece |
| 00:54.40 | jayden | a free IP500 would be great... |
| 00:54.42 | _Sam-- | listed on the site, but not listed in source when i go to make it |
| 00:54.46 | Koshatul | Moc_: and for the little known problem that a cisco 7960 will crash if it connects to an aliased interface on a asterisk box |
| 00:54.53 | Moc_ | IP 600 just have 3 more line + a very nice hires Display |
| 00:55.04 | *** join/#asterisk WilliamK (~wkeller@c-24-0-130-60.client.comcast.net) |
| 00:55.10 | Koshatul | Moc_: so ip 600 it is ? |
| 00:55.11 | jayden | that would be fine too :) |
| 00:55.19 | Moc_ | my cisco crashed on my only 2 time this year.. |
| 00:55.56 | Moc_ | I got 7960, IP 500 and IP 600 (ok BT101 too) |
| 00:56.02 | Koshatul | Moc_: i used to run my asterisk box on internal ip and aliased the external ip, my cisco was on the external network, and everytiem it booted it would crash |
| 00:56.03 | DrPete | Moc_: I guess these things are gonna last me. I dont mind buying a good IP phone, but I dont wanna get a couple and they bring and new standard out or something |
| 00:56.07 | Moc_ | and polycom configurations option are just damn GREAT !!! |
| 00:56.21 | implicit | Moc_ which one looks nicer 7960 or ip600? |
| 00:56.27 | Koshatul | ip600 |
| 00:56.37 | implicit | Koshatul: do you have them both? |
| 00:56.43 | Koshatul | but 7960 has cisco written on it, so it has coolness factor |
| 00:56.47 | Moc_ | The design, 7960... But the Display of IP 600 is hell better |
| 00:56.49 | Koshatul | :P |
| 00:56.52 | implicit | Koshatul: heh, i have 7960 right now |
| 00:56.54 | Moc_ | even the IP 500 is better for the display |
| 00:57.09 | Koshatul | implicit: if only chicks dug the 7960 more :) |
| 00:57.17 | implicit | Koshatul: they do actually |
| 00:57.19 | terrapen | dammit |
| 00:57.28 | Koshatul | 7960 hasa huge flaw, no backlight |
| 00:57.28 | _Sam-- | my thoughts exactly |
| 00:57.28 | implicit | Koshatul: i just dont want it to stop if i get ip600 |
| 00:57.30 | Moc_ | Koshatul, make them watch 24... and put the 24 ringtone on your phone.. |
| 00:57.31 | terrapen | now i have The Price Is Right theme song stuck in my head |
| 00:57.34 | Mother_ | any of you got this bull from distributors that they want your client's details to sell you Cisco VoIP? |
| 00:57.35 | Moc_ | they will love it ;) |
| 00:57.35 | terrapen | doo dooo duh-doooo |
| 00:57.36 | agray | I have some questions regarding symmetric NAT. Anyone have work arrounds for the problems with it. |
| 00:57.36 | terrapen | doo dooo duh-doooo |
| 00:57.38 | implicit | Koshatul: so does polycom? |
| 00:57.57 | Koshatul | i thought the ip600 was colour backlit ? |
| 00:58.07 | terrapen | do da-doodoo da-doodoo da-doodoo doo-doo dah-doo |
| 00:58.17 | Moc_ | Koshatul, no. They are looking to release a updated ip600 with backlit |
| 00:58.37 | Koshatul | Mother_: yup, i got that buying routers |
| 00:58.38 | _Sam-- | nobody knows if this card is still supported? just wondering if i wasted 1200.... |
| 00:58.40 | _Sam-- | D/120JCT-LS 12-port analog + voice |
| 00:58.42 | DrPete | Moc_: where is the cheapest place to buy the polycom in the US? |
| 00:58.47 | Moc_ | and 2.5mm headset jack |
| 00:58.49 | Koshatul | Moc_: my world of polycom has shattered |
| 00:58.55 | terrapen | drpete, yes, i want to know the same |
| 00:59.04 | Moc_ | DrPete, donno, www.voipsupply.com aint TOO BAD.. |
| 00:59.09 | Moc_ | but check www.froogle.com |
| 00:59.14 | Mother_ | hmmmm it sounds iffy...they have the details, then they could go around everyone and give your clients to some partner or whatever |
| 00:59.19 | Mother_ | I don't like it at all |
| 00:59.28 | DrPete | Moc_: I am in the UK, if I buy a US phone will it work? |
| 00:59.29 | Mother_ | and I can't see a real reason behind it |
| 00:59.30 | Moc_ | note, be sure to get the SIP image |
| 00:59.36 | Koshatul | Mother_: yeah dodgy as hell, but i think it's more for the smartnet stuff |
| 00:59.37 | Moc_ | DrPete, yes |
| 00:59.56 | Moc_ | DrPete, well you will need a 12volt power supply for your area |
| 01:00.00 | Moc_ | OR a PoE Adaptor |
| 01:00.03 | Mother_ | Koshatul: could be - I got this when I tried to order a couple of 7960s |
| 01:00.09 | BuddaBoy | does anybody have some /var/log/asterisk/cdr-csv data i could borrow to use as test data for a PHP based app? please. |
| 01:00.13 | DrPete | Moc_: ooh they have poe |
| 01:00.21 | Koshatul | Mother_: with smartnet ? or without ? |
| 01:00.23 | Moc_ | DrPete, there is the IP 300 also |
| 01:00.29 | Moc_ | DrPete, ok IP 600 have PoE |
| 01:00.31 | Mother_ | Koshatul: without |
| 01:00.38 | Mother_ | just the bare phones |
| 01:00.43 | Koshatul | Moc_: i so want a poe for my cisco, just gotta remember which side is which :) |
| 01:00.44 | Moc_ | IP 500 use a Power Injector cable.. but you can get PoE cable for the IP 500 |
| 01:00.45 | DrPete | Moc_: I have a bro in law over in the US, i could get him to send me themover |
| 01:00.55 | Moc_ | I dont think you have PoE on IP 300 atall |
| 01:01.22 | Moc_ | sure.. |
| 01:01.23 | Koshatul | Mother_: hrmz, i stillt think it's a warranty thing, (apparently) with cisco if you buy a device you can call up and check to make sure they guy selling it isn't dodgy (stolen) |
| 01:01.26 | terrapen | are PoE switches expensive? |
| 01:01.33 | *** part/#asterisk stonefly (~Stonefly@209-33-211-198.sg-wireless.infowest.net) |
| 01:01.39 | jayden | y |
| 01:01.45 | jayden | terrapen |
| 01:01.53 | Mother_ | Koshatul: could well be...I'll hope for the best :) |
| 01:01.55 | Koshatul | terrapen: i thought PoE was a inline unit |
| 01:02.06 | Koshatul | Mother_: give them your details |
| 01:02.11 | Mother_ | LOL yeah! |
| 01:02.19 | jayden | but you can also get additional injectore devices, both one off, and in multiport boxes |
| 01:02.29 | jayden | they are cheaper |
| 01:02.35 | Moc_ | For you people interested in polycom IP phone, check this PDF for all the stuff you can configure on the phone, READ it closely ;) http://www.freedomphones.net/polycom/files/Admin_Guide-SoundPoint_IP_SIP_2004-06-16.pdf |
| 01:03.07 | Moc_ | also check this page.. the top include stuff I found interesting on the polycom phone : http://www.voip-info.org/tiki-index.php?page=Polycom%20Phones |
| 01:03.34 | terrapen | what is the soundpoint? |
| 01:03.40 | terrapen | == IP600? |
| 01:04.08 | Moc_ | Soundpoint IP 600, SoundPoint IP 500, SoundPoint IP 300, and SoundStation IP 3000 |
| 01:04.11 | Mother_ | hmmmm I guess I'll have to get that polycom cert thing |
| 01:04.12 | *** join/#asterisk gopinsurg (cashmoney@dialup-4.224.186.22.Dial1.Cincinnati1.Level3.net) |
| 01:04.24 | Moc_ | SoundStation IP 3000 is the Polycom conference phone, but IP |
| 01:04.27 | Mother_ | at least it's free |
| 01:04.38 | epoch | Moc_: it's useless though |
| 01:04.45 | Moc_ | epoch: ?? |
| 01:04.45 | epoch | Moc_: it's not sip :P |
| 01:04.56 | Moc_ | epoch, it h323.. :( but incoming calls work on it ;) |
| 01:05.04 | Moc_ | I couldn't get outgoing call to work yet.. |
| 01:05.07 | epoch | Moc_: ah |
| 01:05.11 | epoch | well that's pretty cool |
| 01:05.20 | epoch | how does it sound? |
| 01:05.29 | Umaro | hey guys, i'm trying to convert my * recordings to mp3 format.. what kind of switches do I need to use with sox to convert it to something a mp3 encoder will encode? |
| 01:05.39 | Umaro | my * recordings are wav49 |
| 01:05.40 | Moc_ | good, Im already used to the soundstation polycom phone at work so.. |
| 01:06.04 | Moc_ | Maybe using MGCP image, I could get better result |
| 01:06.12 | Moc_ | I mean SKINNY or what ever it call |
| 01:06.27 | Connor_ | anyone know of a website I can feed it TOS bits in hex format and it kicks out what it means? |
| 01:06.35 | Moc_ | Only Polycom phone missing in my collection is the IP 300 |
| 01:07.55 | Mother_ | as in IP TOS? |
| 01:08.25 | eschvoca | Hi phone freedom troopers |
| 01:08.29 | Connor_ | Moc_ yes. |
| 01:08.51 | Moc_ | interesting then.. |
| 01:09.01 | Connor_ | Like 0x10 vs 0x60 vs 0x6b vx 0x68 |
| 01:09.13 | eschvoca | I want to write a macro called from different extensions. What extension do I use for the macro?? |
| 01:09.31 | Moc_ | I wish we could do like a Nortel system, conference up to 6 person |
| 01:09.48 | ds2 | the stevens TCP book should have the TOS bits |
| 01:10.59 | Mother_ | hmmm I don't think they are in RFC 790 or 791 |
| 01:11.40 | *** part/#asterisk gabriels (~gabriels@12-216-224-186.client.mchsi.com) |
| 01:11.59 | Mother_ | maybe here |
| 01:12.00 | Mother_ | http://www.iana.org/assignments/ip-parameters |
| 01:13.44 | *** join/#asterisk ennuyeux72 (~ennuyeux7@62.53.79.208) |
| 01:13.58 | Mother_ | Connor: were those the ones? |
| 01:14.25 | Connor_ | I havne't a clue.. I'm so freaking over trying to figure out ToS and QoS I can't see straight |
| 01:14.35 | Mother_ | lol |
| 01:14.41 | *** join/#asterisk doushanes (~Owner@c-67-184-189-220.client.comcast.net) |
| 01:14.53 | Mother_ | the TOS should be those five bits |
| 01:15.05 | Mother_ | in RFC 791 you have the position of the bits inside the header |
| 01:15.07 | Connor_ | All I know Is I see My tcpdumps do 0x10 and 0x60, and I've seen online 0x68 and 0x6b (RTP and SIP) |
| 01:16.46 | DrPete | Moc_: so why is the poly phones better then say the budgettone? |
| 01:17.01 | Chuji | Moc: So you get that 3000 working w/ * yet? |
| 01:17.11 | Mother_ | Connor-: try this one too http://rfc.net/rfc1349.html |
| 01:17.22 | doushanes | can anyone tell me what exactly do they mean by the digium tdm400p cards being half length? Are they the same type pci cards for low profile cases(Micro ATX)? |
| 01:17.25 | *** join/#asterisk stratos (~stratos@c-67-172-224-6.client.comcast.net) |
| 01:17.36 | stratos | in zttool, under IRQ MISSES |
| 01:17.47 | stratos | whats the significance if this number keeps rising? |
| 01:18.07 | Moc_ | Chuji, incoming work, not outgoing |
| 01:18.18 | BoRiS | Anyone have a newer firmware for the Senao SI7800 then 0.003.0008? |
| 01:18.21 | Moc_ | DrPete, you really took the worst phone ;) |
| 01:18.26 | Moc_ | to compare with.. |
| 01:18.50 | DrPete | Moc_: hehe |
| 01:18.52 | Chuji | Moc: Come on man, I'm waiting on you to get that before I shell out the dough! |
| 01:18.57 | Moc_ | lol |
| 01:19.05 | Moc_ | Chuji, Ive stop trying rightnow |
| 01:19.27 | Moc_ | Chuji, check on ebay, I got mine at 279$ US |
| 01:19.34 | Connor_ | Mother_, Okay.. That helps a little, but it looks like I have extra BITS. that 0x68 translates to to more than 4 bits |
| 01:19.54 | stratos | anyone know what part IRQ MISSES plays under the zttool interface? |
| 01:19.55 | Mother_ | of course, that would be a full byte |
| 01:20.50 | Chuji | Moc: Wow, that's not bad |
| 01:21.00 | Connor_ | Mght it be precedence then ? |
| 01:21.11 | DrPete | Moc_: I guess what I am trying to ask is, for the extra money am I going to benifit from a polycom, i am small buiness user, and I dont want to pay for things I am never gonna use, if you see what I mean |
| 01:22.00 | Mother_ | i.e. 0x68 == 01101000, and the TOS are bits 3 to 6, i.e. 0100 |
| 01:23.09 | Connor_ | okay, so what is the other bits for? |
| 01:23.09 | Connor_ | Precendence ? |
| 01:23.09 | Mother_ | the first three are precedence |
| 01:23.09 | BoRiS | file!!!!!!!!!!!!!!!!! |
| 01:23.24 | file[laptop] | BoRiS!!!!!!!!!!!!! |
| 01:23.25 | Mother_ | if the network supports precedence, it can give priority to packets based on it |
| 01:23.42 | BoRiS | File: Are you going to bid of the senao wifi phone on ebay? |
| 01:23.48 | Mother_ | look at page 12 of RFC 791 here http://rfc.net/rfc791.html |
| 01:23.50 | file[laptop] | BoRiS: no |
| 01:24.09 | BoRiS | lol, I already have one |
| 01:24.13 | file[laptop] | eh have two |
| 01:24.15 | Connor_ | Mother_ Well, I'm bascily trying to setup QoS. |
| 01:24.23 | Mother_ | in this case, if the network supported precedence, it would mean Flash (011) |
| 01:24.27 | jayden | Boris, or even better... buy me one! |
| 01:24.38 | file[laptop] | now THERE'S an idea |
| 01:24.51 | BoRiS | lol |
| 01:24.59 | file[laptop] | awwwwwwwwwwwww |
| 01:25.16 | BoRiS | except for file |
| 01:25.25 | file[laptop] | :) |
| 01:25.32 | jayden | :( |
| 01:25.37 | Mother_ | Connor-: then you need to play with the precedence and ToS fields, and make sure the network devices support them |
| 01:25.40 | file[laptop] | actually I'm looking at another phone |
| 01:25.46 | BoRiS | Which one? |
| 01:25.48 | file[laptop] | *trying* to get the source code for it |
| 01:25.51 | jayden | the one for me |
| 01:26.35 | jayden | <sigh> |
| 01:26.39 | blitzrage | well.... back to the hotel anyways :) |
| 01:26.39 | file[laptop] | woot home |
| 01:26.47 | blitzrage | lates |
| 01:26.47 | *** join/#asterisk iceyp (~icepick@firewall.unix.co.nz) |
| 01:27.55 | iceyp | hey guys |
| 01:28.21 | iceyp | I'm looking for a good extensions GUI thats web based, can anyone point me to a free one thats easy to install; |
| 01:28.45 | jayden | that depends, do you have any free phones to spread around :) |
| 01:28.55 | Mother_ | question: do all snom phones work OK with *? |
| 01:28.59 | Mother_ | I need low-cost here |
| 01:29.32 | Umaro | there are low cost snom phones?! god, they must REALLY suck ;) |
| 01:29.55 | Mother_ | well I don't know their cost here yet :) |
| 01:30.12 | stratos | Mother_: we use Snom190s at work and they work great |
| 01:30.20 | Mother_ | I'm budging between Cisco and Polycom, but each one gives me grief in some way to purchase them |
| 01:30.21 | stratos | yeah they are expensive as hell though :D |
| 01:30.32 | stratos | we had a bunch of polycom, they are good paper weights ;D |
| 01:30.32 | Mother_ | really, how much can they go for? |
| 01:30.46 | stratos | here in the states i think we got them new for 280 USD or something like that |
| 01:30.48 | Umaro | stratos: sell me some of your paper weights then |
| 01:30.50 | Mother_ | whoa |
| 01:30.56 | stratos | Umaro: haha ;D |
| 01:31.00 | gambolputty | my snom 190 phone works well with * |
| 01:31.12 | stratos | Umaro: they are only h323 though |
| 01:31.14 | Mother_ | I'll find out local prices then, thanks |
| 01:31.16 | Umaro | stratos: all my polycom phones work awesome |
| 01:31.19 | Mother_ | no SIP? |
| 01:31.26 | stratos | snom does sip |
| 01:31.26 | Umaro | what model? |
| 01:31.34 | stratos | Umaro: good question, i dont know off the top of my head |
| 01:31.44 | Mother_ | the blurb on the 105 says it has SIP |
| 01:31.46 | Moc_ | Umaro, you need the model, and the current bootloader |
| 01:32.05 | Moc_ | I had a IP 500 off ebay, but couldnt get upgraded to SIP |
| 01:32.11 | Umaro | Moc_: oh yeah? |
| 01:32.23 | Moc_ | IP 500 exist for a while.. |
| 01:32.26 | Umaro | Moc_: too many revisions different? |
| 01:32.33 | Umaro | I've got a bunch of ip500css |
| 01:32.38 | stratos | anyone ever have an issue with IRC MISSES before? |
| 01:32.43 | Umaro | they do SIP just fine |
| 01:32.50 | Mother_ | thanks for the info |
| 01:32.50 | stratos | err |
| 01:32.52 | stratos | IRQ ISSES |
| 01:32.54 | stratos | DAMNIT! |
| 01:32.56 | Moc_ | hehe |
| 01:32.56 | stratos | IRQ MISSES |
| 01:32.57 | stratos | sheesh |
| 01:33.02 | Moc_ | you can do it |
| 01:33.11 | Mother_ | now I'll get some sleep, tomorrow I have to put toghether the box & TDM400....wish me luck |
| 01:33.14 | Umaro | C'mon voipjet.. email me back. |
| 01:33.49 | Moc_ | Umaro, it a 1 men, 1 Server buisness, he is probably at school and go no time to help you out ;) |
| 01:34.37 | Umaro | Am I the only one who has DTMF issues with nearly every voip provider? |
| 01:35.04 | file[laptop] | apparently. |
| 01:35.13 | Umaro | supposedly, * sucks at rfc2833. no clue how true that is, but if so, it makes me sad :( |
| 01:35.18 | *** part/#asterisk ennuyeux72 (~ennuyeux7@62.53.79.208) |
| 01:35.30 | Moc_ | Umaro, for incoming or outgoing ? |
| 01:35.46 | Umaro | Moc_: outgoing |
| 01:35.51 | hermie | Moc_, they recently added their server in LA, don't forget! |
| 01:35.56 | Moc_ | Umaro, well Nufone and voiceconduits work fine for outgoing |
| 01:36.03 | Umaro | voiceconduits? |
| 01:36.04 | Umaro | hmm |
| 01:36.14 | file[laptop] | uh oh |
| 01:36.19 | Koshatul | redder86: thanks, the 1.0 connects to the iax provider fine |
| 01:36.20 | file[laptop] | you're not supposed to say we exist |
| 01:36.28 | Moc_ | file, oohh.. |
| 01:36.29 | Koshatul | redder86: but i just had a "19120 Floating point exception" |
| 01:36.35 | Umaro | I already have like 13 voip providers.. :/ |
| 01:36.50 | Umaro | But I did get my rate engine going.. man is it sweet. |
| 01:36.53 | Moc_ | file, Im still waiting my call back on my voicemail I left.. |
| 01:37.26 | Moc_ | are you support ? ;) |
| 01:37.32 | file[laptop] | nah |
| 01:37.41 | Moc_ | well it normal ;) |
| 01:38.45 | Moc_ | my did are weird.. when I call from work, I get no audio, and when I hangup, it NEVER get detected... |
| 01:39.05 | Moc_ | vc keep the channels open... |
| 01:39.18 | file[laptop] | you're getting a direct connection to the carrier |
| 01:39.46 | file[laptop] | you're not doing a reinvite are you? |
| 01:39.58 | hermie | that's not right! I don't practice Santeria... I ain't got no crystal ball! |
| 01:40.30 | Moc_ | I dont know.. I didnt change anything |
| 01:40.44 | Moc_ | normaly, the hangup should come from you ? no |
| 01:40.50 | file[laptop] | what's the number? |
| 01:40.58 | *** join/#asterisk Brixius (Brixius@c-24-118-4-197.mn.client2.attbi.com) |
| 01:41.01 | Moc_ | 5144486164 |
| 01:41.08 | Moc_ | or 5144486165 |
| 01:41.16 | Moc_ | use 6164 for the moment |
| 01:41.16 | *** join/#asterisk W1thdraw (~W1thdraw@ip68-5-125-44.oc.oc.cox.net) |
| 01:41.27 | Connor_ | how do I get asterisk to mark rtp one type of tos and sip another? |
| 01:41.28 | file[laptop] | erm lemme find a phone |
| 01:41.39 | Moc_ | file, Im not at work to reproduce the issue |
| 01:41.40 | BoRiS | calling |
| 01:42.07 | Moc_ | BoRiS, 514 cost 15cent/min ;) |
| 01:42.11 | Moc_ | jk ;) |
| 01:42.19 | file[laptop] | er school? |
| 01:42.22 | file[laptop] | console... |
| 01:42.24 | Moc_ | lol |
| 01:42.27 | Moc_ | I wondered |
| 01:42.48 | file[laptop] | TNC Solutions? |
| 01:42.53 | Moc_ | TMC |
| 01:42.57 | file[laptop] | ah |
| 01:43.04 | file[laptop] | working nicely actually |
| 01:43.10 | *** join/#asterisk cc (~cc@byte.fedora) |
| 01:43.22 | BoRiS | Mine just kept ringing(6164) |
| 01:44.07 | *** join/#asterisk forrestc (~fwc@206.127.78.199) |
| 01:44.32 | file[laptop] | BoRiS: I'm not even seeing you |
| 01:44.55 | BoRiS | I just hung up after the 8th ring |
| 01:46.04 | *** join/#asterisk mtqh (~chatzilla@dsl093-001-038.det1.dsl.speakeasy.net) |
| 01:46.21 | *** part/#asterisk BuddaBoy (~budda@81-86-85-233.dsl.pipex.com) |
| 01:47.42 | *** join/#asterisk zane1 (~zane1@static-64-223-94-14.burl.east.verizon.net) |
| 01:47.49 | *** join/#asterisk Zaw (hidden-use@zaw.subneural.net) |
| 01:48.27 | zane1 | hey everyone |
| 01:50.41 | BoRiS | anyone on the confi? |
| 01:51.15 | zane1 | how come i can see AGI Environment Dumps when starting asterisk with -vvvvvcg but not with any other options with -r? |
| 01:53.50 | kaitseb | kram: are you sure that there's no memory corrupting bugs in zaptel/wct4xxp drivers, my system starts to act fishy after few hours of uptime |
| 01:54.55 | Druken | your stuff is crapping out after a few hours? |
| 01:55.00 | Druken | mine takes like 20 days... |
| 01:56.20 | *** join/#asterisk shmaltz (~chatzilla@69.28.255.210) |
| 01:59.51 | Koshatul | whereis the "native format" set ? or is it just the "first allow=" in codecs ? |
| 02:01.23 | Koshatul | my iax provider uses gsm, but i'm getting errors "Dropping incompatible voice frame on SIP/200-c039 of format gsm since our native format has changed to ulaw" whenever i try to pickup with my cisco 7960 |
| 02:01.23 | zane1 | why does astcc show agi environmnet dumps when asterisk is started with asterisk -vvvvcg and not when using -vvvvr? |
| 02:03.01 | *** join/#asterisk Othello (Othello@nusnet-154-210.dynip.nus.edu.sg) |
| 02:04.57 | *** join/#asterisk criptos (~criptos@dsl-200-78-97-55.prod-infinitum.com.mx) |
| 02:05.13 | Koshatul | can i get the asterisk server to decode and recode ? |
| 02:05.14 | criptos | nightsh! |
| 02:05.45 | Druken | Koshatul: in your iax.conf in the providers user block, put in disallow = all |
| 02:05.46 | criptos | How can I emulate the "flash" to make a transfer, using sendDTMF? what is the character equivalent to flash? |
| 02:05.48 | Druken | allow = gsm |
| 02:06.10 | criptos | and what codecs does iaxy supports? only ulaw? or gsm also? |
| 02:06.28 | shmaltz | bkw_: why not? |
| 02:06.30 | Druken | criptos: flash is not a DTMF |
| 02:06.34 | *** join/#asterisk cbachman (~cbachman@129.105.7.250) |
| 02:06.49 | shmaltz | bkw_ I mean the 2905 bug. |
| 02:08.21 | criptos | that's is my cuestion Druken, then how can I emultate it? to use * as automatic op of and old phone switch... so instead of using another fisical channel, send the rigth secuence to produce the effect of telling the switch to transfer the phone |
| 02:10.21 | Druken | your not exactly making sence.... |
| 02:10.37 | Druken | slow down a bit, and tell me what you want to do |
| 02:11.13 | Druken | you want to use flash on an analog phone with asterisk ? |
| 02:11.31 | *** join/#asterisk firestrm (~vince@S010600047577bccd.gv.shawcable.net) |
| 02:13.31 | *** join/#asterisk ArkyLady (ArkyLady@93-95.hspg-ubr2-blk1.cablelynx.com) |
| 02:14.24 | Koshatul | Druken: already in there |
| 02:14.40 | Koshatul | it's the phone |
| 02:14.52 | Koshatul | cisco 7960 doesn't appear to support gsm |
| 02:15.01 | modulus_ | ip500 is really nice |
| 02:15.05 | modulus_ | my fav phone so far |
| 02:15.10 | modulus_ | speakerphone is outstanding |
| 02:15.38 | Druken | Koshatul: no i don't belive it does... however, the message is simply a warning, telling you asterisk is changing the feed to something your phone understands |
| 02:16.43 | Umaro | hey guys, I heard about someone using something other than soxmix to mix their monitored calls.. and that it put the incoming audio on the left channel, and the outgoing audio on the right channel? |
| 02:17.29 | KleinJonp | Hi people,PLEASE tell me how to specify more then 1 single ip address in sip.conf, host= ????, how do i do it? |
| 02:18.03 | *** join/#asterisk rick_ve (WCDLS-110@201.248.7.169) |
| 02:18.07 | Umaro | oh, gstreamer will do that. hmm. |
| 02:18.09 | rick_ve | hello all |
| 02:18.09 | ZX81 | host=dynamic |
| 02:18.23 | KleinJonp | has anyone connected cisco 54xx to asterisk ? what needs to be in sip.conf ? |
| 02:18.38 | rick_ve | can i ask you a question, maybe you can help me, i try search the web but is a low demand topic |
| 02:18.45 | ZX81 | ~data |
| 02:18.46 | jbot | Don't Ask To Ask. Just ASK |
| 02:18.49 | KleinJonp | ZX81: I need 4 different specific ip adresses... can i do multiple permit=xxx lines ? |
| 02:18.54 | rick_ve | jejejej thanks |
| 02:19.17 | ZX81 | one phone cannot have 4 addresses |
| 02:19.22 | modulus_ | jbot g-g-g-g? |
| 02:19.23 | jbot | G-UNIIT!! |
| 02:19.23 | ZX81 | create either 4 accoutns |
| 02:19.24 | ZX81 | or |
| 02:19.30 | rick_ve | i connect a asterisk to a PBX, to connect ip phones to regular phones and regular phones to regular phone trought talk menu |
| 02:19.30 | ZX81 | use host=dynamic |
| 02:19.40 | ZX81 | and allow the endppint to register it's address |
| 02:19.51 | rick_ve | but when i go to tranfer the call usin flash and senddtmf i got a problem |
| 02:20.22 | rick_ve | anyone do something like that or has experience to PBXs? |
| 02:21.12 | KleinJonp | ZX81: Am i right to understand, that if i have 4 different cisco gateways to connect to my *, i need a separate entry in sip.conf for each one of them ? |
| 02:21.23 | ZX81 | correct |
| 02:21.27 | ZX81 | except I'm away |
| 02:21.28 | ZX81 | :) |
| 02:21.30 | rick_ve | i see that's not the general use for arterisk :'( |
| 02:21.40 | Koshatul | Druken: i get dead air |
| 02:21.41 | KleinJonp | ZX81: Thank you |
| 02:21.43 | *** join/#asterisk illek (~knoppix@ip68-14-167-89.ok.ok.cox.net) |
| 02:21.44 | ZX81 | np |
| 02:22.49 | rick_ve | do you know any software that allow a web windows run a IAX software so can make VoIp call without the software installet on PC' |
| 02:23.46 | jayden | rick_ve: what are you talking about? slow down and try to explain your environment |
| 02:24.38 | rick_ve | all right, one of the advantage of asterisk it's to make VoIP calls |
| 02:24.48 | rick_ve | if you have a laptop can use iaxcomm diax etc |
| 02:24.55 | jayden | sure |
| 02:25.47 | rick_ve | i'm looking for a program, software, X that avoid to install the software and can make VoIp call trought web, like: i open web browser and connect www.voip.edu.ve display a phone and make VoIp call to the extensions! |
| 02:25.51 | Koshatul | Druken: my bad |
| 02:26.07 | Koshatul | Druken: ID10T error, you have to hit # to accept the call |
| 02:26.17 | jayden | xten has a web version... not sure about cost. |
| 02:26.32 | rick_ve | all right let me check, thanks a lot for the information |
| 02:26.34 | Umaro | rick_ve: xten xweb |
| 02:26.34 | jayden | beleive it requires activex (or maybe java) |
| 02:26.35 | Koshatul | Druken: that's why the dead air |
| 02:26.43 | Umaro | pretty cool tho |
| 02:27.00 | wankel | oooh, postgres 8.0 out today |
| 02:27.34 | Umaro | when are we gonna see a native asterisk realtime pgsql? :) |
| 02:27.56 | Umaro | or is the odbc driver sufficient? |
| 02:28.17 | wankel | odbc seems to work |
| 02:28.17 | BoRiS | It uses odbc |
| 02:28.30 | wankel | well enough that i won't bother to add native pgsql support |
| 02:28.41 | wankel | i'll probably end up adding native oci support when i have a chance |
| 02:30.33 | Moc_ | I get to get a hold of Fabe .. |
| 02:30.38 | BoRiS | The only problem I am having with postgres + odbc + realtime is when my database goes down, asterisk/odbc does not reconnect for some reason |
| 02:32.00 | wankel | ugh |
| 02:32.03 | wankel | that doesn't sound good |
| 02:32.17 | BoRiS | No, its not :( |
| 02:32.53 | freat | anyone else have problems with Teliax today? or is it just us... |
| 02:32.55 | rick_ve | ohhh i have another question, anyone can recommend me a VoIp providers that accept conection from my asterisk server????? to PSTN US |
| 02:33.09 | *** join/#asterisk FryGuy- (fryguy@c-67-174-57-164.client.comcast.net) |
| 02:33.53 | freat | kinda sucks... the PRI for originating our number is down |
| 02:33.55 | KleinJonp | Say, does anybody know why accountcode doesn't get written into cdr, if not using username/secret in sip.conf ? |
| 02:34.19 | *** join/#asterisk Evanrude (~david@ip68-96-125-184.lu.dl.cox.net) [NETSPLIT VICTIM] |
| 02:34.19 | *** join/#asterisk ard (ard@gw-uunet-office.telegraaf.net) [NETSPLIT VICTIM] |
| 02:34.19 | *** join/#asterisk Makenshi (~makenshi@2001:630:1c0:2001:280:c8ff:fee2:921f) [NETSPLIT VICTIM] |
| 02:34.19 | *** join/#asterisk psycodad_ (~obiwan@2001:4060:4419:b1:0:0:0:2) [NETSPLIT VICTIM] |
| 02:34.20 | *** join/#asterisk ncjp (~switch@61.206.115.5.user.ad.il24.net) [NETSPLIT VICTIM] |
| 02:38.02 | *** join/#asterisk jterrero (~jterrero@66.28.34.177) |
| 02:41.24 | jterrero | pretty silent in here |
| 02:43.28 | *** join/#asterisk BuzzBud (~adoroar@90.62-97-254.bkkb.no) [NETSPLIT VICTIM] |
| 02:43.28 | *** join/#asterisk Brumle (~brumle@brumle.com) [NETSPLIT VICTIM] |
| 02:43.28 | *** join/#asterisk KryoStoffer (~kri@helium.kri.dk) [NETSPLIT VICTIM] |
| 02:43.52 | KleinJonp | Does anybody know how to force writing "accountcode" into cdr ? |
| 02:44.22 | criptos | How can I make * to make aacll trasfer in ax100p card connectd to a analog swith |
| 02:45.49 | modulus_ | <PROTECTED> |
| 02:45.49 | modulus_ | <PROTECTED> |
| 02:45.53 | modulus_ | 103 % idle |
| 02:45.54 | modulus_ | haha |
| 02:46.08 | modulus_ | freebsd is ninja |
| 02:46.31 | redder86 | criptos: you want to transfer back out the same analog line? |
| 02:47.54 | gambolputty | Can * use another computer as an outbound SIP proxy? |
| 02:48.16 | criptos | yeap... as if the x100p was a analog phone and a user were dialing a transfer atthe switch |
| 02:49.24 | redder86 | criptos: does your telco allow you to transfer calls back out of your line? |
| 02:50.01 | criptos | redder, i''m behind avery oldphone analog swith and I want to create adiigtal operator with * |
| 02:50.57 | redder86 | criptos: okay, but since I don't know, can you tell me how you transfer a call from an oldphone analog phone? |
| 02:51.12 | criptos | so when u call the extennsio 0 or deoepraton rings and I want to act as the operator... So if you want to call to "cripts office" you dial 1 and the box transfer the call from the oepator to the eextnsion 10 at the analog swith... |
| 02:52.09 | redder86 | okay, I understand, but how do you do that without asterisk? |
| 02:53.37 | criptos | you have a people at extension *, she/he answer, and you tell her "please with mr criptos" and then he/she flash and dial 10 and then hanghup. |
| 02:53.44 | criptos | so, mi phone rings.. |
| 02:54.11 | criptos | at extension 0 i mean... |
| 02:54.43 | redder86 | sigh |
| 02:54.51 | redder86 | how do you control the switch? |
| 02:55.26 | criptos | with a normal phone... |
| 02:55.29 | criptos | ? |
| 02:56.02 | redder86 | if you don't know how to control the switch how are we supposed to tell you to make Asterisk control it? |
| 02:57.27 | *** join/#asterisk Corydon76-home (orange@pcp08665860pcs.500ash01.tn.comcast.net) |
| 02:57.34 | criptos | you use the phone.. any analog phone. You can sumulate the flash with a fast hangup/hookoff and then dial the extension... |
| 02:58.15 | criptos | this is what I do to make the transfer... in "newer" phone, still analog,I can use the flash button andd then dial the extension.. |
| 02:58.28 | redder86 | so you want to know how to get Asterisk to flash the analog line through the X100P? |
| 02:58.33 | criptos | The switch is ameribrand brand... |
| 02:59.10 | criptos | Iyeap redder.. |
| 02:59.49 | *** join/#asterisk libpcp (libpcp@210.16.20.5) |
| 03:00.17 | eschvoca | In * can you pattern match an extension inside of a macro? |
| 03:00.36 | eschvoca | So the callers extension is matched? Or is the "s" extension always used? |
| 03:00.56 | redder86 | criptos: make sure transfer=yes in zapata.conf, and you should be able to do it |
| 03:01.01 | criptos | any ideas redder? the send the digits, well i can use senddtmf() but the "flash" part is what I have it missing.. |
| 03:01.20 | criptos | yeap... and how I make that * "send" this flash? |
| 03:02.23 | redder86 | criptos: it should do it automatically when your extensions transfer to an extension that the dialplan recognizes is through the X100P, I would think. |
| 03:02.38 | twisted | OMFG |
| 03:02.39 | twisted | http://a.as-us.falkag.net/dat/bgf/200501/02/bv8.jpg |
| 03:02.49 | BoRiS | twisted: come join us :) |
| 03:03.05 | criptos | but actually the dialplan knows nothing about the extensions at the analig switch... |
| 03:03.36 | criptos | I havee something like this: |
| 03:03.41 | criptos | s,1,Answer |
| 03:04.00 | criptos | s,2,Playback("to-whom-tal") |
| 03:04.06 | criptos | and then |
| 03:04.16 | criptos | (me is option "1") |
| 03:04.17 | Veto | that was painless, phone -> NAT'ed firewall -> firewall -> *, had to open a range for RTP though, 100 ports enough for a small shop? |
| 03:04.44 | *** join/#asterisk andrew` (~andrew@adsl-67-119-26-96.dsl.snfc21.pacbell.net) |
| 03:05.21 | *** join/#asterisk jbot (ibot@apt.bot.TimRiker.active.supporter.pdpc) |
| 03:05.21 | *** topic/#asterisk is Asterisk: The Open Source PBX || DUNDi: http://www.dundi.com #dundi || http://it.slashdot.org/it/05/01/12/1829240.shtml?tid=215&tid=218 |
