| 00:00.02 | *** join/#asterisk jpablo (~jpablo@host-148-244-137-95.block.alestra.net.mx) |
| 00:00.10 | ckruetze | You can use shielded Cat5 for E1, so it should also work for T1 |
| 00:00.11 | clnc | t1 is supposed to be sheilded pairs |
| 00:00.19 | tzanger | it clnc says who? |
| 00:00.29 | tzanger | I don't think any trunk in the world is shielded pair |
| 00:00.32 | AgiNamu | For a very short run, maybe it'll work. Lots of people have problems when they just go use Cat5 for T1. |
| 00:00.32 | tzanger | it's all CAT3 |
| 00:00.34 | tzanger | UTP |
| 00:00.36 | outtolunc | i've run t1 feeds over 100' on cat3 |
| 00:00.47 | clnc | <PROTECTED> |
| 00:00.47 | tzanger | T1 can go I think 5000' on CAT3 |
| 00:01.03 | outtolunc | depends on teh CSU/DSU |
| 00:01.21 | tzanger | it's not DSX1 mind you like the T100P and any small-span equipment but typically speaking T1 is 5000' on CAT3 |
| 00:01.25 | tzanger | with no repeaters |
| 00:01.30 | tzanger | for more distance you use repeaters |
| 00:02.25 | tzanger | like I said trunks are not shielded pair |
| 00:02.30 | *** join/#asterisk mxmasster (~mxm@rottie.media.net) |
| 00:02.32 | mxmasster | hi all |
| 00:02.37 | tzanger | and T1 has run over it for over 40 years |
| 00:02.42 | mxmasster | so, i've hit a wall |
| 00:02.45 | tzanger | well that was D1, which is even harder to get to run :-) |
| 00:02.54 | mxmasster | i've been trying to install the AMP software based on their psd |
| 00:02.56 | mxmasster | err pdf |
| 00:03.03 | clnc | so within the building, from a te card on * to a switch, cat3 is ok? You guys do it that way? |
| 00:03.04 | KalD|Work | mxmasster, you ok? did you hurt the wall? |
| 00:03.11 | tzanger | I'm willing to bet you can't get the run to go because you're splitting the pair |
| 00:03.19 | tzanger | pair 1 is pins 4&5 |
| 00:03.20 | mxmasster | gone through the entire installation, and when i try to start asterisk i get a segmentation fault and a core file |
| 00:03.22 | tzanger | pair 2 is pins 3&6 |
| 00:03.23 | Moc_ | anyone used chan_oh323 instead of chan_h323 ? |
| 00:03.26 | *** join/#asterisk criptos (~leonardox@201.128.220.198) |
| 00:03.26 | tzanger | pair 3 is pins 1&2 |
| 00:03.27 | mxmasster | i'm running whel 3.0r1 |
| 00:03.29 | tzanger | pair 4 is pins 7&8 |
| 00:03.31 | criptos | HI! |
| 00:03.36 | Moc_ | hi |
| 00:03.46 | tzanger | I bet you're running it pair 1 = 1&2, 2=3&4, 3=5&6, 4=7&8 |
| 00:03.58 | mxmasster | how do i figure out what is causing asterisk to core dump? |
| 00:03.59 | tzanger | and you're splitting pairs 2 and 3 when you do that, which are the two pairs that T1 uses |
| 00:04.04 | criptos | what codecs are supported my the iaxy device? I provisioned it with codec=iLBC and it was unable to connect my *box |
| 00:04.09 | tzanger | mxmasster: gdb -c corefile `which asterisk` |
| 00:04.22 | tzanger | then run 'bt' in gdb and post that along with any relevant info to asterisk-users |
| 00:04.50 | Brixius | criptos: Read the docs you got with it again |
| 00:04.53 | redder86 | criptos: I think it only supports ulaw |
| 00:04.59 | ta[i]nted | bkw_: you there? |
| 00:05.14 | mxmasster | tzanger, when i run bt |
| 00:05.16 | mxmasster | i get three lines |
| 00:05.26 | mxmasster | /lib/tls/libpthread.so.0 |
| 00:05.33 | mxmasster | and /lib/tls/libc.so.6 |
| 00:05.37 | criptos | htnaks redder86 |
| 00:05.49 | ta[i]nted | bkw_: i basically have a .call file with "Channel: Local/foo@bar", and in my extensions.conf under [bar] i have Dial(IAX2/client) |
| 00:06.26 | ta[i]nted | bkw_: the incompatible voice frame drops occur on DIAL(IAX2/client) |
| 00:06.47 | mxmasster | tzanger, is there anything i can do above and beyond asking the mailing list? |
| 00:10.39 | *** part/#asterisk shepherd (~nobody@dhcp-157.digium.com) |
| 00:10.45 | *** join/#asterisk doughecka_ (~Doug@adsl-18-113-209.sdf.bellsouth.net) |
| 00:13.07 | *** part/#asterisk mogorman (~mogorman@216.207.244.182) |
| 00:13.15 | *** join/#asterisk Bentley (~bentley@S01060080c8135e6a.cg.shawcable.net) |
| 00:15.25 | *** join/#asterisk expressfone1 (expressfon@80.27.3.23) |
| 00:16.06 | *** part/#asterisk Pinhole (~nuxi@lin-dsl-static-206-222-194-115.inetnebr.com) |
| 00:17.13 | *** join/#asterisk BoRiS (~boris@S01060040ca1e5b54.wp.shawcable.net) |
| 00:17.41 | Moc_ | I emailed someone at polycom, I hope I can get in contact in some cool Polycom guy |
| 00:18.56 | ST-3 | has anyone seen a phone trying to register that isn't configured in * crash *? |
| 00:19.06 | Moc_ | nope |
| 00:19.09 | ST-3 | hrm |
| 00:19.10 | Moc_ | sip phone ? |
| 00:19.12 | ST-3 | yah |
| 00:19.20 | Moc_ | donno |
| 00:19.23 | ST-3 | i got a sipura and a ata186 to do it |
| 00:19.26 | *** join/#asterisk buddah (~hnic@208.179.86.5) |
| 00:19.33 | ST-3 | typoed the username on the sipura |
| 00:19.46 | ST-3 | *poof* |
| 00:19.47 | expressfone1 | i´m selling Cuba +53 Termination (IAX2) $0.50USD/min for this weekend |
| 00:20.06 | Moc_ | and ? |
| 00:20.10 | buddah | should sell me cuba |
| 00:20.14 | ST-3 | expressfone1: Cool, do you happen to have the phone # for the guy in Cuba that is selling the little mini cows? |
| 00:20.38 | expressfone1 | sell to any one will pay it |
| 00:20.38 | buddah | i need some cigars too |
| 00:20.45 | ST-3 | http://msnbc.msn.com/id/5933388/ |
| 00:21.01 | ST-3 | New mini cows perfect for milk? |
| 00:21.01 | ST-3 | Cuban rancher breeds bovines no larger than dogs |
| 00:21.07 | buddah | omg |
| 00:21.51 | ST-3 | get me his #, I will buy some minutes |
| 00:21.51 | buddah | you were serious about the mini cows |
| 00:21.51 | ST-3 | buddah: YES :) |
| 00:21.51 | expressfone1 | Cuban cigars... y have big one here |
| 00:21.51 | buddah | omg |
| 00:21.51 | buddah | that is great |
| 00:21.51 | ST-3 | hahaha |
| 00:21.51 | buddah | i want one |
| 00:21.51 | *** join/#asterisk C11- (~jonas@p50837675.dip0.t-ipconnect.de) |
| 00:21.51 | buddah | maybe even 2 |
| 00:21.51 | buddah | bet they make great stocking stuffers |
| 00:21.51 | buddah | so i'm having this problem here at work |
| 00:21.51 | buddah | trying to register a phone with sip |
| 00:21.55 | buddah | and i'm getting a damn 403 |
| 00:22.17 | buddah | a guy at digium said the config looks fine and he thinks the problem is somethings blocking all the ports from going through |
| 00:22.31 | buddah | but the phones on a public ip, and not behind a firewall or anything, so nothing should be blocking |
| 00:23.01 | ManxPower | buddah: some distros install afirewall by default. "service firewall stop" |
| 00:23.05 | buddah | anyone maybe familiar with that or that could help? |
| 00:23.21 | buddah | so run that on the asterisk box? |
| 00:23.24 | buddah | service firewall stop |
| 00:23.25 | buddah | ? |
| 00:23.36 | KalD|Work | buddah, what distro do you have? |
| 00:23.42 | buddah | i'm probably the wrong person trying to get this fixed, but my boss is a schmooze |
| 00:23.47 | ManxPower | buddah: "ls /etc/rc.d/init.d" and see if iptables, ipchains, or firewall is listed. |
| 00:23.54 | buddah | RH9 |
| 00:23.54 | ST-3 | buddah: what kind of phone? |
| 00:24.04 | KalD|Work | buddah, type this: iptables -F |
| 00:24.08 | buddah | and a polycom soundpoint ip 500 |
| 00:24.10 | ST-3 | hrm |
| 00:24.14 | ST-3 | I had a similar problem with a Sipura |
| 00:24.18 | buddah | k just a sec gotta re log in |
| 00:24.23 | Moc_ | POLYCOM ROCK ;) |
| 00:24.23 | ManxPower | but if you get a 403 then it's a username/password issue |
| 00:24.23 | ST-3 | had to nuke the config and do it over |
| 00:24.55 | Moc_ | except no SIP firmware for SoundStation 3000 :( |
| 00:24.55 | Malthus | oh |
| 00:25.05 | Moc_ | none public anyway |
| 00:25.10 | Malthus | the SIP errors are the same as for HTTP? |
| 00:25.30 | ManxPower | Malthus: Similar |
| 00:25.30 | Damin | Anyone know how to enable pattern based auto-dial on Polycoms? |
| 00:25.32 | KalD|Work | Malthus, close - but not same |
| 00:25.43 | buddah | iptables is listed on ls /etc/rc.d/init.d |
| 00:25.47 | Damin | I'm working with the dialplan maps, but I don't seem to be getting anywhere.. |
| 00:25.50 | Malthus | cool |
| 00:25.57 | ManxPower | Damin: I read the docs. In fact there is a message on the mailig list talking about it too |
| 00:26.05 | KalD|Work | buddah, if this is a username/pw issue don't take down your firewall just yet... |
| 00:26.14 | Damin | ManxPower: I posted the message! ;) But it's not working.. |
| 00:26.21 | Damin | ManxPower: I'm missing something.. |
| 00:26.29 | Malthus | are the Digium TDM cards as flaky as the Slashdot comments (from the story in the topic) suggests? |
| 00:26.37 | expressfone1 | any one here from miami?? |
| 00:26.37 | *** part/#asterisk telme_ (~trillian@sta-207-174-139-178.rockynet.com) |
| 00:26.40 | buddah | KalD|Work: i figured, i just dont see why the u/p isnt transmitting right |
| 00:26.40 | bkw_ | Malthus no |
| 00:26.44 | mikegrb | pensacola |
| 00:26.45 | bkw_ | just the TDM400 |
| 00:26.53 | Damin | ManxPower: Basically, I have this in phone100.cfg: dialplan.1.digitmap="[1-8]xxx|[2-9]xxxxxx|*xx|91xxxxxxxxxx|1[2-9]xxxxxxxxx|[2-9]xxxxxxxxx|911|9911|411|9411|2 |
| 00:26.53 | Damin | 11|9211" |
| 00:26.54 | bkw_ | the T1/E1 stuff is pretty rock solid |
| 00:27.06 | Moc_ | Damin, yes |
| 00:27.14 | file | it's analog, can't expect miracles |
| 00:27.16 | ManxPower | Damin: The FIRST match is what's used. |
| 00:27.21 | Malthus | bkw_ : its the TDM one's I'm talking about |
| 00:27.31 | bkw_ | ya they can be flakey |
| 00:27.35 | bkw_ | if you use a 25 dollar motherboard |
| 00:27.43 | file | I LOVE ANALOG |
| 00:27.44 | file | OMG ANALOG |
| 00:27.44 | Damin | ManxPower: So what does that mean? :) |
| 00:27.48 | Malthus | everywhere says the T1/E1 cards are ok |
| 00:27.48 | file | HOLD ME HOLD ME HUG ME HUG ME |
| 00:27.53 | bkw_ | ANALOG has the world Anal in it |
| 00:27.54 | Damin | ManxPower: Shortest first? |
| 00:27.55 | bkw_ | nuf said |
| 00:27.58 | Malthus | I _HATE_ analog |
| 00:28.02 | file | bkw_: yeah and you like it |
| 00:28.03 | file | ooh la la |
| 00:28.07 | expressfone1 | mikegrb>if you know any one on termination reselling business, +53 for .50 can be profit |
| 00:28.19 | bkw_ | file we should be on the same flight from denver to sjc |
| 00:28.21 | file | bkw_: my plane ticket situation has been sorted out, Dave is getting it |
| 00:28.21 | Malthus | are there any better analog cards? |
| 00:28.23 | bkw_ | ArkyLady get on MSN grrl |
| 00:28.31 | bkw_ | file kewl |
| 00:28.43 | Damin | Hmmm.. |
| 00:28.44 | Damin | Alright.. |
| 00:28.59 | Malthus | I need a setup for a 1/2 line config |
| 00:29.06 | Damin | So.. It appears that DialPlan matching doesn't work when you use the "New Call" soft key. |
| 00:29.21 | Malthus | I can't go with a PRI/T1 for just one line |
| 00:29.23 | file | bkw_: so you should be able to take care of transit from 'da airport to hotel? |
| 00:29.24 | ManxPower | I mean [1-8]xxx|1xxxxxxxxxx will match the 4 digit extension |
| 00:29.41 | file | IT ONLY TAKES A MINUTE GIRL, TO FALL IN LOVE |
| 00:29.54 | waddy | where can i get app valetparking? |
| 00:30.00 | Damin | ManxPower: So everything after that doesn't mean squat? |
| 00:30.10 | AgiNamu | Hey, I have a patch for preferred codec when dialing, lets you set a channel var, and that'll be preferred. Works for both SIP and IAX |
| 00:30.21 | AgiNamu | Can someone review it for me? tell me if I fscked up ? It's my first patch. |
| 00:30.23 | ManxPower | Correct. The ORDER matters a lot, as does planning. |
| 00:30.33 | Damin | AgiNamu: Great. Now make it work on the Atari 2600. :) |
| 00:30.37 | AgiNamu | i dont want negative karma on my first shot |
| 00:30.48 | Damin | ManxPower: Great.. Well.. Hmm.. |
| 00:31.02 | ManxPower | Damin: Want my polcom dial plan? |
| 00:31.06 | redder86 | Is there a way to make it so that a voicemail user cannot change their password and voice greetings? |
| 00:31.09 | Damin | ManxPower: Sure.. |
| 00:31.42 | Damin | ManxPower: Can't the Polycom fall through and re-try on the next match though? I.E. if 4 digit doesn't work, fall through and match 7 digit, then 10? |
| 00:31.42 | ManxPower | 9,1[2-9]xxxxxxxxx|9,[2-9]xxxxxx|*xx|0|[2-8]xxx|9,911||911|xxxx. |
| 00:32.10 | ManxPower | Damin: I highly doubt it. |
| 00:32.24 | Malthus | theme cl |
| 00:32.27 | Malthus | oops |
| 00:32.32 | Damin | ManxPower: That's fucked.. Grandstream checks every digit.. Now their's doesn't WORK, but it is interesting.. |
| 00:32.33 | ManxPower | Damin: Any PBX admin will tell you that dialplans look REALLY simple but are actually REALLY complex |
| 00:33.05 | BoRiS | bkw!!! file!!! |
| 00:33.25 | bkw_ | BoRiS |
| 00:33.42 | BoRiS | :)))) |
| 00:33.45 | BoRiS | Whats up? |
| 00:33.53 | Malthus | any reccomendations for a single or dual FXO device |
| 00:34.03 | *** join/#asterisk nikko (~chatzilla@12-218-243-173.client.mchsi.com) |
| 00:34.04 | Malthus | or any definte no nos? |
| 00:34.11 | ManxPower | Damin: If you CANNOT avoid overlapping patterns, end the overlapping shorter one with a T (timeout) |
| 00:34.22 | redder86 | Malthus: an X100P ? |
| 00:34.22 | file | BoRiS!!!!!!!!! |
| 00:34.24 | Malthus | TCP or PCI |
| 00:34.41 | BoRiS | :))) |
| 00:34.42 | kuj | Damin, don't know how the Polycom implements it, but the rfc said somewhere that as soon as there's a complete match with any pattern it is sent off to the call agent |
| 00:34.43 | *** join/#asterisk fifer_ (~fifer@207.202.227.161) |
| 00:34.52 | Damin | ManxPower: For now, I'm just going to use "[2-9]xxxxxxxxx" ;) |
| 00:34.56 | *** join/#asterisk buddah (~hnic@208.179.86.5) |
| 00:35.00 | bkw_ | I use (x.) |
| 00:35.08 | ManxPower | make them dial 9 for outside line |
| 00:35.09 | bkw_ | in my sipura |
| 00:35.22 | Damin | ManxPower: No. That would confuse them. They are idiots.. |
| 00:35.25 | *** join/#asterisk MikeFRandT (~MikeFRand@70.150.212.100) |
| 00:35.31 | kuj | i worked around it by adding a timeout to my patterns |
| 00:35.33 | Malthus | redder86 : I read on viop-info and the mailing lists that echo is worse on the X100 than the TDM4xx, you know if that is true? |
| 00:35.37 | *** join/#asterisk sturothrock (~stuart@243-73.8-67.tampabay.rr.com) |
| 00:35.41 | Damin | ManxPower: They can't even handle using a Fax machine, much less be trusted to dial a phone. |
| 00:35.46 | redder86 | I don't make my users dial any "9" like prefixes. It's just annoying. |
| 00:36.05 | PTG123 | redder: any update on that fax driver? :) |
| 00:36.11 | ManxPower | Damin: We have related users |
| 00:36.12 | MikeFRandT | Hello all. I just installed asterisk on a fresh linux system with a voice/data/fax modem within it. Is there any advice that I can get on configuration and functions that can make this useful? |
| 00:36.14 | redder86 | Malthus: I don't have a TDM4xx, but I do have an X100P, and I don't have echo problems. |
| 00:36.15 | bkw_ | redder86 i'm with ya |
| 00:36.17 | bkw_ | its anoying |
| 00:36.27 | redder86 | PTG123: I haven't seen Steve mention anything. |
| 00:36.37 | ManxPower | I make all my users dial 9. Better than the 5 - 7 second time I would need for dialing 4 digit extensions |
| 00:36.42 | Damin | ManxPower: Dude.. If you use the New Call Softkey, or just dial the number, pattern matching DOES NOT WORK! Am I missing some other f***ed up xml config parameter now? |
| 00:36.48 | PTG123 | redder86: lame, hylaxfax supports T.38 already right? |
| 00:36.59 | Malthus | redder86 thanks |
| 00:37.06 | redder86 | PTG123: not directly, but through t38modem yes |
| 00:37.13 | Damin | ManxPower: You have to either pick up the phone, or select one of the lines for it to work.. |
| 00:37.19 | lters | fxstest - how does that work? |
| 00:37.36 | bkw_ | redder86 how well does t38modem work? |
| 00:37.46 | Damin | bkw_: It rocks! :) |
| 00:37.57 | nikko | does anyone know if NuFone supports iax trunking? |
| 00:37.58 | redder86 | bkw_: I've used it a number of times - faxing between t38modems, and it works well |
| 00:38.06 | redder86 | bkw_: I've never used it through a gateway, though. |
| 00:38.09 | ManxPower | Damin: I don't know. The polycom has been deployed for a while. |
| 00:38.12 | bkw_ | now if we just had something asterisk could use |
| 00:38.13 | Damin | nikko: Nufone supports IAX2 trunking just fine. |
| 00:38.18 | BoRiS | What supports T38? |
| 00:38.24 | redder86 | t38modem |
| 00:38.28 | bkw_ | can't use t38modem with sip ? |
| 00:38.28 | nikko | thanks - |
| 00:38.32 | Damin | BoRiS: Everything on the planet. :) |
| 00:38.35 | redder86 | OpenH323's t38modem |
| 00:38.49 | BoRiS | So much talking about T38 |
| 00:38.55 | nikko | I'm building a conference bridge, and am having some BW issues |
| 00:38.58 | redder86 | bkw_: my understanding is that t38modem uses H.323 |
| 00:38.58 | Damin | ManxPower: You have one sitting in front of you? |
| 00:39.13 | MikeFRandT | I'm a newbie with astrisk can someone kind of devote about 15 to 30 mins with me here? |
| 00:39.15 | Damin | Moc_: What about you? You seem to be a Polycom dude.. |
| 00:39.17 | nikko | i enabled trunking, but wasn't sure if it would work or not |
| 00:39.29 | Damin | Moc_: BTW: I got the 1.3.4 SIP load and it solved the other issues I was having.. |
| 00:39.44 | redder86 | Damin: for me too |
| 00:39.46 | Damin | So if anyone wants the Polycom SIP 1.3.4 archive, let me know.. |
| 00:40.00 | kuj | i want |
| 00:40.37 | nikko | MikeFRandT - what are you trying to do? |
| 00:40.45 | bjohnson | I'm having trouble finding anything on the wiki about using the cdr systems .. where can I find info about cdr? |
| 00:41.16 | ManxPower | The Polycoms I had have been deployed out to users for a while. |
| 00:41.30 | MikeFRandT | nikko: I have a voice/data/fax modem, nice network and 100% pure linux systems on all computers in the network. I'm thinking about using Astrisk on the voice modem and route the calls to all the system with voip |
| 00:41.33 | redder86 | What I'm not clear on is whether T.38 would be described as a channel type or a codec type. |
| 00:41.41 | kuj | Damin, try adding a "T" (interdigit timeout) to your dialmap. works perfectly for me. Like so: *xxT*xxx.T |
| 00:41.51 | *** join/#asterisk juice (~juice@mo-65-41-96-199.dyn.sprint-hsd.net) |
| 00:42.16 | kuj | sorry, *xxT|*xxx.T |
| 00:42.26 | Damin | kuj: What is that supposed to do? |
| 00:42.27 | nikko | the voice modem is 99.5% guranteed not to work for this. where will inbound calls be coming from? |
| 00:42.54 | Damin | kuj: Will that allow the dialplan to work with the softkey? |
| 00:42.57 | redder86 | MikeFRandT: what kind of "voice/data/fax modem" ? |
| 00:43.37 | ManxPower | Damin: The Polycom admin guide points you to the Dialplan section of the PGCP RFC. |
| 00:43.54 | kuj | does for me. the issue is that the poly appears to send the dialstring to * as soon as it matches the first pattern in the digitmap |
| 00:43.56 | MikeFRandT | nikko: We have 4 lines comming in so far, and the phone systems are a "AT&T Small Business System" basically using anylog and transfers calls between phones with a paging feature via ultra high frequencies like a DSL connection would be used between you and the DSLAM |
| 00:43.56 | *** join/#asterisk soundguy (~soundguy@zeus.soundguy.id.au) |
| 00:44.22 | Damin | ManxPower: Which I've read, and which I posted to the asterisk-users list.. Doesn't mean I actually understand what I read though.. ;) |
| 00:44.55 | kuj | the "T" waits for a timeout, allowing users to add numbers even after the first match |
| 00:45.00 | MikeFRandT | nikko: I don't have any fancey hardware besides that voice modem. It is a 4 line card, capable of handling 4 lines at one time. |
| 00:45.44 | Damin | kuj: Is that the timeout value that is specifed in dialplan.1.digitmap.timeOut |
| 00:45.50 | MikeFRandT | There is "Microshaft" programs that work with that card like FaxTalk Messenger Pro, but that is Microshaft, I don't like making myself more stupid by using that program. |
| 00:46.04 | *** join/#asterisk taller (~dfghdfhg@wsip-24-234-218-94.lv.lv.cox.net) |
| 00:46.09 | nikko | MikeFRandT - you have a few choices to make - combine thse lines onto a T-1 and get a cheap T-1 card, or get a quad FXO card and plug your existing lines into that |
| 00:46.13 | redder86 | FaxTalk isn't made by Microsoft. |
| 00:46.33 | kuj | haven't modified the length of the timeout myself. let me check |
| 00:46.42 | taller | has anyone used the asterisk realtime b4? |
| 00:46.42 | Damin | kuj: So in otherwords, I could do "[2-9]xxxTxxTxxxxT"? |
| 00:46.48 | redder86 | FaxTalk is distributed by Thought Communications. |
| 00:46.56 | taller | has anyone used the asterisk realtime b4? |
| 00:47.01 | taller | it's a bich |
| 00:47.07 | redder86 | FaxTalk is not a PBX, it's an answering machine, fax machine, etc thing. Not a PBX. |
| 00:47.28 | ManxPower | Damin: More like [1-8]xxxT|[2-9]xxxxxx |
| 00:47.37 | kuj | still need to do the "|" in between. try adding the "T" to each of your patterns in your dialplan |
| 00:47.40 | nikko | I use a T-1 and a T100P from digium, nd bring in 4 analog lines over the T-1 |
| 00:48.00 | nikko | then asterisk manages the lines with the ZAP driver |
| 00:48.08 | nikko | Zaptel, that is |
| 00:48.37 | nikko | do you plan on your # of voice lines increasing over time? |
| 00:49.15 | nikko | You can also go with a pure VOIP solution by getting service from an asterisk friendly VOIP provider |
| 00:49.50 | nikko | and all traffic inbound and outbound is via an internet connection |
| 00:50.07 | kuj | Damin, looks like I DID set my timeout in the digitmap.timeOut variable. Set mine to 3 (seconds), so the phone will give me that much time to type the next digit |
| 00:50.30 | *** join/#asterisk Ad-Hoc (~ad-hoc@ppp3-adsl-252.ath.forthnet.gr) |
| 00:50.37 | Ad-Hoc | hello |
| 00:50.44 | kuj | downside is of course that users might have to wait that long for the phone to send the dialed number to * |
| 00:50.48 | nikko | I also use this solution with NuFone, and do outbound calling and Confernce calls via NuFone |
| 00:51.44 | *** join/#asterisk Guest27745 (~ASLII__@n219078223098.netvigator.com) |
| 00:52.06 | nikko | MikeFRandT - that enlighten you any? |
| 00:53.34 | *** join/#asterisk expressfone1 (expressfon@80.27.1.16) |
| 00:53.41 | *** join/#asterisk ArkyLady (nat32@93-95.hspg-ubr2-blk1.cablelynx.com) |
| 00:53.48 | Damin | kuj: Still bugs me that the damn thing doesn't work when you hit "new call" |
| 00:53.54 | Ad-Hoc | does anynone know anything about this kind of message? |
| 00:54.12 | Ad-Hoc | Jan 15 02:14:30 NOTICE[11404]: chan_iax2.c:5718 socket_read: Rejected connect attempt from 65.39.205.121, request 'xxxxxx@fromiaxfwd' does not exist |
| 00:54.22 | Damin | holy shit.. |
| 00:54.24 | Damin | Now it works.. |
| 00:54.25 | Damin | WTF? |
| 00:54.28 | Ad-Hoc | about the request |
| 00:54.37 | kuj | added the "T"? |
| 00:54.40 | Damin | No.. |
| 00:54.45 | Ad-Hoc | incoming call |
| 00:54.47 | expressfone1 | any ss7 solution for *?? |
| 00:55.03 | Damin | kuj: Here is what I'm using: [2-9]xxxxxxxxx|911|9911|1xx|8500|85100|411|9411|*xx |
| 00:55.32 | *** join/#asterisk dizzydiffi (~diseyi@adsl-70-240-183-99.dsl.hstntx.swbell.net) |
| 00:55.33 | *** join/#asterisk Othello (Othello@nusnet-156-21.dynip.nus.edu.sg) |
| 00:55.55 | MikeFRandT | Sorry nikko I was away for a moment |
| 00:56.13 | ManxPower | Damin: You have the "911" pattern too, I see. I call that the "too upset to remember to dial 9" pattern. |
| 00:56.21 | kuj | Damin, that's cause you eliminated the "overlap" from your old dialmap |
| 00:56.34 | dizzydiffi | hello peps |
| 00:56.39 | Damin | kuj: Well.. No.. now it doesn't work.. WTF? |
| 00:56.52 | Damin | kuj: Holy shit.. |
| 00:56.59 | dizzydiffi | is this the asterisk irc |
| 00:57.04 | Damin | kuj: I think I might have found a firmware bug.. |
| 00:57.11 | kuj | in your old map, you had e.g. "[1-8]xxx|[2-9]xxxxxx" |
| 00:57.12 | MikeFRandT | Nikko: You are saying you have to have a T-1 to get Astrisk to work? I have a friend that uses Astrisk over a regular phone line :S... How is this? |
| 00:57.18 | ManxPower | dizzydiffi: yes |
| 00:57.32 | dizzydiffi | great |
| 00:57.38 | nikko | no, you may have missed part of my messages |
| 00:57.44 | Damin | kuj: Just dialing the number, it doesn't auto-call.. |
| 00:58.07 | Damin | kuj: I have to press "New Call" or Pick Up the handset, or press a line button or press the headset button. |
| 00:58.11 | kuj | when trying to dial 7234567 it would match the 4-digit pattern (7234) and send that to * |
| 00:58.12 | nikko | you can use pots lines as well, you just need an interface for asterisk - your voice modem won't work |
| 00:58.14 | dizzydiffi | hey manxpower |
| 00:58.32 | MikeFRandT | What kind of interfaces are you talking about? |
| 00:58.34 | dizzydiffi | have u any experience in CDR |
| 00:58.35 | Damin | kuj: It appears that it only does "match and send" w/ the dialplan if there is a dial-tone.. |
| 00:59.18 | Damin | Holy shit I have to get out of here.. |
| 00:59.18 | kuj | you set it up for multi-line? i have it registered only on one line |
| 00:59.31 | Damin | kuj: Yeah.. I'm registerd to the same extension on both lines.. |
| 00:59.43 | Damin | kuj: Let me try killing the second line registration.. |
| 00:59.45 | nikko | well, digium sells several, depending on what you want to do. a TDM400P will accept 4 POTS lines |
| 00:59.49 | kuj | that might explain that |
| 01:00.15 | MikeFRandT | oh lord, how much do these cost? |
| 01:00.18 | MikeFRandT | lol |
| 01:00.31 | nikko | if you want to expand beyond that, you'll then need to add another card that will handle 4 more lines |
| 01:00.53 | sivana | CAC Channel bank to TE405P = straight through ethernet cable? |
| 01:01.00 | nikko | you have to look at economy - I only pay $500 US for a full t-1 with 4 voice channels and the rest data |
| 01:01.05 | redder86 | sivana: technically no |
| 01:01.10 | Damin | Anyone have a TE405P for sale? |
| 01:01.13 | redder86 | sivana: but it may work |
| 01:01.14 | ManxPower | sivana: Yes, that would generally work in a pinch |
| 01:01.29 | MikeFRandT | Ok so if I get this 4 line card... then what? |
| 01:01.30 | nikko | so after 4 pots lines, it's not real economical to use pots lines |
| 01:01.37 | redder86 | Damin: doesn't Digium? |
| 01:01.42 | ManxPower | But if you have significant problems, get a T-1 wired cable. |
| 01:01.56 | MikeFRandT | I already have a T-1 but data only |
| 01:02.03 | Damin | redder86: Yes, but many of the developers that hang out here have development equipment, and I usually ask here first.. |
| 01:02.14 | redder86 | ManxPower: I always had to make my own, I could never find them ready-made. |
| 01:02.20 | Damin | reffer86: Helps everyone out that way, and I can usually pick it up at a discount.. |
| 01:02.25 | nikko | you justinstall it in your server, plug up your phone lines, build and load the drivers, and set asterisk up |
| 01:02.39 | expressfone1 | you can find link for t1 cable at digium site |
| 01:02.42 | Damin | kuj: Dropping back to a single registration doesn't solve the problem.. |
| 01:02.43 | nikko | asterisk needs to be told where/what types on lines you have |
| 01:03.03 | dizzydiffi | any one set up the CDR in asterisk |
| 01:03.08 | MikeFRandT | So basically there is no real small business solution for this? |
| 01:03.10 | Damin | kuj: I'll post on asterisk-users about this as a followup to my Polycom IP 500 thread.. |
| 01:03.12 | sivana | ManxPower: T1 crossover cable? |
| 01:03.20 | redder86 | MikeFRandT: huh? yes there is. |
| 01:03.24 | sivana | ManxPower: ie: http://www2.adtran.com/support/technotes/t1ddsadptxvr/ |
| 01:03.38 | nikko | well, a very easy way to get started with NO hardware is an Asterisk friendly VOIP provider |
| 01:03.42 | *** join/#asterisk techie (gus@asterisk.horizonte.us) |
| 01:03.48 | redder86 | sivana: the crossover cable is generally not what you need between the channel bank and your TE405P |
| 01:03.54 | nikko | takes about 10 minutes |
| 01:03.57 | ManxPower | sivana: I don't believe any either net cable will work as a t-1 crossover, but I could be wrong. |
| 01:04.00 | nikko | and works really well |
| 01:04.04 | kuj | yup. beats me, though, cause it does work for me (single line, though) |
| 01:04.04 | expressfone1 | t1 crossover cable 1-4 2-5 3-3 4-1 5-2 6-6 |
| 01:04.14 | *** join/#asterisk iMediax (~user@00045a7b37e7.click-network.com) |
| 01:04.23 | sivana | what's the pin out for a T1 cable? |
| 01:04.35 | redder86 | MikeFRandT: you don't need any FXO ports or any FXS ports to use Asterisk in a small business application. |
| 01:04.35 | MikeFRandT | Nikko... What do you mean? |
| 01:04.43 | nikko | We use that solution at several remote offices and for conference bridges and such |
| 01:05.11 | MikeFRandT | eredder86, so what do I need to do guys? |
| 01:05.15 | nikko | you can sign up for an account with a VOIP provider that sends calls to you vai the internet |
| 01:05.16 | redder86 | sivana: I'd typo or something stupid like that if I typed it here. Good diagrams are available on-line. |
| 01:05.28 | nikko | *via |
| 01:05.31 | redder86 | MikeFRandT: I missed your description of what you want to do. |
| 01:05.34 | expressfone1 | sivana> t1 crossover cable 1-4 2-5 3-3 4-1 5-2 6-6 |
| 01:05.41 | redder86 | MikeFRandT: other than "small business application" |
| 01:06.18 | MikeFRandT | redder86 I need 10 extensions, but I only have 4 lines... All 4 lines do roll-over, and the 4th line is for incomming fax as well. |
| 01:06.42 | nikko | do you need local #'s? |
| 01:07.04 | redder86 | MikeFRandT: okay, firstly, forget the terminology of "lines" ... it doesn't really apply to VoIP. You can have unlimited numbers of calls with VoIP so long as your bandwidth permits it. |
| 01:07.37 | redder86 | MikeFRandT: keep one line for fax, Asterisk sucks at analog fax applications right now. then .... |
| 01:08.42 | redder86 | MikeFRandT: you have several options, you can exclusively use VoIP for your external "lines" or you can exclusively use your existing lines for your external calls... or you can combine the two. |
| 01:09.22 | sivana | redder86: ever use a CAC 24 port FXS? |
| 01:09.34 | redder86 | MikeFRandT: you'll be less-expensive with pure VoIP, but maybe with some ... perhaps ... some issues due to your inexperience with VoIP |
| 01:09.42 | redder86 | sivana: no, I haven't. |
| 01:10.46 | redder86 | MikeFRandT: but, if you really want to save some money, then you would drop all of your land lines except for your fax line, and then you would put that money into getting the best broadband internet connection you could get. |
| 01:11.03 | redder86 | MikeFRandT: then you get 10 IP phones, like Polycoms or Grandstreams. |
| 01:11.30 | *** join/#asterisk kludgebox (~bob@12.171.178.194) |
| 01:11.43 | redder86 | MikeFRandT: then you set up an Asterisk server and set up an account with a good VoIP provider and get a toll-free DID from them. |
| 01:11.44 | MikeFRandT | Ok redder86, who do we go through to get this service? |
| 01:11.53 | redder86 | MikeFRandT: which part? |
| 01:11.59 | MikeFRandT | Yeah I do have 3 toll free numbers but I can get them fowarded |
| 01:12.09 | MikeFRandT | The voip service? |
| 01:12.12 | *** join/#asterisk MasterK (~Flash@82.154.41.248) |
| 01:12.41 | MikeFRandT | you said I need external numbers... I'm assuming that someone else has the actual PBX and I use this as sub pbx remotely |
| 01:12.49 | redder86 | MikeFRandT: I use NuFone, but I hear Broadvoice is okay, and others will surely recommend others. Just basically pick one that suits your needs well. They are listed on http://www.voip-info.org |
| 01:13.07 | MikeFRandT | ok |
| 01:13.14 | redder86 | MikeFRandT: I don't know what the difference is between "actual PBX" and "sub PBX". |
| 01:13.27 | redder86 | MikeFRandT: the PSTN lines would terminate at your VoIP provider |
| 01:13.47 | redder86 | MikeFRandT: they would "VoIP" the call and forward them to you via IP. |
| 01:14.25 | redder86 | MikeFRandT: now, once you start wanting to host your own PSTN connections, then you start talking about some hardware expense in the Asterisk PBX. |
| 01:15.05 | redder86 | MikeFRandT: so as long as you can rely upon a VoIP provider you don't have to buy any TDM cards or T1 cards or whatever. |
| 01:15.31 | redder86 | MikeFRandT: the minute that you want to host your own PSTN connections, though, then you need to determine how much you want to host... one channel/line, two, three... etc. |
| 01:15.46 | redder86 | MikeFRandT: with 10 extensions you could, in theory, have 10 simultaneous calls |
| 01:15.47 | MikeFRandT | Ok who can I go through that is cost effective for a PSTN provider? |
| 01:16.16 | nikko | MikeFRandT - do you need local #'s? |
| 01:16.17 | redder86 | MikeFRandT: you generally are limited by your geography... Qwest, Sprint, etc. |
| 01:16.55 | redder86 | MikeFRandT: you are already familiar with the PSTN providers in your area. They give you your existing lines. |
| 01:17.05 | redder86 | well, they sell them to you |
| 01:17.31 | redder86 | anyway, I'm off to dinner, someone else can surely pick up where I left off :-) |
| 01:18.05 | *** join/#asterisk samueltc (~samuel@levinux.UQAR.UQUEBEC.CA) |
| 01:18.55 | Malthus | how bad is the fax suckage? is it all workable? |
| 01:19.57 | Silik0n | anyone started a a Operator panel in Mono yet so its cross platform? |
| 01:20.11 | Malthus | mono? |
| 01:20.36 | Silik0n | yeah the novell sponsored .net clone |
| 01:20.38 | Malthus | you mean with gtk? |
| 01:20.46 | Silik0n | no i mean under mono |
| 01:20.55 | Silik0n | GTK != mono |
| 01:21.00 | Malthus | I know |
| 01:21.11 | nikko | Is it possible to run IAX2 on more than one port per instance of *? |
| 01:21.14 | Silik0n | C# type stuffs |
| 01:21.14 | Malthus | but I am asking with what graphical toolkit |
| 01:21.28 | Silik0n | who cares... heh GTK is fine |
| 01:21.33 | Malthus | win forms are't cross platform |
| 01:21.42 | Silik0n | yeah |
| 01:21.45 | Silik0n | true dat |
| 01:21.51 | Malthus | true dat? |
| 01:21.52 | Silik0n | GTK is included with mono tho |
| 01:22.01 | Silik0n | as winforms being cross platform |
| 01:22.03 | Malthus | where are you from ? |
| 01:22.09 | Silik0n | memphis |
| 01:22.17 | Malthus | oh zeen |
| 01:22.22 | *** join/#asterisk aokaze (aokaze@modem162.supermusca.bigsky.net) |
| 01:23.26 | hermie | It's all hapening at the zoo... I do believe, I do belive it's true |
| 01:24.58 | hermie | it's a rough and tumble journey from the east side to the west |
| 01:28.13 | Moc_ | ~ntp |
| 01:28.15 | jbot | somebody said ntp was nano time parameters |
| 01:28.33 | Malthus | ok |
| 01:28.37 | Moc_ | what is the public US NTP server ? |
| 01:29.57 | ST-3 | anyone ever upgraded the software on a WISIP? |
| 01:30.02 | hermie | tick.usno.navy.mil |
| 01:30.03 | hermie | or tock |
| 01:30.10 | hermie | at NTP is "network time protocol" |
| 01:31.05 | hermie | ~ntp |
| 01:31.06 | jbot | ntp is, like, Network Time Protocol |
| 01:32.29 | sivana | yay... T1 cross over worked |
| 01:33.34 | Silik0n | dead add/drop CDU/DSU ++ |
| 01:34.40 | Moc_ | jbot, no ntp is Network Time Protocol, and a public NTP server is pool.ntp.org |
| 01:34.41 | jbot | okay, Moc_ |
| 01:34.44 | Moc_ | ~ntp |
| 01:34.45 | jbot | ntp is, like, Network Time Protocol, and a public NTP server is pool.ntp.org |
| 01:34.49 | Moc_ | there |
| 01:35.08 | Silik0n | jaj |
| 01:35.19 | Silik0n | ~Moc |
| 01:35.20 | jbot | Perl Script for updating myIP.org Dynamic DNS entries. URL: http://www.nmia.com/~trilia/perl/, or http://www.automated.it/guidetoasterisk.htm |
| 01:35.47 | Moc_ | wtf ? |
| 01:36.11 | Moc_ | jbot, forget moc |
| 01:36.34 | Moc_ | no, moc is the initials of Moc |
| 01:37.20 | Moc_ | jbot, no, moc is the initials of Moc |
| 01:37.21 | jbot | Moc_: okay |
| 01:39.28 | *** join/#asterisk nwhit (~chatzilla@65.107.59.67.ptr.us.xo.net) |
| 01:39.49 | hermie | 2/3 of open Mantis bugs either have a patch or have been fixed in CVS and are being tested |
| 01:39.52 | hermie | amazing |
| 01:40.53 | dude_where | Anyone know of good cheap business like phones to use w/ * that aren't ADSI? |
| 01:40.56 | nwhit | is there anyone here that can help me debug a problem with my pri? |
| 01:41.07 | *** join/#asterisk buddah (~hnic@208.179.86.5) |
| 01:41.26 | hermie | nwhit: plenty of people, what's the deal? |
| 01:41.55 | nwhit | hermie: this is a new pri, I have the outbound working great, but am having a problem with the inbound |
| 01:42.27 | hermie | nwhit: go to nick.pastebin.ca and put up your zapata.conf zaptel.conf and extensions.conf |
| 01:42.29 | nwhit | hermie: i never see the call coming in at all on any of my DIDs, but the telco says my pbx is rejecting the call |
| 01:42.35 | waddy | do cisco 79XX support GSM? |
| 01:42.54 | hermie | nwhit: how many digits of DNIS are they giving you? |
| 01:43.01 | *** join/#asterisk doughecka_ (~Doug@12-220-235-106.client.insightBB.com) |
| 01:43.01 | nwhit | 10 |
| 01:43.20 | hermie | nwhit: well, just start with putting your extensions.conf up |
| 01:44.07 | Yoda-BZH`ZzZ | une bonne nuit je vous souhaite ! / A good night I wish you |
| 01:45.30 | nwhit | hermie: http://nick.pastebin.ca/4208 this is the context where the call is received (my extensions.conf is huge right now) |
| 01:46.14 | nwhit | nwhit: i've tried a few different things... the part that I have commented out was working at one time, until they added more DID's to the line |
| 01:46.22 | Moonwick | heh. yoda must be from quebec. |
| 01:46.45 | Yoda-BZH`ZzZ | no |
| 01:46.46 | Yoda-BZH`ZzZ | :p |
| 01:46.53 | hermie | nwhit: what do you see with 'pri debug on' |
| 01:46.59 | nwhit | hermie: nothing |
| 01:47.30 | *** join/#asterisk doughecka_ (~Doug@12-220-235-106.client.insightBB.com) |
| 01:47.43 | hermie | nwhit: how about a 'pri intense debug'? |
| 01:47.59 | hermie | nwhit: because if you're not seeing anything, they're not sending it... |
| 01:48.00 | nwhit | hermie: periodic stuff... i can post it |
| 01:48.16 | hermie | nwhit: just post what you see when you try calling a DID |
| 01:49.27 | nwhit | hermie: http://nick.pastebin.ca/4209 |
| 01:49.52 | hermie | nwhit: that's all you see? |
| 01:50.04 | nwhit | hermie: should my extensions.conf accept any did the way it's setup? |
| 01:50.11 | hermie | nwhit: I'd say the problem's on their end |
| 01:50.25 | nikko | Hey, how can I determine if * is trunking IAX calls? I do IAX2 trunk debug, andit never shows anything, regardless of how many IAX calls are outbound or inbound |
| 01:50.31 | nwhit | hermie: ok |
| 01:50.38 | hermie | nwhit: the only other thing to try is answering _X., reloading, and trying again |
| 01:51.38 | nwhit | so that would be exten => _X,1,Answer |
| 01:52.19 | hermie | no _X.,1,Answer() |
| 01:52.27 | hermie | the . means match anything following |
| 01:52.34 | nwhit | hermie: ok... let me see |
| 01:54.13 | nwhit | hermie: ok... nope |
| 01:54.38 | hermie | nwhit: their problem, as long as your zaptel.conf is putting the PRI in that context for incoming |
| 01:54.55 | hermie | but based on your PRI debug (or lack thereof) I'd say they screwed the pooch |
| 01:56.46 | nwhit | hermie: i posted http://nick.pastebin.ca/4210 my zapata.conf... does it look right? |
| 01:56.56 | nwhit | hermie: I just want to cover all bases |
| 01:57.16 | hermie | nwhit: looks fine to me |
| 01:57.29 | nwhit | hermie: ok thanks... let's see what happens with them |
| 01:58.22 | hermie | nwhit: I bet they'll tell you they didn't fix anything, but somehow it'll work |
| 01:59.27 | nwhit | yup... i hear that all the time... makes you wonder.. hmmm..... |
| 02:07.42 | hermie | nwhit: it's all a CYA thing... |
| 02:07.57 | nwhit | hermie: CYA? |
| 02:08.20 | hermie | ~cya |
| 02:08.22 | jbot | [cya] Cover Your Ass |
| 02:08.40 | nwhit | :) |
| 02:09.08 | nwhit | hermie: unfortunately they haven't been able to do that yet. i am waiting for a tech call back |
| 02:10.24 | *** join/#asterisk ArkyLady (nat32@93-95.hspg-ubr2-blk1.cablelynx.com) |
| 02:15.04 | Umaro | On the offchance, anyone siemens-experienced consultants? willing to pay big, of course ;) |
| 02:15.57 | *** join/#asterisk sivana (~richard@northbay-pppoe-211.vianet.ca) |
| 02:17.41 | mishehu | boom. |
| 02:17.43 | mishehu | shaka. |
| 02:17.44 | mishehu | laka. |
| 02:17.48 | sivana | does the CAC channel bank use kewlstart or loopstart? |
| 02:23.47 | *** join/#asterisk AgiNamu (~AgiNamu@200.12.43.74) |
| 02:27.48 | AgiNamu | Hi |
| 02:27.56 | AgiNamu | Does anyone here have a fax ? |
| 02:28.02 | AgiNamu | and wouldn't mind doing me a favour? |
| 02:28.06 | AgiNamu | I need to fax in a digium disclaimer |
| 02:29.09 | wolfson | how are you gonna get the sig to the person with the fax? |
| 02:30.11 | *** join/#asterisk heison (~heison@CPE000a01d49e6f-CM00111a59bec2.cpe.net.cable.rogers.com) |
| 02:30.37 | AgiNamu | GIF |
| 02:30.42 | AgiNamu | I printed the fax to TIF |
| 02:30.49 | AgiNamu | then added my sig, saved as gif |
| 02:30.55 | AgiNamu | who deals with paper these days? |
| 02:31.21 | wolfson | anyone dealing with actual legal contracts |
| 02:31.28 | *** join/#asterisk coldshado (~coldshado@adsl-67-117-175-70.dsl.lsan03.pacbell.net) |
| 02:31.37 | coldshado | hey, realize this is o/t, but anyone used colinux? |
| 02:31.44 | AgiNamu | fax doesnt mean any more than a jpeg, since fax is just digital |
| 02:31.58 | bkw_ | you mean tiff |
| 02:32.12 | AgiNamu | yea, tiff |
| 02:32.40 | bkw_ | file you in hell? |
| 02:32.42 | file[laptop] | nope |
| 02:32.47 | bkw_ | why not |
| 02:32.50 | bkw_ | i'm lurking in htere |
| 02:32.52 | file[laptop] | fine |
| 02:33.39 | AgiNamu | I'll pay a few bucks to get it sent off |
| 02:34.11 | *** join/#asterisk tessier__ (~treed@ip-66-181-5-46.cust.i2bnetworks.com) |
| 02:34.19 | tessier__ | Hello all! |
| 02:34.35 | tessier__ | So after a rediculous ordeal I got my replacement TDM401B (4 port FXO) from Digium. |
| 02:34.51 | hermie | Cecilia, you're breaking my heart, you're shaking my confidence daily... |
| 02:35.06 | tessier__ | And guess what? Some hamfisted tech has bent the bracket on the end of the card. |
| 02:35.18 | tessier__ | I pulled it out of the sealed box from digium and it's bent. |
| 02:35.23 | tessier__ | Ok, just cosmetic. |
| 02:35.35 | tessier__ | So I take out the defective card that has been giving me grief for over a month and put in the new one. |
| 02:35.49 | JonR800 | huh??? they sent you a card with a bent bracket? |
| 02:35.53 | tessier__ | When I modprobe the wcfxs module the console streams " |
| 02:36.05 | tessier__ | Unexpected NMI for unknown reason 30 |
| 02:36.09 | tessier__ | JonR800: Yep. |
| 02:36.28 | tessier__ | And unfortunately this asterisk box has no X so I can't really google. |
| 02:36.34 | tessier__ | Anyone happen to know what might cause this? |
| 02:36.39 | JonR800 | That's umm.. surprising and careless. |
| 02:36.45 | tessier__ | Indeed. |
| 02:37.08 | tessier__ | When I send back my RMA card they are going to get a long letter from me explaining how much money their little $500 card has cost me. |
| 02:37.30 | tessier__ | I can't wait to finish this project and wash my hands of them. |
| 02:37.45 | tessier__ | If I can't get this working I get to buy back the whole phone system. |
| 02:38.32 | tessier__ | So....any ideas? |
| 02:38.51 | JerJer | its modprobe wctdm now |
| 02:39.27 | JerJer | your not going to get much help with that attitude |
| 02:39.36 | Faithful | Think my lag problem may be over ... I am using a different ISDN card |
| 02:39.43 | modulus_ | his not |
| 02:40.19 | tessier__ | ~ |
| 02:40.54 | tessier__ | ~ |
| 02:40.56 | tessier__ | oops |
| 02:41.03 | tessier__ | brb |
| 02:41.12 | hermie | no kidding |
| 02:41.15 | hermie | :-) |
| 02:41.24 | AgiNamu | So... no one wants to send a fax for me? :) I'll pay |
| 02:42.30 | JerJer | you push a fax out any number of VoIP providers |
| 02:42.38 | sivana | if I have a channel bank with FXS cards, shouldn't it be able to handle inbound/outbound calls? |
| 02:42.46 | AgiNamu | jerjer, I just dont have any fax hardware |
| 02:44.44 | hermie | AgiNamu: http://www.faxaway.com/ |
| 02:45.23 | *** join/#asterisk tessier__ (~treed@ip-66-181-5-46.cust.i2bnetworks.com) |
| 02:45.30 | hermie | AgiNamu: http://www.ourfax.com/ |
| 02:45.38 | hermie | AgiNamu: All from that one place... Goodle |
| 02:45.42 | hermie | aka Google :-) |
| 02:46.10 | tessier__ | JerJer: I don't recall if I did it or if the zaptel make install does it but wcfxs is aliased to wctdm in modules.conf. Just to be sure I rebooted and did a modprobe wctdm and got the same result. |
| 02:46.21 | AgiNamu | hermie, I tried those. they are ASCII only |
| 02:46.22 | AgiNamu | no sig |
| 02:46.28 | AgiNamu | And FaxAway says I have to send a fax to sign up. |
| 02:46.43 | Faithful | tessier: go into your system bios and do a reset on the hardware interrupts |
| 02:47.01 | tessier__ | Faithful: Wouldn't a power off do that? |
| 02:47.08 | Faithful | and/or move the card to a different slot |
| 02:47.20 | Faithful | tessier: not at all it is stored in nvram |
| 02:47.30 | tessier__ | This card is in the same slot the other card was in which worked fine (except for the hardware problem). |
| 02:47.35 | Faithful | that's why there is an option in the bios ;-) |
| 02:47.47 | Faithful | Pardon? |
| 02:47.48 | tessier__ | To rule out IRQ sharing I assigned this slot IRQ 10. |
| 02:48.06 | tessier__ | Faithful: The card I am installing is replacing a card which is being RMA'd. |
| 02:48.22 | tessier__ | And it did not have this problem of causing NMI's. |
| 02:48.51 | Faithful | Well after you've done 10 cards and bought the system back.... give what I say a try |
| 02:48.59 | tessier__ | An hour ago this system was working perfectly other than the lines dying. I just swapped one TDM401 out for another one. |
| 02:49.01 | Faithful | what brand mobo is it |
| 02:49.29 | Faithful | NMI is an interupt issue |
| 02:49.40 | tessier__ | It's a Dell PowerEdge 750. |
| 02:49.49 | Faithful | Ok |
| 02:49.49 | tessier__ | Indeed it is. Just seems odd that this card causes it and the other one doesn't. |
| 02:50.03 | tessier__ | Certainly I'll poke into the bios and see if that helps though. |
| 02:50.43 | Faithful | You could have another dead card... plug it into another box and see what happens |
| 02:51.29 | Faithful | tessier: you need a spare bogus dev box for checking such annoyances |
| 02:51.56 | Faithful | So it the box a turnkey system from digium? |
| 02:52.04 | Faithful | Who bent the card? |
| 02:52.08 | tessier__ | I have one although not here at the clients site. I bet I can scrounge one up here though. |
| 02:52.16 | tessier__ | Digium did. I took picks of it right after I took it out of the box. |
| 02:52.19 | tessier__ | pics |
| 02:52.27 | tessier__ | I'll be emailing those to digium. |
| 02:52.38 | *** join/#asterisk Yoda_BZH (~yoda-bzh@80.125.248.235) |
| 02:52.42 | tessier__ | The card arrived via FedEx yesterday afternoon. |
| 02:52.58 | Faithful | Disgusting... they should be ashamed of putting hamfisted slackness out into production |
| 02:53.19 | tessier__ | Faithful: I am trying not to say anymore than pure frustration forces me too... |
| 02:53.43 | tessier__ | If this card is dead too... well let's just plug it into another box and see what happens. |
| 02:54.07 | Faithful | I understand but asterisk's reputation needs be protected (and your pocket book too) |
| 02:54.08 | JonR800 | that TDM400 card just seems to make a lot of people run in circles :) |
| 02:54.43 | sivana | if I have a channel bank with FXS cards, shouldn't it be able to handle inbound/outbound calls? |
| 02:54.47 | Faithful | tessier: you've another linux box you can modprobe it on |
| 02:55.05 | JonR800 | Well it's not really Asterisk's fault perse.. but hardware.. |
| 02:56.24 | Umaro | On the offchance, anyone siemens-experienced consultants? willing to pay big, of course ;) |
| 02:56.34 | Faithful | JonR800: they delivered a box with a bent card in it that later failed. Who know what other damage was do in assembly to the mainboard, if they bend cards putting them in (and ship them that way) |
| 02:57.33 | tessier__ | Faithful: We are scrounging up another box. |
| 02:57.34 | JonR800 | Faithful: I totally agree, but I don't necessarily believe that Digium's quality control is a direct representation of the quality of Asterisk itself. :) |
| 02:57.51 | Faithful | JonR800: point taken |
| 02:58.01 | tessier__ | Someone recently recommended to never touch these FXO/FXS cards and just do channel banks. |
| 02:58.18 | tessier__ | I am beginning to think they knew what they were talking about. |
| 02:58.37 | tessier__ | Well, gotta reboot this box to play with the irq's and then pull the card out and put it into another box. bbl |
| 02:58.48 | Faithful | I am just concerned for tessier that he gets this going, doesn't end up with a major cost, a second hand box, and a client that bags asterisk |
| 02:59.52 | JonR800 | true. There's been a lot of FUD around the TDM 400 lately. |
| 03:00.36 | JonR800 | I don't remember this many people complaining before. Could be the whole wctdm mixup |
| 03:00.38 | Faithful | I don't know it is it's not the PRI card I gather |
| 03:00.56 | JerJer | we have quite a few TDM400P systems out there |
| 03:01.09 | JonR800 | JerJer: working fine? |
| 03:01.18 | JerJer | the only trouble we have is from time to time we have to rmmod and modprobe the kernel driver |
| 03:01.43 | JerJer | he is prolly using old ass hardware |
| 03:01.54 | JonR800 | JerJer: Ahh.. he said a poweredge 750 |
| 03:01.59 | bjohnson | what does chan_sip.c:2401 find_call: Call missing call ID from '192.168.2.6' mean? that IP is my SPA 3000 and I'm trying to route a call out of it |
| 03:02.10 | JerJer | we use Dell 400SC |
| 03:02.13 | JerJer | and mini-itx |
| 03:02.29 | JerJer | via EPIA like 1ghz |
| 03:02.40 | JonR800 | I really think for a production environment though.. rmmod/modprobe'ing regularly may not be the way to go. |
| 03:02.43 | hermie | why do people buy Dells? I know there are better system builders than Dell at least by JerJer... |
| 03:02.59 | JonR800 | hermie: comes down to money. |
| 03:03.06 | JerJer | JonR800: i didn't say regularly |
| 03:03.13 | hermie | JonR800: Dells are more expensive |
| 03:03.14 | JerJer | maybe once a month |
| 03:03.14 | AgiNamu | I buy Dell because they're good price and I can somewhat trust them. |
| 03:03.21 | Faithful | hermie: Dells are ok... let's not start a war |
| 03:03.25 | JerJer | dell expensive? |
| 03:03.32 | Faithful | Dells are not expensive |
| 03:03.34 | JonR800 | JonR800: the 400SC is more expensive than what? hehe.. I got a 420SC for $299 last week |
| 03:03.44 | *** join/#asterisk mindCrime (~mindCrime@rrcs-24-106-188-6.se.biz.rr.com) |
| 03:03.45 | hermie | more expensive than a local shop could build the same thing for |
| 03:03.46 | JerJer | i can get dells for less than 300 |
| 03:03.57 | AgiNamu | yea, but will the local shop come in 4 hours when something fails? |
| 03:03.59 | JonR800 | JerJer: sorry to put words in your mouth. |
| 03:04.12 | hermie | AgiNamu: uh, yeah... |
| 03:04.12 | AgiNamu | And will the local shop actual run tests to ensure all the pieces work together and are certified? |
| 03:04.14 | Faithful | 3yr 24/7 4hr response warranty for the same price I can build a white box intel server??? |
| 03:04.16 | hermie | AgiNamu: uh, yeah... |
| 03:04.25 | AgiNamu | wow, some local shop. |
| 03:04.35 | AgiNamu | I dont know of ANY local shops that get, say, MS hardware compat testing. |
| 03:04.35 | hermie | is the 420SC the one with the Celerons? |
| 03:04.48 | JerJer | you can get a real p4 in them |
| 03:05.11 | hermie | for $300? |
| 03:05.16 | JerJer | oh 420 i don't know about 420 |
| 03:05.27 | Faithful | Dell have major R&D, they don't put something out that is going to cost them money |
| 03:05.40 | JerJer | hermie: we don't just order one box off their website |
| 03:05.45 | JonR800 | you can get a real p4 in them for $300 some i believe. |
| 03:05.47 | Faithful | They would be out of business in 1 year |
| 03:06.32 | Faithful | Their bottom end server is a steal.... I would buy them for desktop pcs |
| 03:06.47 | firestrm | dell precision 620's, i get them for <$600.00 cad, with 2 18 gig ultra scsi drives and 1 meg of registered ecc ram, Dual 1 gig xeons. they make awesome web/mail/asterisk/w.h.y servers. cant get that from a local shop |
| 03:06.52 | Faithful | Not much more than a workstation |
| 03:07.04 | hermie | Faithful: a box with a celeron in it is not a server |
| 03:07.26 | JonR800 | lol |
| 03:07.39 | Faithful | hermie: none of their server line in AU have celery |
| 03:07.45 | JonR800 | hermie: says who? if it's running services it is a server correct? |
| 03:08.00 | JerJer | we have no celerons |
| 03:08.05 | JerJer | all P4s or Xeon |
| 03:08.09 | JerJer | all Dell |
| 03:08.35 | hermie | JonR800: my laptop runs services... it's not a server |
| 03:08.57 | Faithful | Man, Dell kick but for the money... I can't compete with them and make a profit |
| 03:09.28 | Faithful | It's so easy when something blows up "call Dell" |
| 03:09.33 | mishehu | I personally don't like dell so much. |
| 03:09.38 | JonR800 | we're getting into technicalities hermie. |
| 03:10.04 | sivana | if I have a channel bank with FXS cards, shouldn't it be able to handle inbound/outbound calls? |
| 03:10.28 | Faithful | mishehu: They're servers are all designed to run linux... and their SATA RAID cards rock!!! |
| 03:10.33 | firestrm | sivana, you need fxo card for outbound |
| 03:10.52 | firestrm | sivana, or fxs->fxo adaptors |
| 03:10.58 | sivana | so an entire fxs channel bank is only outbound? |
| 03:11.11 | JerJer | sivana: that question makes no sense |
| 03:11.17 | JonR800 | bingo |
| 03:11.32 | JerJer | FXS provides battery and dialtone to the line |
| 03:11.41 | sivana | right |
| 03:11.41 | firestrm | sivana, fxo.. connects to an outside line, fxs is what you plug a phone into |
| 03:11.48 | JerJer | FXO expects to take battery and hear a dialtone |
| 03:12.04 | sivana | my fax calls, but never answerfs |
| 03:12.19 | sivana | it's in a channel bank with fxs cards |
| 03:12.34 | JerJer | then the fax machine is not set to answer |
| 03:13.00 | sivana | yes and Asterisk shows the zap channel as ringing |
| 03:13.23 | JerJer | no |
| 03:13.27 | Damin | Alright.. |
| 03:13.38 | firestrm | sivana, so you do have an fxo card? |
| 03:13.45 | bjohnson | what does chan_sip.c:2401 find_call: Call missing call ID from '192.168.2.6' mean? that IP is my SPA 3000 and I'm trying to route a call out of it from a macro. It works when I take the Dial command out of the macro |
| 03:13.53 | sivana | I plugged a phone into it and I don't get ringing |
| 03:13.54 | JerJer | if the channel is ringing the device plugged into it needs to answer it |
| 03:14.14 | sivana | even though asterisk says zap/26 is ringing... |
| 03:14.27 | Damin | There is most definitely a problem introduced into chan_sip of stable that affects Polycom phones.. I'm running 12/26 at work, and had to back off from the 1/8 chan_sip, and I just did a CVSup last night at home, and now my Polycom won't dial. |
| 03:14.27 | file[laptop] | bjohnson: it means the SIP message was bad, incomplete, whateva |
| 03:14.48 | bjohnson | any idea why it would be a problem when inside a macro? |
| 03:14.55 | file[laptop] | nope |
| 03:15.13 | sivana | JerJer: I'm using loopstart.. does that matter? |
| 03:15.17 | JerJer | maybe |
| 03:15.23 | JerJer | what is your channel bank expecting? |
| 03:15.37 | file[laptop] | bjohnson: Try a sip debug and see what 'da messages look like |
| 03:16.00 | sivana | I'm not sure.. is that a dip switch setting? |
| 03:16.02 | JerJer | Damin: run cvs -head |
| 03:16.36 | sivana | I think it'll take loopstart or groundstart |
| 03:16.57 | sivana | I tried both |
| 03:16.59 | JonR800 | Damin: im using 12/28 fine.. so it's something after that. |
| 03:19.07 | *** join/#asterisk Miker (~mike@201.129.119.248) |
| 03:20.40 | Umaro | Anyone here using * with a siemens phone system? trying to figure out why it's sending the calls I send to it to a seemingly random, but always consistant extension on the siemens |
| 03:21.22 | *** join/#asterisk v-office (~acidtrux@amako.team-anomaly.org) |
| 03:23.02 | bjohnson | I guess my problem was the format I was using to set the CIDNUM |
| 03:24.57 | Faithful | Umaro: It would have to be in the extensions.conf or the dial info is not reaching asterisk correctly if it happens often enough, sit and watch the messages on the CLI |
| 03:25.02 | heison | is the latest zaptel (CVS) broken? zaptel.c:45:31: linux/moduleparam.h: No such file or directory |
| 03:25.02 | heison | make: *** [zaptel.o] Error 1 |
| 03:25.31 | Umaro | Faithful: no, * is sending the calls to the siemens, not the other way around |
| 03:25.33 | *** join/#asterisk bigEE (~nospam@res-24-197-115-062.spa.sc.charter.com) |
| 03:25.39 | v-office | heison: it just built for me |
| 03:25.49 | Umaro | the siemens sending calls to * works perfectly, actually |
| 03:25.51 | v-office | you have a problem with your kernelheaders |
| 03:26.19 | Faithful | Umaro: have you ever had it right? |
| 03:26.58 | *** join/#asterisk rees40 (~ericr@92-3.69-92-cpe.cableone.net) |
| 03:27.06 | *** join/#asterisk santiago (~santiago@63.245.86.86) |
| 03:27.46 | rees40 | REGISTER america |
| 03:27.50 | heison | v-office: what kernel r u running? 2.4 / 2.6? |
| 03:28.12 | v-office | both |
| 03:28.51 | *** join/#asterisk tessier__ (~treed@ip-66-181-5-46.cust.i2bnetworks.com) |
| 03:29.18 | tessier__ | Downloading kernel source onto this RH9 box we scrounged up to test the card on. |
| 03:29.37 | WilliamK | heison, open all the files and delete linux/moduleparam.h and you should be fine to compile =) |
| 03:29.54 | Umaro | Faithful: no. I am trying to make it work. |
| 03:30.00 | v-office | heh yeah, bad advice :) |
| 03:30.13 | Faithful | tessier: best of luck... the irq thing didn't help huh? |
| 03:30.22 | heison | WilliamK: what does linux/moduleparam.h provides... it's definitely new as compare to CVS on Jan 5 |
| 03:30.38 | Umaro | Faithful: but when the siemens is connected to the E1 directly, the siemens works great. When the E1 connects to * and then * to the siemens, I have problems with inbound calls |
| 03:30.38 | WilliamK | heison, it's the same as module.h |
| 03:30.46 | WilliamK | which is already there |
| 03:31.04 | heison | WilliamK: why not link module.h to moduleparam.h? |
| 03:31.11 | tessier__ | Faithful: I went into the BIOS and changed the IRQ for the card to something else unused and it had no effect. |
| 03:31.16 | Faithful | Umaro: then if asterisk is working calls in / calls out I would be looking carefully at your extensions.conf |
| 03:31.16 | WilliamK | heison, wish I knew =) |
| 03:31.29 | tessier__ | Faithful: You mentioned something about clearing the IRQ's or something. Did you mean something other than what I did? |
| 03:31.39 | *** join/#asterisk ericbrees40 (~ericr@92-3.69-92-cpe.cableone.net) |
| 03:31.47 | Faithful | tessier: yes I did |
| 03:31.59 | tessier__ | What did you mean exactly? |
| 03:31.59 | Faithful | tessier: clear the nvram |
| 03:32.23 | tessier__ | Does that mean reset the bios to factory defaults? |
| 03:32.28 | Faithful | It has different names on different mobos but is usually found in the PCI configuration section |
| 03:32.29 | tessier__ | Or jumper something to discharge the RAM? |
| 03:32.41 | heison | creating symlink seem to fix the compile... is moduleparam.h something new from recent kernel src? I'm on 2.4 |
| 03:33.20 | Faithful | tessier: no it means that it removes all the IRQ non volitile entries in the bios |
| 03:33.23 | Umaro | Faithful: calls coming in from the E1 go to a context called inbound. inbound has one line, that says _X.,1,Dial(Zap/g4/${EXTEN}), g4 being the zap channels on the PRI to the siemens that are delegated for inbound calls |
| 03:33.28 | bjohnson | anyone use Areski for looking at CDR info? |
| 03:33.42 | WilliamK | I think someone honestly goofed, no idea yet... I know the suggestion earlier was to remove it from the files and then compile it |
| 03:34.40 | criptos | nigths! |
| 03:35.09 | criptos | when you are stating which codecs are allowd using a iax peer, it its allow=iLBC,ulaw or allow=iLBC ulaw ? |
| 03:35.27 | ericbrees40 | I have a question. Has anyone been able to disable call-waiting on Polycom phones? |
| 03:35.36 | Umaro | criptos: allow=ilbc\nallow=ulaw is what I use |
| 03:35.52 | Umaro | criptos: not sure if it works the other way. |
| 03:37.06 | Faithful | Umaro: do any calls go through correct or are they as going to one "wrong" extension? |
| 03:40.03 | Umaro | Faithful: they are all going to one wrong extension |
| 03:42.22 | tessier__ | hmm...trying to compile zaptel on this new box and it can't find linux/moduleparam.h |
| 03:42.34 | Umaro | Faithful: trying to find a siemens tech to come look at the siemens PBX and tell me what info it's getting or not getting that is making it call the wrong extension everytime |
| 03:42.34 | iMediax | Umaro: you try the "s" extension? |
| 03:42.41 | tessier__ | updatedb shows it nowhere on the system so it's not in this kernel I guess. |
| 03:42.42 | Umaro | iMediax: how do you mean? |
| 03:42.57 | iMediax | like s,1,Dial(Zap/g4/${EXTEN}) |
| 03:43.19 | Umaro | no.. would that work? |
| 03:43.37 | Umaro | ${EXTEN} wouldn't just be "s" at that point? |
| 03:43.57 | AgiNamu | Anyone here want to fax my disclaimer into digium for me? Please? :) I'll pay even. |
| 03:44.14 | iMediax | sure if your context is set to "inbound" it would be the fall back extension for the "inbound" context |
| 03:44.21 | Umaro | AgiNamu: fix my problem and i'll deliver it in person ;) |
| 03:44.43 | Umaro | iMediax: well, but the call is getting placed, you realize |
| 03:44.48 | AgiNamu | is it related to the siemens? |
| 03:44.56 | Umaro | AgiNamu: yes :) |
| 03:45.14 | AgiNamu | good luck :P |
| 03:45.27 | Umaro | AgiNamu: I'll even pickup a dozen krispy kreme donuts for kram on the way ;) |
| 03:45.39 | iMediax | your _X. exten for example would take the first number and then dial the following after the first if i remember correctly |
| 03:45.45 | bjohnson | is there a slick way to track incoming voip calls to different CDR accounts without making a separate incoming context for each service? |
| 03:45.58 | Umaro | iMediax: ohyeah?! |
| 03:46.05 | iMediax | bjohnson : accountcode |
| 03:46.17 | iMediax | set accountcode i think it is |
| 03:47.17 | Umaro | bjohnson: you can set the accountcode from the dialplan via the SetAccount app, or you can set it for each peer by accountcode=blah in their sip.conf or iax.conf entry. |
| 03:47.18 | bjohnson | ahh .. I was trying to use SetAccount in extensions.conf .. but I see now there is an accountcode= setting in iax.conf, sip.conf, etc |
| 03:47.47 | Umaro | iMediax: where can I read more about this? |
| 03:47.48 | iMediax | u can do in your dialplan as a specific extension too |
| 03:48.06 | iMediax | Umaro: ill get the wiki link |
| 03:48.22 | Umaro | iMediax: VERY interesting idea |
| 03:48.35 | Faithful | Yay! I think it's official... my lag problems are gone |
| 03:48.42 | *** join/#asterisk brc_ (~brian@ip24-251-178-25.ph.ph.cox.net) |
| 03:48.55 | iMediax | bjohnson: like this ;exten => _1NXXNXXXXXX,5,SetAccount(nufone) |
| 03:49.17 | bjohnson | yes .. I ws doing that |
| 03:49.36 | Umaro | Faithful: woot! |
| 03:49.44 | Toadyus | If I want to use and older analog phone I need to buy an FXS module? |
| 03:49.46 | bjohnson | but am happy I don't have to make an incoming dialplan context for each voip providers just to set the CDR account |
| 03:49.54 | bjohnson | Toadyus: yes |
| 03:49.57 | Toadyus | thnx |
| 03:50.08 | bjohnson | Toadyus: even a newer analog phone :) |
| 03:50.18 | Toadyus | the x100p just gives me FXO access the my POTS line? |
| 03:50.25 | bjohnson | yes |
| 03:50.37 | iMediax | Umaro: it work? |
| 03:50.55 | Toadyus | where can a guy pickup a cheap FXS module? |
| 03:51.07 | file[laptop] | well, you can draw one... that's pretty cheap |
| 03:51.22 | bjohnson | Toadyus: road house bar? |
| 03:51.37 | Umaro | iMediax: not sure yet.. my associate there is in bed atm.. which isn't a big problem, I can get him up and get him over there asap to try this, i'm just trying to get more info before I do. |
| 03:51.39 | *** join/#asterisk CoaxD (coax@shell1.cornernet.com) |
| 03:51.45 | iMediax | Umaro: http://www.voip-info.org/wiki-Asterisk+standard+extensions |
| 03:51.50 | iMediax | read about "s" |
| 03:52.02 | iMediax | under explanations |
| 03:52.03 | Toadyus | bjohnson - I need a real answer |
| 03:52.18 | Faithful | now just to patch my source for broadvoice and rebuild chan_sip.o |
| 03:52.22 | bjohnson | Toadyus: cheapest I've founf is the Sipura SPA 2000 from voxilla $80 for 2 fxs ports(+ $10 off coupon .. enter *users at checkout) |
| 03:52.36 | Toadyus | k sweet |
| 03:52.39 | tessier__ | The zaptel drivers seem to have changed significantly. |
| 03:52.44 | tessier__ | Maybe I just need to upgrade my drivers... |
| 03:52.46 | Toadyus | the sipurran external units work? |
| 03:52.48 | iMediax | Faithful, you using CVS? |
| 03:52.55 | bjohnson | Toadyus: it might be $75 .. I forget |
| 03:53.07 | Faithful | iMediax: no |
| 03:53.09 | iMediax | ok |
| 03:53.11 | bjohnson | Toadyus: I have 2 and have 4 more on the way |
| 03:53.21 | Faithful | iMediax: to scared to |
| 03:53.25 | Toadyus | cool.. |
| 03:53.31 | CoaxD | SPA-2000 is good stuff |
| 03:53.34 | Faithful | infact I am still using 1.0.2 |
| 03:53.38 | bjohnson | Toadyus: I have one 2000 and one 3000. I ordered 2 more of each |
| 03:53.40 | Toadyus | how do they interface with the box?.....through the fxo? |
| 03:53.48 | iMediax | hehe...i have a hard time getting stable play nice...CVS no problem |
| 03:53.51 | CoaxD | Toadyus: sip. |
| 03:54.05 | bjohnson | Toadyus: no .. lan .. just plug them into your lan |
| 03:54.11 | Faithful | has CVS got broadvoice patch in? |
| 03:54.19 | Toadyus | any links of info of installing...I was reading on voip-info couldn't find anything |
| 03:54.24 | iMediax | yep |
| 03:54.37 | Faithful | iMediax: yep to me? |
| 03:54.42 | CoaxD | "Broadvoice Patch"? What the hell did they change |
| 03:54.44 | iMediax | Faithful: yes |
| 03:54.44 | bjohnson | Toadyus: check the wiki. there is a link to a voxilla howto for a 2000 and * |
| 03:54.46 | Umaro | iMediax: hmm.. how does that relate to a PRI though? |
| 03:55.20 | Toadyus | bjohnson - sweet thnx... |
| 03:55.23 | Faithful | Hmmm.... tempting... but I am using .deb and I don't trust asterisk to build for me |
| 03:55.38 | CoaxD | Faithful: You really need to learn how to build asterisk. |
| 03:55.44 | Faithful | I guess I could try on my deb box |
| 03:55.51 | tessier__ | crap |
| 03:55.55 | Faithful | CoaxD: why so? |
| 03:55.55 | tessier__ | I cvs'd the unstable version. |
| 03:56.13 | tessier__ | I really wish digium would list the instructions for getting the stable version first instead of unstable and leaving stable as barely a footnote |
| 03:56.20 | CoaxD | Faithful: because as far as the last time i checked, the asterisk version in debian unstable was very, very old |
| 03:56.44 | Faithful | CoaxD: but it works ;-) |
| 03:57.02 | CoaxD | Faithful: Barely |
| 03:57.13 | Faithful | CoaxD: oh. |
| 03:57.24 | CoaxD | Faithful: "it works" is a copout. Asterisk users should be using CVS. |
| 03:57.25 | iMediax | Umaro, i'm just saying s is used if the exten isnt found |
| 03:57.31 | CoaxD | Faithful: You just don't want to have to learn anything |
| 03:58.08 | CoaxD | Faithful: You can't use asterisk without learning ltos of things; one of which is how to adequately fix bugs as you run into them - by applying latest cvs sources, etc |
| 03:58.20 | Faithful | CoaxD: Not so, it's just I spend sooooo much time compiling drivers / hacking etc that I avoid it when I can |
| 03:58.39 | CoaxD | Faithful: Let me assure you, you can't avoid it. |
| 03:58.59 | CoaxD | Faithful: Sooner than later you're going to get bit with a bug from the old tree, and when you do, your only choice will be to take your shit down and learn |
| 03:59.12 | CoaxD | Faith: Plus, what could be more easy than 'make && make install' in 3 directories? |
| 03:59.34 | Faithful | CoaxD: I CVS most stuff I compile. |
| 03:59.50 | tessier__ | Does *anyone* here have a working TDM400P? |
| 03:59.55 | CoaxD | tessier: Yup |
| 04:00.00 | CoaxD | tessier: with 1 FXS module |
| 04:00.14 | tessier__ | CoaxD: Cool. Nobody had ever popped up and said "works for me". |
| 04:00.19 | *** join/#asterisk justinnnnnn (justin@c211-28-205-205.eburwd1.vic.optusnet.com.au) |
| 04:00.19 | CoaxD | tessier: handle's my mom's desk phone |
| 04:00.22 | CoaxD | note: Thats not latest CVS tree |
| 04:00.24 | *** join/#asterisk expousr (~expousr@66.101.10.152) |
| 04:00.34 | CoaxD | tessier: Whats the problem exactly? |
| 04:01.03 | expousr | anyone have boxen at internap? |
| 04:01.09 | tessier__ | CoaxD: Well, I had a TDM401P card in this box but from the very beginning it would run for a few hours or maybe a day and then it would lock up. Not receieve incoming or make outgoing calls. |
| 04:01.24 | tessier__ | CoaxD: After a lot of talking digium acknowledged it as a known hardware flaw and said they hvae seen this before. |
| 04:01.30 | CoaxD | tessier: Could be driver, could be hardware |
| 04:01.36 | tessier__ | So I arranged an RMA. The new card arrived yesterday. |
| 04:01.44 | CoaxD | tessier: as in, the motherboard you are using |
| 04:01.58 | Faithful | CoaxD: chill, I hand patched the broadvoice stuff... I'm not affraid of work. This is my first asterisk dev box. I have had it running for about 5 days and I'm still figuring issues like 2 second lag on incoming hisax which I just fixed last night at 2am. I am not totally avoiding CVS, I just find it easier to bug fix & learn off of a system that is basically running. If I started with CVS I probably wouldn't have my box working yet. |
| 04:01.59 | CoaxD | tessier: hmmmm. What hardware flaw? I've had this card since...hmmm. i forget now. but its been a while |
| 04:02.24 | CoaxD | Faithful: Ummm, ok. |
| 04:02.36 | tessier__ | I am here to install it today and when I put the card in and boot the system up (without touching any of the hardware) and now it says Uhhuh. NMI received for unknown reason 31 on |
| 04:02.36 | tessier__ | CPU 0. |
| 04:02.40 | CoaxD | Faithful: Have fun with that |
| 04:02.53 | CoaxD | tessier: You can be assured that there's a problem with your motherboard |
| 04:02.56 | tessier__ | CoaxD: They were vague on specifics but I have an email from digium. |
| 04:03.00 | CoaxD | tessier: Try another slot on the board |
| 04:03.07 | tessier__ | CoaxD: This is a brand new Dell Poweredge 750. |
| 04:03.11 | tessier__ | CoaxD: Tried both slots. |
| 04:03.15 | CoaxD | tessier: Hahahaha. And you call that a machine? :) |
| 04:03.27 | tessier__ | It is more than enough to run a 30 user office. |
| 04:03.28 | CoaxD | tessier: That thing is barely a 1U, much less a machine! *lol* |
| 04:03.46 | CoaxD | tessier: bottom of the barrel, etc. |
| 04:03.51 | CoaxD | tessier: Note: I'm not saying it wont work |
| 04:03.58 | tessier__ | I don't buy that for a second. |
| 04:04.06 | CoaxD | tessier: I'm just saying - you'd better test this stuff in another machine just to verify that the same thing happens elsewhere |
| 04:04.12 | Faithful | CoaxD: But I _really_ do appreciate your input, and hope to be of help to others as I learn the tricks of the trade in asterisk |
| 04:04.14 | CoaxD | tessier: I have 2 of the machines here |
| 04:04.22 | tessier__ | I just compiled the driver on another box. |
| 04:04.23 | CoaxD | tessiedr: haven't had a problem with 'em |
| 04:04.41 | CoaxD | tessier: Then again, i havent been trying to plug buggy digium hardware into them either ;) |
| 04:04.51 | Faithful | tessier: are you running redhat in production? |
| 04:05.02 | tessier__ | You say it's gotta be the motherboard and then you say buggy digium hardware? :) |
| 04:05.02 | Faithful | RH9 or something |
| 04:05.03 | AgiNamu | Hey, I just submitted my first patch:http://bugs.digium.com/bug_view_page.php?bug_id=0003346 |
| 04:05.07 | tessier__ | Faithful: Yes, RHEL |
| 04:05.09 | CoaxD | Faithful: Heh. excellent |
| 04:05.10 | AgiNamu | Can someone quickly review me and make sure I didnt seriously fsck up? |
| 04:05.15 | CoaxD | tessier: hehe |
| 04:05.24 | Faithful | tessier: Ah Ok. |
| 04:05.24 | CoaxD | tessier: Could just be a compatibility issue |
| 04:05.56 | AgiNamu | im so nervous |
| 04:06.07 | file[laptop] | AgiNamu: lpc10? you're sick |
| 04:06.07 | CoaxD | AgiNamu: If it doesnt suck, somebody will know it ;) |
| 04:06.11 | tessier__ | CoaxD: Very odd that it mostly worked before, digium acknowledged the bug, and now it doesn't work at all when plugged into the same slot in the same board. |
| 04:06.22 | AgiNamu | file :) |
| 04:06.25 | CoaxD | tessier: Hmmmmmmm |
| 04:06.28 | CoaxD | what is lpc10 |
| 04:06.33 | *** join/#asterisk postel (~canonical@host217-42-82-130.range217-42.btcentralplus.com) |
| 04:06.34 | AgiNamu | LPC10 is a good example: it'd never get chosen otherwise. |
| 04:06.35 | tessier__ | I bet I can plug the old card back in (which I might have to do :() and it will work fine until it dies in a few hours. |
| 04:06.51 | Faithful | It took me 2 months to get into this. 6 weeks reading what I could and 2 weeks of no sleep when I decided to install. |
| 04:06.51 | CoaxD | tessier: Trust me on this; test in another machine |
| 04:06.59 | file[laptop] | CoaxD: it's a very ROBOTIC codec |
| 04:07.03 | AgiNamu | file, but i didnt break some massive rule and I most likely won't get -1000 karma? |
| 04:07.26 | file[laptop] | erm I'm looking at your patch now |
| 04:07.30 | CoaxD | tessier: A desktop machine with a semi-standard board in it. (i.e. not compaq. or packard bell. or something equally as stupid.) |
| 04:07.33 | AgiNamu | thanks! |
| 04:07.38 | Faithful | Then you look back and think "how did it take you so long to figure something so simple" but it isn't to a newcomer |
| 04:07.47 | tessier__ | CoaxD: That is what I am doing right now. This new machine doesn't even seem to see the card but I have only just started tweaking on it. |
| 04:07.57 | CoaxD | Faithful: That indicates that you aren't very good at this. |
| 04:08.13 | Faithful | CoaxD: at what? |
| 04:08.20 | CoaxD | Faithful: (by "this", i mean UNIX, amongst other things.) |
| 04:08.29 | Faithful | CoaxD: How so? |
| 04:08.31 | CoaxD | Faithful: Not that you have to be. But, there's a lot to learn about asterisk, so.. |
| 04:08.33 | file[laptop] | AgiNamu: the sip portion might throw a fit on older GCCs |
| 04:08.53 | AgiNamu | file oh damn, why? |
| 04:08.57 | CoaxD | Faithful: Most of the folks I know who get into asterisk compile and install it, and then learn the config mechanisms over approx. 2 weeks. |
| 04:09.07 | file[laptop] | not declaring the variables at the top of the function |
| 04:09.13 | *** join/#asterisk santiago (~santiago@63.245.86.86) |
| 04:09.18 | AgiNamu | But I saw other code doing this a lot |
| 04:09.23 | AgiNamu | if it's not function scoped |
| 04:09.31 | AgiNamu | you mean you can't do " for(int i ....)? |
| 04:09.37 | Damin | JerJer: You need help manning your booth at Speing Von? |
| 04:09.46 | tessier__ | CoaxD: I have it plugged into a standard whitebox system with an Asus mobo |
| 04:09.47 | CoaxD | Faithful: Unless of course you're new to UNIX too. in which case, yeah, i can see how you might not sleep for 2 weeks trying to get it installed |
| 04:09.49 | Faithful | CoaxD: yep that's what it took me... but I had zero idea about VoIP |
| 04:09.59 | file[laptop] | AgiNamu: that's why I said might |
| 04:10.06 | AgiNamu | hmm :S |
| 04:10.13 | AgiNamu | Other code is like that :).... and it's not function scoped. |
| 04:10.13 | CoaxD | Faithful: Are you new to UNIX? |
| 04:10.17 | AgiNamu | Should I change it around? |
| 04:10.20 | AgiNamu | I dont wanna get naield. |
| 04:10.21 | file[laptop] | AgiNamu: lemme look |
| 04:10.21 | AgiNamu | nailed. |
| 04:10.26 | AgiNamu | thanks a lot |
| 04:10.26 | tessier__ | http://www.mail-archive.com/asterisk-users@lists.digium.com/msg66149.html |
| 04:10.32 | tessier__ | That is *exactly* the problem I am seeing |
| 04:10.53 | CoaxD | tessier: probably a compatibility error between the hardware and the pci slot, if you want the truth |
| 04:11.15 | CoaxD | tessier: Are you still seeing the nmi errors on the asus board?? |
| 04:11.16 | Faithful | CoaxD: at the same time I was dd'ing a drive that had lost it's superblocks that I migh read it with hexedit... I don't know if I am new to unix. quite new compared to some I know. I am not a programmer. |
| 04:11.47 | CoaxD | Faithful: Hmmmmm. Okay. Well, i've been doing this Linux thing for about 10 years now. |
| 04:11.52 | file[laptop] | AgiNamu: go with the flow of the code, if stuff is defined at the top and not inside of it... stick with it... |
| 04:12.03 | CoaxD | Faithful: So a lot of the unixy things that go along with installing an application such as this - is stuff i've been doing for 10 years |
| 04:12.05 | Faithful | CoaxD: maybe about 5 years |
| 04:12.09 | file[laptop] | AgiNamu: it's hard to explain what I mean |
| 04:12.12 | CoaxD | Faithful: Understood |
| 04:12.13 | AgiNamu | file, no its not |
| 04:12.20 | tessier__ | CoaxD: The power connector is not required to be connected when using only FXS, correct? |
| 04:12.24 | AgiNamu | I understand, I was just tryign to make my code concise. |
| 04:12.27 | tessier__ | erl... |
| 04:12.31 | file[laptop] | I don't mean globally, I mean at the top of the function... yeah |
| 04:12.32 | tessier__ | When using only FXO I mean |
| 04:12.35 | tessier__ | To connecto to the telco |
| 04:12.35 | AgiNamu | and didnt think putting extra vars at the top that are conditionally used is... good. |
| 04:12.40 | *** join/#asterisk mhnoyes_ (~mhnoyes@user-2ivfnd9.dialup.mindspring.com) |
| 04:12.46 | AgiNamu | ok, I'll change it |
| 04:12.54 | file[laptop] | I still think GCC 2.95 might throw a slight fit at those |
| 04:12.59 | file[laptop] | let's go try and see |
| 04:13.05 | CoaxD | uh |
| 04:13.08 | AgiNamu | ok, tell me if i should |
| 04:13.11 | tessier__ | Coax |
| 04:13.14 | CoaxD | tessier: You need the frickin power connector no matter what you do |
| 04:13.15 | file[laptop] | AgiNamu: I'll get back to you |
| 04:13.18 | tessier__ | CoaxD: This new box is not even seeing the card. |
| 04:13.21 | file[laptop] | in like 10 minutes |
| 04:13.22 | AgiNamu | thanks |
| 04:13.33 | CoaxD | tessier: It powers the modules |
| 04:13.35 | file[laptop] | aha, a 2.95.3 box |
| 04:13.54 | tessier__ | CoaxD: I believe that is only to generate ring voltage on fxs cards. |
| 04:14.12 | tessier__ | Because I never had it hooked up on the old card and it worked just fine. |
| 04:14.17 | tessier__ | I'm pretty sure I googled this before. |
| 04:14.17 | *** join/#asterisk aminorex (~tony@c66.191.69.132.dul.mn.charter.com) |
| 04:14.21 | CoaxD | uh |
| 04:14.35 | CoaxD | tessier: JUST PLUG IT IN! |
| 04:15.30 | CoaxD | tessier:It obviously didnt "work just fine". it died after 3 hours |
| 04:15.48 | CoaxD | tessier: The card has the connector for a reason.use it. they may not have tested without it. |
| 04:15.53 | tessier__ | http://www.archivum.info/asterisk-users@lists.digium.com/2004-12/msg04703.html |
| 04:16.00 | CoaxD | dont spit urls at me |
| 04:16.31 | tessier__ | Sorry, just STFW'ing |
| 04:16.35 | CoaxD | i'm on a goddamn dumb terminal. i'm not going to type that in. :) |
| 04:16.54 | tessier__ | I'm pretty sure I asked digium about this too. |
| 04:17.12 | CoaxD | havefun with that |
| 04:17.32 | CoaxD | tessier:If you're incapable of plugging it in, testingfurther is a moot point |
| 04:17.49 | tessier__ | The Dell box does not have an extra power connector for this. |
| 04:17.56 | CoaxD | tessier: So geta goddamn splitter |
| 04:17.58 | file[laptop] | kick your legs up in the air |
| 04:18.00 | Silik0n | dumb terminals are so 1970 |
| 04:18.02 | file[laptop] | kick it like you just don't care |
| 04:18.02 | CoaxD | tessier: $2 at any computer store |
| 04:18.07 | Faithful | CoaxD: I heard about asterisk... so I typed apt-get install asterisk... say "Oh great" went into /etc/asterisk and said "man alive, what's all this" jumped in #asterisk asked "where to start" ... silence ... read the Asterisk Handbook, then I was a little confused. Got stuck into trying to get it to work and learn as I went (about 2 nights). Discovered the Hitch Hikers guide... ah, this is what I needed in the first place, that helped |
| 04:18.07 | Faithful | <PROTECTED> |
| 04:18.31 | Silik0n | ~urls |
| 04:18.50 | CoaxD | tessier: Guaranteed, the PE750 can handleit |
| 04:18.52 | file[laptop] | dreamer dream, take me in your dreams |
| 04:18.57 | file[laptop] | take me any way you please you're making me scream |
| 04:19.08 | CoaxD | tessier: Plus yer testing in a standard mobo right now. see if the failure mode happens with it plugged in. |
| 04:19.39 | wankel | anyone here use two sip devices behind the same linksys firewally before? it seems to be trying to be smart and help the NAT work by always using outside port 5060, but with two devices it... gets a little stupid. the second device get nat'd to port 15060... sometimes. sometimes they get swapped. it's confusing the hell out of asterisk, of course. |
| 04:19.42 | CoaxD | tessier: (You should be able to have the card recognized. only on insmodding zaptel stuff should it break.) |
| 04:20.00 | CoaxD | wankel: You'll need to set your sip client to a different port manually |
| 04:20.05 | Faithful | CoaxD: now I ponder things like "hisax driver must have a buffer issue with HFC cards to make asterisk lag 2 seconds" |
| 04:20.10 | CoaxD | wankel: Whether that client is a sipura2000 or x-lite, you can set it |
| 04:20.28 | file[laptop] | only you... can make me feel so true |
| 04:20.29 | CoaxD | wankel: But yes, i do it. |
| 04:20.33 | file[laptop] | I'll never never be so blue, as long as I have you |
| 04:20.42 | wankel | coax: it's outbound nat'd the linksys is picking the outside port. |
| 04:20.51 | Brixius | Whatcha singing file? |
| 04:21.03 | file[laptop] | AgiNamu: you broke pre-3 GCC compatibility |
| 04:21.05 | CoaxD | wankel: Heh :) |
| 04:21.12 | wankel | or is the linksys doing something stupid like trying to use the same port? |
| 04:21.25 | CoaxD | wankel: I'mtelling you,just set the sip port on each device to different ones, manually |
| 04:21.27 | file[laptop] | Brixius: Only You by Captain Jack |
| 04:21.35 | wankel | all i want is just normal outbound nat! i hate it when cheap crappy firewalls TRY to be smart and screw it up. |
| 04:21.37 | CoaxD | wankel: its what i had to do to get a linksys etherfast to work right[ |
| 04:21.47 | Faithful | CoaxD: I must admit my first six weeks was sporadic reading mags about VoIP convincing myself to suffer yet another learning curve. |
| 04:21.59 | CoaxD | Faithful: heh :) Reading can be fun! |
| 04:22.00 | wankel | coax: you only have this problem with linksys? |
| 04:22.05 | *** join/#asterisk _TripleF_ (kvirc@Toronto-HSE-ppp3883989.sympatico.ca) |
| 04:22.13 | CoaxD | wankel: No. anything NAT |
| 04:22.18 | wankel | really? |
| 04:22.21 | CoaxD | wankel: Remember, the linksys doesnt have a fscking clue what you're trying to do |
| 04:22.32 | wankel | the NAT at work uses randomly assigned outbound ports |
| 04:22.37 | AgiNamu | file, ok I'll fix it |
| 04:22.38 | wankel | s/outbound/outside/ |
| 04:22.58 | wankel | i think it's one of the fw-1 boxes that does the final NAT |
| 04:23.51 | CoaxD | wankel: Hm.well, i dunno. i'm not all that good at debugging nat problems |
| 04:23.51 | wankel | most nat boxes i've ever seen just pick random outside ports for each udp "conversation" |
| 04:23.51 | file[laptop] | fly up high to the sky oh so blue |
| 04:23.51 | wankel | bleah. this is bloody stupid! |
| 04:23.51 | wankel | coax: what other firewalls have you seen it with? |
| 04:23.51 | CoaxD | wankel: I just know what i had to do |
| 04:23.51 | CoaxD | wankel: me? I've used itjust on linksys |
| 04:24.08 | wankel | yeah, that sounds like that may work for the linksys, at the least. this is going to make it fun to deploy that 300-phone job. well, luckily they won't be using a linksys :) |
| 04:24.11 | CoaxD | wankel: i suck when it comes to widespread experience with this |
| 04:24.16 | Faithful | CoaxD: now "though a little flakey" I have working VoIP I can show my friends... however it has precipitated another issue... traffic shaping... yet another learning expedition |
| 04:24.16 | AgiNamu | file, any other comments, besides that? |
| 04:24.26 | file[laptop] | AgiNamu: good luck getting it into CVS |
| 04:24.31 | CoaxD | faithful: why traffic shaping |
| 04:24.39 | CoaxD | faithful: Why you need that? |
| 04:24.53 | Faithful | ADSL with 7 servers hanging off |
| 04:24.59 | CoaxD | Faithful: Um |
| 04:25.02 | *** join/#asterisk kludgebox (~bob@adsl-70-243-202-249.dsl.rcsntx.swbell.net) |
| 04:25.02 | AgiNamu | file, why's that? |
| 04:25.07 | file[laptop] | AgiNamu: because it's evil |
| 04:25.13 | file[laptop] | AgiNamu: plus SIP already has SIP_CODEC |
| 04:25.13 | AgiNamu | why? |
| 04:25.17 | AgiNamu | that's on answer |
| 04:25.19 | Faithful | websites, RDP all evil to VoIP |
| 04:25.24 | AgiNamu | I could not make it work on outgoing |
| 04:25.27 | CoaxD | Faithful: you could use QoS if the adsl router supports it |
| 04:25.36 | Brixius | yea, someone looked at my bug and wrote a patch.... |
| 04:25.41 | CoaxD | faithful: but thats only gonna work for outbound |
| 04:25.48 | file[laptop] | AgiNamu: but still, good luck... it's just evil doing codec stuff from the dialplan |
| 04:25.50 | Faithful | sometimes it's worse than a cell phone |
| 04:25.53 | CoaxD | faithful: someone pingfloods you and your voice call goes buh bye |
| 04:26.01 | CoaxD | Faithful: It can be, if improperly set up and latencies suck |
| 04:26.13 | AgiNamu | file, its a limitation if you cant decide the codec on the fly. What if you dial a cell phone and thus want to switch to GSM? |
| 04:26.21 | AgiNamu | why should one HAVE to edit their configs? |
| 04:26.34 | CoaxD | Faithful: With a sipura2000 that is local LAN to asterisk, i make a voip call to a local extension and i cannot tell the difference between voip and real POTS |
| 04:26.37 | Faithful | CoaxD: I don't run a mickey mouse adsl router... my adsl modem is in bridge mode |
| 04:26.51 | CoaxD | Faithful: Oh, to linux or somesuch? |
| 04:26.53 | AgiNamu | I'm sick of having my dialplan look like "Dial(SIP/Provider-G729)" and "SIP/Provider-ULAW" |
| 04:27.08 | file[laptop] | AgiNamu: it'll be interesting to see if it gets in |
| 04:27.26 | Brixius | no matter what router you are using if it's not sip aware, good luck. |
| 04:27.39 | Faithful | FreeBSD here but usually linux, and rate limit ping floods etc. |
| 04:27.47 | CoaxD | Faithful: Most excellent. |
| 04:27.58 | CoaxD | ipf GOOD |
| 04:28.18 | CoaxD | doesnt do any good to ratelimit pingfloods tho |
| 04:28.19 | Faithful | I love looking at firewall logs... most enjoyable |
| 04:28.27 | aminorex | tc-fu |
| 04:28.30 | CoaxD | bythe time it gets to your firewall, the bandwidth is used already |
| 04:28.31 | bjohnson | AgiNamu: can't you just allow both codecs in the config for that provider? |
| 04:28.46 | CoaxD | the only thing you can do is make sure the packets dont make it to a box behind the fw |
| 04:28.57 | Faithful | CoaxD: yeah generallly I do netfilter |
| 04:29.10 | bkw_ | AgiNamu |
| 04:29.18 | bkw_ | AgiNamu your patch doesn't work |
| 04:29.28 | bkw_ | it goes about it the wrong way also |
| 04:29.31 | Faithful | CoaxD: anyhow just drop pings if it is an issue |
| 04:29.34 | bkw_ | for IAX that is |
| 04:30.16 | CoaxD | Faithful: dropping them is irrelevant; my point is, someone can steal all your bandwidth whether you have a firewall in place or not |
| 04:30.28 | CoaxD | faithful: YOu have to open the packet to find out that it is icmp so you can drop it. :) :) |
| 04:31.17 | Faithful | CoaxD: given |
| 04:31.19 | iMediax | just pray you pipe is bigger :) |
| 04:31.25 | AgiNamu | ok thanks guys :S. I'll go back to the coding board. |
| 04:31.47 | file[laptop] | ooooh you're overwriting the native formats.. |
| 04:31.56 | Faithful | Well that's hard with ADSL 1.5/256 the other guy always wins |
| 04:32.08 | CoaxD | Faithful: Duh] |
| 04:32.29 | CoaxD | welp. bbl |
| 04:32.39 | Faithful | adios |
| 04:34.26 | Brixius | I heard there was codec prefrence order code that was written for iax and sip but Mark decided to pull it. Anyone know why? |
| 04:35.12 | CoaxD | downtime in 25 minutes. |
| 04:35.20 | file[laptop] | T-Minus Pi! |
| 04:36.23 | wankel | coax: holy crapy the linksys is broken. i moved the second phone to 5061, but it's still xlating the first phone to 15060 (it seems to go 5060, 15060, 25060, ...). i had to reboot the stupid linksys. |
| 04:36.48 | Faithful | wankel: no calm down... |
| 04:37.18 | *** join/#asterisk drkool (~drkool@210.211.144.224) |
| 04:37.36 | Brixius | Wankel: shouldn't that have been, "Holy Packets Batman, the Linksys is broken!!!" |
| 04:38.08 | CoaxD | wankel: Hehe |
| 04:38.19 | CoaxD | wankel: Linksys firewalls are, for the most part, very dumb |
| 04:38.37 | CoaxD | keep this in mind tho: The wrt54g's are small linux boxes. those are a LOT more powerful |
| 04:38.48 | *** join/#asterisk blankman (~blankman@h000d88a1570c.ne.client2.attbi.com) |
| 04:38.49 | wankel | that's what i have |
| 04:38.52 | wankel | but it does the same thing |
| 04:38.54 | CoaxD | wankel: hmm. |
| 04:39.02 | wankel | well, i haven't tried with two |
| 04:39.22 | CoaxD | wankel: I dunno. i've never configured one. allo i know is that if that is what you are using, you have a prayer in hell of fixing it |
| 04:39.23 | wankel | maybe the support for more than one sip client works on the wrt54g |
| 04:39.34 | CoaxD | who knows |
| 04:39.34 | wankel | if it was REALLY dumb it would work fine |
| 04:39.57 | Faithful | wankel: the router does mac magic, and if you fudge the connections buy messing with hardware the router "becomes" the device behind it (in practice) just wait a while and it all comes good... but the more you plug and unplug and fuss about the longer it takes, unless ofcourse you poweroff your linksys |
| 04:40.03 | blankman | Hey, anyone know how to make the spooler files just connect a call and not wait for the one end to 'pickup' before the other end does? |
| 04:40.09 | firestrm | anyone know why sox makes my sound files sound like darth vader when i convert to gsm? |
| 04:40.34 | Faithful | CoaxD: no way linksys rock!!! Linux WAPs with open source |
| 04:40.40 | blankman | firestrm, you need to fiddle with the encoding, google for an answer ... there are lots of hints on how to fix that :-) |
| 04:41.03 | firestrm | blankman, thanks.. going google.. |
| 04:41.06 | blankman | Has anyone tried the new zigbee kits? |
| 04:41.18 | Faithful | CoaxD: and great 3rd party firmware to make cisco routers jealouse |
| 04:41.20 | blankman | We were thinking it would be cool to have a "zigbee" phone call :-) |
| 04:41.33 | wankel | faithful: uh, yeah. anything with linux is cool, even if it doesn't work right. |
| 04:41.40 | Umaro | Woot. I bought my first package of squid jerky today. |
| 04:41.46 | wankel | cisco routers jealous. heh. hehehe. |
| 04:41.55 | drkool | k |
| 04:42.17 | iMediax | squid jerky? |
| 04:42.58 | blankman | So, back to the original question ... anyone here know how to make the spooler not call serially? The Wiki makes "explicit" mention of that fact that this is the out come for the example path listed ... but then doesn't show "how" to make the par. call? |
| 04:43.01 | drkool | Hi! could somebody please tell me if Digium would ship the ZAP cards to india |
| 04:43.13 | Umaro | iMediax: yes.. apparently it's a korean thing, found it at a korean food shop here |
| 04:43.17 | Faithful | wankel: have heaps of phones haning off wrt54g no probs... you just hit the mac magic "did you ever wonder just how a wap 'bridges' networks transparently"? |
| 04:43.46 | blankman | drkool, If they wont someone will ... or just pay someone to bring it in ... last I checked, India isn't on the no commerce list :-) |
| 04:44.24 | wankel | there's no bridging involved here. this is a layer 3 problem, not layer 2. |
| 04:44.30 | blankman | Umaro, isn't it also Japanese? |
| 04:44.31 | wankel | and no wap involved |
| 04:44.47 | Umaro | blankman: asian as a whole, probably |
| 04:44.48 | blankman | ... well I guess cuttle fish isn't squid ... but same family ;-) |
| 04:45.00 | Faithful | wankel: I thought you said it wast a wrt54g? |
| 04:45.15 | blankman | Bummer ... no one can help me ? |
| 04:45.17 | Umaro | Still haven't actually tried it yet, though.. wondering if I should crack it open before or after I eat my wife's spaghetti. ;) |
| 04:45.25 | blankman | heheheheeh |
| 04:45.26 | wankel | one of them is. one is just a linksys cable modem router. |
| 04:45.45 | blankman | Hey anyone here used the MediaMVP from HP? |
| 04:45.57 | drkool | blankman trouble is I am not sure if those will pass the customs..therefore I am little anxious before ordering. I was hoping there was somebody here with some prior exp in shipping those to india |
| 04:45.59 | blankman | Ops that should have been clearer ... Hauppenauge(sp?); |
| 04:46.18 | Toadyus | is there a way to split a 23b pri line into 2 different pbx's |
| 04:46.19 | *** join/#asterisk coppice (~chatzilla@234.168.17.210.dyn.pacific.net.hk) |
| 04:46.35 | iMediax | blankman: yes I have 3 |
| 04:46.44 | blankman | drkool ... I am sure they will pass customs ... the problem is how much will you have to pay ... I have sent Intel (dialogic) boards to India before ... |
| 04:46.52 | Umaro | Toadyus: T1 or PRI? |
| 04:46.56 | Toadyus | pri |
| 04:47.05 | coppice | does anyone know if * provides any support for free calls? |
| 04:47.06 | blankman | iMediax, so have you tried having it be your "caller" announcer yet? |
| 04:47.09 | Umaro | Toadyus: well, then you can't just splice it |
| 04:47.20 | blankman | iMediax, and have you tried MythPhone yet? |
| 04:47.25 | *** join/#asterisk Inv_arp (junya@adsl-8-230-153.mia.bellsouth.net) |
| 04:47.37 | Umaro | blankman: caller announcer? sorry, what is it you are talking about? |
| 04:47.41 | iMediax | nope |
| 04:47.44 | blankman | coppice, that is a wierd ? what exact you asking? |
| 04:47.47 | blankman | iaxtel? |
| 04:47.48 | drkool | blankman..would you have an idea how much was charged for those ..perhaps that would give me a ballpark figure |
| 04:47.49 | Toadyus | «Umaro» what about a T1...I can get a PRI line for cheaper than a T1..in canada both are tariffed...t1's are 900 and pri is only 350 |
| 04:48.05 | Faithful | blankman: I would be good if mythphone popped up ontop of the TV image :) |
| 04:48.09 | PatrickDK | heh, nothing is free |
| 04:48.13 | Toadyus | but my dad's office doesn't need 23 lines |
| 04:48.21 | PatrickDK | it takes time and money to even install asterisk |
| 04:48.21 | blankman | Umaro, basicly it is to pipe the incoming voicemessage to the tv :-) |
| 04:48.30 | coppice | blankman: does it ever detect or send a free call flag regardless of the signalling type |
| 04:48.32 | Umaro | blankman: lol |
| 04:48.34 | PatrickDK | and thn it's not much good unless you pay for internet access, or run your own wan |
| 04:48.36 | iMediax | really! |
| 04:48.46 | blankman | drkool, it was like three years ... sorry ... no help there ... look it up on their commerce site. |
| 04:48.47 | *** join/#asterisk datareactor (datareacto@203.81.192.33) |
| 04:48.56 | Umaro | Toadyus: so just order a PRI with less than 23 b channels? |
| 04:49.22 | drkool | thank anyway blankman..you give me hope. Thank you very much |
| 04:49.29 | Toadyus | it's not cost effective...it's cheaper to take all 23 |
| 04:49.50 | blankman | coppice, that I don't know the answer to ... my guess if you look at the channels it would explain :-) |
| 04:49.52 | PatrickDK | heh, the breaking point around here is about 10lines before 23 is cheaper |
| 04:50.02 | Umaro | hmm.. so your dad's office is going to share it with another office? |
| 04:50.07 | Toadyus | yes |
| 04:50.11 | blankman | drkool, if you can't find someone to ship there ... you would have a good reason to start a company :-) |
| 04:50.26 | Toadyus | but I dunno how a clec adds two main lines onto one pri |
| 04:50.31 | Toadyus | is that even possible |
| 04:50.33 | coppice | blankman: I can't see anything about free calls, but I might be missing something |
| 04:50.35 | drkool | lol |
| 04:50.50 | blankman | drkool, have the cards made in china then shipped to india ... that I am for sure you can do cause I just did something like that for other card types :-) |
| 04:50.55 | PatrickDK | toadyus, what do you mean two main lines? |
| 04:51.03 | PatrickDK | you just mean two numbers? |
| 04:51.08 | blankman | ummm ... are you looking for a flag in which channel? |
| 04:51.09 | Umaro | Toadyus: what if you had an * server that took the whole PRI in, and then had another T1 card in the * box and forwarded whatever ports weren't yours out the other card to their phone system? your phone system wouldn't mess with the calls at all, just forward them |
| 04:51.17 | drkool | blankman do you mean the digium cards.. |
| 04:51.19 | Faithful | coppice: VoIP is free in the sense to other VoIP users |
| 04:51.20 | blankman | sorry, that was for coppice |
| 04:51.22 | Toadyus | there is two offices in his building when I talked to sprint they said they could only attach the pri to one line not both... |
| 04:51.35 | blankman | Nope, it as an embedded h.323 project. |
| 04:51.39 | Faithful | I don't know anyone offering "free" land line callse |
| 04:51.44 | Toadyus | «Umaro» ya that's an idea |
| 04:51.45 | blankman | again for drkool. |
| 04:51.56 | Toadyus | they would just have to keep there main line and cancel all there numbers |
| 04:52.07 | Toadyus | «Umaro» good idea |
| 04:52.14 | *** join/#asterisk stupidnic (~foo@cs68201147-69.sw.rr.com) |
| 04:52.24 | PatrickDK | toadyus, you can easily split out the pri, it might be a problem with what phone numbers they have though |
| 04:52.24 | coppice | Faithful: I'm not concerned abotu VoIP calls. Its PSTN calls that matter |
| 04:52.32 | Damin | Welp.. |
| 04:52.37 | Toadyus | «PatrickDK» ya that |
| 04:52.41 | drkool | blankman..if the specs for the digium cards are avaialble I can get them built in india .. |
| 04:52.48 | Umaro | Toadyus: or you could sell them phone service, lol |
| 04:52.50 | blankman | Toadyus, buy a re-used adtran and then route over the t1 cards in it ... |
| 04:52.50 | Toadyus | «PatrickDK» ya that's what I was thinking might have to pay for the main line.. |
| 04:52.52 | Faithful | coppice: then I guess the answer is no |
| 04:53.10 | Toadyus | then I can just give them 10 lines off the main pri |
| 04:53.16 | blankman | drkool, no such luck .. that is their business :-) |
| 04:53.26 | Damin | chan_sip + Polycom problem doesn't appear in 1/1/2005 version of chan_sip from cvs-stable.. |
| 04:53.27 | Toadyus | I need to keep one or myself...not going to pay for phone service again haha |
| 04:53.31 | blankman | You have to find someone to do the engineering to make the cards ... the drivers ... etc |
| 04:53.32 | blankman | :-) |
| 04:53.38 | Moc_ | Damin, what problem ? |
| 04:53.54 | PatrickDK | toadyus, channel bank will split the pri into 23 normal phone lines |
| 04:54.06 | Moc_ | I havent had problem with my polycom phone (except this h323 conference phone ;) |
| 04:54.07 | PatrickDK | I know my cisco stuff will split it into phone lines, or two other pri lines |
| 04:54.27 | Moc_ | but I got incoming to work on my Polycom SoundStation IP 3000 |
| 04:54.38 | Moc_ | but outgoing still doesnt work |
| 04:54.40 | blankman | Toadyus, what PatrickDK said, but it is cheaper to take a refirb'd adtran :-) |
| 04:54.58 | Toadyus | «PatrickDK» is there special pci cards for the asterisk box to use a pri or do those quad t1 cards work |
| 04:55.01 | drkool | blankman lol actually i too think it would make a lot more sense to just order from Digium and pay the custom duties. |
| 04:55.15 | Toadyus | «blankman» how much are refurb'd adtrans? |
| 04:55.20 | PatrickDK | toadyus, as far as asterisk is concerned, t1=pri |
| 04:55.24 | Toadyus | wicked |
| 04:55.27 | blankman | Toadyus, we use a model Total Access 850. |
| 04:55.33 | PatrickDK | the wire is the same, just the software the drives it is different |
| 04:55.42 | Toadyus | I'm going to have DID's all over alberta too.. |
| 04:56.16 | blankman | I think with the cards it was like usd1800 per ... but I think you can do it cheaper by "handing out" a frac. t1 instead of the 23 pots we do :-) |
| 04:56.53 | Toadyus | i saw a 24 port zhone fxs on ebay for like 600 bux |
| 04:57.04 | *** join/#asterisk afrosheen (~afro@c-24-0-136-170.client.comcast.net) |
| 04:57.16 | afrosheen | someone please tell me how they have their network, asterisk is killing me |
| 04:57.35 | blankman | afrosheen, huh? |
| 04:57.40 | afrosheen | uh I mean tell me how your network is setup..i.e. in a business environment |
| 04:57.42 | PatrickDK | netowkr? |
| 04:57.57 | afrosheen | pulling my hair out over our implementation |
| 04:58.00 | blankman | afrosheen, many boxes, many t1's, many people :-) |
| 04:58.13 | PatrickDK | oh, for voip, layer2 and layer3 switchs are a must :) |
| 04:58.16 | blankman | No ... really what is your ? |
| 04:58.22 | Toadyus | I still need to get a box in production...been reading for last 2 days straight at work...need to get a fxo card |
| 04:58.23 | blankman | How the IP or the PSTN is setup? |
| 04:58.26 | afrosheen | ok check it |
| 04:58.41 | afrosheen | the asterisk box has a 10.x.x.x address on one port of the firewall |
| 04:58.57 | blankman | Toadyus, There are a couple of card makers now ... look the wiki |
| 04:59.07 | afrosheen | 1 to 1 nat and port forwarding is enabled so externally the asterisk box has a 64.x.x.x address |
| 04:59.32 | blankman | ... before you go to far, what is your up time? |
| 04:59.35 | afrosheen | now, the pc's are on another network port on 192.x.x.x on the lan |
| 04:59.44 | afrosheen | uptime is good, the asterisk box is stable |
| 04:59.57 | blankman | afrosheen, no, what is your sla for that box? |
| 05:00.16 | afrosheen | sla for this box? fuck if I know, we built it |
| 05:00.19 | blankman | It always needs to be up ... you need it to be as cheap as possible ... well those are the extrems. |
| 05:00.39 | afrosheen | don't derail me, there's more to the story :) |
| 05:00.48 | blankman | afrosheen, that is what I mean ... is this the bet your business on it .. ie your a call center ... or does everyone use cell phones and it is simply a "nice to have"... |
| 05:00.52 | Damin | Moc_: Calls don't complete... Call starts to progress, and then you get fast busy that you can't get rid of on the phone.. It's happeneing with multiple phones on different firmware revisions. With 1/1/2005 chan_sip.c from cvs-stable, problem is not apparent. |
| 05:01.07 | afrosheen | blankman: it's our company phone system |
| 05:01.14 | afrosheen | blankman: for a small company |
| 05:01.17 | blankman | afrosheen, here is my point. You should load balance your * box then. |
| 05:01.21 | stupidnic | Can somebody clue me in on this error? "ZT_CHANCONFIG failed on channel 1: No such device or address (6)" is that a config error or a hardware/device error (udev)? |
| 05:01.28 | afrosheen | blankman: we don't need it yet |
| 05:01.35 | blankman | k :-) |
| 05:01.41 | afrosheen | 30 or so employees, 6 pstn lines and 2 tdm400 cards |
| 05:01.43 | blankman | back to the clients :-) |
| 05:01.50 | PatrickDK | heh, everyone needs it |
| 05:01.54 | PatrickDK | just most won't pay for it |
| 05:01.55 | blankman | 192.xxxx |
| 05:02.06 | blankman | heeheh |
| 05:02.28 | PatrickDK | hell, I just got done loadbalancing dhcpd of all things |
| 05:02.28 | blankman | We have to pay for it, it takes us four boxes for evey "server" we put in prod :-) |
| 05:02.53 | blankman | So afrosheen you were saying? |
| 05:03.28 | Damin | Moc: I'm trying to pinpoint it down to a specific day.. Right now, it is between 1/1/2005 and 1/4/2005, but I'm in the process of trying 1/1/2005 now. |
| 05:03.30 | afrosheen | ok soooo |
| 05:03.36 | blankman | stupidnic, I think that may be a hardware issue with the zaptel drivers ... do you have rights to the instance you are running? |
| 05:03.49 | afrosheen | since the phones are on 192.x.x.x and the server is on 10.x.x.x I have nat issues. |
| 05:04.03 | blankman | afrosheen, this is all in your corp? |
| 05:04.09 | afrosheen | blankman: yes |
| 05:04.10 | stupidnic | blankman: I am not sure what you mean by "rights to the instance" |
| 05:04.10 | blankman | afrosheen, one box? |
| 05:04.15 | afrosheen | not one box |
| 05:04.17 | blankman | afrosheen, meaning one * box? |
| 05:04.18 | PatrickDK | afrosheen, use vlans |
| 05:04.20 | afrosheen | yes one * |
| 05:04.32 | blankman | stupidnic, in /dev/zaptel |
| 05:04.33 | PatrickDK | put all the phones onto one vlan |
| 05:04.34 | stupidnic | are you say that I need to roll back the CVS I pulled down? |
| 05:04.38 | PatrickDK | and shove asterisk on it also |
| 05:04.47 | blankman | PatrickDK, that is one way to solve it :-) |
| 05:04.54 | PatrickDK | that would be the best solution, then you can solve lag problems too |
| 05:05.01 | stupidnic | blankman: I added both the udev entries and permissions |
| 05:05.02 | blankman | afrosheen, what are you using? sip, h323? |
| 05:05.09 | afrosheen | sip :( |
| 05:05.40 | blankman | afrosheen, ... switch to iax2 it will save you money trust me ... and since you have complete control it will be okay ... I promise you of this. |
| 05:05.53 | afrosheen | so, with nat=yes the clients can make calls to each other |
| 05:05.58 | drkool | ppl..will using a USB headset instead of a soundcard for xten's eyebeam help reduce lag..( however small a factor) |
| 05:06.01 | stupidnic | blankman: should the modified udev entries be zap? or zaptel? the README says one thing, but the errors seem to indicate something else. |
| 05:06.22 | blankman | stupidnic, ummm .. give me more data ... what are you running? |
| 05:06.28 | afrosheen | too late to switch to iax2, I have 20 polycom sip phones already |
| 05:06.29 | blankman | stupidnic, hardware wise? |
| 05:06.43 | stupidnic | blankman: TDM400P with 2FXS and 1FXO |
| 05:06.47 | stupidnic | on Fedora Core 3 |
| 05:06.51 | blankman | can you put an asterisk server on the 192.x.x.x lan? |
| 05:06.54 | afrosheen | PatrickDK: yeah vlans are probably a good idea since the phones and client pc's are on the same wire |
| 05:07.04 | PatrickDK | afroseen, I would very highly not recommend you run sip across nat |
| 05:07.06 | afrosheen | blankman: yes I can but then I have nat issues with the outside world |
| 05:07.08 | blankman | stupidnic, hang on let me check something ... |
| 05:07.13 | PatrickDK | if you must, put one asterisk box with the net the phones are on |
| 05:07.22 | blankman | let me look the thing up for stupidnic .... |
| 05:07.22 | PatrickDK | and run iax2 between that box and your current one |
| 05:07.38 | blankman | PatrickDK, has it :-) |
| 05:07.58 | afrosheen | so I need another * box? ah jeez |
| 05:08.13 | PatrickDK | that, or completely change your net setup |
| 05:08.35 | PatrickDK | if your net setup contains wans, that could be hard to do |
| 05:08.36 | afrosheen | PatrickDK: that's possible, we have complete control over it, but the phones and the client pc's MUST share the same wire |
| 05:08.41 | blankman | stupidnic, it should be zap. |
| 05:08.51 | PatrickDK | afroseen, look up vlans |
| 05:08.57 | PatrickDK | if you want them to share the same wire |
| 05:09.04 | stupidnic | blankman: damn, okay... that is what I have |
| 05:09.16 | afrosheen | PatrickDK: yeah we have our cisco router and 2 3500xl switches that are vlan-aware |
| 05:09.21 | afrosheen | PatrickDK: so that's do-able |
| 05:09.39 | PatrickDK | ya, just set the phones up to use vlan # |
| 05:09.54 | afrosheen | PatrickDK: but what do we do about sip/nat on the firewall..have an * server on a public routable IP _next_ to the firewall? |
| 05:10.13 | stupidnic | blankman: I am getting (what I think are indications of the problem) in messages when I modprobe zaptel |
| 05:10.16 | PatrickDK | oh heh, asterisk is outside the firewall? |
| 05:10.16 | Brixius | Afro: What type of firewall? |
| 05:10.24 | afrosheen | Brixius: clarkconnect 2.2 |
| 05:10.35 | Brixius | never heard of it. |
| 05:10.42 | blankman | stupidnic, pastebin the error |
| 05:10.43 | iMediax | CC rocks |
| 05:10.45 | afrosheen | it's a linux distro/firewall, clarkconnect.org |
| 05:11.02 | afrosheen | it's very cool but the user forums are full of morons |
| 05:11.04 | Brixius | so ipf based? |
| 05:11.06 | stupidnic | blankman: something to the effect of wait_for_sysfs and then it references /class/zaptel/zaptimer, etc |
| 05:11.09 | PatrickDK | afrosheen, do you plan to make alot of sip connections to other people? |
| 05:11.20 | iMediax | not ipf |
| 05:11.22 | afrosheen | PatrickDK: you never know..we have some users that have sip phones at home |
| 05:11.33 | afrosheen | PatrickDK: like the CEO for example..and myself |
| 05:11.52 | PatrickDK | dunno, two box might be best for firewall |
| 05:12.02 | PatrickDK | sip and firewalls don't get along |
| 05:12.06 | PatrickDK | sometimes you can make it work |
| 05:12.18 | afrosheen | PatrickDK: yeah no shit but you don't see Polycom selling iax2 phones either |
| 05:12.19 | *** join/#asterisk doughecka_ (~Doug@12-220-235-106.client.insightBB.com) |
| 05:12.19 | wankel | do you guys know of anyone who offers multi-market DIDs with IAX or SIP service for a reasonable price? |
| 05:12.19 | blankman | stupidnic, that sounds like your build is wierd ... did it go clean? |
| 05:12.43 | PatrickDK | afro, that is cause the phones where never meant to go across firewalls |
| 05:12.46 | afrosheen | PatrickDK: asterisk does ok with one nat but not with two nats |
| 05:12.58 | *** join/#asterisk gambolputty (~gambolput@cblmdm65-221-51-80.buckeye-express.com) |
| 05:13.05 | stupidnic | blankman: as far as I can tell, yes... I ahd some problems with the 2.6.10 kernel so I roleld it back (behind a firewall anyways) and tried it again. |
| 05:13.14 | afrosheen | i.e. home user behind his goofy netgear router calls to asterisk, that works but no voice gets across due to rtp issues |
| 05:13.19 | PatrickDK | afro, iax2 should work across many messed up nats |
| 05:13.21 | stupidnic | blankman: the errors are here: http://slim.expresshosting.net/error.phps |
| 05:13.33 | blankman | afrosheen, you are playing with fire if you are letting the CEO "dial" into the server from home ... there is no encryption yet. |
| 05:13.49 | afrosheen | blankman: we can do that with a vpn tunnel through clarkconnect |
| 05:15.05 | blankman | k ... I haven't found the the QoS through vpn's is good enough ... enough of the time :-) |
| 05:15.07 | blankman | stupidnic, looking .... |
| 05:15.18 | afrosheen | blankman: if all else fails there's ssh tunneling also |
| 05:15.18 | iMediax | afrosheen: our CC box is on our10.x.x.x subnet, and my location is on another subnet and It works |
| 05:15.35 | stupidnic | mmm ssh tunnels... I love them so |
| 05:15.36 | afrosheen | iMediax: can you get to it from the outside? |
| 05:15.58 | afrosheen | iMediax: cause we have it working fine like that with nat=yes for all extensions (phones) |
| 05:16.01 | blankman | stupidnic, your build didn't work right for the given platform ... we use suse, so I will not be of much help here ... someone else will have to dive in ... |
| 05:16.17 | iMediax | outside of the firewall? |
| 05:16.22 | afrosheen | iMediax: yes |
| 05:16.31 | blankman | stupidnic, I can tell you that the error is saying that the zaptel timer hasn't been symlinked though ... |
| 05:16.38 | iMediax | yes. I can use it say from home etc.. |
| 05:16.48 | afrosheen | iMediax: how? 10.x isn't routable to the outside |
| 05:17.01 | iMediax | via the public IP |
| 05:17.12 | afrosheen | iMediax: uh..the firewall's external ip? |
| 05:17.13 | iMediax | it's nat IP eth1 is 10x.x.x |
| 05:17.21 | afrosheen | iMediax: right |
| 05:17.23 | iMediax | that would be outside |
| 05:17.24 | *** join/#asterisk greendisease (~jack@greendisease.fedora) |
| 05:17.50 | afrosheen | iMediax: did you use nat=yes for your clients on 192.x? |
| 05:18.04 | iMediax | yes |
| 05:18.06 | *** join/#asterisk jcims (~chatzilla@cpe-69-135-121-57.columbus.rr.com) |
| 05:18.14 | stupidnic | blankman: where would then be symlinked to? |
| 05:18.28 | stupidnic | blankman: I show the entries in the /dev/zap/ |
| 05:18.38 | blankman | afrosheen, just so you are clear, we have a seven * box deep config for some of our phones ... so I promise you it will work with iax2 |
| 05:18.40 | afrosheen | iMediax: is port forwarding or any 1 to 1 nat setup on your firewall? |
| 05:18.57 | stupidnic | I have to assume it means a more internal (kernel level perhaps) linking |
| 05:18.59 | blankman | stupidnic, hang on I will look ... you said tdm400 not 410p? |
| 05:19.06 | stupidnic | tdm400 |
| 05:19.13 | stupidnic | I appreacite your assistance |
| 05:19.31 | iMediax | ok i c... our * is on the firewall |
| 05:19.50 | afrosheen | iMediax: what do you mean 'on the firewall', actually installed ON the cc box? |
| 05:19.55 | iMediax | yep |
| 05:19.57 | afrosheen | oh man |
| 05:20.22 | iMediax | haha |
| 05:20.27 | afrosheen | I have 4 nics already |
| 05:20.34 | blankman | stupidnic, guess that is wrong ... now that I am looking at the /dev/zap on my box ... |
| 05:20.35 | afrosheen | guess I could yank 2 of them out |
| 05:21.05 | afrosheen | iMediax: I had considered putting * on the firewall though, but how does CC affect it's traffic |
| 05:21.49 | iMediax | 2.2 was a nightmare.... i risked loading 3beta4 and used shaping..... love it |
| 05:22.28 | iMediax | i set SIP using the highest priority so it always has the bandwidth now |
| 05:23.13 | afrosheen | iMediax: the only thing is, the firewall is on amd64 hardware, wonder if there will be any issues |
| 05:23.39 | stupidnic | blankman: are there any known problems with zaptel out of CVS the past couple of days? |
| 05:23.51 | blankman | stupidnic, ... not sure here ... someone on his platform know anything? |
| 05:24.05 | iMediax | we're using ulaw only to avoid excess cpu usage... but it's a dual xeon |
| 05:24.15 | blankman | stupidnic, not sure there either ... |
| 05:24.41 | stupidnic | I might just scrap it and go to Debian like I planned on doing before. that is what I get for trying something new. :) |
| 05:25.08 | datareactor | can i use net2phone account with asterisk |
| 05:25.11 | Damin | Alright.. |
| 05:25.14 | iMediax | just DMZ the asterisk box on CC |
| 05:25.16 | Damin | I narrowed it down.. |
| 05:26.24 | blankman | stupidnic, debian I could help with, suse I could help with ... anything else ... not so much help :-) |
| 05:26.37 | afrosheen | I know mandrake :) |
| 05:26.56 | afrosheen | iMediax: if I dmz it the lan clients can't use it |
| 05:27.11 | stupidnic | blankman: I am sure Debian will work properly. I should never second guess myself like that. :) |
| 05:27.13 | afrosheen | iMediax: I think it'll end up with 1 way traffic |
| 05:27.32 | afrosheen | iMediax: or are you suggesting that the DMZ get real, public IP's |
| 05:27.36 | blankman | So, one last time anyone on know how to make the pbx spooler make a call that connects the channels at the same time? |
| 05:27.47 | afrosheen | no |
| 05:27.48 | iMediax | yes real ip |
| 05:27.58 | blankman | Damin ... you know? |
| 05:28.01 | iMediax | asterisk box have 32 nics? |
| 05:28.03 | afrosheen | iMediax: ok that's one thing I haven't tried yet |
| 05:28.04 | iMediax | err 2 |
| 05:28.14 | blankman | hehehe |
| 05:28.17 | afrosheen | iMediax: sorta, onboard and one extra |
| 05:28.20 | blankman | I was like that is impressive :-) |
| 05:28.54 | blankman | 32 was an eight slot backplane with quad-port nics ... that would be a box :-) |
| 05:29.02 | iMediax | haha |
| 05:29.04 | Damin | blankman: The problem shows up between 1/1/2005 chan_sip.c and 1/2/2005 chan_sip.c in cvs-stable. |
| 05:29.06 | afrosheen | blankman: well having a firewall with a total of 5 nics, 4 of them gigabit, on amd64 hardware with a gig of ram is kinda impressive |
| 05:29.23 | iMediax | 5? |
| 05:29.25 | Damin | blankman: I've got a diff of the two versions and it isn't a lot of code.. |
| 05:29.41 | blankman | Damin, that makes it easy to back out then huh ;-) |
| 05:30.05 | blankman | Did you ever get your lvs working 100% |
| 05:30.09 | blankman | that was for damin |
| 05:30.17 | Damin | blankman: Well, yeah.. if someone has a clue.. which I don't.. ;) |
| 05:30.36 | blankman | We ended up using external balancers ... |
| 05:30.38 | Damin | blankman: No.. never followed through on it.. |
| 05:30.39 | afrosheen | iMediax: one onboard (gigabit), 4 nics, 3 of which are intel ee1000pros |
| 05:31.09 | iMediax | i've just got 2 on my CC box |
| 05:31.23 | blankman | Damin, they were way easier to deal with ... and since they already have all the fail-over (active/passive) it worked out well ... |
| 05:31.27 | afrosheen | iMediax: that's really all it needs, but we wanted to experiment a little and don't trust onboard crap in linux |
| 05:31.40 | blankman | Damin, we have been looking at a Redline Networks box ... |
| 05:31.57 | Damin | blankman: What make/model external balancers did you go with? |
| 05:32.48 | blankman | Damin, also, we have been looking at using https for our tunnels ... it is really kinda cool ... we ended up making http1.1 type tunnel, but we are using the coyotepoints for now for the ssl exceleration we need ... it is loooking pretty promising :-) |
| 05:33.05 | blankman | Damin, looks like this: |
| 05:33.59 | blankman | <phones>--<lb>--<*>--<fw>--<Inet>--<fw>--<lb>--<*>--<phones> |
| 05:34.31 | afrosheen | ouch |
| 05:34.58 | afrosheen | wait you only have one prospective nat there, that looks better than what I'm looking at :) |
| 05:35.05 | blankman | Where the lbs are doing the https tunnels ... we create two connections that are "open" using chunking ... seems to work out pretty well.... we have tested 100 calls with ulaw and no issues. |
| 05:36.33 | afrosheen | iMediax: so, your asterisk box shares the gateway IP for your firewall or did you create a virtual interface on the external with a public ip? |
| 05:36.41 | blankman | Damin, from talking to redline rep, it is something that will only get better with there "backplane", as it is highly oppimized for that kind of stuff :-) |
| 05:36.44 | iMediax | same |
| 05:36.51 | afrosheen | ? |
| 05:37.17 | *** join/#asterisk Guest27745 (~JapanCocu@n219079037057.netvigator.com) |
| 05:37.18 | iMediax | eth0 is public, eth1 private |
| 05:37.27 | iMediax | * listens on eth0 |
| 05:37.41 | afrosheen | iMediax: ok so lemme get this straight |
| 05:37.59 | *** join/#asterisk [TripleF] (kvirc@Toronto-HSE-ppp3884232.sympatico.ca) |
| 05:38.00 | afrosheen | iMediax: *'s ip is the firewall's external, public ip |
| 05:38.06 | iMediax | yes |
| 05:38.13 | afrosheen | iMediax: that way outside clients don't have to deal with nat coming in |
| 05:38.20 | iMediax | correct |
| 05:38.30 | afrosheen | iMediax: and you just nat=yes all your internal clients |
| 05:38.37 | iMediax | yep |
| 05:38.51 | afrosheen | iMediax: now, how does CC shape the traffic if it's running ON the firewall? |
| 05:39.07 | iMediax | have u played w/ 3beta4? |
| 05:39.18 | blankman | Damin, you want to write for a new journel me and a friend started ... it is dedicated to realtime * and iax. |
| 05:39.20 | afrosheen | iMediax: beta4/5 is crap from what we've seen |
| 05:39.24 | iMediax | u can shape by port. |
| 05:39.31 | iMediax | it's been stable for me |
| 05:39.34 | afrosheen | it gets the external and lan interfaces backwards from the start, that pissed us off |
| 05:40.05 | afrosheen | everyone said 'just switch the cables' and I was like wtf |
| 05:40.06 | iMediax | there is a bug using more that 2 nics |
| 05:40.18 | iMediax | i tried adding a 3rd and everything was backwards |
| 05:40.20 | afrosheen | oh really? more than 2? so if I yanked some cards it'd work? |
| 05:40.26 | iMediax | ya |
| 05:40.53 | afrosheen | iMediax: how much of a pain is it to build asterisk on redhat |
| 05:40.57 | iMediax | ya 2 seems solid...there are still known bugs... but none i care about... like the CUPS setup crap |
| 05:41.26 | iMediax | apt-get cc-devel and compile *.. done! |
| 05:42.02 | iMediax | oh and read the forum for the *correct* kernel source |
| 05:42.13 | iMediax | i did make a mistake on that |
| 05:42.55 | datareactor | what i have to setup for musiconhold in extention.conf i have configured musiconhold.conf |
| 05:43.11 | Faithful | tessier: how's it going? |
| 05:43.45 | *** join/#asterisk Brixius (Brixius@c-24-118-4-197.mn.client2.attbi.com) |
| 05:44.10 | afrosheen | iMediax: does your cc box have any tdm cards or anything? |
| 05:44.38 | iMediax | yes, but we arent using them yet...only as timing for now |
| 05:44.49 | afrosheen | iMediax: so the modules built and installed ok? |
| 05:44.53 | blankman | datareactor, you have to turn it on for the dial or call the music on hold app. |
| 05:44.55 | iMediax | ya |
| 05:45.19 | iMediax | tdm400 we're using now |
| 05:45.20 | afrosheen | iMediax: hmm ok I might have to give this a shot then..it's making alot of sense, but dmz with public ip's may be an option |
| 05:46.22 | iMediax | even asterisk on cc, and then IAX to your existing * box might work |
| 05:46.44 | iMediax | depending on your provider |
| 05:47.37 | *** join/#asterisk Josh1 (Joshwah@adsl-8-161-173.mia.bellsouth.net) |
| 05:47.48 | Josh1 | hey guys |
| 05:47.50 | Josh1 | i have a question |
| 05:48.07 | blankman | We may have an answer :-) |
| 05:48.40 | Josh1 | im getting this when im trying to get broadvoice to register with asterisk |
| 05:48.48 | datareactor | Blankman what i want that when caller press callonhold from xlite they hear some music |
| 05:48.52 | Josh1 | Jan 15 00:48:34 NOTICE[24412]: chan_sip.c:4018 sip_reg_timeout: -- Registration for '9543371424@sip.broadvoice.com' timed out, trying again |
| 05:49.05 | afrosheen | iMediax: I appreciate your help, it's giving me some hope ) |
| 05:49.12 | Josh1 | anderiv, i copied everything correctly from their site |
| 05:49.16 | Josh1 | and* |
| 05:49.31 | blankman | So there are lots of examples for that did you google already? |
| 05:49.37 | blankman | That was for datareactor. |
| 05:50.52 | Josh1 | alright |
| 05:51.12 | datareactor | blankman OK thanks |
| 05:51.19 | *** join/#asterisk jonas (jonas@jonas.probe-networks.host.freenode) |
| 05:53.21 | *** join/#asterisk ptblank (~MURDER1@67.21.12.21) |
| 05:53.35 | *** join/#asterisk cluv (~trillian@209.169.111.95) |
| 05:54.33 | *** join/#asterisk BoRiS (~boris@S01060040ca1e5b54.wp.shawcable.net) |
| 05:54.37 | cluv | What are the standard RTP ports for *? |
| 05:54.59 | blankman | same as for any other system I think ... did you look at the wiki ... they are all listed there. |
| 05:55.13 | cluv | i saw 8000 in most posts |
| 05:55.29 | cluv | but i have seen a couple in the 5000 range |
| 05:55.48 | blankman | cluv ... if you search on ports in the wiki, it iwll give you the "defaults" that asterisk "ships" with. |
| 05:55.52 | Faithful | man X is chewing 376Mb ram...? |
| 05:56.01 | Faithful | that's a bit rich |
| 05:56.01 | blankman | Why are you runnign x? |
| 05:56.12 | iMediax | aren't they in rtp.conf |
| 05:56.35 | jcims | X is the original bloatware |
| 05:56.54 | blankman | iMediax ... might be, but I don't use anything but iax so ... I only have one port :-) |
| 05:57.42 | cluv | iMediax they sure are there...looks like 10000 to 20000 |
| 05:57.48 | iMediax | ;) |
| 05:57.50 | Josh1 | dang |
| 05:57.54 | Josh1 | ive tried everything |
| 05:57.56 | Josh1 | still now working |
| 05:57.56 | Josh1 | sip_reg_timeout: -- Registration for '9543371424@147.135.0.128' timed out, trying again |
| 05:58.15 | iMediax | voip provider? |
| 05:58.27 | iMediax | BV? |
| 05:58.28 | Josh1 | yeah |
| 05:58.34 | stupidnic | hehe 954 |
| 05:58.39 | iMediax | use sip.broadvoice.com |
| 05:58.44 | Josh1 | alright ill try |
| 05:58.49 | stupidnic | Ft Liquordale |
| 05:59.39 | Josh1 | stupidnic, yup yup :) |
| 05:59.57 | stupidnic | born and raised there myself :) |
| 06:00.02 | stupidnic | then.. I left. :) |
| 06:00.20 | Josh1 | heh |
| 06:00.25 | jcims | Asterisk: The Insomniac's PBX |
| 06:00.35 | stupidnic | You know it |
| 06:00.35 | Josh1 | well, i got rid of that timeout it seems |
| 06:00.41 | Josh1 | but im still getting my voicemail |
| 06:00.43 | coppice | Asterisk: The Vampire's PBX |
| 06:01.15 | stupidnic | mainly because I pay the bills with my daylight hours... and try to make my non-payable stuff work at night |
| 06:01.23 | iMediax | and u want to use "mia" 147.135.4.128 |
| 06:01.42 | jcims | i'm just a junkie |
| 06:01.44 | Josh1 | hmm alright |
| 06:01.54 | Josh1 | iMediax, thats for the host and the register right? |
| 06:01.55 | Josh1 | Jan 15 01:01:11 NOTICE[5175]: chan_sip.c:7379 handle_request: Unable to create/find channel |
| 06:01.55 | Josh1 | Jan 15 01:01:17 WARNING[5175]: chan_sip.c:695 retrans_pkt: Maximum retries exceeded on call 46832c4d-1572447a-58a854d2@192.168.1.95 for seqno 22794 (Non-critical Response) |
| 06:01.58 | Josh1 | cuz thats what i get |
| 06:02.03 | stupidnic | armed with my Debian ISO I am off to wipe out this damn Fedora Core 3 server... |
| 06:02.12 | iMediax | u need to add an extry to /etc/hosts |
| 06:02.16 | iMediax | entry* |
| 06:02.31 | Josh1 | hmm |
| 06:02.32 | Josh1 | for? |
| 06:02.39 | jcims | giddyup. i did the same thing about two weeks ago. |
| 06:02.46 | iMediax | add 147.135.4.128 sip.broadvoice.com to /etc/hosts |
| 06:02.51 | jcims | i'm loving apt-get |
| 06:02.51 | Josh1 | alright |
| 06:02.56 | stupidnic | Man my server closet needs some ventalation |
| 06:03.02 | blankman | coppice, did your problem get fixed? |
| 06:03.07 | stupidnic | I open that door and it is like a mini furnace in there |
| 06:03.15 | stupidnic | stupid SCSI drives heating up the place |
| 06:03.23 | coppice | blankman: which of my many problems? :-) |
| 06:03.45 | blankman | Don't remember that was an hour ago :-) |
| 06:04.00 | Josh1 | iMediax, still cant call out :/ |
| 06:04.12 | coppice | then its probably well down the pushdown stack by now :-) |
| 06:04.13 | iMediax | one sec ill show the sip.conf |
| 06:04.16 | Josh1 | Jan 15 01:04:05 WARNING[5175]: chan_sip.c:695 retrans_pkt: Maximum retries exceeded on call 558d3f8-356e7cae-28fb4e37@192.168.1.95 for seqno 888 (Critical Response) |
| 06:05.09 | Josh1 | kk |
| 06:06.41 | iMediax | Josh1: http://pastebin.ca/4220 |
| 06:07.16 | Josh1 | thanks iMediax |
| 06:07.21 | iMediax | np |
| 06:07.50 | datareactor | Josh1 i also get this message on my asterisk |
| 06:08.20 | afrosheen | jcims: you'd probably like urpmi better :) |
| 06:08.36 | jcims | what's that? |
| 06:08.41 | *** join/#asterisk jaiger (~jaiger@65.86.55.130) |
| 06:08.47 | afrosheen | jcims: mandrake's equivalent to apt-get |
| 06:08.52 | jcims | ahh. |
| 06:08.55 | afrosheen | works about the same, better some might say :) |
| 06:09.20 | jcims | haven't tried mandrake yet. |
| 06:09.29 | ZX81 | PTG123: yah |
| 06:09.41 | afrosheen | jcims: it's very debian-ish in some spots, redhat-ish in others |
| 06:10.14 | afrosheen | stupidnic: save on power bills and put wet laundry in that closet |
| 06:10.38 | Josh1 | Jan 15 01:09:46 WARNING[12480]: chan_skinny.c:2584 reload_config: Unable to get our IP address, Skinny disabled |
| 06:10.39 | Josh1 | Jan 15 01:09:46 WARNING[12480]: chan_oss.c:434 soundcard_init: Unable to open /dev/dsp: Permission denied |
| 06:10.39 | Josh1 | Jan 15 01:10:16 NOTICE[12992]: chan_sip.c:7379 handle_request: Unable to create/find channel |
| 06:10.39 | Josh1 | Jan 15 01:10:22 WARNING[12992]: chan_sip.c:695 retrans_pkt: Maximum retries exceeded on call 70776b90-16e34838-3b7f3012@192.168.1.95 for seqno 6568 (Non-critical Response) |
| 06:10.39 | Josh1 | Jan 15 01:10:24 WARNING[12992]: chan_sip.c:695 retrans_pkt: Maximum retries exceeded on call 70776b90-16e34838-3b7f3012@192.168.1.95 for seqno 6568 (Critical Response) |
| 06:10.41 | afrosheen | stupidnic: you've got plenty of hot air blowing in there |
| 06:10.46 | afrosheen | AAAAHH FLOOD |
| 06:10.56 | blankman | Josh1, learn to use pastebin. |
| 06:11.05 | Josh1 | heh |
| 06:11.27 | wankel | afrosheen: bah, that was nothing. it wasn't even in bold or flashing. |
| 06:11.33 | jcims | afrosheen: i think the redhat heritage is why i never tried it...seemed like i could just 'get the real thing'. i did take another look at it when i went with debian, but didn't know it had taken it's own approach to pkg mgmt |
| 06:11.47 | Josh1 | iMediax, still not working *tear* |
| 06:11.54 | wankel | mandrake still exists? |
| 06:11.57 | *** join/#asterisk phenix (phenix@216.253.66.149) |
| 06:12.03 | afrosheen | still exists..feh |
| 06:12.07 | iMediax | some of those warnings u can ignore |
| 06:12.08 | jcims | my 'floods' are pretty anemic, i just now figured out how to do multiline in chatzilla |
| 06:12.15 | iMediax | brb |
| 06:12.18 | phenix | hi. so im having trouble getting asterisk to dial out with broadvoice. |
| 06:12.29 | stupidnic | afrosheen: Yeah... it is pretty hot in there... and to think people are paying big bucks for a vertical dryer theses days... I have one and I only paid a couple of bucks for the 15K RPM SCSI drives |
| 06:12.41 | afrosheen | jcims: they extracted most of the redhat 'heritage' (read sabotage) years ago |
| 06:12.43 | Josh1 | phenix, same here |
| 06:12.47 | phenix | i keep getting um, timeouts from broadvoice, and then asterisk is sending a 403 back to the softphone |
| 06:13.10 | afrosheen | phenix: is broadvoice hosted with internap or who |
| 06:13.12 | Josh1 | im not even using a softphone :P |
| 06:13.15 | jcims | afrosheen: good on them :) |
| 06:13.20 | Josh1 | yeah internap is down |
| 06:13.24 | Josh1 | and they are pingable |
| 06:13.26 | Josh1 | but i dunno |
| 06:13.35 | phenix | i dont think its related |
| 06:13.42 | phenix | it was working ok directly from the softphone to them |
| 06:13.48 | afrosheen | internap took a big shit today, livejournals are dead |
| 06:13.54 | Josh1 | yep |
| 06:13.57 | stupidnic | internap isn't down for me |
| 06:14.00 | afrosheen | godzilla snapped the powerlines again |
| 06:14.04 | afrosheen | that asshole |
| 06:14.18 | jcims | does anyone running voip-info freqent here? i've got a tikiwiki question for them... |
| 06:14.32 | Josh1 | gah i cant make outgoing calls from my phone :/ |
| 06:14.41 | iMediax | bv timeouts? |
| 06:14.46 | ZX81 | use your fish then |
| 06:14.48 | ZX81 | phones sux |
| 06:14.53 | ZX81 | :-) |
| 06:14.54 | phenix | josh, its working for incoming? |
| 06:14.56 | afrosheen | babelfish? |
| 06:14.59 | ZX81 | hehe |
| 06:15.02 | ZX81 | Cod |
| 06:15.04 | ZX81 | Snapper |
| 06:15.07 | ZX81 | blue cod best |
| 06:15.08 | afrosheen | tuna |
| 06:15.09 | ZX81 | :-) |
| 06:15.13 | ZX81 | hehe |
| 06:15.18 | afrosheen | chicken of the sea |
| 06:17.57 | blankman | Pork the other white meat. |
| 06:18.16 | Josh1 | someone please help |
| 06:18.34 | afrosheen | Josh1: don't worry, you can call your girlfriend tomorrow |
| 06:18.39 | iMediax | haha |
| 06:18.43 | Josh1 | lol |
| 06:18.46 | Josh1 | Jan 15 01:18:31 NOTICE[12992]: chan_sip.c:7379 handle_request: Unable to create/find channel |
| 06:18.46 | Josh1 | Jan 15 01:18:37 WARNING[12992]: chan_sip.c:695 retrans_pkt: Maximum retries exceeded on call 447a71ac-3dc96336-798e664c@192.168.1.95 for seqno 24159 (Non-critical Response) |
| 06:18.54 | datareactor | :) |
| 06:19.00 | phenix | josh |
| 06:19.07 | phenix | whats your extensions.conf look like |
| 06:19.14 | datareactor | Josh1 i get this error but i still able to call |
| 06:19.15 | blankman | Pastebin it :-) |
| 06:19.43 | Josh1 | pastebin? |
| 06:19.51 | blankman | Hey, so anyone on now know how to help me with the pbx_spooler? on how to make it make a serial call? |
| 06:20.06 | datareactor | yes pastbin.ca is site where u can paster yr config |
| 06:20.18 | datareactor | pastebin.ca |
| 06:20.28 | afrosheen | mavis beacon |
| 06:21.00 | Josh1 | http://pastebin.com/229109 |
| 06:21.21 | iMediax | no |
| 06:22.37 | datareactor | Josh1 this is very incomplete info what is your sip.conf |
| 06:22.59 | Josh1 | datareactor, hold on a sec, iMediax is trying :) |
| 06:23.15 | datareactor | Josh1 OK |
| 06:23.39 | iMediax | ya his dialplan is all wrong |
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| 06:24.50 | blankman | nite all. |
| 06:25.03 | Josh1 | night |
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| 06:25.46 | phenix | humm |
| 06:25.48 | phenix | datareactor |
| 06:25.51 | phenix | want to look at mine? :) |
| 06:26.19 | datareactor | Yes phenix :) |
| 06:26.38 | phenix | i used the uh, sample configs, so its ugly. |
| 06:27.04 | bkw_ | GO TO HELL PEOPLE!!! PAGING EVERYONE TO HELL!!! |
| 06:27.36 | gambolputty | ? |
| 06:27.39 | iMediax | haha |
| 06:27.44 | afrosheen | last train to hell, all aboooooard! |
| 06:27.51 | bkw_ | those that know |
| 06:28.02 | datareactor | phenix never mind everybody starts like that |
| 06:28.27 | phenix | http://pastebin.com/229113 |
| 06:29.17 | phenix | the softphone is supposed to register extension 8001 |
| 06:31.11 | Nugget | you can't register an extension. |
| 06:31.37 | phenix | what does, regexten do? |
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| 06:34.36 | phenix | great. |
| 06:34.36 | afrosheen | kaboom |
| 06:34.36 | Josh1 | :P |
| 06:34.36 | datareactor | i think whole ISP is rebooted |
| 06:34.36 | stupidnic | ah damn |
| 06:34.36 | iMediax | the hell bus |
| 06:34.36 | bkw_ | BoRiS |
| 06:34.36 | bkw_ | where are you |
| 06:34.36 | bkw_ | oh BoRiS |
| 06:34.36 | bkw_ | file |
| 06:34.36 | stupidnic | our side of the netsplit was the "cool side" |
| 06:34.36 | stupidnic | we had strippers and booze on our side |
| 06:34.36 | Brixius | Well that was fun |
| 06:34.36 | Qwell | we had bundt cake |
| 06:34.37 | phenix | anyway |
| 06:35.30 | stupidnic | damn... I wanted a cake |
| 06:35.30 | afrosheen | bundt cake and strippers, awesome |
| 06:35.30 | afrosheen | that sounds like a good time |
| 06:35.30 | Brixius | I wanted the stripper's to stay, but no you had to shoo them out before everyone got back... |
| 06:35.30 | stupidnic | Well I screwed up my Debian install and I have the infamous 01 scrolling... |
| 06:35.30 | stupidnic | so I am going to take a queue and go to bed |
| 06:35.30 | afrosheen | stupidnic: yay, a hosed bootloader |
| 06:35.30 | stupidnic | Yeah... I know |
| 06:36.26 | stupidnic | damn IDE drive probably messing with things |
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| 06:36.27 | stupidnic | So I can take hint... |
| 06:36.27 | stupidnic | night |
| 06:36.27 | phenix | so im getting service unavailable on the phone now |
| 06:37.09 | phenix | and a not found from the peer |
| 06:38.49 | iMediax | phenix: what's "sip show registry" say |
| 06:39.05 | phenix | the broadview server and the softphone |
| 06:39.17 | phenix | actual, just broadvoice |
| 06:39.36 | phenix | registered |
| 06:39.58 | coppice | kram: does * have any support for free calls? |
| 06:40.07 | kram | for free calls? |
| 06:40.24 | brc_ | coppice, eh?.. |
| 06:40.42 | coppice | some PRI and T1RBS lines allow you to control if the call is charges. All R2 and SS7 lines do |
| 06:40.49 | brc_ | ahh |
| 06:40.57 | iMediax | phenix is sip now registered to BV? |
| 06:41.34 | phenix | bravo*CLI> sip show registry |
| 06:41.34 | phenix | Host Username Refresh State |
| 06:41.34 | phenix | sip.broadvoice.com:5060 adam3584 Registered |
| 06:41.47 | iMediax | ok that part is fixed |
| 06:42.07 | iMediax | now dialplan |
| 06:42.12 | phenix | http://pastebin.com/229113 is most likely where the problem is |
| 06:42.37 | kram | coppice: you can ust not answer the line |
| 06:42.39 | iMediax | is that your current sip.conf? |
| 06:42.48 | phenix | excerpts from sip and extensions |
| 06:43.30 | phenix | the phone and broadvoice are registered as peers |
| 06:43.47 | sturothrock | greetings all! any tips for successfully compiling ijx for the 2.6 (fedora 3) kernel? |
| 06:43.54 | coppice | kram: with T1RBS you can (sometimes), with other signalling its different. The important bit is does * have any way to tell a channel if the call should be free? |
| 06:44.42 | kram | on PRI if you send PROGRESS with the "information available" bit usually you can transmit whatever audio youw ant |
| 06:45.47 | coppice | kram: yeah, yeah, I know how to make the signalling do free calls. what I don't have right now is an indication from * to the channel driver to tell it the call should be free. |
| 06:46.07 | kram | just don't answer the line :) |
| 06:46.17 | *** part/#asterisk drkool (~drkool@210.211.144.224) |
| 06:48.05 | *** join/#asterisk Guest^DJ (~some@218.111.35.203) |
| 06:48.26 | Guest^DJ | hi, is any one free to help me on my IVR ? |
| 06:48.50 | ZX81 | ~data |
| 06:48.51 | jbot | Don't Ask To Ask. Just ASK |
| 06:48.51 | coppice | kram: that doesn't work with things like R2, but it isn't the point. Some calls need to be free, and some do not. I need an indicator to tell the channel, based on information the channel doesn't have. I could put some ranges fo free call numbers in the channel's config file but I was hoping there was a more generic solution |
| 06:49.49 | Guest^DJ | i do not know how to end my ivr, it keep loop foever. i was told setvar but nothing on setvar i could find in wiki |
| 06:50.09 | ZX81 | Guest^DJ: I'll help you in private chat |
| 06:51.40 | Damin | http://bugs.digium.com/bug_view_page.php?bug_id=0003348 |
| 06:51.48 | ZX81 | saw it |
| 06:51.50 | ZX81 | you sure? |
| 06:51.59 | ZX81 | :-) |
| 06:52.01 | bkw_ | OMG its kram |
| 06:52.01 | ZX81 | I guess so |
| 06:52.09 | ZX81 | didn't see the diff Damin |
| 06:52.15 | kram | yah i guess |
| 06:52.19 | ZX81 | :-) |
| 06:52.27 | phenix | media? |
| 06:52.32 | iMediax | yes |
| 06:52.37 | ZX81 | Super-Duper-Programmer!!!!!!! |
| 06:52.38 | phenix | anything special? :) |
| 06:52.58 | *** join/#asterisk JimVanM (~JimVanM@HSE-MTL-ppp76783.qc.sympatico.ca) |
| 06:53.03 | ZX81 | :-) |
| 06:54.40 | brc_ | Guest^DJ, you'll need to use both SetVar(${loops} + 1) and GoToIf($[${loops} > 3]?hanguppriority) |
| 06:54.52 | brc_ | you'll have to play with it |
| 06:55.02 | *** part/#asterisk JimVanM (~JimVanM@HSE-MTL-ppp76783.qc.sympatico.ca) |
| 06:55.14 | brc_ | actually |
| 06:55.17 | ZX81 | :-) |
| 06:55.21 | brc_ | nevermind, you get the idea |
| 06:55.22 | ZX81 | he has other probs |
| 06:55.31 | ZX81 | no extensions |
| 06:55.31 | brc_ | eh |
| 06:55.38 | ZX81 | and an i exten that goes back |
| 06:55.44 | ZX81 | to the start |
| 06:55.50 | brc_ | :) |
| 06:56.02 | ZX81 | :-) |
| 06:56.25 | *** join/#asterisk JimVanM (~jimvanm@HSE-MTL-ppp76783.qc.sympatico.ca) |
| 06:56.25 | brc_ | ~brc_ |
| 06:56.26 | jbot | you are, like, Brian is available for custom Asterisk consulting...Contact brian@summitcompany.com for details...Paypal @ paypal@brc007.com, or lame, or excellent with IVRs |
| 06:56.42 | ZX81 | and interviews |
| 06:56.43 | ZX81 | :-) |
| 06:56.44 | brc_ | =) |
| 06:56.44 | ZX81 | sooon |
| 06:56.46 | ZX81 | soz |
| 06:56.49 | ZX81 | :-) |
| 06:56.50 | brc_ | eh? |
| 06:56.57 | ZX81 | gonna interview you |
| 06:57.02 | ZX81 | for daily asterisk news |
| 06:57.31 | brc_ | I haven't exactly been keeping up with the cvs updates have I |
| 06:57.38 | brc_ | so much time, so little to do |
| 06:57.57 | ZX81 | :-) |
| 06:58.05 | ZX81 | all good |
| 06:58.43 | Guest^DJ | brc_: there was nothing on SetVar in wiki |
| 06:58.48 | brc_ | oh comon people |
| 06:59.09 | brc_ | I can't believe nobody noticed I said that backwards |
| 06:59.09 | bkw_ | http://www.cnn.com/2005/LAW/01/14/daughter.killed.ap/index.html |
| 06:59.10 | brc_ | oh well |
| 06:59.13 | bkw_ | thats soooo wrong |
| 06:59.20 | ZX81 | :-) |
| 06:59.25 | brc_ | Guest^DJ, and you tried "show application setvar" on the asterisk console? |
| 06:59.47 | ZX81 | bkw_: harsh |
| 06:59.48 | brc_ | Guest^DJ, and, try searching the wiki with the google search box on the left side of the wiki....the built in search is *USELESS* |
| 07:00.03 | brc_ | sad |
| 07:00.06 | Guest^DJ | brc_: roger |
| 07:01.25 | Damin | kram: Hey there.. |
| 07:01.46 | Damin | kram: Has russell dropped 1.0.4 yet? |
| 07:02.00 | ZX81 | don't think so |
| 07:02.05 | ZX81 | must be soon though |
| 07:02.08 | ZX81 | :-) |
| 07:02.18 | Damin | I don't think he should until we can get this resolved: http://bugs.digium.com/bug_view_page.php?bug_id=0003348 |
| 07:02.21 | ZX81 | some changelogs are in |
| 07:02.38 | kram | damin: how about a sip debug |
| 07:02.52 | ZX81 | :-) |
| 07:02.53 | Damin | Otherwise, it's possible that there will be a LOT of people with polycom phones complaining.. |
| 07:02.57 | ZX81 | I was just typing that |
| 07:02.58 | ZX81 | lol |
| 07:03.21 | Damin | kram: Gladly.. how should I proceed? I tried "sip history" but it didn't seem to have anything.. |
| 07:03.31 | kram | it's "sip debug" |
| 07:03.33 | kram | then make the call that fails |
| 07:03.41 | kram | do one with the failed call and one with the working one |
| 07:04.06 | Damin | kram: Give me 5 minutes and I'll have them up there.. |
| 07:04.25 | Damin | kram: Damn! :) I just posted that 10 minutes ago.. ;) |
| 07:21.04 | *** join/#asterisk PinG (~I_Love_Yo@81.213.226.0) |
| 07:21.35 | Damin | kram: Alright.. working/non-working SIP debugs posted.. |
| 07:24.42 | phenix | and then a 404 not found from the server |
| 07:25.00 | phenix | and then asterisk sends a service unavailable to the phone |
| 07:25.54 | Josh1 | baaahhhh |
| 07:25.55 | Josh1 | Jan 15 02:25:36 NOTICE[28576]: chan_sip.c:7379 handle_request: Unable to create/find channel |
| 07:25.55 | Josh1 | Jan 15 02:25:41 WARNING[28576]: chan_sip.c:695 retrans_pkt: Maximum retries exceeded on call 7dce0c-3d0f7493-414816e8@192.168.1.95 for seqno 20622 (Non-critical Response) |
| 07:26.07 | Corydon76-home | Damin: remove the mailbox subscription from the Polycom config file |
| 07:26.31 | Corydon76-home | Rely only on sip.conf to tell the Polycom which mailboxes have voicemail |
| 07:26.54 | Corydon76-home | That's how we fixed our Polycom 500s |
| 07:27.31 | Corydon76-home | The problem is that the mailbox subscription turns the line into "Shared" and makes calls impossible, as the line is already in use |
| 07:28.58 | iMediax | is BV patch included in the ftp download of 1.0? |
| 07:30.07 | afrosheen | what's this about ip500's having problems? |
| 07:30.30 | *** join/#asterisk Josh1 (~josh@adsl-8-161-173.mia.bellsouth.net) |
| 07:30.48 | kram | damin: can you confirm? |
| 07:31.30 | Damin | Corydon-w: Let me take a look at that and see.. |
| 07:32.33 | phenix | thanks |
| 07:32.53 | bkw_ | I recall this being an issue hermie pointed out for someone else last week |
| 07:33.02 | bkw_ | the polycom has some option you need to turn off |
| 07:33.04 | Damin | Coyrdon: So you want me to get rid of mwi msg.mwi.1.subscribe="100" msg.mwi.1.callBackMode="contact" msg.mwi.1.callBack="85100" |
| 07:33.05 | bkw_ | or something |
| 07:33.13 | bkw_ | ya get rid of that |
| 07:33.15 | bkw_ | thats it |
| 07:35.49 | Damin | bkw_: I'm trying it to confirm, but I believe that is going to totally break 1-touch voicemail access on the phone. |
| 07:35.51 | *** join/#asterisk andrew` (~andrew@adsl-67-119-27-177.dsl.snfc21.pacbell.net) |
| 07:37.24 | Damin | Well, that seems to work.. |
| 07:37.41 | Damin | But, as expected, it breaks One Touch Voicemail.. |
| 07:37.53 | Damin | Which sucks.. |
| 07:38.56 | kram | so what's the root cause then? |
| 07:39.05 | kram | other than the polycom being a POS |
| 07:39.07 | afrosheen | Damin: remap the key then |
| 07:39.27 | afrosheen | Damin: it's in the admin guide, make the key dial *98(extension) or whatever |
| 07:40.20 | afrosheen | for the life of me I haven't gotten the hang of loading alternate ringers into these polycoms but someday... |
| 07:40.53 | Damin | afrosheen: You can't do that on the IP 300. |
| 07:41.17 | afrosheen | Damin: what, remap keys? |
| 07:41.34 | afrosheen | or load tones |
| 07:41.55 | Damin | afrosheen: There is no "Voicemail Key" on the IP 300 like there is on the 500 and 600. You have to press "Menu" then "2" |
| 07:42.16 | *** join/#asterisk gambolputty (~gambolput@cblmdm65-221-51-80.buckeye-express.com) |
| 07:42.37 | afrosheen | Damin: huh, yeah you're right, just tested it |
| 07:42.47 | *** join/#asterisk flewid (~flewid@CPE0050ba8c9a95-CM000f9fac6da2.cpe.net.cable.rogers.com) |
| 07:42.49 | flewid | sup |
| 07:43.10 | afrosheen | Damin: and the ip300 blows because it doesn't have any keys to remap really |
| 07:43.13 | Damin | afrosheen: But my question is what about the simple changes in chan_sip from 1/1 to 1/2 are causing the problme? |
| 07:43.35 | Damin | afrosheen: Yes, but that isn't the issue. ;) It worked just fine up until the 1/2 chan_sip updates in cvs-stable. |
| 07:43.47 | afrosheen | Damin: I'm assuming you've already loaded 1.3.x on the phones |
| 07:43.57 | Damin | afrosheen: 1.3.4.003 |
| 07:44.10 | afrosheen | Damin: send it to me please |
| 07:44.16 | afrosheen | :) so hard to obtain |
| 07:44.31 | Damin | Email? |
| 07:44.51 | flewid | is anyone here using spandsp? |
| 07:44.57 | afrosheen | Damin: dcc? |
| 07:47.52 | *** join/#asterisk fleeeb (~basin@pc-24-151-28-122.newm2.ct.charter.com) |
| 07:48.01 | ta[i]nted | how do i supress NOTICES in * CLI? |
| 07:48.26 | flewid | edit your logger.conf and/or change the amount of -vv's when you load asterisk. |
| 07:55.59 | Damin | OK.. |
| 07:56.08 | kram | so what's our story? |
| 07:56.46 | Damin | The story is that I am going to see if there is a way to make one-touch voicemail work with current chan_sip. |
| 08:01.00 | flewid | finally got my netweb-301 to register with asterisk (sorta) |
| 08:01.28 | flewid | i can call zap devices or sip devices with it, but can't call from sip/zap to the iax phone, and asterisk always says 'unreachable' for the iax phone... |
| 08:01.37 | flewid | at least it's a start =) |
| 08:04.21 | kram | damin: still here? |
| 08:06.21 | Damin | kram: Yep... |
| 08:06.47 | Damin | kram: Just reading the Polycom manual (mostly crap) and waiting for my phone to reboot. I THINK I have a solution... |
| 08:06.52 | Damin | kram: But I'm testing.. |
| 08:07.20 | Damin | kram: Yep.. |
| 08:07.28 | Damin | kram: Solved the problem... |
| 08:07.57 | Damin | kram: You CANNOT have the msg.mwi.1.subscribe tag set. |
| 08:08.06 | Damin | kram: I had my Voicemailbox number in there.. |
| 08:08.10 | modulus_ | polycom is the best ip phone i've used yet |
| 08:08.20 | flewid | Damin: ah, yeah, cause tyhe phone will 'always be busy' |
| 08:08.41 | *** join/#asterisk NeScHe`mUziK`diN (~SeLma_-@81.212.12.215) |
| 08:08.58 | Damin | kram: So, from Corydon's suggestion, + the manual + the http://www.voip-info.org/wiki-Getting+MWI+on+Polycom+Phones+to+work+with+Asterisk page, I've got it all working as it should.. |
| 08:10.28 | brc_ | could ya post that to the bug for future reference?... |
| 08:12.06 | Damin | brc_: Already done.. |
| 08:12.17 | brc_ | :) |
| 08:13.00 | Damin | kram: I'm going to "resolve" the bug, but put the caveat that you CANNOT use mwi subscribe w/ the phone. |
| 08:13.03 | brc_ | so should it be closed?...or is there a problem with chan_sip... |
| 08:13.38 | kram | okay great |
| 08:13.41 | *** join/#asterisk Che (~GuNDuZ@81.212.12.97) |
| 08:13.49 | kram | well yes, lets clarify that |
| 08:14.01 | kram | what about what i mentioned in the bug tracker? |
| 08:14.14 | kram | can we compare the pre- and post- bug subscribe logs |
| 08:15.04 | brc_ | you're up late :) |
| 08:15.27 | *** join/#asterisk jmls (~jmls@host81-134-142-184.in-addr.btopenworld.com) |
| 08:16.37 | Damin | kram: So you want me to get a current sip-debug and post it as post-mwi-subscribe-disable? |
| 08:19.11 | *** join/#asterisk dolson_ (~dana@Sudbury-HSE-ppp3977810.sympatico.ca) |
| 08:21.47 | brc_ | ~SEEN coppice |
| 08:21.48 | jbot | coppice is currently on #asterisk (3h 35m 29s). Has said a total of 12 messages. Is idling for 1h 32m 57s |
| 08:21.48 | coppice | no |
| 08:21.48 | brc_ | jbot, seen coppice |
| 08:21.49 | jbot | coppice is currently on #asterisk (3h 35m 30s). Has said a total of 13 messages. Is idling for 1s |
| 08:21.49 | brc_ | :p |
| 08:21.49 | brc_ | howdy |
| 08:21.57 | brc_ | well thankyouverymuch jbot |
| 08:22.07 | coppice | tea break |
| 08:22.35 | *** join/#asterisk h3x (~Justino@nv-65-40-157-57.sta.sprint-hsd.net) |
| 08:23.10 | *** join/#asterisk Zeeek (~Zeeek@80.125.80.38) |
| 08:23.37 | brc_ | was that you that posted something about t.38 to the mailing list awhile ago? |
| 08:23.42 | brc_ | h3x, whatup! |
| 08:23.53 | brc_ | h3x, I can't believe it man |
| 08:24.02 | coppice | i'm working on t.38 |
| 08:24.11 | brc_ | ahh okay |
| 08:24.13 | Zeeek | ice t.38 |
| 08:24.58 | coppice | no. T.38 is Pu Erh. You don't drink that iced :-) |
| 08:25.04 | brc_ | heh |
| 08:25.09 | brc_ | so how's it going? :p |
| 08:25.47 | coppice | let me check the file dates....... 11th Dec. Oh, its over a month since I last touched it :-( |
| 08:25.55 | brc_ | aww |
| 08:26.20 | brc_ | how much have you done? |
| 08:27.18 | brc_ | more or less then 1/4 |
| 08:27.32 | brc_ | or too hard to say |
| 08:27.46 | coppice | well, my modems obviously give me one end, and I have the other end that talks T.38 packets going. The bit is the middle is missing |
| 08:28.06 | Damin | kram: I've updated the bug and documented the changes in my polycom config file, and I also uloaded a working-no-mwi sip-debug file for comparison. |
| 08:29.00 | Corydon76-home | We oughta tell Polycom it would be useful to have the source code to diagnose problems like this... ;-) |
| 08:29.14 | afrosheen | we oughta tell polycom to support their users a little bit |
| 08:29.18 | Damin | Corydon-w: And How. ;) |
| 08:29.56 | afrosheen | their phone's built in configuration tool is missing so many things |
| 08:30.13 | Damin | afrosheen: No.. it's missing everything! ;) |
| 08:30.22 | brc_ | sure would be nice if faxes would just go away |
| 08:30.47 | brc_ | I don't suppose you've heard anything about when sipura will release the t.38 firmware for the 2100 eh? |
| 08:31.54 | coppice | one thing I really need at the moment is some kit that supports T.38 for testing. The only thing I have is t38modem from openh323, and that is decidedly iffy. |
| 08:32.17 | brc_ | ok |
| 08:32.43 | brc_ | ~t38 |
| 08:32.58 | brc_ | ~t38 is see http://www.brooktrout.com/whitepapers/pdf/fax_over_ip.pdf for a decent overview of how it all works |
| 08:32.59 | jbot | okay, brc_ |
| 08:34.13 | *** join/#asterisk _TripleF_ (kvirc@Toronto-HSE-ppp3771018.sympatico.ca) |
| 08:34.17 | coppice | I can sum up how it works - nastily :-) |
| 08:34.39 | brc_ | haha |
| 08:35.35 | coppice | T.38 looks OK..... until you start to implement it. |
| 08:36.06 | brc_ | if I understand it correctly...if asterisk is used as a gateway ("normal" fax machine ==pstn=> asterisk ==t.38==> t.38 speaking device/software) it'll still need to decode t.30 correct? |
| 08:37.27 | brc_ | then t.38 wraps t.30, and you stuff it all into sip |
| 08:37.35 | coppice | basically. T.38 appears to not need the gateway to understand much of what it is doing. as you implement ot you find it needs most of a T.30 engine |
| 08:37.49 | coppice | T.38 has nothing to do with SIP |
| 08:38.00 | brc_ | right |
| 08:38.06 | Damin | Alright.. |
| 08:38.18 | Damin | Now that I've solved this little problem, I'm going to bed.. |
| 08:38.19 | Damin | Night all! |
| 08:38.20 | brc_ | you just need an "an IP call control protocol"...of which sip is one |
| 08:38.24 | brc_ | night Damin |
| 08:39.19 | *** join/#asterisk Jameno123 (~james@ddsl-216-68-219-38.fuse.net) |
| 08:39.20 | brc_ | and being that it needs 'most' of a T.30 engine...you have to do all kinds of magic to work with non-compliant fax machines no? |
| 08:39.27 | *** join/#asterisk crazy1 (~user@S01060004e2a79973.gv.shawcable.net) |
| 08:40.00 | *** join/#asterisk oDin (~lolitaaa@n219079193241.netvigator.com) |
| 08:40.06 | coppice | You need a call control protocol to set things up, but T.38 has no interest in how that is done |
| 08:40.20 | brc_ | I see |
| 08:40.45 | coppice | The need for a T.30 engine probably has various implications for non-compliant FAX machines. |
| 08:40.56 | _TripleF_ | ?? |
| 08:41.12 | brc_ | _TripleF_, yes? |
| 08:41.16 | coppice | T.38 doesn't care how the call was set up, just like RTP doesn't care |
| 08:41.28 | Damin | You know.. |
| 08:41.28 | brc_ | ahh |
| 08:41.32 | Damin | Asterisk just freaking rocks.. |
| 08:41.46 | brc_ | except when it sucks |
| 08:41.49 | Damin | And you guys all rock.. |
| 08:41.56 | Zeeek | I don't rock |
| 08:41.59 | brc_ | except when we don't |
| 08:42.07 | Zeeek | but thanks anyway |
| 08:42.12 | brc_ | you're welecome |
| 08:42.21 | *** join/#asterisk meebey_ (meebey@meebey.net) |
| 08:42.31 | Zeeek | what are you guys doing up at this hour anyway? |
| 08:43.04 | afrosheen | plotting our next conspiracy |
| 08:43.15 | afrosheen | sip that traverses NAT easily! madness! |
| 08:43.16 | coppice | should I be having a siesta at this time in the afternoon, then? :-) |
| 08:43.24 | Zeeek | more like discussing asterisk when you should be gently stoking someone or some thing :) |
| 08:43.29 | brc_ | afrosheen, eh? |
| 08:43.41 | Zeeek | coppice no, YOU should be perfecting your thang |
| 08:44.00 | Zeeek | coppice - did you know that somone once defined genius as |
| 08:44.27 | Zeeek | the ability to keep working and prefecting something infinitely ? (or something close to that) |
| 08:44.27 | coppice | knowing how to hide your sources? |
| 08:44.39 | Zeeek | Ford Prefect heh |
| 08:44.46 | coppice | I think that was Einstein |
| 08:44.52 | Zeeek | lateral thinking if you will |
| 08:45.01 | coppice | the hide your sources one, I mean |
| 08:45.05 | Zeeek | "ok it works, now how can I make it work better" |
| 08:45.25 | Zeeek | which I guess is more or less what do |
| 08:45.48 | coppice | that's how windows got where it is today. |
| 08:45.57 | Zeeek | yyyyyyes.... |
| 08:46.05 | Zeeek | quite. |
| 08:46.09 | Zeeek | next subject? |
| 08:46.10 | coppice | "its screwing the public" |
| 08:46.11 | coppice | "how can we screw them more" |
| 08:46.25 | Zeeek | I won't go there because it isn't productive |
| 08:46.46 | Zeeek | I rather hear amusing jokes that OS or distro bashes |
| 08:47.05 | Zeeek | but I digress... |
| 08:47.12 | Zeeek | anyone here know much about IAX2 ? |
| 08:47.17 | brc_ | no |
| 08:47.22 | coppice | no |
| 08:47.26 | Zeeek | I don't that's fer sure |
| 08:47.34 | coppice | nobody knows anything about IAX2 |
| 08:47.47 | Zeeek | well, do you know much about WIndows then ? |
| 08:47.50 | jmls | The word "Windows" is a word out of an old dialect of the Apaches. It means: "White man staring through glass-screen onto an hourglass..." |
| 08:48.05 | Zeeek | sorry that applies more to Mac |
| 08:48.09 | brc_ | what's that? Interactive Astronomical eXchange version 2? |
| 08:48.14 | Zeeek | I'll state my IAX thing anyway for grins |
| 08:48.22 | brc_ | I am getting a mac |
| 08:48.27 | brc_ | I swear |
| 08:48.31 | brc_ | if it's the last thing I do |
| 08:48.40 | Zeeek | every few days and sometimes a couple days in a row... in a galaxy not far... |
| 08:49.01 | Zeeek | [there is a new cigar box Mac for $499 - just add screen kbd & mouse] |
| 08:49.08 | brc_ | no |
| 08:49.10 | brc_ | buzzzz |
| 08:49.11 | brc_ | wrong |
| 08:49.32 | jmls | Q: How do you make a Mac User's eyes sparkle? |
| 08:49.32 | Zeeek | every few days I get ten minutes of frantic IAX2 packets back and forth wich add 50,000 lines to the log |
| 08:49.33 | jmls | A: Shine a flashlight in their ears. |
| 08:49.54 | brc_ | it's |
| 08:49.56 | jmls | sorry guys. just a bit of fun |
| 08:49.59 | brc_ | damnit |
| 08:50.00 | Zeeek | these packets are to and from a single server which I am not at liberty to name |
| 08:50.01 | brc_ | I can't remeber |
| 08:50.10 | brc_ | it's like BYOKVM |
| 08:50.13 | brc_ | but that's not it |
| 08:50.29 | Zeeek | just thought I'd ask |
| 08:51.15 | Zeeek | Breaking News: the newest voIP operator here just has asterisk powered vmail and the public LOVE IT! |
| 08:51.43 | Zeeek | people on the forum are gushing about it (they think the provider invented it) |
| 08:52.00 | Zeeek | but the provider, Wengo, clearly said in the forum it was asterisk |
| 08:52.12 | Zeeek | I think that's good news |
| 08:54.06 | ZX81 | what |
| 08:54.07 | ZX81 | news? |
| 08:54.11 | ZX81 | huh? |
| 08:54.13 | ZX81 | where? |
| 08:54.17 | ZX81 | :-) |
| 08:54.24 | Zeeek | ZX81 yeah you might actually wanna cover that |
| 08:54.39 | ZX81 | where is it? |
| 08:54.55 | Zeeek | want me to message you ? (so as not to flood again)? |
| 08:54.58 | Guest^DJ | hi Zeeek |
| 08:55.01 | ZX81 | yup |
| 08:55.08 | h3x | brc |
| 08:55.10 | Zeeek | Hi DJ |
| 08:55.13 | Guest^DJ | ZX81: you are back |
| 08:55.16 | ZX81 | :-) |
| 08:55.16 | ZX81 | yup |
| 08:55.20 | ZX81 | temporarily |
| 08:55.21 | ZX81 | :-) |
| 08:55.24 | *** join/#asterisk B4 (~B4@202.69.48.245) |
| 08:55.31 | firestrm | another fine day learning asterisk.. thanks all.. g-nite |
| 08:56.13 | Guest^DJ | hey, i am now 7 day old with * |
| 08:56.28 | brc_ | h3x, |
| 08:56.31 | Guest^DJ | or should i say young |
| 08:56.35 | h3x | what up holmes |
| 08:56.38 | crazy1 | anybody know of a way to configure asterisk so that a user can press something like ** to break out of wherever they are and default back to a known starting point? eg: user is in MeetMe or Directory, presses ** and busts back to the beginning of an IVR menu |
| 08:56.53 | brc_ | you hurt me man |
| 08:57.02 | h3x | ? |
| 08:57.07 | brc_ | you didn't stop by |
| 08:57.13 | h3x | ?? |
| 08:57.20 | brc_ | I was there for ces |
| 08:57.29 | h3x | i didnt go to ces |
| 08:57.42 | brc_ | you didn't miss much |
| 08:57.49 | h3x | heh |
| 08:58.09 | coppice | brc_ the booth babes not up to much this time? :-) |
| 08:58.19 | brc_ | heh |
| 08:58.35 | brc_ | most of the stuff was protoware |
| 08:58.51 | brc_ | or release-date-unknownware |
| 08:59.00 | brc_ | or asia-only-ware |
| 08:59.13 | brc_ | all those cool cellphones |
| 08:59.20 | coppice | what's wrong with asia-only-ware? |
| 08:59.33 | crazy1 | i'd like to be able to dial in on my land line, check voice mail, hit **, dial out on my voip line, hit **, chat in a conference room, hit **, make another outbound call, etc. without having to redial in to the asterisk box each time |
| 08:59.34 | h3x | coppice: it comes with a free dose of SARS :) |
| 08:59.40 | brc_ | samsung had some nice plasmas tho |
| 08:59.46 | *** join/#asterisk HjemmeRoyK (~roy@110.80-203-29.nextgentel.com) |
| 08:59.50 | coppice | SARS is so 2003 |
| 09:00.04 | brc_ | mad cow? |
| 09:00.10 | coppice | the 102" one is impressive |
| 09:00.11 | brc_ | yup |
| 09:00.16 | brc_ | coppice, did you go to ces? |
| 09:00.20 | coppice | brc: you mean Mrs Thatcher? |
| 09:00.41 | coppice | brc_: nope |
| 09:01.11 | brc_ | crazy1, I'm pretty sure you can do that...probably not with the "**" key, but there is a way to exit from voicemailmain |
| 09:02.20 | brc_ | hm...not sure if you can get out of meetme though |
| 09:02.30 | brc_ | syn? |
| 09:02.43 | crazy1 | yeah, i can break out of voicemailmain by hitting 0, but i was hoping there was a way to have asterisk listen in on whatever i'm doing and recognize that i want to go back to the IVR menu from any application or call |
| 09:02.54 | brc_ | nope |
| 09:03.01 | brc_ | would be nice if there was |
| 09:03.31 | brc_ | on nice...RoR (Ruby on Rails) got a new site http://www.rubyonrails.org/ |
| 09:03.33 | crazy1 | i've figured out how to get out of meetme |
| 09:04.00 | crazy1 | it seems like each application has a specific way of getting out though |
| 09:04.04 | h3x | im still trying, a year later, to get fiber installed in my place from 3 more carriers... blah. |
| 09:04.05 | brc_ | yup |
| 09:04.35 | jmls | Mrs Thatcher was not mad. Just misunderstood |
| 09:04.57 | coppice | yeah, like Charles Manson |
| 09:04.58 | afrosheen | and ugly |
| 09:05.31 | crazy1 | unfortunately many of them need # to get out and whenever i press # i get a voice saying "transfer" and from there any extension i choose is "invalid" |
| 09:05.51 | jmls | Dennis married her - oh, wait ... |
| 09:06.32 | brc_ | crazy1, that's because you have # transfer enabled |
| 09:06.41 | brc_ | remove the tT from your dial options... |
| 09:06.44 | brc_ | unless you're using that |
| 09:07.07 | crazy1 | i don't have tT in my dial options |
| 09:08.05 | jmls | are there any other boxes similar to the spa3000 (FXO and FXS ports) but not as complicated to set up with * ? The spa2000 is wonderful but I needed both types of port. |
| 09:09.23 | *** join/#asterisk Othello (Othello@nusnet-156-21.dynip.nus.edu.sg) |
| 09:09.47 | brc_ | there are fxo sip devices |
| 09:09.49 | brc_ | google... |
| 09:09.59 | brc_ | somebody makes an 8 port |
| 09:10.02 | brc_ | check atacomm.com |
| 09:10.17 | *** join/#asterisk hardstone (~chatzilla@202.61.44.207) |
| 09:11.44 | jmls | I was wanting one for the parents - I was going to set up a simple * server, but plug in their existing phone - change is bad, espcially for mum :) |
| 09:12.28 | brc_ | so get a 3000... |
| 09:12.43 | brc_ | or ia-92 (x100p) card |
| 09:13.16 | *** join/#asterisk scardinal (~supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 09:27.21 | *** join/#asterisk psywar (psywar@rasterburn.org) |
| 09:27.46 | *** join/#asterisk Mike_TK (~Mike_TK@213.180.245.62) |
| 09:27.55 | psywar | I have dialtone! |
| 09:27.58 | *** join/#asterisk formatioreticula (~formatior@J8629.j.pppool.de) |
| 09:28.14 | psywar | I got a SPA-2000 and it works |
| 09:28.21 | psywar | So, thanks. |
| 09:28.32 | psywar | Word to your respective mothers. |
| 09:28.54 | psywar | Asterisk is the shizzle. |
| 09:29.19 | ZX81 | indeed |
| 09:30.11 | coppice | I wonder if that means its good or its bad. it sounds nasty :-) |
| 09:30.23 | HjemmeRoyK | morning, copp |
| 09:30.25 | psywar | ANyone know why I'd have two ringing sounds, one that never ends |
| 09:30.28 | psywar | coppice: hehe |
| 09:30.52 | coppice | the one that never ends it called tinitus |
| 09:31.03 | psywar | damn |
| 09:31.13 | ZX81 | ahahaha |
| 09:31.23 | ZX81 | coppice: :-) |
| 09:32.33 | *** join/#asterisk Fabe (~Fabe@p50858490.dip0.t-ipconnect.de) |
| 09:32.44 | psywar | asterisk has some mighty config files |
| 09:33.10 | psywar | 120% more powerful than sendmail |
| 09:33.22 | flewid | <PROTECTED> |
| 09:33.22 | flewid | <PROTECTED> |
| 09:33.22 | flewid | <PROTECTED> |
| 09:33.25 | flewid | grr |
| 09:33.38 | psywar | ? |
| 09:33.43 | flewid | misclick :) |
| 09:33.46 | flewid | hence the grr |
| 09:33.49 | Yoda-BZH | Le bonjour je vous souhaite / Hi ppl |
| 09:34.53 | psywar | flewid: ah the joys of pasting under X |
| 09:36.15 | psywar | "Oh, you wanted to paste what you just highlighted? Hmm, maybe I will, maybe I won't!" |
| 09:37.26 | psywar | Seriously though, synthetic ring sound - any hints as to where I disable this? |
| 09:37.48 | psywar | I'll read all the config files this weekend, I guess. |
| 09:38.42 | psywar | also, is the RTP port fixed at 16482, or does it change? |
| 09:40.51 | *** join/#asterisk emk (~emk@62.24.101.29) |
| 09:42.10 | *** join/#asterisk [TripleF] (kvirc@Toronto-HSE-ppp3771509.sympatico.ca) |
| 09:53.44 | emk | need help: I'm trying to setup an asterisk demo for the first time using 4 machines on a LAN: 1 asterisk server and 3 clients using GnoPhone and earphone-headsets. |
| 09:54.40 | psywar | it's not easy to configure |
| 09:54.52 | psywar | have you read the PDF manual? |
| 09:55.04 | *** join/#asterisk gmc (~gmc@sonolo.xs4all.nl) |
| 09:56.36 | psywar | neat |
| 09:57.57 | psywar | emk: I'm talking to you |
| 09:58.46 | psywar | Ok nevermind then. |
| 09:59.23 | *** join/#asterisk kanak (~passionfr@203-217-24-222.dyn.iinet.net.au) |
| 09:59.28 | kanak | hi |
| 10:05.18 | *** join/#asterisk ernst (~emk@62.24.101.29) |
| 10:12.05 | *** join/#asterisk jradford (~jradford@hoss.npl.com) |
| 10:16.21 | *** join/#asterisk qiu (~qiu@andrei.digicom.ro) |
| 10:16.23 | kanak | Hi is the "THE Asterisk" channel? |
| 10:18.56 | *** join/#asterisk meppl (mephisto@p54853BD9.dip.t-dialin.net) |
| 10:19.17 | *** join/#asterisk OnuR (LadyxXx_@212.77.218.44) |
| 10:20.02 | Makenshi | kanak, it's probably the best known Asterisk channel |
| 10:20.28 | kanak | thanks for replying, I was not sure my program was working |
| 10:21.26 | kanak | I need some help for a problem I have with my 1st * installation. Am I able to get some help here? |
| 10:21.40 | Makenshi | kanak, certainly, what problem do you have? |
| 10:22.43 | kanak | thanks, the issue I have is with the transfer of calls. I don't seem to be able to transfer an incoming or outgoing call using the "#" key |
| 10:24.19 | *** join/#asterisk rainmkr (~rainmkr@wsip-68-15-106-188.ok.ok.cox.net) |
| 10:24.29 | Makenshi | kanak, ok, what version of asterisk are you running? |
| 10:24.45 | *** part/#asterisk rainmkr (~rainmkr@wsip-68-15-106-188.ok.ok.cox.net) |
| 10:24.57 | kanak | just a second, I connect to the server. I diodn't do the installation, but I am setting up the dial plans etc... |
| 10:26.15 | *** join/#asterisk Guest38411 (YeLKeN@210.186.104.10) |
| 10:26.39 | kanak | sorry, but how do you get the version from the CLI? |
| 10:27.10 | *** join/#asterisk rowterius (~Drake@201.133.209.110) |
| 10:27.13 | kanak | it has the Dundi function so I guess it can't be that old |
| 10:28.45 | *** join/#asterisk SoS (~sinemmmmm@pD9E49B5C.dip.t-dialin.net) |
| 10:30.52 | *** join/#asterisk jetscreamer (~jetscream@adsl-64-219-216-41.dsl.hstntx.swbell.net) |
| 10:33.04 | *** join/#asterisk Gh0sty (~Ghosty@ip-81-11-138-173.dsl.scarlet.be) |
| 10:34.45 | *** join/#asterisk bjohnson (~bjohnson@ip226-181.tor.istop.com) |
| 10:40.27 | Torq | question: is there a softmodem for asterisk? i want to accept 300baud calls |
| 10:42.35 | datareactor | can use to go2call with asterisk |
| 10:42.51 | Torq | sorry? |
| 10:58.53 | *** join/#asterisk Che (~CenaZe@81.212.12.170) |
| 11:07.21 | *** join/#asterisk af_ (~af@ip-148-227.sn1.eutelia.it) |
| 11:15.52 | Torq | Has anyone used tddmode in asterisk? |
| 11:16.43 | *** join/#asterisk jetscreamer (~jetscream@adsl-64-219-216-41.dsl.hstntx.swbell.net) |
| 11:17.12 | *** join/#asterisk ellvis (~ellvis@83.103.31.163) |
| 11:17.19 | ellvis | hi people |
| 11:17.33 | ellvis | may i have a question about ISDN problem? |
| 11:18.02 | B4 | ISDN problem! heh |
| 11:18.41 | B4 | what seems to be bothering you? |
| 11:18.55 | ellvis | well, i am not abble to use the isdn line in asterisk |
| 11:19.06 | B4 | with which card? |
| 11:19.14 | Makenshi | kanak, `asterisk -V` will give you the version |
| 11:19.23 | kanak | ok i try tkx |
| 11:20.41 | kanak | It gave me this: Asterisk CVS-HEAD-12/17/04-16:28:43 |
| 11:21.01 | *** join/#asterisk Ping (~E-sHeee@62.162.14.87) |
| 11:23.01 | ellvis | B4: AVM C4 |
| 11:23.27 | B4 | ellvis: whats the exact issue? |
| 11:23.31 | ellvis | B4: my problem is, that i am getting "DISCONNECT_IND PLCI=0x102 REASON=0x3302" |
| 11:23.34 | *** part/#asterisk berrayahkamel (~berrayahk@82.101.128.6) |
| 11:23.50 | ellvis | B4: as far as i know, it's because sw and hw configuration is different |
| 11:24.06 | ellvis | B4: all was working for few months untill yesterday |
| 11:24.16 | B4 | you can not get incoming or outgoing calls to work? |
| 11:24.21 | Makenshi | kanak, one thing to check is that your Dial statements have the 't' option |
| 11:24.32 | B4 | upgraded to new cvs? |
| 11:24.38 | Makenshi | CVS HEAD is not the latest stable release, so its possible that not all features are working properly |
| 11:24.44 | kanak | Makenshi, they have the t and T |
| 11:24.57 | kanak | ok, what is the stable release? |
| 11:25.05 | ellvis | B4: if i know, that isdn itself is configured to point-to-point, how can i change the sw configuration? |
| 11:25.31 | B4 | hmm no clue ellvis |
| 11:25.37 | *** join/#asterisk ManxPower (~eric@dsl-208-164-150-160.datasync.com) |
| 11:26.04 | ellvis | B4: cos the support of my provider work only for the home users during the weekend... |
| 11:26.27 | ellvis | B4: mno, okaj, i'll try to find it, thanks for listening me anyway |
| 11:27.07 | Faithful | ellvis: what are you trying to do??? VoIP over TCP/IP via ISDN or use ISDN card for Telephony? |
| 11:27.13 | B4 | have you looked at error code desc? |
| 11:27.23 | ellvis | Faithful: i am sending faxes over isdn |
| 11:27.40 | B4 | ellvis: http://home.t-online.de/home/Peter.Zwosta/msk0306.htm |
| 11:27.48 | Faithful | isdn as normal phone line? |
| 11:27.54 | ellvis | Faithful: yes |
| 11:28.27 | Faithful | point-to-point has nothing to do with it then |
| 11:28.44 | ellvis | ups, sorry, point-to-multipoint |
| 11:28.55 | Faithful | ellvis: You are using a fritz card? |
| 11:29.01 | ellvis | because there were some people playing with new pbx box and changed it |
| 11:29.07 | ellvis | just a second |
| 11:29.09 | Faithful | ellvis: it has nothing to do with it |
| 11:29.14 | ellvis | be right back |
| 11:29.26 | ManxPower | Useful Asterisk Docs (BOOKMARK THEM!): http://www.digium.com/index.php?menu=documentation (look at the "Unofficial Links") and http://www.voip-info.org/wiki-Asterisk and http://www.fnords.org/~eric/asterisk/ (my site) and http://asteriskdocs.org/ |
| 11:30.18 | *** join/#asterisk Zeeek (~Zeeek@80.125.80.38) |
| 11:30.28 | Faithful | ellvis: so I gather you have BRI isdn |
| 11:31.11 | ellvis | ManxPower: i am consulting everything with the documentation before i ask here, just i am in a bit time-stress now :( |
| 11:31.51 | Faithful | ellvis: it is a BRI ? |
| 11:32.39 | ellvis | Faithful: yes |
| 11:32.42 | ellvis | BRI line |
| 11:33.37 | B4 | ellvis: maybe someone changed some layer2 parameters on your pbx? |
| 11:33.43 | Faithful | can you make a normal call out at the moment so you know the line is synced ... you shouldn't mess with p-t-p or p-t-mp that is determined by your carrier |
| 11:33.54 | ManxPower | ellvis: Not a lot of people use BRI with Asterisk. There are some, but none in the USA as far as I know. |
| 11:34.13 | B4 | < uses bri with AVM fritz |
| 11:34.16 | Faithful | ManxPower: ellvis I use BRI with asterist |
| 11:34.41 | Fabe | you find a lot of bri * docs at ip-phone-forum.de in german |
| 11:34.44 | ManxPower | Faithful: Using I4L, mISDN, CAPI, ZapHFC, or ZapBRI? |
| 11:34.58 | Faithful | I4L at the moment |
| 11:35.08 | B4 | < CAPI |
| 11:35.36 | Faithful | i have a hisax card |
| 11:35.45 | HjemmeRoyK | B4: s/</\// |
| 11:35.48 | B4 | I tried to use mISDN but I can not get it to work on fedora c3 :( |
| 11:36.01 | ManxPower | BTW, the order of preference is (best to worst) ZapBRI, ZapHFC, CAPI, I4L, mISDN |
| 11:36.04 | Faithful | mISDN looks permature to me |
| 11:36.11 | ManxPower | With I4L and mISDN being pretty much dead last. |
| 11:36.24 | ellvis | B4: yes, that's very possible |
| 11:36.34 | B4 | what I wanted to do was to use mISDN with chancapi not chanmisdn |
| 11:36.34 | Faithful | I4L has been ok but you can't fax |
| 11:36.38 | ellvis | ManxPower: i am in italy |
| 11:36.50 | ManxPower | bristuffed seems to be the most common and best option for cards that it supports. |
| 11:37.12 | B4 | ellvis: your error code says Layer 2 protocol error |
| 11:37.32 | ellvis | B4: i know |
| 11:37.46 | ellvis | B4: i am just a bit of lamer, that's my biggest problem |
| 11:38.02 | B4 | lol k |
| 11:38.27 | ellvis | B4: i am new to asterisk and things like that, just 2 months of experiences... i am keeping alive already installed box and don't have much time to install my new one for learning purposes |
| 11:38.49 | B4 | cool |
| 11:39.01 | B4 | ellvis: have you tried mISDN? |
| 11:39.22 | ellvis | B4: no, i have no gcc on the box |
| 11:39.36 | B4 | oh |
| 11:39.45 | ManxPower | ellvis: You should seriously look at CAPI, ZapHFC (or maybe ZapBRI). |
| 11:39.48 | B4 | install gcc :) heh |
| 11:40.10 | ellvis | B4: it was done by a company and we denied support from them (don't ask me why, i am just a technological slave in this company) |
| 11:40.15 | ManxPower | I don't use ISDN BRI, but I've been here long enough to know that people using those three have the least problems. |
| 11:40.27 | ManxPower | Still, if you have a layer 2 error there's nothing we can do for you. |
| 11:40.30 | *** join/#asterisk FaithX (~Faithful@202-6-145-116.ip.adam.com.au) |
| 11:40.38 | B4 | Manx, which is the least cost card I can use with ZapHFC or ZapBRI? |
| 11:40.57 | ellvis | ManxPower: okay. i just need to be kicked sometimes. |
| 11:41.03 | ManxPower | B4: I don't know, but look at kapejod's site. |
| 11:41.07 | ellvis | all: thanks for your trying! |
| 11:41.12 | B4 | hehehe technological slave is a new one :) |
| 11:41.16 | B4 | which company? |
| 11:41.27 | ellvis | freefile.it |
| 11:41.41 | ManxPower | B4: HFC cards seem to be well supported and I think Fritz! is an HFC card. |
| 11:41.47 | Fabe | B4: just get any cheap hfc-s card from ebay.. a list is in the voip-info wiki. |
| 11:41.58 | B4 | nope fritz is not HFC :( |
| 11:42.00 | Fabe | fritz is no hfc-s i think... |
| 11:42.02 | ellvis | i am working in slovak office, just i am here for a week to keep asterisk alive, i am leaving tomorow |
| 11:42.24 | ManxPower | BRI people go here! http://83.137.99.170/jn/relaunch/asterisk/ |
| 11:42.33 | *** join/#asterisk Kanak2 (~passionfr@203-217-24-222.dyn.iinet.net.au) |
| 11:43.08 | B4 | slovak ... neat |
| 11:43.11 | ManxPower | Also... |
| 11:43.15 | ManxPower | To search the Asterisk mailing list archive go to www.google.com and put site:lists.digium.com in addition to your other query terms. |
| 11:43.32 | ManxPower | Lots of various discussion of BRI on the mailing list archives. |
| 11:43.55 | ManxPower | ALSO check the Wiki |
| 11:43.57 | ManxPower | Useful Asterisk Docs (BOOKMARK THEM!): http://www.digium.com/index.php?menu=documentation (look at the "Unofficial Links") and http://www.voip-info.org/wiki-Asterisk and http://www.fnords.org/~eric/asterisk/ (my site) and http://asteriskdocs.org/ |
| 11:45.35 | ellvis | ManxPower: thx, bookmarking |
| 11:46.31 | ManxPower | ellvis: BTW, EVERYONE that comes here that has BRI has issues setting it up. |
| 11:47.07 | B4 | hehehe |
| 11:47.26 | B4 | well my AVM fritz and CAPI work pretty good |
| 11:47.34 | ellvis | well, okaj, i'll work on remowing BRI and send it to some black hole :) |
| 11:47.35 | B4 | I4L is useless |
| 11:47.51 | ManxPower | B4: *nod* Before the ZapHFC and ZapBRI existed, CAPI was the best for Asterisk. |
| 11:48.05 | ManxPower | I4L has latency and DTMF problems. |
| 11:48.12 | B4 | yep echo |
| 11:48.19 | ManxPower | The DTMF problem MAY have been fixed in more recent Linux kernels |
| 11:48.50 | coppice | you mean someone finally got around to giving them a decent DTMF decoder? |
| 11:48.59 | B4 | I want to be able to use the w6692 I have laying around |
| 11:49.33 | ManxPower | coppice: It was purely speculation |
| 11:49.53 | B4 | DTMF problems are still ther |
| 11:50.04 | ManxPower | All hail coppice the spandsp god! |
| 11:50.48 | B4 | < bows to coppice |
| 11:51.21 | coppice | the DTMF decode we use is mostly floating point. it needs turning into fixed point to put it in the kernel. that was suggested to the developers of i4l who were whinning about how hard DSP is, but I don't know if they listened |
| 11:53.24 | lters | how do the Sangoma t1 cards compare to the TDM410's |
| 11:53.27 | ManxPower | coppice: Did you hear that I'm closing down the fnords.org/~eric/asterisk site soon. I donated the entire thing to the asterisk docs project. |
| 11:55.01 | coppice | lters: if you are using E1s, then very badly :-) |
| 11:55.21 | lters | coppice, what about just t1's |
| 11:55.30 | coppice | well, * could use some coherent docs |
| 11:56.04 | *** join/#asterisk pkwong (~stimmy@ool-44c087de.dyn.optonline.net) |
| 11:56.05 | coppice | lters: the sangoma drivers are less mature. the hardware functionality should be similar |
| 11:56.07 | pkwong | hi all. |
| 11:56.27 | pkwong | how big a box do i need to support 8 t1s? |
| 11:56.38 | ManxPower | ellvis: http://www.voip-info.org/wiki-Asterisk+CAPI+Channels |
| 11:56.50 | ManxPower | ellvis: http://www.voip-info.org/wiki-zaptelBRI |
| 11:57.05 | ManxPower | ellvis: http://www.voip-info.org/wiki-Asterisk+zaphfc |
| 11:57.28 | ManxPower | pkwong: Dual Xenon SMP as fast as there is on the market. |
| 11:57.32 | ManxPower | And don't use G729 |
| 11:57.38 | ManxPower | or iLBC or Speex |
| 11:58.27 | pkwong | well.. i'm using the boxes to route calls only.. g729a |
| 11:58.45 | pkwong | gonna do other stuff (ivr, vm, etc) on other asterisk boxes. |
| 11:58.57 | pkwong | do i still need such a monster box? |
| 11:59.37 | ManxPower | ellvis: http://www.voip-info.org/tiki-index.php?page=Asterisk%20AVM%20Fritz%20CAPI%20Driver%20Install |
| 11:59.45 | pkwong | so the t1 box only routes traffic to the pstn. |
| 11:59.53 | pkwong | err.. to/from |
| 12:00.05 | ManxPower | pkwong: Is it going to use ANY codec other than ulaw or alaw? |
| 12:00.33 | pkwong | well.. can i do t.38 on g729? |
| 12:01.05 | ManxPower | pkwong: 8 T-1s is 12Mbps IN REALTIME. |
| 12:01.12 | ManxPower | pkwong: Asterisk does not support T.38 |
| 12:01.17 | pkwong | i know.. |
| 12:01.26 | ManxPower | pkwong: G729 transcoding licenses are $10/channel |
| 12:01.33 | pkwong | yeah.. i know. |
| 12:01.42 | ManxPower | pkwong: G729 transcoding sucks up LOTS of CPU. |
| 12:01.58 | ManxPower | Don't expect to do a lot of G729/ulaw calls at a time. |
| 12:02.00 | pkwong | what about doing the transcoding off that box? |
| 12:02.07 | pkwong | just sending the traffic there. |
| 12:02.25 | ManxPower | pkwong: If you are transcoding outside of asterisk then you have MUCH less CPU power being used. |
| 12:03.01 | pkwong | i was thinking of doing the transcoding on another asterisk box and sending the traffic to the pstn asterisk box via iax. |
| 12:03.58 | ManxPower | pkwong: That would help. |
| 12:04.12 | ManxPower | Your best bet is two boxes, each with 4 T-1s. |
| 12:04.19 | pkwong | hmmm.. ok. |
| 12:04.29 | pkwong | single xeon for each? |
| 12:04.43 | ManxPower | Then add more transcoding only boxes as needed. |
| 12:04.56 | ellvis | ManxPower: how long are you working with asterisk and things around? |
| 12:04.57 | ManxPower | pkwong: Remember Digium cards don't have any buffering or DSP on them. |
| 12:05.02 | pkwong | or can i get away with p4 regular boxes.. |
| 12:05.09 | ManxPower | ellvis: I've been using asterisk for about 1.5 years. |
| 12:05.14 | pkwong | yeah.. i know.. if i want the dsp, i would go dialogic. |
| 12:05.21 | pkwong | right? |
| 12:05.31 | ManxPower | pkwong: There is no way to be SURE. There are really too many variables involved. |
| 12:05.44 | pkwong | yeah.. i figured. |
| 12:07.45 | pkwong | with asterisk |
| 12:08.09 | ManxPower | pkwong: Dialogic is not well supported in Asterisk. See the digium hardware page for a list of "supported" cards. The Asterisk drivers for Dialogic are #10/channel |
| 12:08.33 | pkwong | yeah.. i was there.. didn't see much. |
| 12:08.40 | pkwong | nothing big enough for my needs. |
| 12:09.01 | ManxPower | pkwong: Look into MAX TNT |
| 12:09.32 | ManxPower | They are "cheap" on ebay, MAY support SIP and MAY let you transcode lots of channels. |
| 12:09.38 | ManxPower | Then just use Asterisk as a PBX |
| 12:10.28 | pkwong | ahhh.. good idea. |
| 12:10.29 | pkwong | :) |
| 12:10.31 | pkwong | thanks :) |
| 12:10.38 | lters | ManxPower, how does the MAX TNT work? |
| 12:10.50 | lters | like channel bank? |
| 12:11.07 | pkwong | right.. and i sip connect to it right? |
| 12:11.43 | *** join/#asterisk h4mm3r` (~h4mm3r@81-208-60-202.fastres.net) |
| 12:11.55 | lters | I've been trying to figure out how to get ss7 traffic too * |
| 12:12.43 | lters | cisco has as5350/pgw which will do ss7 to sip/rtp but it is big $ |
| 12:13.06 | lters | or it can do ss7 <--> pri |
| 12:15.01 | *** join/#asterisk scubasteve (~steve@rdu88-251-252.nc.rr.com) |
| 12:15.26 | pkwong | manx: with the max tnt, where does the transcoding happen? |
| 12:15.35 | pkwong | on the asterisk box or the tnt? |
| 12:17.02 | ManxPower | lters: The MAX TNT started out as an ISP core router/access server. They added a lot of VoIP stuff before Lucent bought the company. |
| 12:17.07 | coppice | lters: how much money do you have? |
| 12:17.17 | ManxPower | pkwong: It would be pretty useless if the TNT didn't do transcoding. |
| 12:17.37 | pkwong | heh.. ok.. so i can really save the pennies going this route then.. |
| 12:18.06 | ManxPower | lters: There is no free way to interconnect SS7 with Asterisk. PRI service has enough "SS7 features" for almost all people. |
| 12:18.08 | pkwong | i bought a few tnts at a former job when tnts were an ascend product, but it's been such a long time.. |
| 12:18.15 | coppice | what happened to the TNT, and why are the second hand ones so very cheap? |
| 12:18.20 | pkwong | i don't know much about them now.. |
| 12:18.31 | ManxPower | pkwong: Ascent was bought by Lucent. |
| 12:18.35 | ManxPower | ..er..Ascend. |
| 12:18.39 | pkwong | yeah.. i remember that. |
| 12:18.45 | ManxPower | Livingston was also bought by Lucent |
| 12:18.55 | coppice | ManxPower: you might be surprised how wrong you are about PRI suiting people |
| 12:18.56 | pkwong | god.. remember the portmasters? |
| 12:19.14 | coppice | so what happened to the TNT? |
| 12:19.16 | ManxPower | coppice: Yes, I know. But I refuse to admit it. |
| 12:19.31 | *** join/#asterisk ozJames79 (~james@CPE20320889-1842-1.gex.ncable.net.au) |
| 12:19.36 | *** join/#asterisk Guest38411 (WARDA@210.186.104.7) |
| 12:19.41 | coppice | A large number of people must use SS7. No choice |
| 12:19.45 | ManxPower | coppice: I THINK Lucent killed off a lot of the Ascend and Livingston products |
| 12:20.15 | ManxPower | coppice: It's weird. The local telco here won't even TALK to you aboue SS7 unless you are a carrier. |
| 12:20.42 | coppice | why are they so cheap? They are huge complex boxes, which now seem to have real value to people with the growth of VoIP |
| 12:20.58 | coppice | ManxPower: "carrier" is the magic word |
| 12:21.47 | ManxPower | pkwong: I think like a lot of used Cisco gear, the TNT is no longer made and so there's a LOT of people buying and selling the used equipment. |
| 12:21.47 | *** part/#asterisk Kanak (~passionfr@203-217-24-222.dyn.iinet.net.au) |
| 12:21.47 | pkwong | hmmm.. ok. |
| 12:21.59 | coppice | ManxPower: in a lot of places, if you are considered a carrier of some form you can only use SS7 |
| 12:22.05 | pkwong | the tnt will do caller id and caller name id i assume? |
| 12:22.14 | ManxPower | coppice: I suspect the TNT was too "carrier class" for Lucent. i.e. they wanted their own original carrier class stuff. |
| 12:23.19 | ManxPower | I'm a BIG fan of used Cisco hear. Lots of the stuff out there so replacement parts are not a problem, upgrades are not a problem. We are looking at the Cisco 5000 series of switches. |
| 12:23.34 | ManxPower | LOTS of ports, not lots of money. |
| 12:23.34 | lters | what for? |
| 12:23.51 | lters | I know the 4000 and the 6000 |
| 12:23.59 | ManxPower | lters: We need 86 100Mpbs switch ports, but don't need fiber or GigE |
| 12:24.22 | lters | do you have discount status with cisco? |
| 12:24.24 | ManxPower | lters: The 5000 series was replaces by the 4000 and 6000 series, but those are like 4x the cost of the 5000 series |
| 12:25.13 | lters | our company has gotten a large discount from cisco cause we are using their oc192 stuff for a large ring |
| 12:25.15 | meebey | strange, the CAPI here does not work when asterisk is run as non-root the user I tell asterisk to change to via -U is in a group which has acccess to /dev/capi20 |
| 12:25.17 | ManxPower | no, we do not get heavy discounts with Cisco. We don't buy enough Cisco stuff. We buy like three 2621s a year. |
| 12:25.26 | lters | ah |
| 12:25.41 | lters | the 2651 is a power ss7 piece. |
| 12:25.55 | ManxPower | We have 1 PRI |
| 12:26.01 | meebey | 13636 open("/dev/capi20", O_RDWR) = -1 EACCES (Permission denied) |
| 12:26.05 | meebey | thats what strace tells me |
| 12:26.06 | lters | ManxPower, I might be able to get you some discount... |
| 12:26.34 | *** join/#asterisk eeek (~Zeeek@80.125.80.38) |
| 12:27.12 | pkwong | hmm.. as5300.. i'll need the voip card with it in addition to the modems right? |
| 12:27.37 | lters | pkwong, what are you going to/from? |
| 12:27.44 | pkwong | or just the octal card? |
| 12:29.35 | ManxPower | lters: Catalyst 5-slot chassis $125 http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&category=51206&item=5743278427&rd=1 |
| 12:31.36 | lters | used? |
| 12:31.44 | ManxPower | of course, used. |
| 12:32.02 | ManxPower | http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&category=51260&item=5743107961&rd=1 13-slot Catalyst $1,300 |
| 12:32.23 | Torq | Has anyone here used the TDD support in asterisk? |
| 12:32.42 | ManxPower | Here is the seller's store, lots of used Cisco Gear. |
| 12:32.45 | ManxPower | http://stores.ebay.com/BEYOND-AUTOMATION_W0QQsspagenameZl2QQtZkm |
| 12:32.56 | ManxPower | Torq: Asterisk supports TDD? |
| 12:34.07 | Torq | yup :) |
| 12:34.15 | Torq | I was surprised myself. see tdd.c in the source |
| 12:34.28 | Torq | i cant find any docs on how to use it anywhere though |
| 12:34.40 | *** join/#asterisk dtwilson (~dave@cpc1-blfs2-5-0-cust219.blfs.cable.ntl.com) |
| 12:35.50 | dtwilson | hi - I'm getting constant errors and need some help - have googled to no avail |
| 12:35.55 | Torq | I would really like this TDD to work, if someone can figure out how it works |
| 12:36.12 | dtwilson | the error is: |
| 12:36.13 | dtwilson | WARNING[114695]: chan_sip.c:7690 in sip_send_mwi_to_peer: Unable to build sip pvt data for MWI |
| 12:37.02 | dtwilson | any ideas? |
| 12:38.28 | ManxPower | dtwilson: What does "show version" say? PASTE it. |
| 12:40.21 | Zeeek | if the error messages were in one file, someone could go through them and make them more clear |
| 12:40.30 | ManxPower | Torq: Ask on the mailing list and check the mailing list archives. It's an uncommon enough question you should not expect anyone here to know about it. |
| 12:40.37 | Torq | ok |
| 12:41.51 | Zeeek | my favorite: "ERRORr[323432]: This shouldn't ever happen!" |
| 12:44.23 | *** join/#asterisk ckruetze (~ckruetze@i3ED63B25.versanet.de) |
| 12:44.31 | coppice | TDD support was part of the very original zapata code |
| 12:44.33 | ckruetze | Hi |
| 12:44.36 | pkwong | manx: have you used the tnts? |
| 12:45.02 | ManxPower | pkwong: I used the Max Classic in my ISP days. |
| 12:45.16 | ManxPower | pkwong: But the TNTs have been discussed on the mailing list. |
| 12:45.22 | pkwong | hmm.. ok. :) |
| 12:45.24 | *** join/#asterisk [TripleF] (kvirc@Toronto-HSE-ppp3884548.sympatico.ca) |
| 12:45.36 | Torq | coppice: you have any idea how it works? |
| 12:46.28 | coppice | Torq: not these days |
| 12:47.05 | ManxPower | You DO realize that %80 of what I know I learned from the mailing list? |
| 12:47.22 | ManxPower | The other %20 (the most useful) I learned by DOING. |
| 12:47.56 | coppice | ManxPower: English? arithmetic? presumably you never learned any logic or common sense :-) |
| 12:49.49 | *** join/#asterisk zotz (~zotz@24.244.133.136) |
| 12:50.02 | Zeeek | ManxPower so where the f did the three or four people who know everything on the mailing list (but usually answer with "it's in the list archives") learn everything? |
| 12:51.57 | dtwilson | ManxPower: Asterisk CVS-HEAD-07/16/04-14:42:45 built by root@ldev2.lan.autosols.com on a i686 running Linux |
| 12:52.52 | Zeeek | dtwilson that's back a ways! |
| 12:52.55 | dtwilson | I know it's old, but it's normally stable - we have a Py-Asterisk module running on it |
| 12:53.21 | Zeeek | I used to run STABLE from around then, it was um, pretty STABLE |
| 12:53.24 | dtwilson | and the guy who set it all up has left us in the lurch - old sysadmin guy |
| 12:53.40 | Zeeek | they always look a lot older than they are you know |
| 12:53.52 | dtwilson | lol |
| 12:53.53 | Zeeek | he's prolly out banging some 20 year old ring now |
| 12:54.11 | Zeeek | what is your problem with MWI ? |
| 12:54.11 | dtwilson | actually he was 21 - banging an 18 yr old |
| 12:54.27 | Zeeek | oh so you meant "former" when you said old |
| 12:54.33 | dtwilson | Zeeek scroll up |
| 12:54.34 | dtwilson | yup former |
| 12:54.39 | *** join/#asterisk Fabe (~Fabe@p50858D7B.dip0.t-ipconnect.de) |
| 12:54.52 | Zeeek | as a friend of mine said years ago when he was 47, "47 don't go into 20 too many times" |
| 12:55.03 | Zeeek | I saw the message but not the symptoms |
| 12:55.42 | Zeeek | I see the error msg |
| 12:55.55 | dtwilson | ok well - here's the deal - our dsl provider had a major multi-source network failure yesterday |
| 12:56.20 | Zeeek | which if translated to English looks like it means "I can't read the voicemail data" |
| 12:56.29 | dtwilson | and after several hours finally got our dsl back up - then discovered our phones were broke |
| 12:56.39 | Zeeek | all phones ? |
| 12:56.45 | dtwilson | and the mwi error message is all I know |
| 12:56.57 | Zeeek | and mwi not working anywhere |
| 12:57.26 | dtwilson | we're doing two PSTN lines via X101p's into the office - a dozen or so sip extensions |
| 12:57.47 | dtwilson | ummm, I don't even know what mwi is at this point |
| 12:58.00 | Zeeek | Mail waiting indicator |
| 12:58.06 | Zeeek | and here is your first clue: |
| 12:58.07 | Zeeek | http://lists.digium.com/pipermail/asterisk-users/2004-October/066720.html |
| 12:58.22 | coppice | message waiting to be fixed indicator :-) |
| 12:58.32 | dtwilson | yeah I saw that post |
| 12:58.42 | dtwilson | but it's not related |
| 12:58.44 | dtwilson | 2 secs |
| 12:58.45 | Zeeek | ok |
| 12:59.17 | dtwilson | 2005-01-15 13:05:30 WARNING[114695]: rtp.c:855 ast_rtp_new_with_bindaddr: Unexpected bind error: Cannot assign requested address |
| 12:59.32 | Zeeek | as I said one BIG step for asterisk would be to have all error msgs in one file and get someone to go through them and make the more obscure ones understandable |
| 12:59.55 | Zeeek | dtwilson did your linux box crash or vomit when the disconnect took place? |
| 13:00.00 | dtwilson | does that just mean asterisk can't bind the the configured ip address on the machine? |
| 13:00.37 | dtwilson | Zeeek: I accidentally rebooted it, yes - too many ssh windows open :P |
| 13:00.52 | Zeeek | so isn't there a chance that this is file system related ? |
| 13:00.58 | dtwilson | was intending to reboot our router |
| 13:01.04 | Zeeek | because it's very unlikely that you will be able |
| 13:01.10 | dtwilson | nah - it was cleanly shutdown |
| 13:01.14 | Zeeek | to drag the old sysop off the 18 year old |
| 13:01.32 | dtwilson | Zeeek: that's precisely why he became "old" :) |
| 13:01.34 | Zeeek | ok, no fs damage |
| 13:02.05 | Zeeek | what about looking at some file mod dates? |
| 13:02.29 | dtwilson | which files? |
| 13:02.54 | Zeeek | I don't have the source in front of me (not that I'd know much) but you could see what pvt sip data means to whomever wrote that catchy phrase |
| 13:03.00 | Zeeek | I meant any asterisk files |
| 13:03.16 | Zeeek | maybe there's a typical comment like: |
| 13:03.34 | Zeeek | "// can't get pvt SIP data" |
| 13:03.47 | Zeeek | printf("Unable to get...") |
| 13:04.25 | Zeeek | or /**** tell them they're fucked ***/ |
| 13:06.18 | Zeeek | ok, I ssh'ed in and I'lm looking just for fun |
| 13:07.03 | Zeeek | the message is sent after being unable to allocate memory |
| 13:07.28 | Zeeek | <PROTECTED> |
| 13:08.23 | Zeeek | dtwilson ya following here? |
| 13:08.50 | dtwilson | ok - I'm not very strong with linux - I follow ya yup - just no idea how I'd go about sorting it |
| 13:09.21 | Zeeek | The point is to get close and then hope some kind soul will jump in :) |
| 13:10.08 | Zeeek | so it would appear that when asterisk is in channel_sip looking for messages for each subscriber it makes a request for the data |
| 13:10.13 | *** join/#asterisk Nuttah (~andrew@amber2.interdart.co.uk) |
| 13:10.24 | Nuttah | mornin :) |
| 13:10.35 | dtwilson | afternoon (UK) nuttah |
| 13:10.38 | Zeeek | that request dies either because the malloc fails - which on a system running all this time is pretty rare |
| 13:10.51 | Nuttah | hey i was being polite to the US ppl :) |
| 13:10.51 | Zeeek | Nuttah you know anything (at all) ? |
| 13:11.01 | Zeeek | (about anything) |
| 13:11.11 | dtwilson | he kows it's morning in US :) |
| 13:11.40 | Zeeek | but especially sip_pvt |
| 13:11.42 | Nuttah | something busting ya balls this day zeek? |
| 13:11.48 | Nuttah | or ya usually this anal? :) |
| 13:11.51 | dtwilson | Zeeek: top shows very minimal mem usage etc |
| 13:12.02 | Zeeek | not me out friend here whose ex-sysop is balling some young chick |
| 13:12.22 | Zeeek | damn him |
| 13:12.23 | *** join/#asterisk Josh1 (Joshwah@adsl-8-161-173.mia.bellsouth.net) |
| 13:13.03 | Nuttah | anyone here has any experience with the 1.0.5.16 firmware for the budgettones |
| 13:13.07 | Zeeek | what could losing connectivity do to a running system? What does the log look like? Did you look? |
| 13:13.10 | Nuttah | for the 101/102 only |
| 13:13.16 | Zeeek | Nuttah yes plenty |
| 13:13.24 | Zeeek | I can't get it working |
| 13:13.28 | Nuttah | me neither :P |
| 13:13.34 | Torq | WARNING[25442]: chan_sip.c:8711 reload_config: Failed to bind to 0.0.0.0:5060: Address already in use |
| 13:13.36 | Nuttah | well a lie |
| 13:13.37 | Zeeek | try 18,20,21 |
| 13:13.40 | Nuttah | i can get everything working |
| 13:13.41 | dtwilson | Zeeek: our box isnt dependent on the dsl connectivity |
| 13:13.43 | Nuttah | but calling out |
| 13:13.50 | Torq | i have nothing running on 5060. and netstat -a shows nothing listening?!? |
| 13:13.56 | ManxPower | Torq: That means something is already using that port. Are you running a softphone on the system |
| 13:14.05 | Torq | Manx: no... |
| 13:14.11 | Zeeek | dtwilson (Sherlock?) but what is the connection between broken asterisk and that incident? |
| 13:14.20 | ManxPower | Torq: does ps show an Asterisk already running? |
| 13:14.27 | Torq | no... |
| 13:14.29 | Nuttah | zeeek, I thought the 1.0.5.16 is the latest?... |
| 13:14.39 | Zeeek | Nuttah I need to talk to you - I have the exact same prob and they tell me"you are the only one having this problem" |
| 13:14.44 | dtwilson | Zeeek: only connection is that I sent shutdown -r now to the wrong ssh window :S |
| 13:14.44 | ManxPower | Torq: Then undortunatly you are living in a different universe. |
| 13:14.57 | Torq | something is listening on 5060, i cant find it |
| 13:15.01 | Nuttah | certainly m8 |
| 13:15.03 | Zeeek | Nuttah does the phone do DNS? Does the time show correctly? |
| 13:15.08 | Nuttah | oh yes |
| 13:15.12 | Nuttah | very pleased with that |
| 13:15.12 | Torq | udp 0 0 mythtv:5060 *:* |
| 13:15.23 | Nuttah | an then i checked the outbound |
| 13:15.24 | ManxPower | Torq: There ya go! |
| 13:15.29 | Nuttah | no tone.. |
| 13:15.32 | Zeeek | Nuttah my problem is simple - after 5.11 the phone does no DNS lookup at all |
| 13:15.42 | Nuttah | mines the revese |
| 13:15.42 | Torq | manx: yes but what has the port.... i cant find anything obvious... |
| 13:15.46 | Zeeek | I got tone but it can't see the server ip |
| 13:15.51 | Nuttah | before .11 i never got ntp to work properlly |
| 13:15.54 | Zeeek | if I put in the ip it wokrs |
| 13:16.08 | ManxPower | what is mythtv? |
| 13:16.08 | Zeeek | we should trade phones :) |
| 13:16.16 | Nuttah | lol does sound like it |
| 13:16.17 | Torq | mythtv is my box name |
| 13:16.20 | Zeeek | like MTV but for Greek Gods ? |
| 13:16.23 | ManxPower | Torq: Ah. |
| 13:16.24 | Nuttah | is the .16 the latest? |
| 13:16.35 | Zeeek | no .20 or .21 and 18 is recommended |
| 13:16.46 | Torq | this is fedora core 2, anything in there by default that would grab 5060? |
| 13:16.49 | Nuttah | hmm they dont list those from the website |
| 13:16.52 | Zeeek | http://fm.grandstream.com/gs/ |
| 13:16.57 | Zeeek | how's this: |
| 13:16.59 | Nuttah | appreciated |
| 13:17.01 | dtwilson | Torq: I doubt it |
| 13:17.27 | Zeeek | 21 is for 486 only sorry |
| 13:17.29 | ManxPower | Torq: You realize that running graphics on the system will usually screw up Asterisk, right? |
| 13:17.44 | Nuttah | i'll try .20 and let you know what happens |
| 13:17.48 | Torq | oh. no i didnt know. i wanted to run it on the same box as my PVR. |
| 13:17.50 | Zeeek | no 18 |
| 13:17.53 | *** join/#asterisk sadie (~sadie@193.17.41.120) |
| 13:17.55 | Zeeek | 19 is for ATA as well |
| 13:17.58 | Torq | why would it screw up asterisk? |
| 13:17.59 | Nuttah | 18 more reliable? |
| 13:18.08 | ManxPower | Torq: Well don't expect to be able to use and Digium cards with it. |
| 13:18.10 | Zeeek | 18 is what GS told me to load - but I'm back on .11 |
| 13:18.16 | Nuttah | k |
| 13:18.18 | dtwilson | Torq: cos it tends to screw up everything eventually :) |
| 13:18.21 | Zeeek | ANd I think they gave up on solving my prob |
| 13:18.23 | Torq | i dont intend to. Just using it for sip routing |
| 13:18.29 | ManxPower | Torq: Graphics generally lock interrupts for a very long time. |
| 13:18.53 | Torq | this is pure VOIP. no posts |
| 13:18.56 | Torq | no pots |
| 13:19.21 | Zeeek | Nuttah - 20 also has a version for phone: b21p1....20 |
| 13:19.24 | ManxPower | Torq: Try it and see, but don't be suprized if you have crappy audio. |
| 13:19.32 | Torq | I have registered phone numbers in different area codes with different sip providers, so friends and family can contact me for the price of a local call. |
| 13:19.35 | dtwilson | Zeek - how do I check for fs corruption? (Debian) |
| 13:20.00 | Zeeek | ManxPower can tell you that |
| 13:20.00 | Josh1 | NOTICE[26690]: pbx_dundi.c:430 reset_global_eid: No ethernet interface found for seeding global EID You will have to set it manually. |
| 13:20.01 | Josh1 | ? |
| 13:20.02 | Nuttah | hmm i'll grab both and have a look |
| 13:20.16 | Torq | dammit, something has grabbed my 5060. no idea what it could be |
| 13:20.21 | dtwilson | am praying my old sysop had committed a working copy to our svn server |
| 13:20.26 | *** join/#asterisk xpasha (~pavel@195.146.99.163) |
| 13:20.27 | *** join/#asterisk MuSTaFa (~Temp@dsl81-214-7186.adsl.ttnet.net.tr) |
| 13:20.29 | ManxPower | Torq: do a "netstat -an | grep 5060" |
| 13:20.33 | xpasha | hi ppl |
| 13:20.50 | Torq | udp 0 0 192.168.1.102:5060 0.0.0.0:* |
| 13:20.56 | xpasha | how to make in asterisk call preferences like in cisco dial peer preferences |
| 13:21.11 | Torq | theres nothing in ps that remotely looks like a softphone or asterisk |
| 13:21.17 | xpasha | for example is one peer is failed then call goes to other |
| 13:21.24 | xpasha | for example if one peer is failed then call goes to other |
| 13:22.10 | Nuttah | btw zeek, when i spoke to GS about my ntp problems and transfer issues. I got the distinct impression they didnt know their arse from their elbow |
| 13:22.24 | Torq | I think i might have sip support compiled into mythtv |
| 13:23.08 | ManxPower | Torq: lsof | grep -i 5060 |
| 13:23.45 | Torq | mythfront 2939 mythtv 14u IPv4 5748 UDP mythtv:5060 |
| 13:23.46 | Zeeek | Nuttah you talk to Cindy ? |
| 13:23.47 | ManxPower | xpasha: This was discussed on the mailing list this week or last week. Too bad you missed it. |
| 13:23.53 | Torq | gee. seems mythfrontend does sip |
| 13:24.13 | ManxPower | Torq: Your question has now been answered. |
| 13:24.19 | Torq | yes, thanks |
| 13:24.42 | Nuttah | cant remember... but it was via email only, aint waiting on the phone from the UK fro some smuck to ask me inane questions |
| 13:25.18 | Zeeek | Nuttah : basically after requesting and getting full tcp dumps, Grandstream says "it must be your setup, you need to use STUN or a proxy server" |
| 13:25.21 | ManxPower | Nuttah: search the mailing list archives for "hellophone" for other GS firmware locations |
| 13:25.38 | Nuttah | lol rings a bell m8 |
| 13:25.46 | ManxPower | Is ANYONE here on the mailing list? |
| 13:25.55 | Zeeek | so I asked why 5.11 works then? |
| 13:26.04 | Zeeek | mailing list? What mailing list? |
| 13:26.30 | Zeeek | ntw anyone know anything about VPN hardware? |
| 13:27.04 | dtwilson | Zeeek: old I386 with Debian is your answer |
| 13:27.07 | Nuttah | as in...? |
| 13:27.26 | Nuttah | VPN routers? freeswan? |
| 13:27.37 | Zeeek | ya hw routers for small VPN |
| 13:27.41 | dtwilson | freeswan is deprecated |
| 13:27.45 | Zeeek | Bintec - Arkoon ? |
| 13:27.48 | Nuttah | Draytek are my fave VPN routers |
| 13:27.55 | Nuttah | for small offices |
| 13:28.02 | Josh1 | http://pastebin.ca/4229 |
| 13:28.06 | dtwilson | openvpn for ipsec, or simply use poptop for easy windows road warrior setup |
| 13:28.10 | Josh1 | help :( |
| 13:28.29 | Zeeek | this is all windows to windows |
| 13:29.17 | Zeeek | Josh1 FreeBSD ? |
| 13:29.19 | ManxPower | Josh1: You sure know how to screw up Asterisk |
| 13:29.22 | Zeeek | heh |
| 13:29.33 | Josh1 | heh |
| 13:29.34 | Josh1 | no |
| 13:29.38 | Josh1 | Redhat 9 |
| 13:29.42 | Zeeek | there is at least one really clear message about fixing a missing = sin |
| 13:29.51 | ManxPower | Josh1: First error: 002 Jan 15 08:24:56 WARNING[25123]: config.c:356 cfg_process: No '=' (equal sign) in line 388 of /etc/asterisk/indications.conf |
| 13:29.53 | ManxPower | fix it |
| 13:30.06 | dtwilson | ManxPower: did you follow my thread with Zeek earlier? re mwi and sip_pvt probs? reckon he's on the money with possible fs corruption? |
| 13:30.06 | Zeeek | you heard the Manx, FIX IT! NOW! |
| 13:30.16 | ManxPower | Josh1: Can you ping other machines from that machine? |
| 13:30.25 | Josh1 | yes |
| 13:30.33 | Josh1 | its a remote machine :P |
| 13:30.35 | sadie | anyone can point me to docs on scaling aix/sip ---> zap setup? |
| 13:30.50 | ManxPower | Josh1: also take out bindaddr= lines in your config files. |
| 13:31.00 | ManxPower | Josh1: How many network cards are installed? |
| 13:31.08 | Josh1 | one |
| 13:31.13 | Josh1 | 4 ips are binded to it |
| 13:31.18 | Nuttah | .5.18 works.. but no ntp |
| 13:31.28 | ManxPower | sadie: There are no docs for that. See the mailing list archives. |
| 13:31.40 | Zeeek | Nuttah, it calls out ? |
| 13:31.47 | Nuttah | aye |
| 13:31.54 | Zeeek | behind NAT ? |
| 13:31.57 | sadie | ManxPower: any pointers, subjects that I should pay attention too? |
| 13:31.59 | Nuttah | lol no |
| 13:32.02 | Zeeek | o |
| 13:32.16 | Zeeek | :0 |
| 13:32.16 | *** join/#asterisk h3x0r (~Justino@nv-65-40-157-57.sta.sprint-hsd.net) |
| 13:32.23 | Nuttah | ISP here.. have loads of IPs :P |
| 13:32.49 | Zeeek | :0 | rm /grandstream |
| 13:32.49 | ManxPower | sadie: No idea. |
| 13:33.03 | ManxPower | Josh1: Did it ever work? |
| 13:33.12 | ManxPower | Josh1: Are all the ip addresses in /etc/hosts? |
| 13:33.12 | Josh1 | yes |
| 13:33.15 | Josh1 | yes |
| 13:33.23 | Josh1 | it worked until i used the CVS version |
| 13:33.30 | ManxPower | Josh1: Well stop using it then. |
| 13:33.32 | sadie | ManxPower: well, gotta read the archives, thanks |
| 13:33.37 | ManxPower | Josh1: CVS-HEAD or stable. |
| 13:33.45 | Josh1 | ManxPower, im using CVS-HEAD |
| 13:33.52 | Josh1 | stable didnt seem to want to work |
| 13:33.57 | ManxPower | CVS-HEAD and (maybe stable) have been broken for a while. |
| 13:34.02 | ManxPower | 2 or 3 days, I think. |
| 13:34.14 | ManxPower | Get 1.0.3 and see if that works. |
| 13:34.41 | Josh1 | see |
| 13:34.48 | coppice | CVS-HEAD tends to be rather less satisfying than other kinds of head :-) |
| 13:34.50 | Zeeek | Nuttah what Draytek model for a small office? |
| 13:34.50 | Josh1 | im trying to get the version that works with broadvoice |
| 13:35.02 | Zeeek | coppice tu m'étonnes ! |
| 13:35.14 | Torq | it was mythphone in mythtv. i changed it to use 5059 |
| 13:36.10 | *** part/#asterisk ManxPower (~eric@dsl-208-164-150-160.datasync.com) |
| 13:36.25 | tih | Hey, folks... |
| 13:36.28 | *** join/#asterisk bjohnson (~bjohnson@ip226-181.tor.istop.com) |
| 13:36.34 | tih | Everybody having fun? |
| 13:36.37 | Josh1 | no :P |
| 13:36.38 | Zeeek | tih - just the man... don't worry I won't ask |
| 13:36.49 | tih | Zeeek: just sent you email... |
| 13:36.51 | Nuttah | Zeeek, what sorta vpns ya need? and in what direction |
| 13:37.11 | tih | Zeeek: you've got bad packets being repeated ad nauseam. |
| 13:37.20 | Josh1 | is it ok to keep switching versions like this? |
| 13:37.22 | Zeeek | very embarassing Nuttah - a cust wants to be able to dabble in the filesystem on the road - Windows only |
| 13:37.22 | tih | Zeeek: every time I've seen this, it's been a switching loop. |
| 13:37.25 | Josh1 | with make; make install, etc.? |
| 13:37.46 | Nuttah | so he is likely to be dynamic IP via windows? |
| 13:37.50 | Nuttah | christ |
| 13:37.50 | *** join/#asterisk Yatta (~yatta@port0056-adsl11.cwjamaica.com) |
| 13:38.07 | Zeeek | tih I'm gonna read your mail - thanks a lot for taking the time |
| 13:38.33 | Zeeek | Nuttah - the whole thing sucks but I want to make suck the least possible and they do ask my advice |
| 13:38.42 | Nuttah | is it dynamic? |
| 13:38.46 | Nuttah | ip wise |
| 13:38.51 | Zeeek | ya see we provide their email (FreeBSD) |
| 13:39.10 | Zeeek | I told them if you wanna do VPN you gotta get a fixed ip. Did I do ok ? |
| 13:39.25 | Zeeek | well I didn't say gotta |
| 13:39.30 | Nuttah | heh ya right ish |
| 13:39.37 | dtwilson | Zeeek: not strictly true, but that'll buy you time :) |
| 13:39.38 | Nuttah | it is a hell of a lot easier statically |
| 13:40.02 | Zeeek | seems to me it'd be easier to install and maintain, esp since they wanna send email - don't get me started |
| 13:40.03 | Nuttah | i'd go for a Vigor 2600 then matey |
| 13:40.23 | Zeeek | ok I'll look at it but it isn't in the proposal they got from an installer |
| 13:40.31 | dtwilson | I set up pptp VPN the other day - took under 2 hours of fiddling with settings - works a dream - and no need for clients to have static ip |
| 13:40.39 | *** join/#asterisk DannyF (~wizardon@h66n6c2o848.bredband.skanova.com) |
| 13:40.44 | Nuttah | pptp is one of the few that dont need it |
| 13:40.54 | DannyF | lo folks |
| 13:40.56 | Nuttah | ipsec you can do, but ideally you need to do via ssl |
| 13:41.12 | dtwilson | Nuttah: yup, that's why I chose it - for road warriors |
| 13:41.12 | Josh1 | hello |
| 13:41.15 | Nuttah | whcih in windows |
| 13:41.20 | Nuttah | is a fucking nightmare |
| 13:41.38 | tih | OpenVPN is also great for quick and easy VPN. |
| 13:41.54 | *** join/#asterisk [Yatta] (~not@port0056-adsl11.cwjamaica.com) |
| 13:41.56 | tih | NAT friendly, no need for permanent addresses. |
| 13:41.59 | Nuttah | aye |
| 13:42.05 | DannyF | any of you guys used ipcb.net? |
| 13:42.08 | dtwilson | tih: yeah we have OpenVPN too, but for some reason the windows clients won't work |
| 13:42.16 | Zeeek | tih thanks again - I will set the asterisk box on a hub and use ethereal next time - I've tried to notify the pther party but... nevermind |
| 13:42.35 | dtwilson | we use OpenVPN for linux-linux and pptp for windows->linux |
| 13:42.36 | Zeeek | I have openvpn installed on my win at the office but haven't been able to use it yet |
| 13:42.54 | Nuttah | bbl its lunch time |
| 13:43.04 | Zeeek | bon apétit |
| 13:43.13 | Zeeek | with two 'p' |
| 13:43.14 | Nuttah | merci |
| 13:43.33 | Zeeek | my espresso machine is b0rken ! |
| 13:43.39 | tih | Zeeek: are you completely sure you don't have a possible switching loop? |
| 13:43.55 | Zeeek | the setup is simple: a Linksys WAG54G |
| 13:44.02 | Zeeek | three or four boxes |
| 13:44.11 | Zeeek | DSL on the other side |
| 13:44.25 | Zeeek | might be the provider though... who knows |
| 13:44.55 | Zeeek | a quick look at the log size alone tells me it hasn't happened since :) |
| 13:45.21 | Zeeek | since Jan 12 20:48:22 |
| 13:45.33 | tih | Something is very very wrong, though. Not only are packets being retransmitted over an over -- it's happening with a packet with a bad checksum as well. |
| 13:45.34 | Zeeek | when I cleared the huge log |
| 13:45.52 | Zeeek | As little as I know, that did make me wonder |
| 13:45.59 | ellvis | ok, i did it, it is working |
| 13:46.14 | ellvis | capi is a good idea:) |
| 13:48.13 | ellvis | xpasha: hey, it's saturday, this is not a good day for working!:) |
| 13:48.16 | Zeeek | someone remind me what this means : zt hook failed: Device or resource busy |
| 13:48.45 | xpasha | ellvis :) |
| 13:48.46 | xpasha | hi |
| 13:49.14 | Zeeek | tih you work today ? |
| 13:51.02 | Josh1 | ugh |
| 13:51.03 | Josh1 | now |
| 13:51.08 | Josh1 | i installed 1.0.3 |
| 13:51.11 | Josh1 | and this is what i get now |
| 13:51.11 | Josh1 | http://pastebin.ca/4230 |
| 13:51.38 | ellvis | Zeeek: zt? like "zeroteam"?:) http://zeroteam.sk |
| 13:51.47 | tih | Zeeek: nope, I'm at home. |
| 13:51.51 | Josh1 | now it wont even start |
| 13:52.05 | DannyF | ouch |
| 13:52.05 | DannyF | did you manually clear the modules dir? |
| 13:52.09 | Zeeek | tih ok. Got a minute? (different subject) |
| 13:52.13 | Josh1 | no |
| 13:52.22 | Josh1 | what do you mean? |
| 13:52.27 | DannyF | try it then redo make install |
| 13:52.48 | DannyF | you had an older version installed previously i figure |
| 13:52.48 | Josh1 | wheres the module dir? /var/lib/asterisk? |
| 13:52.51 | Josh1 | yeah |
| 13:53.01 | DannyF | in /usr/lib/asterisk/modules/ |
| 13:54.46 | Josh1 | that worked |
| 13:54.51 | DannyF | :) |
| 13:55.04 | DannyF | good idea todo when switching versions |
| 13:55.10 | Zeeek | tih bythe way I'll know more if this keeps happening and I can get more dumps because if it is always the same src, that will narrow things down I think. |
| 13:56.30 | Josh1 | Jan 15 08:56:17 NOTICE[13153]: chan_sip.c:7295 handle_request: Unable to create/find channel |
| 13:56.30 | Josh1 | Jan 15 08:56:23 WARNING[13153]: chan_sip.c:685 retrans_pkt: Maximum retries exceeded on call 4a7c143d-33ae48ab-7753d98@192.168.1.95 for seqno 8707 (Non-critical Response) |
| 13:57.37 | Zeeek | Josh1 sip show peers |
| 13:57.59 | Josh1 | k |
| 13:58.15 | Zeeek | don't paste it all in here pls |
| 13:58.19 | Josh1 | alright |
| 13:58.28 | Josh1 | http://pastebin.ca/4231 |
| 13:58.33 | tih | Zeeek: gotta go - let me know if you find out something interesting, though... |
| 13:58.44 | Zeeek | ok thanks again tih |
| 13:58.57 | *** join/#asterisk JimVanM (~jimvanm@HSE-Toronto-ppp3490015.sympatico.ca) |
| 13:59.15 | Zeeek | Josh1 what is the error message above from - whenyou try to dial a number ? |
| 13:59.22 | Zeeek | or at reload? |
| 13:59.30 | Josh1 | i get that error |
| 13:59.33 | Josh1 | when i try to dial on my phone |
| 13:59.47 | Zeeek | paster the dial line here - just the DIal |
| 13:59.55 | Josh1 | ? |
| 14:00.03 | Zeeek | the line in extensions that dials |
| 14:00.11 | Josh1 | oh ok |
| 14:00.17 | Zeeek | the Dial() application ? |
| 14:00.27 | *** join/#asterisk burton27 (mimx@w201.ljudmila.org) |
| 14:00.32 | Josh1 | [outgoing] |
| 14:00.32 | Josh1 | exten => _1NXXNXXXXXX,1,Dial(SIP/${EXTEN}@sip.broadvoice.com,60) |
| 14:01.15 | Zeeek | isn't BV the provider everyone is having trouble with ? |
| 14:01.22 | Josh1 | i dont know |
| 14:01.24 | Zeeek | have you read the mailing list |
| 14:01.27 | Josh1 | nope |
| 14:01.43 | Zeeek | may want to consider doing so |
| 14:02.06 | Zeeek | also.... |
| 14:02.08 | Zeeek | http://www.voip-info.org/tiki-index.php?page=Asterisk+Broadvoice+patch |
| 14:03.07 | Josh1 | should i do the -p0 -R when patching? |
| 14:03.19 | sadie | is there sth like SER but for IAX ? |
| 14:03.29 | dtwilson | how do I find what address sip_alloc is trying to assign? |
| 14:04.01 | Zeeek | Josh1 here is my final thought: go to google and type : site:lists.digium.com asterisk broadvoice |
| 14:04.20 | Zeeek | there are 900 posts you may want to skim? |
| 14:04.35 | datareactor | can i configure my grandstream through web interface ? |
| 14:04.46 | dtwilson | datareactor: yes |
| 14:04.52 | Zeeek | datareactor that's what it's for |
| 14:04.54 | *** join/#asterisk WizardOne_ (~wizardon@h236n9c1o848.bredband.skanova.com) |
| 14:05.51 | DannyF_ | asrghl i'm alregic agaionst "&%# dsl lines |
| 14:05.55 | datareactor | zeek which default ip it have on its PC interface |
| 14:06.13 | Zeeek | the one you give it or the one it gets from DHCP |
| 14:06.16 | DannyF_ | Josh1 how did it go |
| 14:06.41 | Josh1 | not good |
| 14:07.03 | dtwilson | datareactor: press <menu> then the down arrow once, then press <menu> again - should display the ip address |
| 14:07.27 | Josh1 | just patched it, going to try make and make install again |
| 14:07.33 | DannyF_ | k |
| 14:07.48 | Josh1 | hmm |
| 14:07.50 | Josh1 | chan_sip.c: In function `handle_response': |
| 14:07.51 | Josh1 | chan_sip.c:6803: warning: too many arguments for format |
| 14:08.20 | datareactor | dtwilson should i have give the same class ip to used the system as well |
| 14:09.09 | dtwilson | datareactor: do you have a dhcp server? |
| 14:09.21 | dtwilson | for your lan |
| 14:09.36 | dtwilson | if so, then set the phone to grab ip from there |
| 14:09.42 | Josh1 | bah patched and same thing |
| 14:09.50 | DannyF_ | hum |
| 14:09.59 | dtwilson | if not, then you'll have t manually configure ip, mask, gw etc to fit your lan |
| 14:10.00 | datareactor | dtwilson the grandstream currently connected to cable modem |
| 14:10.07 | DannyF_ | u sure about versions for the patch or is the patch for HEAD? |
| 14:10.10 | Zeeek | datareactor you datareactor if you have a router it has a web interface and usualy a page to show what ip are given on the LAN side |
| 14:10.34 | *** join/#asterisk ctooley ([U2FsdGVkX@199.89.146.18) |
| 14:10.39 | Zeeek | datareactor in that case it already has an ip probably |
| 14:10.56 | datareactor | Zeek i want to connect Pc to pc interface |
| 14:11.14 | datareactor | Zeek it has public ip |
| 14:11.23 | dtwilson | fatareactor: you can do that - it just acts as a pass-thru |
| 14:11.36 | Zeeek | datareactor with no asterisk? |
| 14:11.38 | dtwilson | sp/fatareactor/datareactor |
| 14:11.42 | Josh1 | DannyF, no its the patch for 1.0.3 like you told me to get |
| 14:12.09 | ellvis | anyone have experience with Voismart IP Ranger phones? |
| 14:12.45 | datareactor | should i configure the PC with What ip to access asterisk ? |
| 14:12.51 | DannyF_ | uhm did i tell u to get a patch!? |
| 14:13.25 | Josh1 | well still |
| 14:13.30 | Josh1 | you need in im sure |
| 14:13.38 | Josh1 | otherwise broadvoice wouldnt put it on their website |
| 14:13.47 | DannyF_ | prolly so |
| 14:14.03 | *** join/#asterisk fafnir (~hello@tdds-gw.Moscow.gldn.net) |
| 14:14.05 | DannyF_ | dont use BV so i wouldnt know |
| 14:14.37 | *** join/#asterisk DrPete (~Pete@82-40-26-136.cable.ubr04.uddi.blueyonder.co.uk) |
| 14:14.41 | Zeeek | Josh1 here is my final thought: go to google and type : site:lists.digium.com asterisk broadvoice |
| 14:14.43 | DannyF_ | but * is a pretty fast running target so might be fixed already |
| 14:14.47 | datareactor | what is default password of grandstream phone ? |
| 14:14.53 | DannyF_ | Zeeek good idea |
| 14:14.55 | Zeeek | admin ? |
| 14:15.00 | ellvis | datareactor: admin |
| 14:15.02 | Josh1 | Zeeek, im looking through it |
| 14:15.05 | Josh1 | not really finding anything |
| 14:15.20 | Zeeek | well unless someone here now uses it no one can really help |
| 14:15.37 | Zeeek | BV has been spoken of many times as not working well with asterisk |
| 14:15.40 | DannyF_ | Josh1 dont give up thou ;) |
| 14:15.42 | datareactor | Zeek Thanks |
| 14:15.57 | DannyF_ | although i'we heard they improved alot |
| 14:16.02 | Zeeek | Josh1 don't give up but find the solution |
| 14:16.14 | DrPete | ello everyone |
| 14:16.15 | Zeeek | someone has posted about this |
| 14:16.27 | Zeeek | The dentist has arrived! |
| 14:16.29 | DannyF_ | lo DrPete |
| 14:18.34 | FaithX | What advantage over i4l has zapHFC ? |
| 14:19.17 | Josh1 | it seems like im registered |
| 14:19.31 | Josh1 | so i dont think its broadvoice |
| 14:19.35 | arrgh | my SPA-841 showed yesterday. |
| 14:19.42 | Josh1 | i think its that my incoming and outgoing parameters arent set correctly |
| 14:20.04 | arrgh | for being nearly the same cost as the grand's, its an improvement. |
| 14:20.42 | FaithX | any know how tessier got on? |
| 14:21.18 | DrPete | anyone know any UK viop provders that are good? |
| 14:22.18 | dtwilson | datareactor: google for grandstream user guide - it's available as pdf and very easy to follow |
| 14:22.34 | dtwilson | DrPete Magrathea telecom |
| 14:22.53 | Umaro | Hey guys, anyone here familiar with siemens phone systems? trying to get * connected to one, but having some problems sending calls to it. |
| 14:22.55 | dtwilson | magrathea.co.uk - speak to Linus Surguy - good guy |
| 14:23.03 | dtwilson | he does IAX2 peering as well |
| 14:23.37 | Josh1 | Umaro, i have a siemens too |
| 14:23.40 | Josh1 | HiNet LP 5100 |
| 14:23.41 | dtwilson | and profit sharing with 08** nmbers |
| 14:23.52 | Umaro | Josh1: cool.. mine is a HiCom 3000 |
| 14:23.56 | Josh1 | cool |
| 14:24.01 | Umaro | Josh1: do you have it working with *? |
| 14:24.08 | Josh1 | well.. it registers |
| 14:24.13 | Josh1 | you have to use Gateway Routing |
| 14:24.22 | Umaro | Josh1: oh, the HiNet is a voip one, I see |
| 14:24.26 | Josh1 | and make sure your e164 number is your * extension |
| 14:24.31 | Umaro | I'm connecting to my hicom 3000 via PRI |
| 14:24.35 | Josh1 | ohh |
| 14:25.29 | *** join/#asterisk ariel_ (~Ariel@dsl-20-177.cofs.net) |
| 14:25.56 | ariel_ | hello everyone. |
| 14:26.03 | Umaro | Having an odd problem that maybe you can help me with, though.. when I try to make a call from * to the siemens, like Zap/g4/4155777, it rings extension 300 on the siemens, even though it's supposed to ring say, 415. If I call any of my 100 DIDs, it always goes to that same, wrong 300 extension |
| 14:26.09 | Zeeek | arrgh so how's the phone working with asterisk ? |
| 14:26.28 | fafnir | is the handbook good enough for a newb to setup their asterisk? |
| 14:27.18 | dtwilson | fafnir - that depends on what other knowledge you have |
| 14:27.19 | Josh1 | dunno.. still trying to get mine working |
| 14:28.03 | fafnir | dtwilson: nice ambigous answer |
| 14:28.13 | dtwilson | my pleasure :) |
| 14:28.19 | ariel_ | fafnir, yes plus the wiki and other sites like:http://www.fnords.org/~eric/asterisk/ and the wiki: http://www.voip-info.org/wiki-Asterisk |
| 14:28.48 | dtwilson | what I meant is - if you're linux savvy, then it's sufficient, otherwise imho it's not really good enough |
| 14:29.01 | dtwilson | on its own |
| 14:29.17 | ariel_ | Umaro, your group 4 first one to dial might be ext 300 |
| 14:29.18 | fafnir | well, im fbsd savvy, which is the enviroment i compiled it in |
| 14:29.32 | dtwilson | ok fairynuff - should hopefully be ok for you then |
| 14:29.59 | dtwilson | still a bit of a learning curve to the telephony side of things |
| 14:30.11 | *** join/#asterisk bonbon-home (~happy@81-86-185-223.dsl.pipex.com) |
| 14:31.45 | ariel_ | dtwilson, telephony side sometimes is hard but in time it will be logical. Besides we are here to help |
| 14:35.37 | dtwilson | ariel_: thanks - I think I've sussed a lot of it now anyways - was just giving insight for newb :) |
| 14:37.02 | fafnir | it says that its configured to work out of the box as a voip pbx, what if i want to voip to it and make outgoing calls on the pstn? is there a specific resource i should read? |
| 14:37.50 | DrPete | dtwilson: I will check it out, thanks |
| 14:37.56 | Josh1 | i think its my phone : |
| 14:38.00 | Josh1 | :|* |
| 14:38.03 | dtwilson | fafnir: voip-info.org has most of the answers - also the mailing list |
| 14:38.07 | ariel_ | fafnir, if you see the sample at http://www.fnords.org/~eric it will give you a starting base to begin. |
| 14:38.15 | fafnir | okay thanks |
| 14:38.42 | *** join/#asterisk expressfone1 (expressfon@80.27.5.184) |
| 14:38.44 | dtwilson | and like ariel said earlier, there's plenty people here to help as well :) |
| 14:39.10 | expressfone1 | hi all |
| 14:39.25 | dtwilson | ariel: you wouldn't happen to know much about debugging rtp and mwi errors? |
| 14:39.49 | expressfone1 | any one from miami here??? |
| 14:40.18 | FaithX | What advantage over i4l has zapHFC ? |
| 14:41.35 | *** join/#asterisk scardinal (~supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 14:43.15 | DrPete | dtwilson: there is not pricing info on there site, can you give me an idea of the prices? |
| 14:44.40 | *** join/#asterisk Yatta (~yatta@port0056-adsl11.cwjamaica.com) |
| 14:44.48 | Yatta | When i do 'iax2 show peers' and under name/Username do those names correspond to the names that are in your iax.conf? |
| 14:45.02 | dtwilson | DrPete: sorry I've no idea of prices, I just had brief dealings with them about 2 years ago - doing some testing - though have heard others being very pleased with their service also |
| 14:45.16 | Yatta | for eg. i have iaxfwd and iaxtel under name/username |
| 14:45.32 | Yatta | i also see those heading in my iax.conf |
| 14:45.49 | dtwilson | if you mail them, they'll no doubt be happy to provide pricing for whichever services you require |
| 14:47.28 | Josh1 | hmm progress |
| 14:47.34 | Josh1 | im able to dial into my pbx now |
| 14:47.36 | Josh1 | and it shows it |
| 14:47.41 | Josh1 | under sip debug |
| 14:47.43 | Josh1 | but |
| 14:47.47 | Josh1 | its not routing to my ip phone |
| 14:47.50 | Josh1 | at ext. 1000 |
| 14:48.47 | DannyF_ | <-- is still looking for someone that use ipcb.net |
| 14:49.29 | tzanger | DannyF_: what's that |
| 14:49.41 | DannyF_ | interconnect market |
| 14:49.48 | DannyF_ | buying selling routes |
| 14:49.50 | *** join/#asterisk bjohnson (~bjohnson@ip226-181.tor.istop.com) |
| 14:50.14 | JerJer | DannyF_: bleh |
| 14:50.20 | DannyF_ | bleh? |
| 14:50.43 | DannyF_ | please elaborate juuust a tad ;) |
| 14:51.04 | ariel_ | expressfone1, I am from Miami |
| 14:51.24 | JerJer | DannyF_: so you are going to trust your traffic to someone who plays middle? |
| 14:51.32 | Josh1 | me too, ariel_ |
| 14:51.36 | JerJer | how can you control QoS? |
| 14:51.44 | DrPete | dtwilson: thanks. Do you know any good pay as you go services, just so I am can play with asterisk, I am new to it all and keen to get it working :) |
| 14:51.50 | DannyF_ | well have you tried them? |
| 14:52.06 | Josh1 | To: 1954xxxxxxx <sip:1954xxxxxxx@207.58.143.217>;tag=as6823eabd |
| 14:52.09 | Josh1 | i think thats why |
| 14:52.09 | JerJer | this tells me they suck: A non-refundable, monthly Membership Fee of US $250.00 allows a company to gain access to VoIP Termination Service. |
| 14:52.18 | tzanger | DannyF_: I've tried a NUMBER of VOIP providers who resell |
| 14:52.22 | czero | unless your network has a physical pipe to them you can't control QoS anyway |
| 14:52.24 | tzanger | it all ends in tears |
| 14:52.25 | Josh1 | it should say sip:1954xxxxxxx@sip.broadvoice.com |
| 14:52.36 | tzanger | deal with someone who does the termination themselves and knows their shit |
| 14:52.43 | JerJer | so a $250 monthly privilege fee |
| 14:52.45 | czero | but yeah a 250$ / month 'membership' fee should be a red flag |
| 14:53.06 | JerJer | above and beyond any termination fees or other fees they subject you to |
| 14:53.16 | DannyF_ | think the $250 is just a newbie filter |
| 14:53.23 | JerJer | its a stupid filter |
| 14:53.29 | JerJer | anyone that pays it is stupid |
| 14:53.36 | DannyF_ | well sure would cut down on support costs |
| 14:53.36 | DrPete | lol |
| 14:53.59 | tzanger | yup it's an idiot tax |
| 14:54.02 | tzanger | like "list price" |
| 14:54.08 | tzanger | DannyF_: hardly |
| 14:54.13 | DannyF_ | well JerJer show me an global interconnect market thats free then |
| 14:54.18 | JerJer | just like they say in the teams... Your operation is only as strong as the weakest link |
| 14:54.22 | tzanger | how many smart people with no money vs how many dumb people iwth too much money do you know? |
| 14:54.36 | Torq | Jan 15 14:56:14 NOTICE[25691]: chan_sip.c:7531 handle_request: Registration from 'Asterisk <sip:xlite1@asterisk>' failed for '192.168.1.1' |
| 14:54.50 | JerJer | DannyF_: we don't charge our customers a $250 monthly privilege fee and we terminate calls globally |
| 14:54.54 | Torq | why?!? this is pissing me off :( i cant figure out why xlite wont talk to my asterik |
| 14:55.08 | JerJer | Torq: you need a type=peer with host=dynamic and a valid secret=blah |
| 14:55.14 | DannyF_ | JerJer well what if i want to sell you routes? |
| 14:55.22 | JerJer | we don't buy routes, that is the whole point |
| 14:55.30 | DannyF_ | heh |
| 14:55.37 | DannyF_ | so youre a telco then |
| 14:55.46 | Torq | JerJer: got all that |
| 14:55.49 | tzanger | DannyF_: you can certainly resell nufone's service by buying minutes and reselling them but now you're one of the bad guys :-) |
| 14:55.52 | Torq | Oh. I have type=friend |
| 14:55.57 | JerJer | you cannot guarantee QoS unless, as tzanger said, you have a direct pipe |
| 14:56.05 | DannyF_ | tzanger ;) |
| 14:56.44 | DannyF_ | well i can guarantee my termination points |
| 14:56.49 | JerJer | we use the Internet as our last mile |
| 14:56.53 | DannyF_ | but of course not the ipcb stuff |
| 14:56.56 | DrPete | I am new to all this, but as a buiness cutomer, I would not pay 250 if I could avoid, it, that to me would just make me think the company was taking the piss |
| 14:56.59 | dtwilson | anybody any ideas on the following errors? http://www.dtwilson.com/asterisk.htm |
| 14:57.08 | *** join/#asterisk humberto (~hav@201.128.177.84) |
| 14:57.14 | *** join/#asterisk Damin (~damin@nucleus.nacs.net) |
| 14:57.16 | tzanger | DannyF_: you can't guarantee shit unless you're terminating to PSTN yourself |
| 14:57.18 | JerJer | and there is no way in hell we will push any of our traffic back out to the Internet for termination |
| 14:57.28 | tzanger | DrPete: exactly |
| 14:57.49 | *** join/#asterisk Druken (Druken@CPE00023f0862f9-CM000e5cde4ca2.cpe.net.cable.rogers.com) |
| 14:57.51 | *** join/#asterisk FreeSTyLe (ThaLia_go_@81.212.138.33) |
| 14:58.26 | DannyF_ | tzafrir well using a CLEC to terminate |
| 14:58.32 | DannyF_ | to PSTN |
| 14:58.40 | JerJer | then you are not terminating it yourself |
| 14:58.43 | JerJer | the CLEC is |
| 14:58.52 | Torq | no luck with xlite and asterisk |
| 14:58.52 | DannyF_ | of course |
| 14:59.03 | tzanger | DannyF_: as in PRI to the CLEC or VOIP to CLEC |
| 14:59.10 | JerJer | Torq: no is no luck with your config |
| 14:59.17 | DrPete | High cost does not allways mean high service. I dont mind paying for what I want, but I dont like feeling ripped off. And if the only way a company can give a good serive is to charge over the odds then in my mind they are doing something very wrong |
| 14:59.20 | DannyF_ | VOIP |
| 14:59.28 | JerJer | LoL |
| 14:59.32 | tzanger | hahaha |
| 14:59.32 | Damin | Morning peeps.. |
| 14:59.33 | JerJer | so your playing middle |
| 14:59.37 | DannyF_ | yes |
| 14:59.41 | JerJer | find a clue |
| 14:59.56 | tzanger | PRI term to CLEC is what I would call you terminating to PSTN |
| 14:59.59 | DannyF_ | that site aint open yet ;) |
| 15:00.11 | tzanger | VOIP to CLEC is man-in-the-middle |
| 15:00.12 | DannyF_ | tzanger havent gotten that far yet |
| 15:00.26 | tzanger | technically PRI to CLEC is man in the middle too but I'd consider it close enough |
| 15:00.38 | tzanger | since there's no issues with QoS or anything |
| 15:00.43 | DannyF_ | tzafrir true |
| 15:00.49 | DannyF_ | tzanger* |
| 15:00.50 | DrPete | Anyone know any good UK VOIP services that I can setup quick to test asterisk |
| 15:01.03 | FaithX | I've been hunting around but cannot seem to find what advantage there is using zapHFC over i4l, anyone? |
| 15:01.05 | DannyF_ | DrPete voipuser |
| 15:01.08 | Damin | DrPete: I've heard that SimpleTelecom has good UK rates. |
| 15:01.38 | tzanger | FaithX: I don't know if there is significant difference |
| 15:01.46 | DannyF_ | JerJer well I have to start somehere dont i ;) |
| 15:01.58 | JerJer | not by playing middle |
| 15:02.02 | DrPete | ooh, ok I will check them out. I already have calls comming to my linux box and xlite answering and I am very impressed. I need to get dialing out now, heh, then get some IP phones |
| 15:02.10 | JerJer | buy a building |
| 15:02.14 | JerJer | buy a switch |
| 15:02.15 | dtwilson | DrPete: I'd let you test via our box, if I could get the damned thing fixed :) |
| 15:02.17 | JerJer | get CLEC status |
| 15:02.22 | JerJer | interconnect with either another CLEC or the LEC |
| 15:02.26 | JerJer | cake |
| 15:02.30 | JerJer | and/or |
| 15:02.31 | FaithX | DrPete, Besides the fact that UK seems bottlenecked to the rest of the world... US has some providers with good world rates... like unlimited world land line calls |
| 15:02.32 | DannyF_ | so i'll just call my dedicated Cisco guy and order a few M in equipment and get started then *sigh* |
| 15:02.33 | tzanger | JerJer: that's ground-up... if he wants to start somewhere, get a PRI and start reselling it |
| 15:02.38 | Damin | JerJer: You heard of the VoIP Peering Fabric up there in the Windy City? |
| 15:02.42 | JerJer | lol |
| 15:02.43 | JerJer | its a joke |
| 15:02.45 | expressfone1 | any one from miami here??? |
| 15:02.56 | *** join/#asterisk sgagers (~sgagers@151.97.29.77) |
| 15:03.11 | JerJer | Damin: its another middle play |
| 15:03.16 | DannyF_ | JerJer ah I see youre assuming im in the US ;) |
| 15:03.27 | Damin | JerJer: Yep... |
| 15:03.41 | JerJer | Damin: that simply wants your money to do what anyone with half a brain can do for themselves |
| 15:04.00 | JerJer | DannyF_: its the same no matter where you are at |
| 15:04.07 | DannyF_ | Damin same thing as the ipcb.net stuff |
| 15:04.17 | JerJer | ipcb is just stupid |
| 15:04.33 | DannyF_ | JerJer not quite the same, regulatory stuff |
| 15:04.36 | JerJer | voip peering fabric at least has their own network operations |
| 15:04.38 | DrPete | FaithX: dam you are kidding, lol I did not know I was bottlenecked, lol Yeah, if I paided US company it would be very cheap for me, as the exhange rate is good |
| 15:04.41 | tzanger | JerJer: it's what PHBs like to see, nothing more |
| 15:04.47 | tzanger | it's perceived value |
| 15:04.50 | JerJer | DannyF_: every country has lawyers |
| 15:04.53 | JerJer | its the same |
| 15:04.56 | tzanger | something I'm having a hard time learning to understand :-) |
| 15:05.06 | Druken | do we have any canadian voip providers in tha house? :) |
| 15:05.19 | tzanger | Druken: I haven't found any relaly good ones yet |
| 15:05.28 | DrPete | dtwilson: thanks, whats up with your box? |
| 15:05.33 | Josh1 | would this be a problem |
| 15:05.34 | Josh1 | if |
| 15:05.35 | Zeeek | tzanger no joy with calgary ? |
| 15:05.37 | *** join/#asterisk jaiger (~jaiger@65.86.55.130) |
| 15:05.39 | JerJer | because canada is not exactly friendly to competition |
| 15:05.43 | Druken | tzanger: well, that's not very assuring |
| 15:05.45 | czero | tzanger hold on then, we'll be offering serivce in CAnada very soon |
| 15:05.46 | tzanger | Zeeek: their businesssupport email address bounced :-) |
| 15:05.51 | *** join/#asterisk Mike_TK (~Mike_TK@213.180.245.62) |
| 15:05.51 | DannyF_ | JerJer well depends on how much $ you invest in it... someplaces still keep it out of reach for anyone but the filthy rich |
| 15:05.54 | Josh1 | i have a phone, connecting through my router, to my box in anohter state? |
| 15:06.00 | Zeeek | tzanger - never a great sign |
| 15:06.03 | Josh1 | because on * |
| 15:06.05 | Damin | JerJer: Shhh.. we should be encouraging as many people as possible to become men in the middle.. Let them go out, do the marketing, get the customer base, fragment the market and build the minutes, and when the entire thing collapses, we can do a roll up. ;) |
| 15:06.09 | Josh1 | it says 192.168.1.95 |
| 15:06.12 | Josh1 | and i think that may be a problem |
| 15:06.18 | tzanger | Zeeek: no... really it forwarded to someone and THAT email addy bounced, so it could have just been a personnel change |
| 15:06.21 | tzanger | I ws gonna call and never got to it |
| 15:06.23 | czero | JerJEr why worry about freind, business is business, we've not hard a hard time setting up in canada todate |
| 15:06.31 | czero | its jsut not as cheep as the US |
| 15:06.36 | Zeeek | tzanger write to Mike@ same domain |
| 15:06.36 | dtwilson | DrPete: I only wish I knew - getting the errors shown at http://www.dtwilson.com/asterisk.htm |
| 15:06.46 | Druken | tzanger: where you are lookin for termination? |
| 15:06.48 | tzanger | yeah you were aying that Zeeek :-) |
| 15:06.52 | JerJer | czero: show me in writing here you have the rights to resell elements |
| 15:07.01 | file[laptop] | dtwilson: it can't use the ip address |
| 15:07.04 | JerJer | here = where |
| 15:07.06 | tzanger | JerJer: I'm still trying to get that from Group Telecom for you |
| 15:07.10 | tzanger | I don't think they think I'm serious |
| 15:07.19 | tzanger | Telus should be able to resell UNEs though |
| 15:07.22 | czero | do not deal with GT |
| 15:07.22 | Druken | GT blows ass |
| 15:07.27 | Druken | they never return calls |
| 15:07.28 | czero | I know guys that worked there |
| 15:07.29 | Zeeek | only one voIP provider has ever answered a first email: that was ICH with an idiotic cut and paste |
| 15:07.33 | czero | and won;t use them |
| 15:07.36 | DrPete | dtwilson: I wish I could help, google not turn up anything? |
| 15:07.38 | Josh1 | hey it worked :) |
| 15:07.46 | dtwilson | file: where can I find where to set * to bind to address? |
| 15:07.46 | czero | Druken you here in the GTA |
| 15:07.47 | DannyF_ | congratz Josh1 |
| 15:07.52 | tzanger | I'e had good success with GT ... it's all with the right people |
| 15:07.58 | file[laptop] | dtwilson: mayyyyyyyyybe sip.conf |
| 15:08.04 | Druken | czero: no, i'm an hour north... |
| 15:08.06 | Damin | Josh1: Cool.. you finally went poopy on the potty? :) |
| 15:08.06 | tzanger | mind you they've been bought and rebought a few times so I'm no longer sure, all my experiences are 5 years old |
| 15:08.14 | DannyF_ | tzanger getting a good contact is massively important |
| 15:08.16 | Josh1 | lol |
| 15:08.16 | czero | Druken kewl kewl |
| 15:08.17 | tzanger | Damin: don't laugh, my son's working on that right now :-) |
| 15:08.25 | Damin | tzanger: So is my duaghter.. |
| 15:08.29 | JerJer | we've had three seperate hot shot lawyers attempt to get UNE rights in cananda and not one of them could get it |
| 15:08.32 | tzanger | Damin: right on |
| 15:08.37 | czero | JerJer I don;t show contracts, sorry |
| 15:08.39 | tzanger | JerJer: rreally |
| 15:08.43 | Druken | czero: why do you ask ? |
| 15:08.43 | tzanger | that is odd |
| 15:08.50 | DannyF_ | JerJer just spend more money then... |
| 15:08.55 | JerJer | canada is not friendly to that kind of competition |
| 15:09.01 | czero | Druken noticed you are on rogers |
| 15:09.03 | JerJer | plain and simple |
| 15:09.11 | tzanger | JerJer: yeah, just spend more money... don't let common sense and price stand in the way. :-p |
| 15:09.12 | Damin | UNE is dead. Long live UNE! |
| 15:09.12 | czero | and sice most of the rogers users are in GTA :) |
| 15:09.16 | Druken | don't deal with GT, go with allstream, ask for their "voip PRI" :) |
| 15:09.16 | DannyF_ | more places like that asI said JerJer |
| 15:09.17 | JerJer | czero: because you cannot show that |
| 15:09.27 | tzanger | Druken: allstream I've had shitty service from |
| 15:09.27 | JerJer | ... you have proper rights to resell elements |
| 15:09.31 | Druken | czero: not most.. but a good portion |
| 15:09.34 | tzanger | they're quick to bill but support is a joke |
| 15:09.42 | Druken | tzanger: agreed |
| 15:09.47 | Druken | but... |
| 15:09.54 | dtwilson | file: thanks - now I'm getting somewhere :D |
| 15:09.56 | czero | druken yeah most (hint I'm responsible for thier network:) ) |
| 15:09.57 | Druken | at least you can get the innitial install done |
| 15:10.01 | tzanger | VOIP PRI? what the fuck is that, a frame T1 carrying ni2 signalling or something? |
| 15:10.03 | JerJer | we've even tried setting up as a n actual Canadian company and still couldnt' get the proper rights |
| 15:10.06 | JerJer | ala Shido6 |
| 15:10.17 | *** join/#asterisk santiago (~santiago@63.245.86.86) |
| 15:10.32 | Josh1 | hmm |
| 15:10.34 | Druken | czero: great, that means you can unlock my modem and make it 5 meg bi directional |
| 15:10.38 | Josh1 | now i cant pickup the calls |
| 15:10.40 | Josh1 | call* |
| 15:10.47 | Damin | Hey.. here is a kick in the ass from the RBOC.. We purchased the assets of a competitor, that had a remote POP served by a pair of ICB T1's run betwen latas and between CARRIERS on the SBC Metro Blitz pricing promotion.. |
| 15:10.48 | DannyF_ | JerJer well Canadian authorities are very similar to Swedish ones *sigh* |
| 15:11.03 | Damin | Obviously, it is totally illegal... |
| 15:11.12 | czero | JerJer youcan think I have or not makes no diffrence to me, but the more people that think they can't do business here and stay away the better for me, then we won't see people selling shit cheep like in the US so better margins for me |
| 15:11.24 | Damin | So we wanted to move the circuits onto our own DS3 for Long-Haul.. |
| 15:11.27 | Druken | tzanger: a voip pri is a special product of allstream, allows DNIS, 911 access, free setting is CID etc... |
| 15:11.30 | *** join/#asterisk Miker (~mike@201.129.119.248) |
| 15:11.38 | DannyF_ | czero hehe |
| 15:11.42 | czero | Druken I do core, I don;t even have 5M serivce |
| 15:11.50 | Damin | Fucking SBC said "No.. that circuit is in violation of so many tarrifs it isn't funny. We can't move it. We have to terminate it NOW" |
| 15:12.15 | Druken | czero: even better, drag a big pipe into my office :) |
| 15:12.19 | czero | :) |
| 15:12.25 | Damin | And I'm like "Wait a second.. You fuckers put it in in the first place, and WE are trying to respect the freaking tarriff by running the last mile as FCC2." |
| 15:12.39 | czero | gave my notice yesterday so I doubt I can even get into the core anymore :) |
| 15:12.42 | tzanger | Druken: tell me more... what's it got over a regular PRI from Bell, where I have DNIS, 911 and CID name+number setting |
| 15:12.57 | Damin | tzanger: It's got a cooler name.. :) |
| 15:13.02 | *** join/#asterisk gambolputty (~gambolput@cblmdm65-221-51-80.buckeye-express.com) |
| 15:13.16 | Druken | tzanger: i'm not sure on all the details... prolly cheaper |
| 15:13.24 | czero | no its more then bell |
| 15:13.25 | DrPete | FaithX: Can you surgest any good US provider that has good UK rates etc. I will also be calling the US a bit, will the US calls be free? |
| 15:13.34 | czero | but not by much |
| 15:13.38 | Damin | DrPete: Try NuFone.. I hear they rock. |
| 15:14.13 | Druken | czero: are you doind toronto termination? |
| 15:14.18 | JerJer | Damin: have your legal department go in front of a jugeto stop the termination (injunction) then petition the PUC for a ruling |
| 15:14.35 | tzanger | I have heard good things about nufone too, just do your research before asking questions. :-) |
| 15:14.42 | czero | Druken not retail yet, but soon, so far only bulk |
| 15:14.56 | JerJer | yes the cocksuckers at NuFone get pissed off easily |
| 15:15.02 | Zeeek | we noticed |
| 15:15.09 | DrPete | Ohh, so much to look at eheh. Do most have a pay as you go system, or can you get providers that send you a bill. I would hate to be cut off half way though a call |
| 15:15.14 | Druken | wuts yer blocks? |
| 15:15.35 | tzanger | Druken: almost all have pay as you go, but I have never had it cut me off mid-call... |
| 15:15.45 | Zeeek | DrPete I have only pay as you go - some will do auto renew but I don't trust most forthat |
| 15:15.49 | Damin | JerJer: Not worth the time. I got them to leave it up until we run the new circuits in. Funny thing is, I wouldn't have worried about it and I would have left things alone, but it was a 12 mile circuit, when we have a DS3 terminated in the CO a half mile away. |
| 15:15.53 | tzanger | it'll go negative and perhaps not allow further calls, or may be let you go so far negative before cutting off... depends on the provider |
| 15:15.57 | Druken | pay as you go? |
| 15:16.01 | DannyF_ | just curious about how many minutes everyone is terminating a month... anyone wanna share? |
| 15:16.08 | Zeeek | provisioned account to the geeky |
| 15:16.17 | Damin | JerJer: In the end, it's cheaper just to drop the damn things onto our DS3 at the CO. |
| 15:16.21 | Zeeek | Druken ^^^ |
| 15:16.37 | Damin | JerJer: Best of all, SBC is screwing themselves out of like $20,000 in revenue on the circuits. |
| 15:16.50 | Damin | JerJer: And our cost is going down significantly.. |
| 15:17.00 | Damin | JerJer: Never buy ISPs.. ;) |
| 15:17.01 | DannyF_ | Damin odd SBC policy ;) |
| 15:17.03 | Torq | telecom is a total scrum |
| 15:17.10 | Damin | DannyF_: Everyting about SBC is odd.. |
| 15:17.11 | JerJer | Damin: hell no |
| 15:17.13 | DrPete | Zeeek: yeah I agree, do you find it a problem running out of credit, or do thet notify you? |
| 15:17.18 | JerJer | only sell ISPs :) |
| 15:17.22 | DannyF_ | Damin so i'we come to understand as well |
| 15:17.36 | Zeeek | DrPete I keep track of my credit. They all have web interfaces |
| 15:17.40 | *** join/#asterisk TheEmperor (TheEmperor@218.111.50.6) |
| 15:17.44 | Josh1 | anything wrong with this? |
| 15:17.45 | Josh1 | http://pastebin.com/229182 |
| 15:17.50 | Damin | JerJer: Hehehehe.. As I said at Astricon, I've been doing it for ten years.. ;) |
| 15:17.57 | Josh1 | because.. when i pickup my phone, i dont hear anything |
| 15:18.01 | *** join/#asterisk kludgebox (~bob@12.171.178.194) |
| 15:18.03 | Damin | JerJer: This last one was interesting.. |
| 15:18.33 | Damin | JerJer: The week after we got verbal commitment, one of the partners had a stroke.. He was lying in his house, paralyzed for 18 hours.... |
| 15:18.42 | JerJer | lol |
| 15:18.49 | JerJer | rock on |
| 15:18.56 | JerJer | no Letter of Commit? :) |
| 15:19.05 | Damin | JerJer: Then, the week after we took ownership, the other partner went into the hospital for a stomach obstruction.. |
| 15:19.11 | JerJer | even better |
| 15:19.28 | *** join/#asterisk JonR800 (jr@pcp05013027pcs.plyntv01.mi.comcast.net) |
| 15:19.31 | DrPete | Zeeek: thats ok then. Is it worth having more than one company incase one is down? |
| 15:19.52 | Damin | JerJer: Yeah, but that came after.. ;) The guy had dropped off the radar, so we knew something was up,. but the other owner didn't tell us until the day after we signed everything.. |
| 15:19.54 | JerJer | DrPete: sure |
| 15:19.58 | JerJer | use the power of asterisk |
| 15:20.08 | JerJer | or even Least Cost Routing, if you are that hardcore |
| 15:20.20 | Zeeek | DrPete up to you I've been testing about 10 companies for quality and "service" for the last 6 months |
| 15:20.22 | JerJer | Damin: wonderful |
| 15:20.30 | Damin | JerJer: No big deal, because they needed to get out of the business anyway.. But there is a third partner involved.. Ever seen "Final Destination"? I'd be shitting bricks if I were the last guy.. :) |
| 15:20.40 | JerJer | lol |
| 15:20.47 | JerJer | you cannot escape death |
| 15:20.47 | expressfone1 | any one here need traffic for Cuba +53 (IAX2)??? |
| 15:20.50 | JerJer | it will find you |
| 15:21.22 | Moc_ | morning |
| 15:21.28 | czero | morning moc |
| 15:21.34 | Druken | well, castro doesn't usually take my calls... so no thanks |
| 15:22.00 | expressfone1 | drunken> call hem "mariconson" |
| 15:22.25 | Torq | express: how much? |
| 15:22.29 | FaithX | How do I get a CVS version that is not going to leave me bleeding? |
| 15:22.32 | DrPete | Zeeek: ooh, whats the best for me in the UK? If i go with a US company, will it matter then I am going allt the way to the US to come back again? |
| 15:22.32 | expressfone1 | .50 usd |
| 15:22.42 | Zeeek | expressfone1 you could do a Tony Soprano in Miami |
| 15:22.49 | Torq | drpete: Im using sipgate.co.uk |
| 15:23.06 | DrPete | Torq: are they good?? |
| 15:23.26 | Torq | Asterisk ok, just cant get the goddamn xlite softphone to reg to asterisk as an extension :( |
| 15:23.26 | Damin | Moc_: I solved the problem w/ Polycom and Chan_Sip. |
| 15:23.41 | Moc_ | yea I just looked at the bug report |
| 15:23.44 | Torq | can someone with xlite and asterisk help me in PM? |
| 15:23.45 | Damin | Moc_: msg.mwi.1.subscribe="" |
| 15:23.50 | Moc_ | yea ;) |
| 15:23.54 | czero | DrPete depnds on the latency to the provider in the US |
| 15:24.25 | Torq | hmm, anyone here terminate in the phillipines at sensible latency? |
| 15:24.43 | jaiger | Damin: what was the problem? |
| 15:25.03 | ctooley | I keep getting this message, I'm guesing it's not important: Request to schedule in the past?!?! |
| 15:25.09 | expressfone1 | any one know a irc chat where ppl sell traffic??? |
| 15:25.11 | DrPete | Torq: man, they can give me a local number, and I have just got one elseware, lol dam |
| 15:25.11 | JerJer | ctooley: cvs update |
| 15:25.20 | Torq | drpete: heh |
| 15:25.27 | Zeeek | #cocaine |
| 15:25.32 | Torq | drpete: I have four local numbers ;) |
| 15:25.43 | Moc_ | Damin, I got incoming to work on my Polycom, but no outgoing via h323 :| well it better than nothing |
| 15:25.46 | ctooley | JerJer, since today? |
| 15:26.03 | Damin | ; Fucking Nelson's Photography Won't Leave Us Alone! |
| 15:26.04 | Damin | exten => 2169203021/4403335061,1,Answer |
| 15:26.04 | Damin | exten => 2169203021/4403335061,2,Wait(2) |
| 15:26.04 | Damin | exten => 2169203021/4403335061,3,Playback(discon-or-out-of-service) |
| 15:26.04 | Damin | exten => 2169203021/4403335061,4,Hangup |
| 15:26.05 | JerJer | i haven't seen that error message in many many many moths |
| 15:26.08 | Damin | Fuck.. sorry.. |
| 15:26.09 | DrPete | Torq: hehe, at least the one I have is not tied anywhere and they will let me do what I want with it, point it anywhere etc |
| 15:26.21 | DannyF_ | JerJer btw how do you guys terminate in foregin countries? via LEC? |
| 15:26.22 | Torq | Drpete: really? how so? |
| 15:26.24 | JerJer | AHHH anti-ex girlfriend logic |
| 15:26.47 | JerJer | Damin: tip... don't answer the line, then it won't cost you :P |
| 15:27.00 | Damin | JerJer: It doesn't cost me anyway. :) |
| 15:27.04 | JerJer | Playback(discon-or-out-of-service|noanswer) |
| 15:27.05 | Damin | JerJer: It's at home.. |
| 15:27.07 | JerJer | oh :( |
| 15:27.16 | JerJer | i'm soooo used to PRI even for my home line :P |
| 15:27.32 | tzanger | JerJer: me too |
| 15:27.36 | ctooley | JerJer, this is CVS-HEAD-12/21/04 with a "cvs -q update -d" in the directory and rebuilt this morning. |
| 15:27.55 | DrPete | Torq: can I connect sipgate to mt asterisk box then? |
| 15:28.09 | DannyF_ | DrPete yes you can |
| 15:28.46 | Damin | jaiger: |
| 15:28.48 | Damin | It is more than likely a bug w/ Polycom's SIP SUBSCRIBE messages, and not neccessarily an Asterisk problem. The workaround is to simply disable MWI Subscribe on the Polycom and let Asterisk notify the phone to set the MWI light when appropriate. End result is the same, and actually asterisk knows better than the phone when the MWI should be turned on, so it all works in the end.. |
| 15:28.48 | JerJer | ctooley: crazy...something isn't right |
| 15:29.05 | ctooley | JerJer, I kinda figured that. ;) |
| 15:29.28 | JerJer | ctooley: with your system |
| 15:29.29 | *** join/#asterisk n4y (~tmalkut@ms.orasoft.net.pl) |
| 15:29.29 | DrPete | DannyF_: thanks Does it use sip or iax to connect the the viop provider, is iax better? |
| 15:29.42 | *** part/#asterisk n4y (~tmalkut@ms.orasoft.net.pl) |
| 15:29.52 | JerJer | you will get a biased answer on that question from many in this room |
| 15:29.55 | JerJer | espcially me |
| 15:30.01 | Damin | Hmm.. |
| 15:30.02 | Umaro | anyone germans here at that german voip conference that's going on there? |
| 15:30.09 | ctooley | So, the question remains, WTF is it? |
| 15:30.13 | JerJer | nien |
| 15:30.19 | Damin | Anyone have a "renee.gsm" file from Allison? |
| 15:30.41 | rajo | Umaro: go to #asterisk-drinkers :) |
| 15:30.45 | DannyF_ | DrPete sip I believe (I do anyways) |
| 15:30.54 | DannyF_ | I prefer iax2 |
| 15:31.10 | DannyF_ | but i'm not very religious |
| 15:31.15 | DrPete | lol |
| 15:31.21 | DannyF_ | as JerJer can attest to *grin* |
| 15:31.23 | *** join/#asterisk avidal45679 (~avidal@80.28.67.185) |
| 15:32.10 | ctooley | Damin, Have you seen any problems with Polycom SIP and Asterisk CVS since the 11th? |
| 15:32.29 | DrPete | well, my incomming calls to my box use iax2, but I guess it dont matter who i go with outbound. I see alot give me free numbers, I already have one, so can I just not bother configuring that?> |
| 15:32.45 | Damin | ctooley: I'm running cvs-stable.. |
| 15:33.09 | DannyF_ | DrPete yupp |
| 15:33.10 | Damin | ctooley: And now that I've fixed my MWI problem, it seems to work, but I've only had it working since last night.. |
| 15:33.45 | DannyF_ | Damin was the polycom thread eh? |
| 15:33.50 | ctooley | Damin, I was getting "Line used remotely" on the Polycom after registering to Asterisk, even though Asterisk thinks all is well, when updating to HEAD after 01-11-2005 |
| 15:34.09 | ctooley | Damin, so, stick with stable |
| 15:34.24 | *** join/#asterisk LigHt (~CooL_girl@pD9FF68A0.dip.t-dialin.net) |
| 15:34.38 | Damin | ctooley: Hmm.. I was getting that as well on stable.. but I don't seem to be getting after my CVS update from last night of stable.. |
| 15:35.14 | DrPete | Hmm, sip gate think vat is 16% and its not its 17.5% thats not good :( |
| 15:35.50 | DrPete | Zeeek: so who is the best company you have found |
| 15:38.14 | *** join/#asterisk doughecka_ (~Doug@adsl-18-113-218.sdf.bellsouth.net) |
| 15:40.31 | fafnir | whats some windows software that use the h323 protocol? |
| 15:40.35 | DrPete | If i have multi provider, can i get it to dail though one, and if it fails use another? |
| 15:40.48 | DannyF_ | fafnir: netmeeting? |
| 15:41.05 | DrPete | fafnir: msn too i think |
| 15:41.07 | tzanger | DrPete: of course |
| 15:41.09 | Moc_ | netmeeting use h323v2, wont connect to * |
| 15:41.16 | fafnir | k |
| 15:41.20 | DrPete | tzanger: good stuff, heh |
| 15:41.22 | tzanger | DrPete: you watch DIALSTATUS and HANGUPCAUSE |
| 15:41.24 | DannyF_ | DrPete just saw a thread on the list about that, try a serach |
| 15:41.52 | fafnir | moc., all versions ot just the latest? |
| 15:41.53 | tzanger | DrPete: if you get CHANUNAVAIL you try the next provider |
| 15:41.54 | DrPete | DannyF_: oki, thanks I will check it out. I will be happy when I can choose 1, lol |
| 15:42.13 | Zeeek | DrPete as I have told you several times it's time to start doing a little reading |
| 15:42.15 | Zeeek | The answers to the questions you've been asking are probably here: |
| 15:42.15 | Zeeek | http://www.onlamp.com/pub/a/onlamp/2003/07/03/asterisk.html |
| 15:42.15 | Zeeek | http://iheavy.com/modules.php?op=modload&name=News&file=article&sid=35&mode=thread&order=0&thold=0 |
| 15:42.15 | Zeeek | http://www.onlamp.com/pub/a/onlamp/2004/01/22/asterisk2.html |
| 15:42.15 | Zeeek | http://automated.it/guidetoasterisk.htm |
| 15:42.16 | Zeeek | http://www.asteriskdocs.org |
| 15:42.17 | Zeeek | http://www.asteriskdocs.org/modules/tinycontent/content/docbook/current/docs-html/book1.html |
| 15:42.21 | DannyF_ | DrPete what tzanger said works very very well |
| 15:42.25 | DrPete | tzanger: ahh, makes sense |
| 15:42.57 | DrPete | Zeeek: I have been reading, i like to know others experiennes with companies :) |
| 15:42.58 | *** join/#asterisk tekati (~captain@cpe-66-75-215-63.bak.rr.com) |
| 15:43.15 | DannyF_ | bbl enjoy ya all |
| 15:43.58 | DrPete | is there a way to get irssi to save the buffer, i need all this info |
| 15:45.49 | Zeeek | DrPete I'm referring to the same "can asterisk..." that I've seen for days, which you would know if you had read the docs I refer to, but go ahead as long as there's someone to listen |
| 15:46.59 | Druken | wut can asterisk? |
| 15:47.05 | Druken | i don't see one |
| 15:48.47 | DrPete | Zeeek: heh oh right, i thought you were gonna tell me your best provider so far? Yeah I have been reading as I go, but to be honest i am inpatinet so I usually enquire if it can be done then read how to do it. Thanks for the links i will get though them |
| 15:51.35 | Moc_ | fafnir, why you want h323 ? It the most evil protocol Ive used so far.. |
| 15:51.50 | *** join/#asterisk WizardOne (~wizardone@h236n9c1o848.bredband.skanova.com) |
| 15:52.04 | *** join/#asterisk newmember (~newmember@S010600a0c93dce87.cg.shawcable.net) |
| 15:53.17 | avidal45679 | someone using * into a PPC computer? |
| 15:53.18 | DrPete | anyone use the handytone 486. I am going to get one of the budget phones, but i also need a cordless, so I was thinking of using the handy to connect it. Does it work well? |
| 15:53.20 | Yatta | bjohnson, Wassup? |
| 15:54.11 | Yatta | When i tried to call my * from FWD website i get this error msg. 'Jan 15 10:31:54 WARNING[5017]: app_voicemail.c:1539 leave_voicemail: No entry in voicemail config file for '1000'' |
| 15:54.34 | file[laptop] | Yatta: read the error and you'll figure out what it is |
| 15:55.02 | Yatta | trying to edit voicemail.conf now... lookingon voip-inof fo context meaning |
| 15:58.55 | JerJer | Yatta: if you are using a context other than default you have to do 1000@your_context_name |
| 15:59.13 | Yatta | k |
| 16:00.06 | *** join/#asterisk kietlak (~kietlak@p50858D7B.dip0.t-ipconnect.de) |
| 16:00.13 | FaithX | asterisk goes belly up compiling when it gets to chan_zap.c "deferencing point to incomplete type" ... anyone? |
| 16:04.56 | avidal45679 | hi JerJer.... i have a h323 compile problem "cc1plus: invalid option `arch=ppc'" |
| 16:05.22 | Moc_ | avidal45679, you using a x86 cpu ? |
| 16:05.56 | avidal45679 | not, using a PPC(YDL running into PowerPC Apple) |
| 16:08.11 | Chuji | Moc: No luck on your soundpoint yet? |
| 16:08.25 | Moc_ | soundstation ;) |
| 16:08.28 | Moc_ | my soundpoint work fine |
| 16:08.34 | Chuji | Moc: I want one of those, but I'm letting you test the waters |
| 16:08.38 | Moc_ | I got incoming to work, but that it |
| 16:08.38 | Chuji | Yeah, that is what I meant |
| 16:08.38 | Josh1 | wtf is with the "407 Proxy Authentication Required"? |
| 16:09.20 | *** join/#asterisk r0d3nt|m (~RatMan@4.19.77.194) |
| 16:09.36 | file[laptop] | Josh1: asterisk sends it back for user/pass authentication request |
| 16:09.56 | Josh1 | and i used the right username and passwords |
| 16:10.02 | Josh1 | so what could be the problem? |
| 16:10.16 | file[laptop] | it's your configuration on your device |
| 16:10.51 | Josh1 | hm |
| 16:10.53 | JonR800 | Josh1: it's probably not registering, what device? |
| 16:11.00 | Josh1 | well both of them |
| 16:11.06 | Josh1 | im trying with x-lite and my siemens phone |
| 16:11.21 | Moc_ | atacomm finally came to sence to lower his polycom phone price alittle |
| 16:11.31 | ellvis | i had exactly problem like this and it was network error |
| 16:11.51 | Josh1 | ellvis, what did you do to solve it? |
| 16:12.20 | ellvis | Josh1: i had to change one hub |
| 16:12.55 | Josh1 | well the thing is |
| 16:12.58 | ellvis | Josh1: generally it look like a problem or with registration (check the phone settings) or with the SIP replies |
| 16:12.58 | Josh1 | im on one network |
| 16:13.08 | Josh1 | and my asterisk box is on another in another state |
| 16:13.13 | BoRiS | DrPete: My HT 486 works well... The router portion of it kinda sucks but just make sure you update the firmware to the latest version |
| 16:13.21 | BoRiS | file!!!!!!!!!!!!! |
| 16:13.30 | file[laptop] | BoRiS!!!!!!!!!!!!!!!! |
| 16:13.41 | BoRiS | Whats up? |
| 16:13.42 | Moc_ | you lovers... |
| 16:13.47 | file[laptop] | not too much, you? |
| 16:13.57 | Josh1 | why do you put 3D in front of all of your attributes? |
| 16:14.02 | Josh1 | what purpose does that serve? |
| 16:14.15 | file[laptop] | 3D? |
| 16:14.34 | Josh1 | yeah |
| 16:14.45 | file[laptop] | they're sent as actions versus regular text |
| 16:14.46 | BoRiS | Just waking up |
| 16:14.54 | Josh1 | like secret=3D1234 |
| 16:15.00 | Josh1 | file, would that help? |
| 16:15.09 | file[laptop] | oh you're talking about config |
| 16:15.17 | BoRiS | file: Is that a bad habit to just wake up and *instantly* go to the computer before anything else? |
| 16:15.20 | BoRiS | hehe |
| 16:15.31 | Josh1 | yeah |
| 16:15.32 | Moc_ | Chuji, I wish it had SIP... Really.. |
| 16:15.32 | file[laptop] | Josh1: you don't have to... |
| 16:15.34 | DrPete | BoRiS: Yeah, i was planning to turn that off, as I have a linux box doing that |
| 16:15.47 | file[laptop] | Josh1: in fact if you put them in front of all the attributes, nothing would work |
| 16:15.59 | DrPete | BoRiS: Sounds good thou. I will have to find a place selling them cheap in the UK |
| 16:16.00 | file[laptop] | BoRiS: welllll I do it... I leave my laptop beside my bed for when I awake |
| 16:16.43 | file[laptop] | Moc_: remind me before March 6th and I'll pester the Polycom folks for a SIP image ;) |
| 16:16.51 | BoRiS | File: Is your laptop like your baby? |
| 16:17.00 | Josh1 | see |
| 16:17.02 | Josh1 | i think |
| 16:17.12 | Josh1 | that the * box is seeing my phone as on ITS netowkr |
| 16:17.13 | Josh1 | work* |
| 16:17.13 | Josh1 | Via: SIP/2.0/UDP 192.168.1.95 |
| 16:17.21 | file[laptop] | Josh1: so put nat=yes |
| 16:17.25 | Josh1 | alright let me try |
| 16:17.29 | file[laptop] | do a sip reload, and see what happens |
| 16:18.00 | BoRiS | Doesn't Polycom give out their SIP firmwares or is it like cisco where you have to PAY for it seperately? |
| 16:18.26 | Moc_ | BoRiS, you need to ask your reseller |
| 16:18.33 | BoRiS | Ugh |
| 16:18.40 | Moc_ | but no liscence is need |
| 16:18.42 | JonR800 | It's stupid. |
| 16:18.49 | BoRiS | That is really stupid |
| 16:18.57 | Josh1 | nope still asking for Proxy Authentication on 192.168.1.95 |
| 16:19.21 | BoRiS | Why don't manufacturers just put their firmwares on their website. Plain and simple |
| 16:19.46 | file[laptop] | Josh1: register or call? pastebin your sip.conf too |
| 16:19.54 | Josh1 | ok |
| 16:19.56 | JonR800 | BoRiS: it probably cuts down on support calls. |
| 16:19.59 | Josh1 | file its acall out |
| 16:20.10 | file[laptop] | yeah I figured it was a call |
| 16:20.14 | Moc_ | the thing is you can't call polycom for support |
| 16:20.26 | Moc_ | polycom phone are sold and supported by the reseller |
| 16:20.39 | JonR800 | ahh |
| 16:20.48 | BoRiS | Sounds kinda like sipura |
| 16:21.39 | BoRiS | Moc: Does that mean don't buy a Polycom? :-p |
| 16:21.56 | Josh1 | http://pastebin.com/229203 |
| 16:21.58 | Josh1 | thats my sip.conf |
| 16:22.09 | *** join/#asterisk randu (~randu@207.144.31.103) |
| 16:22.57 | JonR800 | lol.. well if we were judging by support, then don't buy anything. |
| 16:22.57 | Moc_ | BoRiS, that mean buy a polycom from a * friendly reseller, .. OR you need more friends ;) |
| 16:22.57 | file[laptop] | you do realize you have no codecs there? |
| 16:22.57 | Moc_ | Well if you have a better option than polycom, please let me know.. |
| 16:22.57 | BoRiS | Moc: lol!!! |
| 16:22.57 | Torq | http://pastebin.ca/4237 - why why why? driving me nuts. can anyone see anything? |
| 16:23.00 | randu | What does this mean: Rejected connect attempt from <ipaddress>, request '393-612@outgoing' does not exist |
| 16:23.00 | Moc_ | I love Polycom phone... until I find a better one |
| 16:23.13 | Josh1 | file, but that shouldnt be the cause of the proxy authentication error should it? |
| 16:23.43 | file[laptop] | Josh1: no I'm just saying, take out the defaultip... insecure... username... extension... and regexten |
| 16:23.50 | Moc_ | my progress on finding the best phone is. BT101->Cisco 7960->Polycom IP500->Polycom IP600 |
| 16:23.54 | file[laptop] | randu: as for you - you didn't say hi, and the error is right there |
| 16:23.56 | Josh1 | ok |
| 16:24.07 | BoRiS | lol file! |
| 16:24.52 | file[laptop] | I refuse to understand the error message for you when it's plain and simple |
| 16:24.54 | BoRiS | randu: Someone is trying to call you but in in your dialplan, you dont have a 393-612 dial plan in your outgoing context |
| 16:25.03 | randu | lol sorry I was rude :-) Hi from the beautiful cost of Shallotte NC :-) |
| 16:25.06 | Torq | updated pastebin with config bits http://pastebin.ca/4238 |
| 16:25.24 | Josh1 | SIP/2.0 487 Request Terminated |
| 16:25.24 | Josh1 | Via: SIP/2.0/UDP 192.168.1.95;received=65.8.161.173;rport=50021 |
| 16:25.25 | *** join/#asterisk blaisen1 (~blaisen1@tightcode.ofpower.net) |
| 16:25.25 | Moc_ | I never care to look at snom because they are expensive, and doesn't look that great. Also Im sure handfree is not as good, display is not that great. But they have a console phone... |
| 16:25.26 | file[laptop] | Torq: your username is the part in the [ ] |
| 16:25.36 | Josh1 | and before that i still got the proxy authentication required |
| 16:25.40 | randu | Boris: Thanks I will do some research on that. |
| 16:25.45 | blaisen1 | argh... what gives with iax DID providers... livevoip and iax.cc keep changing their offerings/rates |
| 16:25.46 | file[laptop] | Josh1: of course you'll get the proxy authentication required |
| 16:25.54 | Josh1 | why? |
| 16:25.58 | file[laptop] | Josh1: asterisk challenges for the username/password |
| 16:25.59 | Torq | file: my username is 12345???!? |
| 16:26.03 | file[laptop] | Torq: yes |
| 16:26.14 | Torq | file: even though i said username=xlite1 |
| 16:26.17 | file[laptop] | Josh1: your device then sends another invite with the authentication credentials in it |
| 16:26.19 | Torq | oh for heavens sake... |
| 16:26.22 | file[laptop] | Torq: yes, that's used for calling your device |
| 16:26.27 | Josh1 | file, right.. but my phone specifies it |
| 16:26.33 | Josh1 | and sends it out |
| 16:26.43 | file[laptop] | Josh1: is there an Authorization header in the first INVITE? |
| 16:27.17 | file[laptop] | because if not, you're wrong - and it's not sending it in the first INVITE so asterisk challenges it for it |
| 16:27.22 | BoRiS | Moc_: Have you tried the SPA-841? Very cute and nice looking phone |
| 16:28.04 | *** join/#asterisk kpfleming (~chatzilla@ip68-3-230-141.ph.ph.cox.net) |
| 16:28.47 | Torq | http://pastebin.ca/4239 is a new error. much more happens now, but it isnt working |
| 16:29.15 | *** join/#asterisk oej (~oej@64-151-42-78-dhcp-kc.everestkc.net) |
| 16:29.41 | file[laptop] | Torq: okay skim down that sip friend, take away regexten - username - and insecure |
| 16:29.54 | file[laptop] | do a sip reload, then the X-Lite should register |
| 16:30.19 | file[laptop] | cause I've got an X-Lite right here and it's peachy |
| 16:30.19 | JonR800 | what's up with everyone using insecure? hah |
| 16:30.36 | file[laptop] | I don't know |
| 16:30.52 | Moc_ | BoRiS, last report I heard on them is they are cheap like a budgetone |
| 16:30.54 | file[laptop] | don't try to make some big elaborate friend entry, just the basics... |
| 16:31.31 | file[laptop] | I should go make food |
| 16:31.32 | Josh1 | http://pastebin.com/229207 |
| 16:31.45 | Torq | waiting for it to try to register |
| 16:32.02 | Torq | hmm, still forbidden |
| 16:32.34 | Moc_ | get some freezed food in the oven .. |
| 16:32.35 | file[laptop] | Josh1: now setup your codecs |
| 16:32.39 | BoRiS | Moc_: I dont think anyone will make the same mistake to make a phone like budgetone :-p |
| 16:32.39 | Josh1 | alright |
| 16:32.45 | Silik0n | freezed food? |
| 16:32.47 | Josh1 | allow=ulaw right? |
| 16:32.50 | file[laptop] | disallow=all |
| 16:32.51 | file[laptop] | allow=ulaw |
| 16:32.56 | Silik0n | is that anything like frozen food? |
| 16:32.56 | Josh1 | in [1000]? |
| 16:32.57 | file[laptop] | in [general] and in the friend entry |
| 16:32.59 | Josh1 | alright |
| 16:33.08 | file[laptop] | and that's all I'm helping with |
| 16:33.12 | Torq | yes yes yes! it works! after 5 hours!!! |
| 16:33.12 | *** join/#asterisk mhnoyes (~mhnoyes@user-38lc189.dialup.mindspring.com) |
| 16:33.12 | file[laptop] | BoRiS: you're taking over |
| 16:33.30 | BoRiS | setting up a SIP account? |
| 16:33.35 | Torq | file: I want your babies |
| 16:33.44 | BoRiS | LOL!!!!! |
| 16:33.50 | file[laptop] | O.o |
| 16:33.50 | BoRiS | Have 'em |
| 16:33.54 | file[laptop] | Paypal is welcome. :p |
| 16:33.54 | BoRiS | hehe |
| 16:34.06 | file[laptop] | for the "File going to VON shopping account" |
| 16:34.31 | *** join/#asterisk bprice20 (~bprice20@alb-69-200-26-118.nycap.rr.com) |
| 16:34.50 | bprice20 | mornin all |
| 16:36.15 | file[laptop] | BoRiS: bkw and I will be on the same flight from Denver to San Jose ;) |
| 16:37.45 | Josh1 | file[laptop], |
| 16:37.46 | Josh1 | Jan 15 11:37:32 NOTICE[17195]: chan_sip.c:7379 handle_request: Unable to create/find channel |
| 16:37.46 | Josh1 | Jan 15 11:37:38 WARNING[17195]: chan_sip.c:695 retrans_pkt: Maximum retries exceeded on call b304dd7-1d4c1e69-2bc6dad@192.168.1.95 for seqno 27577 (Non-critical Response) |
| 16:37.57 | Moc_ | I wish I could go too |
| 16:39.28 | BoRiS | file: Both of you will have alot of fun! |
| 16:40.47 | file[laptop] | no foooooooooooood no foooooooooooood |
| 16:41.00 | kpfleming | Moc_: that "sip notify" thing for Polycom phones is a wonderful thing :-) |
| 16:41.19 | Silik0n | *yawn* |
| 16:41.36 | Josh1 | file[laptop], any ideas? |
| 16:41.45 | file[laptop] | ask BoRiS |
| 16:42.12 | Moc_ | kpfleming, yea, it great in deed |
| 16:42.19 | Josh1 | BoRiS, any ideas? :) |
| 16:42.39 | Moc_ | too bad it dont work with cisco phone |
| 16:42.55 | *** join/#asterisk [Latre] (~latre@148.233.19.133) |
| 16:43.10 | kpfleming | hopefully someone will figure out a way to make it work... i may spend some more time on it |
| 16:43.14 | Moc_ | I tryed to hack it to send the 'correct' cisco string, but it never worked... cisco use some crazy notify command.. |
| 16:43.30 | Moc_ | I love polycom ;) |
| 16:43.39 | Moc_ | it simple ... |
| 16:43.47 | Moc_ | and effective .. anyway hehe |
| 16:43.57 | kpfleming | i think the problem may be that the NOTIFY is being sent as part of an existing call (same branch/tag/CSeq), and the Cisco phones don't like that, so if the NOTIFY was sent with a new set of branch/tag/CSeq it might work |
| 16:43.59 | *** join/#asterisk zwi (~zwi@216.88.131.43) |
| 16:44.37 | Moc_ | yes your right, it the part that I wasnt beable to do.. |
| 16:45.10 | Moc_ | I tryed to hack the CSeq but .. ehehe no luck |
| 16:45.11 | Merlin | Moc: what codec do you use for the polycom? |
| 16:45.31 | Moc_ | I got bandwidth, so I use ulaw |
| 16:45.54 | Merlin | i see |
| 16:46.29 | kpfleming | i use only G.729 with them works great |
| 16:47.10 | kpfleming | unlike Cisco, Polycom phones will even 3-way conference using G.729 |
| 16:47.19 | Moc_ | yes I know, it great.. |
| 16:47.25 | wolfson | uyyu |
| 16:47.28 | Moc_ | you saw that you can have multiple call on hold too |
| 16:47.43 | Moc_ | cisco is limited to 2 call max I beleive |
| 16:47.51 | Torq | whats the number for the asterisk demo? |
| 16:48.06 | *** part/#asterisk avidal45679 (~avidal@80.28.67.185) |
| 16:48.17 | bprice20 | I take the time to learn about iaxtel and set it up only to learn it doesn't work |
| 16:48.18 | bprice20 | nice |
| 16:48.36 | Moc_ | I havent use iaxtel for a while |
| 16:48.44 | bprice20 | fwd is a little better but can't dial the 1800 number i need it to dial for testing |
| 16:48.47 | Moc_ | you can look at fwd |
| 16:48.58 | bprice20 | looking for a decent iax provider |
| 16:49.07 | Merlin | kpfleming: i not interested in paying some vendor too much for their codec |
| 16:49.08 | Moc_ | bprice20, get a account with nufone |
| 16:49.30 | Moc_ | 10$ for * liscence is not that bad |
| 16:49.41 | Moc_ | it active calls too.. |
| 16:49.43 | Merlin | i thought it was $10 per station |
| 16:49.45 | *** join/#asterisk DevilFish (DevilFish@216.163.40.2) |
| 16:49.47 | kpfleming | Merlin: when the hard pones support something other than G.729 we'll switch, but for now it's easy and cheap |
| 16:49.48 | czero | <PROTECTED> |
| 16:49.55 | *** join/#asterisk gbs (~GBS@adsl-5062c3cb.monradsl.monornet.hu) |
| 16:50.06 | Merlin | kpfleming: i hear ya |
| 16:50.09 | Moc_ | Merlin, no, it 10$ per active decoding/encoding channel |
| 16:50.27 | Merlin | Moc: so with a company of 55 phones, you might need 35 licenses? |
| 16:50.28 | Merlin | something like that? |
| 16:50.42 | Moc_ | Merlin, well if you have 35 person on the phone at the same time yes |
| 16:50.48 | Merlin | wow |
| 16:50.53 | Merlin | i totally misunderstood their pricing |
| 16:50.53 | Merlin | thank you |
| 16:50.55 | Moc_ | but generally, for 55 phon,e you dont have that much people on the phone |
| 16:51.24 | jaiger | unless you're a call center |
| 16:51.30 | Moc_ | ofcourse |
| 16:51.44 | DevilFish | anyone ever see a line like this in asterisk debug? |
| 16:51.44 | DevilFish | Jan 14 12:53:00 DEBUG[4290]: Bridge stops because we're zombie or need a soft hangup: c0=SIP/302-ec8e, c1=SIP/sip_proxy-out-f81c, flags: No,No,No,Yes |
| 16:53.00 | kpfleming | Merlin: you only need licenses for * for the number of channels that you have to outside your PBX, everything else (internal) would just be pass-through and wouldn't require licenses on * since it would not have to be transcoded... an office with 55 phones that has 18 PSTN lines would only need 18 licenses |
| 16:53.15 | Moc_ | yes that too |
| 16:53.32 | Merlin | oh wow |
| 16:53.35 | Merlin | that's even better |
| 16:53.40 | *** join/#asterisk Nix (~Nix@p50858D7B.dip0.t-ipconnect.de) |
| 16:53.42 | Moc_ | * will passthrought if both end use g729 and * dont need to be in the middle of the call |
| 16:53.42 | *** join/#asterisk florz_ (nobody@I857a.i.pppool.de) |
| 16:53.59 | jaiger | how about voicemail and other asterisk features that may transcode? |
| 16:54.02 | Merlin | so you're almost paying for the decoing? |
| 16:54.03 | kpfleming | there are other things though... if you want to record internal calls, or use Meetme on internal calls, then you will need licenses for those channels too |
| 16:54.26 | kpfleming | jaiger: you set up voicemail and all your sound files to be available in ulaw/alaw _and_ G.729, then it never needs to transcode |
| 16:54.28 | jaiger | Merlin: you are paying for the decoding |
| 16:54.34 | Merlin | jaiger: that makes sense |
| 16:54.38 | jaiger | kpfleming: ahh |
| 16:54.54 | jaiger | so you pre-transcode the files |
| 16:54.57 | kpfleming | yep |
| 16:55.12 | *** join/#asterisk dizzydiffi (~diseyi@adsl-70-240-183-99.dsl.hstntx.swbell.net) |
| 16:55.18 | kpfleming | when voicemail is recording, it will use a license to do that, but then it doesn't need one for playback |
| 16:55.30 | Merlin | hh |
| 16:55.31 | Merlin | ahhh |
| 16:55.33 | Merlin | makes sense |
| 16:56.15 | jaiger | what happens if you don't have enough licenses? does it fail gracefully? |
| 16:56.30 | Moc_ | Merlin, if you wish, you could convert your gsm file into g729 file, for voicemail |
| 16:56.51 | Moc_ | or other * sounds file you use offen from g729 phone.. |
| 16:57.02 | kpfleming | jaiger: no, not gracefully, it can't do what it needs to do if there are no licenses |
| 16:57.02 | Merlin | Moc: and that will save licenses, right? |
| 16:57.06 | Moc_ | yes |
| 16:57.17 | Moc_ | since it will do passthrought from the g729 file to you |
| 16:57.18 | tclark | way ot but what are ppl fav muti-media players for linux |
| 16:57.30 | Moc_ | my fav is vlc |
| 16:58.24 | *** join/#asterisk ChrisP (~me@pcp04634037pcs.gambrl01.md.comcast.net) |
| 16:58.33 | Moc_ | kpfleming, it nice isnt it ? ;) |
| 16:58.39 | kpfleming | way, way cool |
| 16:58.48 | DevilFish | kpfleming: are you having any rouble with your polycoms? |
| 16:58.53 | Moc_ | I love beable to update config remotly |
| 16:59.11 | kpfleming | DevilFish: occasionally they get "confused", but generally they work very well |
| 16:59.26 | DevilFish | mine can't hold a call for more than 1hr |
| 16:59.39 | kpfleming | i have done two hour conf calls with an IP500 a few times |
| 16:59.46 | Torq | when i dial 600 from xlite, it is now giving me busy tone. isnt there supposed to be an echo test there? |
| 16:59.52 | Moc_ | DevilFish, some VoIP provider cut call after 1 hours .. ;) |
| 17:00.03 | Moc_ | I know, I freeked out when it happen 3 time to me.. |
| 17:00.06 | DevilFish | I am the VoIP provider :) |
| 17:00.11 | DevilFish | thats the prob |
| 17:00.18 | ChrisP | Stupid issue - I'm trying to test and show asterisk to my boss, and the demo I'm putting together, firefly on his laptop to my linux asterisk server sounds horrible |
| 17:00.35 | *** join/#asterisk ManxPower (~eric@dsl-208-164-150-160.datasync.com) |
| 17:00.37 | DevilFish | I get this crap after 1 hour |
| 17:00.38 | DevilFish | Jan 14 12:53:00 DEBUG[4290]: Bridge stops because we're zombie or need a soft hangup: c0=SIP/302-ec8e, c1=SIP/sip_proxy-out-f81c, flags: No,No,No,Yes |
| 17:00.41 | ChrisP | All the recorded messages come across really crackely |
| 17:00.42 | ManxPower | Useful Asterisk Docs (BOOKMARK THEM!): http://www.digium.com/index.php?menu=documentation (look at the "Unofficial Links") and http://www.voip-info.org/wiki-Asterisk and http://www.fnords.org/~eric/asterisk/ (my site) and http://asteriskdocs.org/ |
| 17:00.45 | ManxPower | To search the Asterisk mailing list archive go to www.google.com and put site:lists.digium.com in addition to your other query terms. |
| 17:00.48 | bprice20 | ChrisP get a iaxy or a sip phone |
| 17:00.52 | bprice20 | it impresses them |
| 17:01.19 | ChrisP | Good idea!... just worried about the sound quality right now |
| 17:01.34 | Moc_ | DevilFish, hehe |
| 17:01.36 | ChrisP | has anyone had similar problems? |
| 17:01.40 | DevilFish | only happens with my polycoms, xlite is fine |
| 17:01.42 | Moc_ | never saw that |
| 17:01.57 | Moc_ | what firmware do you use ? |
| 17:02.00 | festr_ | any new information about jitter for sip? |
| 17:02.02 | DevilFish | 1.3.4 |
| 17:02.13 | DevilFish | latest cvs as of yesterday |
| 17:02.23 | jaiger | DevilFish: I've only been on my ip500 for under 1 hr so far |
| 17:02.27 | DevilFish | I'm trying anything, I really want to use these phones |
| 17:03.06 | ManxPower | DevilFish: CVS stable or CVS-HEAD? |
| 17:03.15 | DevilFish | I'm getting worried a little cause I haven't spoke to anyone whos seen that debug message or been on a polycom SIP for more than 1:30 |
| 17:03.21 | DevilFish | cvs stable |
| 17:03.26 | ellvis | bye all |
| 17:03.26 | ManxPower | DevilFish: I assume you've been following the polycom thread on asterisk-dev? |
| 17:03.36 | DevilFish | I have noe |
| 17:03.42 | DevilFish | I have not... can you point me? |
| 17:03.46 | Moc_ | DevilFish, Ive being in conference for HOURS with this polycom phone |
| 17:03.48 | ManxPower | too bad. It might have fixed your problem |
| 17:04.04 | Moc_ | I event forgot to hangup onetime, and it was up to 10 hours of active call on bkw conference |
| 17:04.06 | DevilFish | ok, I'm listening..... |
| 17:04.36 | ManxPower | Subject: Re: [Asterisk-Dev] chan_sip cvs-stable post 1/1/2005 polycom bug |
| 17:04.39 | DevilFish | my IP600 will hold a call inside of asterisk forever, its when I call the PSTN that it croaks after 1 hr |
| 17:04.50 | Moc_ | your using stable ??? bad boy .. |
| 17:05.00 | ManxPower | Moc: Hush you! |
| 17:05.04 | Moc_ | ;) |
| 17:05.06 | ManxPower | stable = good |
| 17:05.07 | DevilFish | everyone said use stable |
| 17:05.12 | DevilFish | thats what I thought |
| 17:05.26 | jaiger | DevilFish: I had problems with pstn (channel bank fxo) at 1m mark but rxgain tweaks seemed to fix it |
| 17:05.29 | ManxPower | DevilFish: Unless you NEED a feature in -HEAD, use 1.0.x |
| 17:05.55 | ManxPower | DevilFish: If you can call internally fine then the discussion I'm thinking of doesn't apply to you. |
| 17:06.04 | *** join/#asterisk zigman (~zigman@irc.zigman.de) |
| 17:06.06 | bonbon-home | I've compiled zaptel on a box but no zttool binary is created? Anyone know why? |
| 17:06.16 | ManxPower | DevilFish: Document the problem, post to the mailing list |
| 17:06.20 | Moc_ | I keep seeing people with problem, but finally it turn out they use stable !! |
| 17:06.57 | DevilFish | yeah I'm gonna have to post I think cause I cannot seem to resolve |
| 17:07.07 | czero | anyone want to come pack my apt for me :) |
| 17:07.07 | DevilFish | this happens with 1.0.3 and cvs stable |
| 17:07.19 | ManxPower | DevilFish: Eventually uou may have to file a bug report. |
| 17:07.48 | DevilFish | I'm gonna have to dig deeper I guess... dang it |
| 17:08.01 | DevilFish | they all can't be simple config probs I suppose :) |
| 17:08.33 | DevilFish | I'll see if I can post this weekend I guess |
| 17:08.55 | Corydon76-home | bonbon-home: no libnewt-devel installed? |
| 17:09.01 | randu | Hello Everyone! What do I need to look at for this error: No Channel type registered for 'IAX' Unable to create channnel of type 'IAX' |
| 17:09.13 | ManxPower | randu: You need to use IAX2 |
| 17:09.55 | *** join/#asterisk cRaZyGirL`kafayi (~RaKam@n219079036113.netvigator.com) |
| 17:10.12 | *** join/#asterisk Joshipoo (~Joshwah@adsl-8-175-99.mia.bellsouth.net) |
| 17:10.14 | *** join/#asterisk adker (~adker@70-97-139-119.dsl1.glv.ny.frontiernet.net) |
| 17:10.17 | Joshipoo | :P |
| 17:10.19 | *** join/#asterisk datareactor (datareacto@203.81.214.119) |
| 17:10.32 | bonbon-home | Corydon: yes I think that's the problem. I'm using gentoo, and the emerge doesn't pick it up |
| 17:10.35 | randu | ManxPower: is that I type I made do you think or do I have to go edit a config file? |
| 17:11.07 | ManxPower | randu: You are prolly doing a Dial(IAX/blah and you need to Dial(IAX2 |
| 17:11.18 | randu | ok thanks. |
| 17:11.40 | *** join/#asterisk aminorex (~tony@c66.191.69.132.dul.mn.charter.com) |
| 17:11.41 | bonbon-home | Corydon: any idea what libnewt is called under gentoo? |
| 17:12.43 | czero | errr why dont real estate agnet put pics on line |
| 17:13.25 | *** part/#asterisk newmember (~newmember@S010600a0c93dce87.cg.shawcable.net) |
| 17:14.16 | file[laptop] | OMG BECKY |
| 17:14.54 | ManxPower | czero: Uh, they do. |
| 17:14.59 | randu | ManxPower: yep that was it. thanks! |
| 17:15.07 | czero | sorry, I mean why don;t they for ALL properties |
| 17:15.12 | czero | ) |
| 17:15.13 | czero | :) |
| 17:15.32 | PTG123 | wow czero is awake |
| 17:15.35 | *** join/#asterisk yEsiL (~Sa[Go]Pa@81.212.12.168) |
| 17:15.44 | PTG123 | in az they do |
| 17:15.58 | czero | awake is a relative term preston :) |
| 17:16.09 | blitzrage | afternoon all |
| 17:16.15 | czero | hey blitzrage |
| 17:16.18 | PTG123 | the real estate agents put everything in MLS so other agents can see it |
| 17:16.25 | PTG123 | but the consume can go to a special site and see it as well |
| 17:16.27 | czero | they usully do here too |
| 17:16.33 | czero | but aLOT don't still |
| 17:16.42 | PTG123 | I don't know of any that don't here |
| 17:16.42 | datareactor | hi blitzrage |
| 17:16.46 | *** join/#asterisk Josh1 (~Joshwah@adsl-8-175-108.mia.bellsouth.net) |
| 17:16.47 | PTG123 | you just pay you monthly dues |
| 17:16.50 | PTG123 | and MLS does it for you |
| 17:16.50 | czero | and less then 1/f of the montreal listings are english |
| 17:16.51 | ManxPower | czero: As far as I can tell they only do it for properties they are actually INTERESTED in selling. |
| 17:16.53 | PTG123 | they agent does nothing |
| 17:16.55 | PTG123 | but answer the phone |
| 17:16.58 | czero | and it hurts to read french this time o f day |
| 17:17.02 | PTG123 | which is amazing that they make like $10k for that |
| 17:17.03 | ManxPower | PTG123: Don't get me started about the MLS. |
| 17:17.14 | ManxPower | The Louisiana MLS uses ActiveX for everything |
| 17:17.15 | PTG123 | its such a big scam |
| 17:17.21 | PTG123 | i will never use an agent to sell my house |
| 17:17.31 | datareactor | can i use go2call with asterisk |
| 17:17.33 | PTG123 | or an agent to buy a house |
| 17:17.45 | *** join/#asterisk ardor (~ardorgof@ip68-227-38-164.lv.lv.cox.net) |
| 17:17.52 | ManxPower | datareactor: Does it support a protocol Asterisk supports? |
| 17:17.53 | czero | yeah u can ge tmls listings cheep |
| 17:18.02 | ardor | Hi, how is everyone doing. |
| 17:18.14 | datareactor | ManxPower i dont know :( |
| 17:18.37 | ManxPower | datareactor: Seems to me that you need to find out before asking here. |
| 17:18.43 | DevilFish | ManxPower: so is asterisk-dev where I'd want to be posting? |
| 17:19.06 | ManxPower | DevilFish: Try -users first confirm it's not some oddball config problem |
| 17:19.12 | DevilFish | gotcha |
| 17:19.13 | DevilFish | thanks |
| 17:19.27 | datareactor | Manxpower thats y i am asking here because i dont know |
| 17:19.42 | *** join/#asterisk Darwin_35 (~darwin35@c-24-3-241-22.client.comcast.net) |
| 17:20.39 | datareactor | may be someone already is using it |
| 17:20.46 | *** join/#asterisk ToyMan (~stuq@user-12lcqq2.cable.mindspring.com) |
| 17:21.12 | randu | Hello Everyone. I see there are a bunch of asterisk config gui's. Does anyone have any recommendations for Newbie's like myself? |
| 17:21.12 | blitzrage | ManxPower: want some pictures of Toronto? |
| 17:21.18 | Nugget | randu: vim. |
| 17:21.32 | ManxPower | blitzrage: I have 200 pics of Toronto. |
| 17:21.46 | blitzrage | randu: you're still going to have to learn the dialplans... none of the GUI's (that I'm aware of) really let you get away from that. |
| 17:21.50 | ManxPower | But if they are very naughty pics I might be interested. |
| 17:22.00 | PTG123 | czero: your moving to canada, i want to move to a tropical island :) |
| 17:22.00 | blitzrage | well... some pictures of a large shaft... |
| 17:22.07 | randu | oh ok. |
| 17:22.09 | blitzrage | ManxPower: but its the CN Tower :) |
| 17:22.10 | czero | already in canada |
| 17:22.13 | datareactor | randu i have tried ACTOS it is quite easy |
| 17:22.14 | ManxPower | CN Tower? I have pics of those too. |
| 17:22.20 | czero | I look ou tmy widwo and see the CNtowe |
| 17:22.22 | ManxPower | Even one looking down thru the glass floor |
| 17:22.23 | czero | tower |
| 17:22.30 | blitzrage | ManxPower: but you don't have the panoramic pictures I took with my camera... or the view from my balcony :) |
| 17:22.32 | randu | datareactor: Thanks |
| 17:22.42 | Darwin_35 | ManX did it I saw him |
| 17:22.46 | ManxPower | blitzrage: Perhaps when I get my greencard. |
| 17:22.54 | blitzrage | ManxPower: hehe |
| 17:26.34 | *** join/#asterisk thefallen_ (PolarBear@thefallen.user) |
| 17:29.54 | datareactor | today i have test WISIP IP mobile phones the voice quality is very good |
| 17:30.23 | EvlHimeko | are they full roaming phones? |
| 17:30.53 | BoRiS | datareactor: I was on my Senao 7800 wifi phone and some people said I sounded like on a cell phone. Good but still kinda cell phone like |
| 17:31.07 | datareactor | EVlHImeko they work on 2.4 Ghz and i am using Dlink access point |
| 17:31.22 | file[laptop] | BoRiS: you sounded fine to me |
| 17:31.28 | file[laptop] | BoRiS: does that thing really do G726? |
| 17:31.39 | EvlHimeko | but if you move to a different ap will it attached on it's own |
| 17:31.48 | datareactor | the sound quality even better than grandstream |
| 17:31.53 | ardor | If i have exten => h,1 and exten => h,2 They should both run when someone hangs up right? |
| 17:32.20 | blitzrage | BoRiS: you got the Senao? How do you like it? |
| 17:32.59 | ManxPower | ardor: In theory |
| 17:33.06 | ardor | damn it cuz its not working |
| 17:33.11 | Shido6 | ? |
| 17:33.45 | datareactor | Boris how was the sound quality i was using g711 , ilbc |
| 17:35.05 | ManxPower | ardor: I believe that h is only run when destination hangs up, not the source |
| 17:35.55 | *** join/#asterisk glLoadIdentity (~tuyan@dsl81-214-9283.adsl.ttnet.net.tr) |
| 17:36.16 | randu | Hello again! anyone ever get this? Call Rejected by 69.39.205.121: No Authority Found i am dealing with freeworld dialup at the moment |
| 17:36.28 | *** part/#asterisk glLoadIdentity (~tuyan@dsl81-214-9283.adsl.ttnet.net.tr) |
| 17:36.36 | *** part/#asterisk twisted (~twisted@twisted.active.supporter.pdpc) |
| 17:36.36 | *** join/#asterisk twisted (~twisted@twisted.active.supporter.pdpc) |
| 17:36.36 | *** mode/#asterisk [+o twisted] by ChanServ |
| 17:37.06 | ManxPower | randu: That means that user/pass/context triad in sip|ias.conf does not match what wfd is sending |
| 17:37.59 | ardor | ManxPower: Ya both h,1 and h,2 run if i let it time out. (no time out rule set) |
| 17:38.11 | ardor | but when I disconnect just h,1 runs |
| 17:38.17 | Shido6 | what does h1 do? |
| 17:38.24 | Shido6 | show me the exten |
| 17:38.32 | *** join/#asterisk Josh1 (~Joshwah@adsl-8-175-108.mia.bellsouth.net) |
| 17:38.35 | Shido6 | exten => h,1,..... |
| 17:38.36 | Josh1 | hey guys |
| 17:38.36 | Shido6 | ? |
| 17:38.38 | Josh1 | im getting this |
| 17:38.38 | ardor | exten => h,1,agi,call_log_end.agi|${PS_NAME}${PS_ID}:${UNIQUEID}:${CALL_TYPE}:Disconnected:${RECORD_FLAG}:${LOGSTEP} |
| 17:38.39 | Josh1 | Jan 15 12:38:08 WARNING[17195]: chan_sip.c:612 __sip_xmit: sip_xmit of 0x8140e9c (len 436) to 65.8.175.108 returned -1: Invalid argument |
| 17:38.54 | Shido6 | whats in call_log_end.agi ? |
| 17:39.02 | randu | ManxPower you meant iax.conf not ias.conf right? |
| 17:39.10 | ManxPower | randu: Yeah. |
| 17:39.13 | randu | ok |
| 17:39.15 | ardor | mysql commands that shit the logs into mysql. |
| 17:39.16 | randu | thanks |
| 17:39.35 | datareactor | :) |
| 17:39.39 | DevilFish | ManPower: could defaultexpirey=3600 in my sip.conf be killing my phone after one hour? |
| 17:39.53 | ManxPower | DevilFish: Yes. |
| 17:39.58 | DevilFish | what the hell then |
| 17:39.59 | ManxPower | why do you have that??? |
| 17:40.13 | Josh1 | anyone know what that means? |
| 17:40.20 | DevilFish | from looking at other sip configs |
| 17:40.22 | DevilFish | I guess |
| 17:40.23 | ManxPower | Josh1: It means something terrible happened. |
| 17:40.26 | Josh1 | heh |
| 17:40.29 | Josh1 | i kinda figured that |
| 17:40.39 | datareactor | :) |
| 17:40.42 | DevilFish | what should default expiry be set gor? |
| 17:40.50 | DevilFish | gor = for oops |
| 17:41.13 | datareactor | DevilFish i no need of default expiry |
| 17:41.16 | *** join/#asterisk Fabe (~Fabe@p50858D7B.dip0.t-ipconnect.de) |
| 17:41.17 | ManxPower | DevilFish: Either learn to use as few settings as you can or be prepared for a life of misery and pain. |
| 17:41.29 | DevilFish | yeah I can see that |
| 17:41.31 | file[laptop] | oooooooooooh pain |
| 17:41.37 | file[laptop] | ooooooooh Swiss Chalet |
| 17:41.41 | *** join/#asterisk nullogic (~nullogic@66.240.64.121) |
| 17:41.43 | DevilFish | leeme try with out defaut expiry then |
| 17:41.44 | ManxPower | most of my SIP entries have like 5 items |
| 17:41.56 | ManxPower | and for testing I use as little as three lines |
| 17:42.03 | ManxPower | username, secret, context |
| 17:42.22 | Shido6 | yep |
| 17:42.31 | DevilFish | well deaultexpiry is in the [general] section of sip.conf |
| 17:42.35 | Shido6 | and encryption method |
| 17:42.36 | ManxPower | maybe a disallow=all and allow=ulaw |
| 17:42.52 | ManxPower | DevilFish: In [general] I have VERY few options as well |
| 17:43.05 | Josh1 | ugh im so confused |
| 17:43.22 | ManxPower | Josh1: Can you ping the remote device. Is the remote device behind a firewall? |
| 17:43.31 | Josh1 | yes |
| 17:43.37 | DevilFish | okee dokee then, gonna try with out, I thought my asterisk box had to maintain registry with the voip switch ever hour or so |
| 17:43.38 | Josh1 | and no |
| 17:43.45 | Josh1 | i can ping it |
| 17:43.50 | Josh1 | but they arent behind firewalls |
| 17:44.05 | ManxPower | DevilFish: The defaults are correct for most people |
| 17:44.16 | Josh1 | i was able to call the phone before, but something happened to my sip.conf and i cant remember what i did |
| 17:45.22 | datareactor | Joshi is agi script is in place |
| 17:45.36 | Josh1 | hmm? |
| 17:45.41 | Josh1 | it should be |
| 17:45.44 | DevilFish | been playing with my configs for so damned long now, I forgot what a default one looks like, I'm gonna whittle this sucker down like you said and test again b4 I post then |
| 17:51.20 | Josh1 | anyone? |
| 17:53.43 | *** join/#asterisk wankel (nobody@ohno.mrbill.net) |
| 17:53.47 | ManxPower | The default one is in asterisk/configs/sip.conf.sample |
| 17:54.31 | *** join/#asterisk [TripleF] (kvirc@Quebec-HSE-ppp225196.qc.sympatico.ca) |
| 17:55.20 | datareactor | here site for wisip http://voipstore.pulver.com/product_info.php?products_id=35&osCsid=b9d9ea7934c1bab136ff9309d9c3439b |
| 17:56.11 | ManxPower | Josh1: Does "sip show peers" show the phone's IP address? |
| 17:57.12 | Josh1 | yes |
| 17:57.17 | Josh1 | it says unreachable, but i dont know why |
| 17:57.56 | Josh1 | there we go |
| 17:58.15 | Josh1 | the phone is connected it seems |
| 17:58.23 | Josh1 | but, now i cant dial in or out |
| 17:58.42 | ManxPower | Josh1: If it's UNREACHABLE then the phone is not responding to Asterisk |
| 17:58.49 | Josh1 | its not now |
| 17:58.53 | Josh1 | its "Unmonitored" |
| 17:58.59 | ManxPower | If you can ping it then the phone is responding to ICMP, but not to SIP requests. |
| 17:59.19 | ManxPower | Josh1: Unmonitered means you don't have the optional qualify= line for the entry for the phone in sip.conf |
| 17:59.37 | Josh1 | ill add it |
| 18:00.51 | *** part/#asterisk datareactor (datareacto@203.81.214.119) |
| 18:00.52 | Josh1 | now it says "UNKNOWN" |
| 18:01.26 | ManxPower | Josh1: As I said. Asterisk cannot communicate with your phone. |
| 18:01.38 | Josh1 | hmm |
| 18:01.41 | Josh1 | thats so wierd though |
| 18:01.42 | ManxPower | Josh1: maybe you had a fit of insanity and put a bindaddr= in your sip.conf. |
| 18:01.51 | randu | ManxPower: is there a way for me to see what context FWD is sending? |
| 18:02.06 | ManxPower | randu: For SIP calls a context is not sent |
| 18:02.29 | ManxPower | randu: FWD supports both IAX and SIP and you never said which one you were using in your initial message. |
| 18:02.39 | randu | oh IAX |
| 18:02.49 | Josh1 | ManxPower, nope.. no bindaddr= |
| 18:02.57 | Josh1 | 1000/xxxx (Unspecified) D 255.255.255.255 0 UNKNOWN |
| 18:03.03 | ManxPower | Josh1: |
| 18:03.18 | ManxPower | I ASKED if the IP address of the phone was listed for the peer and you said yes. |
| 18:03.37 | ManxPower | Josh1: Your phone is not registering with asterisk. Until you get it registering you CANNOT call the phone. |
| 18:03.43 | ManxPower | The phone might be able to call out, however. |
| 18:04.03 | ManxPower | <PROTECTED> |
| 18:05.11 | Josh1 | so, what are a few ways to fix that? |
| 18:05.13 | Damin | I think I am smoking crack! |
| 18:05.43 | ManxPower | Josh1: Tell the phone to register. |
| 18:05.44 | JonR800 | oh my josh. |
| 18:05.45 | Damin | I can't get an X100P to freaking send calls for channel 1 to the inbound context. |
| 18:05.59 | Shido6 | take your time :) |
| 18:06.08 | Shido6 | read the zaptel.conf and zapata.conf from top to bottom |
| 18:06.10 | Josh1 | ManxPower, it'd be nice if my phone said how ;P |
| 18:06.25 | nullogic | Josh1: monitor the communication between your sip phone and asterisk with something like asterisk -rvvvc |
| 18:06.27 | ManxPower | Josh1: Yes, it would. What phone are you using? |
| 18:06.50 | Damin | Shido6: Done and done.. |
| 18:06.52 | ManxPower | Josh1: I assume you are not seeing any registration errors on the console? |
| 18:07.41 | *** join/#asterisk BLaSt (~joN@pD9FF6DB4.dip.t-dialin.net) |
| 18:08.06 | Josh1 | ManxPower, no |
| 18:08.13 | Josh1 | erm.. its a Siemens HiNet LP 5100 |
| 18:08.33 | ManxPower | Never heard of it. You are on your own unless you want to buy me one of them. |
| 18:09.06 | Josh1 | heh |
| 18:09.50 | *** join/#asterisk ckruetze (~ckruetze@i3ED63B25.versanet.de) |
| 18:10.46 | ardor | Well is there anyway to combined h,1 to include 2 commands? |
| 18:11.00 | ardor | well I know how i could do it |
| 18:11.10 | ardor | its going to be bad style eather way... |
| 18:11.16 | ManxPower | ardor: make h,1, NoOp and see if the second priority is run. If so, something in your AGI is doing something bad |
| 18:11.38 | ardor | ok |
| 18:11.43 | ManxPower | ..er... exten => h,1,NoOp(I don't do anything) |
| 18:12.04 | ardor | w0rd |
| 18:12.33 | *** join/#asterisk SuPrSluG (~SuPrSluG@pool-70-16-40-16.buff.east.verizon.net) |
| 18:12.48 | DrPete | whats the convetion for numbering extensions? |
| 18:12.58 | SuPrSluG | hello |
| 18:12.59 | ardor | oh fuck you MaxPower, It works now. |
| 18:14.04 | ardor | thanks though, but that really sucks. |
| 18:14.16 | Nugget | heh |
| 18:14.29 | Nugget | a heisenbug. |
| 18:15.34 | Brixius | DrPete: ??? What do you mean? |
| 18:15.53 | ManxPower | DrPete: That is a local policy decision. |
| 18:16.07 | SuPrSluG | Brixius: u get dundi working? |
| 18:16.39 | Nugget | the only universal extension convention is that the tech setting up asterisk gets to choose a "good" extension number for himself. |
| 18:17.00 | Brixius | Ya, dundi itself is working, but my dialplan need's some work to format my extensions to e164 format before passing it to dundi. |
| 18:17.03 | ManxPower | For us it's 0000-1999 = Local only extensions for dev and testing, 2000-2999 = non-company owned systems, 3000-7999 = company owned PBXs, 8000-8999 = manual routing codes, and 9000-9999 are reserved |
| 18:17.08 | randu | ManxPower: ManxPower I am using IAX. Is there a way for me to find out what context they are sending? I know the user and pass are correct so I must have messed something up with the context. |
| 18:17.20 | ManxPower | randu: IAX2 debug |
| 18:17.26 | Damin | 036 -- Executing Goto("Zap/1-1", "default|s|1") in new stack |
| 18:17.40 | randu | ManxPower: Thanks |
| 18:18.22 | Brixius | SuprSlug: do you have it working yet? |
| 18:18.32 | SuPrSluG | yes and no |
| 18:18.52 | *** join/#asterisk nitcom (~nitcom@3ffe:8240:8030:0:0:0:0:2) |
| 18:19.12 | *** join/#asterisk oej (~oej@64-151-42-78-dhcp-kc.everestkc.net) |
| 18:19.26 | DrPete | I just wondered if there was a best way to avoid any problems etc. |
| 18:19.30 | *** join/#asterisk SuperMMan (~graphic@d205-206-30-139.abhsia.telus.net) |
| 18:19.34 | SuPrSluG | Brixius: i can dial out on zap channels remotely but i go into a loop and crash asterisk when i dial a remote internal extension |
| 18:19.56 | DrPete | Do i have 1 2,3 4, or five digit etc |
| 18:19.57 | ManxPower | DrPete: Don't use extensions beginning with 9 or 1 or 0 |
| 18:20.14 | DrPete | right oki |
| 18:20.15 | Brixius | SuPrSluG: ok, so you are having dialplan issues |
| 18:20.19 | ManxPower | DrPete: No matter what you do you'll run out of extensions. I like 4 digit extensions, but 3 digit is common too |
| 18:20.40 | DrPete | oki |
| 18:20.43 | ManxPower | I like that the last 4 digits of the person's DID is the same as their extension |
| 18:20.44 | SuPrSluG | Brixius: Executing Goto("OSS/dsp", "200|1") in new stack |
| 18:20.45 | SuPrSluG | <PROTECTED> |
| 18:20.47 | SuPrSluG | <PROTECTED> |
| 18:20.48 | SuPrSluG | <PROTECTED> |
| 18:21.01 | Nugget | ManxPower: yeah, that's a great way to do it if you have unique DIDs. |
| 18:21.13 | Nugget | s/unique/individual/ |
| 18:21.16 | SuPrSluG | Brixius: until KABOOM |
| 18:21.19 | Damin | Hmmm.. |
| 18:21.23 | DrPete | ManxPower: oki thanks, I think I will have 801 802 803 etc, the missus likes the number 8, lol |
| 18:21.25 | ManxPower | SuperMMan: You have an extensions.conf LOOP |
| 18:21.43 | ManxPower | 10 goto 20 and 20 goto 10 like that. |
| 18:21.47 | SuperMMan | ManxPower, nopper never needed it so i never used it |
| 18:22.01 | SuPrSluG | Brixius: doesn't seem to know how to actually dial |
| 18:22.09 | ManxPower | SuperMMan: Sorry, that was for SuPrSluG |
| 18:22.14 | SuperMMan | oh ok np |
| 18:22.19 | ManxPower | SuPrSluG: You have an extensions.conf LOOP |
| 18:23.30 | ManxPower | Looks like your macro is running exten 200,1 |
| 18:24.10 | DrPete | I am looking at simpletelecom and nufone, anyone use them? they any good? |
| 18:24.21 | SuPrSluG | ManxPower: yeah. it worked at one time but ... i had just had fix something that wasn't broke |
| 18:24.47 | blitzrage | DrPete: I'm using simpletelecom, their pretty good. Doesn't like setting my callerID at the moment though. NuFone is good too. |
| 18:25.39 | SuPrSluG | ManxPower: that's the remote # i want to dial. now i need it to goto stdexten and dial |
| 18:26.03 | wankel | i can't find anyone IAX or SIP with a good selection of multi-market DIDs. nufone only has one in their area. i'm pulling in four multi-market origination PRIs for now. |
| 18:26.05 | DrPete | blitzrage: yeh, i need to look at how caller id all works. what should I look for in a provider? |
| 18:26.49 | Brixius | SuPrSluG: in macro-dunki-e164, try getting rid of the exten => s,1,Goto(${ARG1},1) |
| 18:27.08 | Brixius | s/dunki/dundi |
| 18:27.22 | SuPrSluG | Brixius: i'll try that |
| 18:27.47 | wankel | what's * do when you have a loop in extensions? does it actually crash or does it warn and drop the call? |
| 18:28.07 | Brixius | I'm assuming you are using the sample stuff and modifying it to your needs, |
| 18:28.28 | DevilFish | you gouy operate your asterisk behind nat mostly? how about your sip phones? just real curious here if alot of my troubles will go away if I quit trying to fight nat.. |
| 18:28.41 | wankel | devilfish: things work fine for me behind nat. |
| 18:28.59 | wankel | though i keep the * server on a public ip |
| 18:28.59 | DevilFish | both asterisk and phones behind same nat? |
| 18:29.07 | *** join/#asterisk Yoda-BZH (~yoda-bzh@pro.game.admin.at.yoda-bzh.net) |
| 18:29.16 | DevilFish | server public and phones natted then? |
| 18:29.34 | *** join/#asterisk RaYmAn-B1 (user@213.237.12.147.adsl.vby.tiscali.dk) |
| 18:29.49 | wankel | yeah. the server can be behind nat, too, but you really need static nat entries for it then |
| 18:30.05 | Yoda-BZH | le bonsoir je vous souhaite / Hi ppl |
| 18:31.21 | wankel | the biggest nat problem i've had so far is that linksys routers handle outbound nat for multiple sip clients in a very bad way. the sip clients need to use different source ports if they're behind the same linksys router. |
| 18:31.43 | SuPrSluG | Brixius: i'm using blitzrage's paper |
| 18:32.10 | ManxPower | Useful Asterisk Docs (BOOKMARK THEM!): http://www.digium.com/index.php?menu=documentation (look at the "Unofficial Links") and http://www.voip-info.org/wiki-Asterisk and http://www.fnords.org/~eric/asterisk/ (my site) and http://asteriskdocs.org/ |
| 18:32.21 | ManxPower | get to the FNORDS site quick, it will be going away in a week |
| 18:32.27 | *** join/#asterisk Dibbler (~Dibbler@snaddy.plus.com) |
| 18:32.30 | Damin | Can you pass an extension argurment to the context setting in zapata.conf? I.E. if I want a specific channel to goto inbound context, extension 1234 can I represent that as "context=inbound/1234"? |
| 18:32.44 | ManxPower | Damin: no |
| 18:32.51 | Brixius | I just remembered I printed that, but never turned on the printer, time to turn it on and get my printout :) |
| 18:33.37 | *** join/#asterisk DevilFish (DevilFish@216.163.40.2) |
| 18:33.38 | Damin | ManxPower: Well, then how the hell do I Anti-Ex-Girlfriend Logic on an X100P? |
| 18:33.44 | Damin | ManxPower: Check this out.. |
| 18:34.05 | ManxPower | Damin: Easily. |
| 18:34.18 | bjohnson | Damin: needs callerid |
| 18:34.19 | ManxPower | ex-GF relies on CALLERID. |
| 18:34.43 | Damin | ManxPower: Yeah.. I've GOT callerid.. |
| 18:34.50 | Damin | ManxPower: Check this.. |
| 18:35.00 | ManxPower | Damin: If you flood you get to mop it up. |
| 18:35.02 | bjohnson | Damin: so look at extension page on wiki for girlfried example |
| 18:35.21 | Damin | context=inbound |
| 18:35.28 | bjohnson | Damin: like s/gfcallid,1,.... |
| 18:35.32 | twisted | Damin, set all calls to come to incoming, then priority 1 is a simple exten => _.,1,Wait(2) |
| 18:35.34 | twisted | then move on from there |
| 18:35.45 | ManxPower | Damin: Yes. So in [inbound] in extensions.conf the call will go to exten => s,1,Blah |
| 18:35.50 | twisted | that will make sure we give plenty of time for the fsk burst |
| 18:35.57 | Moc_ | fucking stupid tech removed my FTP server from the phone config... too bad for them, they phones wont work anymore |
| 18:36.00 | ManxPower | Do you need exten => s/bitchfromhellnumber,1,blah |
| 18:36.29 | twisted | you could use s,1,Wait(2) |
| 18:36.31 | ManxPower | twisted: on analog he doesn't need to wait for it since asterisk will wait for it |
| 18:36.35 | Damin | So this is wrong then: http://pastebin.ca/4245 |
| 18:36.43 | twisted | ManxPower, i have seen this solve the issue before. |
| 18:36.52 | ManxPower | twisted: Really? |
| 18:37.05 | twisted | ManxPower, asterisk doesn't wait on the fsk burst, it just waits for the second ring |
| 18:37.40 | twisted | Damin, that's fine, as long as you have DNIS on the line |
| 18:37.41 | ManxPower | exten => s,1,Goto(default,s,1) add exten => s,1/2125551212,Goto(evilpeople,s,1) |
| 18:37.43 | Nugget | is there anyone from nufone in the channel at the moment? |
| 18:38.13 | ManxPower | eek! |
| 18:38.19 | Nugget | I just realized that I made a boneneaded mistake. |
| 18:38.25 | Nugget | boneheaded, even. |
| 18:38.27 | ManxPower | exten => s/2125551212,1,Goto(evilpeople,s,1) |
| 18:38.31 | ManxPower | THAT'S better. |
| 18:38.39 | twisted | hehe |
| 18:38.45 | twisted | ManxPower, i've done that before on a production box |
| 18:38.56 | twisted | pbx_config.so doesn't care too much for it, tho :P |
| 18:39.06 | Nugget | I just created a new nufone account and I didn't notice that my auto password generator dealie has generated a password which contains an @ sign. |
| 18:39.13 | Nugget | this is breaking my register => |
| 18:39.28 | Nugget | there doesn't appear to be a way to change my password on the nufone site |
| 18:39.48 | mikegrb | Nugget: ja, I believe you've got to email support@ |
| 18:39.53 | Nugget | looks that way. |
| 18:39.59 | Nugget | man I feel stupid. what a dumb thing. :) |
| 18:40.06 | mikegrb | I quit smoking 3 days, 1 hour, 40 minutes, and 7 seconds ago. During that time, I would have smoked 42 cigarettes. (That's like smoking a 0.00 mile-long cigarette) By quitting, I've saved $7.35! I've avoided inhaling 1 grams of tar, 67 mg of nicotine, and 672 mg of carbon monoxide. |
| 18:40.11 | mikegrb | :D |
| 18:40.16 | iMediax | we all do it |
| 18:41.06 | phenix | yawn. |
| 18:41.11 | Makenshi | WizardOne, you should set the list of permitted characters, not the other way around :) |
| 18:41.28 | phenix | ok. media you ready to get down and dirty? :) |
| 18:41.29 | WizardOne | Makenshi yupp |
| 18:42.37 | WizardOne | hmm btw anyone heard good or bad things about RNK Telecom? |
| 18:42.42 | wankel | bad enough when they turn the ringer off and tell me they don't get calls... |
| 18:42.50 | Makenshi | I hope what i've heard about asterisk supporting an ldap backend for sip registrations comes to fruition |
| 18:43.20 | WizardOne | Makenshi would be grand, since most email and other apps do |
| 18:43.22 | *** join/#asterisk nullogic (~nullogic@66.240.64.121) |
| 18:43.40 | wankel | 'twould be easy to add. i didn't figure anyone would actually have a use for it. |
| 18:43.42 | WizardOne | many folks use * with radius? |
| 18:43.49 | Makenshi | wankel, no?? |
| 18:44.15 | WizardOne | wankel hysterically usefull in a enterprise environment |
| 18:44.31 | Damin | Voila! http://pastebin.ca/4247 |
| 18:44.35 | Damin | Now it works excellent.. |
| 18:44.36 | WizardOne | ;) |
| 18:44.41 | Makenshi | Even with a Microsoft infrastructure, as Active Directory is just a tarted up LDAP server :) |
| 18:44.46 | bjohnson | Makenshi: I'm more interested in ldap for user auth for vm, user to handset extension mapping, and user followme config storage |
| 18:44.56 | wankel | ah, true. if you're using ldap to hit AD that makes sense. |
| 18:45.25 | WizardOne | unfortunately maaany enterprises are in the MS fold ;) |
| 18:45.31 | wankel | last time i saw it used in an enterprise environment was for a netscape enterprise rollout i did at NYC dep of sanitation. netscape messaging server, calendar server, web server, etc. everything was ldap so it was spiffy enough. |
| 18:45.40 | Makenshi | bjohnson, those would all be tremendously useful to have |
| 18:45.42 | phenix | ok |
| 18:45.46 | phenix | im getting a proxy authentication required now |
| 18:45.57 | Makenshi | bjohnson, have you explored XCAP? I haven't looked into it myself, wonder it could take off |
| 18:46.04 | wankel | verisign is totally M$ internally so i have to deal with it all day. works... well enough, i guess. not as well as it should be not as bad as most geeks pretend. |
| 18:46.05 | phenix | but the phone is registered as a peer |
| 18:46.12 | WizardOne | wankel a hole bunch use the CommunigatePro thing too |
| 18:46.24 | WizardOne | didnt they do some SIP thing aswell? |
| 18:46.30 | Makenshi | wankel, i'm a big fan of active directory, makes things like software deployment and security a lot easier to manage |
| 18:47.39 | WizardOne | wankel well not all MS products is a conspiracy to commit genocide ;) |
| 18:47.40 | Makenshi | Where i work now, most things are run off openldap |
| 18:47.41 | wankel | makenshi: yeah. not a bad idea. just gets mis-used a lot and can make life difficult. shutting down a workstation that's on the AD takes about 5 minutes to sync. |
| 18:48.01 | Brixius | SuPrSluG: there's definatly a loop in this doc. |
| 18:48.17 | wankel | almost that long to log in sometimes, too. |
| 18:48.39 | SuPrSluG | Brixius: now to top it off my isp has changed my ip |
| 18:48.44 | Darwin_35 | ok Manx get over here and fix my trunking |
| 18:48.54 | SuPrSluG | Brixius: DOH |
| 18:49.28 | SuPrSluG | Brixius: where's the loop. and how to correct it? |
| 18:49.41 | Brixius | SuPrSluG: that sucks, use dynamic dns and advertise that in your dundi contexts. |
| 18:49.55 | wankel | wizardone: yeah, most ms-haters are just kinda silly. all my servers are linux (used to be solaris when linux sucked more) but i use windows for a desktop, and i think it works well there. |
| 18:50.13 | SuPrSluG | Brixius: one has static. i'm testing w/ dynamic. |
| 18:50.59 | *** join/#asterisk Fabe (~Fabe@p50858D7B.dip0.t-ipconnect.de) |
| 18:51.11 | WizardOne | wankel never been religious myself ,) I use and deploy whats required... |
| 18:51.38 | wankel | wizardone: exactly. software is just tools. i got over being passionate about it years ago. i guess i'm just old. :) |
| 18:51.40 | *** join/#asterisk markit (~marco@host119-245.pool8172.interbusiness.it) |
| 18:51.49 | WizardOne | :) same here |
| 18:52.21 | DevilFish | don't have the luxury of working or liking just one thing anymore |
| 18:52.29 | Makenshi | Slightly OT, I'm looking for a H323 client i can deploy through AD, pre-configured to work with my * server (ie, server address) |
| 18:52.36 | *** join/#asterisk berrayahkamel (~berrayahk@82.101.128.6) |
| 18:52.44 | *** join/#asterisk dg1nsw_ (~dg1nsw@saturn2.franken.de) |
| 18:52.46 | Brixius | SuPrSluG: I think if you just get rid of the goto it should work. I don't exactly know what the goto is supposed to do. |
| 18:52.49 | WizardOne | Makenshi neetmeeting kinda thing? |
| 18:53.20 | Brixius | SuPrSluG: in this document that is. |
| 18:53.32 | Makenshi | WizardOne, hmm, didn't think of that :) |
| 18:53.40 | bjohnson | Makenshi: I don't know anything about xcap |
| 18:53.48 | JerJer | avoid H.323 |
| 18:53.51 | wankel | sip.conf's layout is fine for a config file but is just NOT a good database design |
| 18:53.56 | WizardOne | wb JerJer ;) |
| 18:53.57 | Makenshi | jerjer, I do hate h323 |
| 18:54.00 | SuPrSluG | Brixius: that puzzled me also. |
| 18:54.08 | Makenshi | jerjer, but i work in academia where it's popular |
| 18:54.16 | JerJer | Makenshi: show them the light |
| 18:54.24 | WizardOne | hehe |
| 18:54.31 | wankel | makenshi: but ASN.1 is great! yay BER! :) |
| 18:55.00 | Makenshi | jerjer, oh i will, i've been invited to join the janet voice advisory group |
| 18:55.14 | WizardOne | ahh time for conf call :) damn * sure saves money *grin* |
| 18:55.25 | Makenshi | jerjer, but the majority of videoconferencing installations are bawsed on h323 |
| 18:55.40 | Makenshi | jerjer, in uk academia i mean |
| 18:55.42 | wankel | well, h.323 was designed for videocon, so it kinda makes sense there. |
| 18:56.05 | randu | Hello! I was told that vonage does not offer all the features that should come with VOIP. For the example given was multi incoming calls on the same line feeding into PBX server. Does anyone have any recommendations for me to check out? |
| 18:56.12 | WizardOne | wankel hint: jerjer is a tad religious :) |
| 18:56.22 | Brixius | now if they would just update h.323 to get the ip address out of the data portion of the packet, everyone would be happer. |
| 18:56.33 | Makenshi | wankel, true, although sip was designed for ip, and video is a natural extension of it, hmm, interesting argument there :) |
| 18:56.47 | wankel | was a stupid idea to re-use h.323 for voip. way too many assumptions about everything running on circuit-switched networks. |
| 18:56.52 | Makenshi | i think h323 is better supported with isdn |
| 18:56.59 | Makenshi | yip |
| 18:57.20 | wankel | yeah, anything involving a packet-switched network and h.323 is bound to suck. they just don't match. |
| 18:58.07 | wankel | back when i ran a toll bypass provider we use it because there wasn't any other choice. MGCP was just starting to make waves and no one had heard of SIP. |
| 18:58.27 | wankel | but it sucked then, too. we used cisco's proprietary stuff whenever possible. |
| 19:00.18 | bjohnson | randu: check the wiki. nufone, voipjet, aleph-com.net .. most of the multi line voip providers do not provide packages .. but do the calcs for your usage, you may still be better off |
| 19:00.43 | *** join/#asterisk heath__ (~heath@12-215-32-191.client.mchsi.com) |
| 19:01.09 | bjohnson | randu: the biggest pita with vonage is the hardware they supply is locked so you can use it for nothing but connecting to them (and can't be used at all if you are using asterisk to connect to them) |
| 19:01.20 | *** join/#asterisk sjobeck3 (~Sjobeck@london.sjobeck.com) |
| 19:01.36 | bjohnson | randu: some of the package voip providers still allow 2 concurrent calls .. I think voicepulse and broadvoice do |
| 19:02.20 | sjobeck3 | hi all. newbie here. how can I see the incoming call ringing on my zap/1 ? |
| 19:02.37 | randu | bjohnson: Thanks :-) |
| 19:03.19 | sjobeck3 | nufone allows (i think) unlimited simul calls |
| 19:03.23 | ManxPower | sjobeck3: look at the asterisk console |
| 19:03.29 | wankel | if it's analog, you can put the wires in your mouth. you'll see a flash when the 100V hits. |
| 19:03.40 | sjobeck3 | nize :) |
| 19:03.44 | randu | an another note I finally figured out that I could not have quotes around my FWD password in extensions.conf so now I am not getting any errors in the logs when I call but all it does is ring and ring. any ideas? |
| 19:03.46 | ManxPower | Most of the unlimimited plans only allow 1 call, maybe 2 calls at a time, most of the per min plans allow unlimited calls |
| 19:03.49 | *** join/#asterisk labo (~ariel@hankster.caliente.com.mx) |
| 19:04.22 | Yoda`Mange | un bon appetit je vous souhaite |
| 19:04.30 | *** join/#asterisk L|NUX (~linux@202.63.215.42) |
| 19:04.38 | Moc_ | tu mange encore ? |
| 19:04.41 | sjobeck3 | manxpower: on a tangent: how to tell from outside the box/case which are my FXO & FXS? |
| 19:04.58 | sjobeck3 | manxpower: are jacks identical? |
| 19:05.21 | Yoda`Mange | Moc ouais, j'ai tout le temps faim :p |
| 19:06.00 | Syncros | une bonne sandwitch au bacon avec dla mayo ya rien de mieux pour commencer la journee |
| 19:06.22 | Yoda`Mange | lol |
| 19:06.28 | sjobeck3 | i could go for a BLT right now too |
| 19:06.34 | kram | un croissant beurre pas trop cuit |
| 19:06.41 | bjohnson | randu: those 5 are my current favs .. but bv needs a patch to the sip code .. info on the wiki |
| 19:06.58 | randu | bjohnson: ok. |
| 19:07.00 | sjobeck3 | any way to tell the FXO from the FXS from outside the case ? |
| 19:07.04 | Moc_ | moi ca ete 2 toste au beure de penute, pis du pate chinois |
| 19:07.19 | Syncros | restants de pizza froide |
| 19:07.23 | twisted | kram: lol |
| 19:07.39 | Moc_ | j'ai encore faim... |
| 19:07.57 | Moc_ | emule calibrate the download of multiple files or what ?? |
| 19:09.21 | *** join/#asterisk synco (~synco@c-24-98-180-64.atl.client2.attbi.com) |
| 19:09.36 | twisted | parlez anglais au sujet des croissants, puis d'autres peuvent apprécier la conversation |
| 19:09.48 | Moc_ | kram, what do you think of bug 3337 ? |
| 19:10.21 | Moc_ | twisted, heille la tete carre :) |
| 19:10.46 | kram | moc: i think it's a start, i just wonder if there is more than can be done, maybe a better structure |
| 19:10.50 | synco | i just moved to a new house and have a slightly different network setup here. now when i make an outgoing FWD call it seems to connect properly but I can't get incoming audio. i'm sure it's a NAT problem but I never had this problem before with NAT. |
| 19:10.57 | bjohnson | mmmm .. buttered buns |
| 19:11.05 | bjohnson | oops .. wrong chan |
| 19:11.11 | kram | for example could it be a structure with menus etc |
| 19:11.12 | twisted | :P |
| 19:11.18 | Moc_ | kram, it what I thought too, I was thinking about maybe using flags to enable or disable options for playback |
| 19:12.58 | Moc_ | menus ? |
| 19:13.15 | bjohnson | synco: iax or sip? try set verbose 3 and watch the messages .. check the codecs |
| 19:13.15 | kram | yes, so that for example the entire application of voicemail could be structured |
| 19:13.33 | ManxPower | Maildir! |
| 19:13.41 | sjobeck3 | how has the disaster relief fundraising been going? |
| 19:14.03 | synco | bjohnson: sip. i saw that iax pretty much solves all the nat problems but i wasn't quite sure how the config should go for fwd-iax and couldn't find an example on wiki. |
| 19:14.26 | sjobeck3 | synco: check the FWD site about their IAX connecting option. |
| 19:15.10 | SuPrSluG | Brixius: now the peer is unreachable. i ping the host @ 65ms but under dundi show peers it's over 2000ms. |
| 19:15.16 | sjobeck3 | all: I dont know why it shows me thusly, this is "sjobeck". |
| 19:15.53 | Moc_ | kram, so like make a struct and assign a key on phone for some playback option, then when you call the playback, you specify what option you want to hear ? |
| 19:16.31 | synco | sjobeck3: not finding info on the fwd site. seems to just be an advertisement more than any kind of support. |
| 19:16.41 | sjobeck3 | synco: 1 sec |
| 19:16.44 | Moc_ | so like 'Press','1','for more options'=Option 1 |
| 19:16.45 | synco | unless it's taht plug thing at the top that my old POS browser won't load. |
| 19:16.57 | synco | *plugin |
| 19:17.02 | kram | well, i don't know that it's that simple... |
| 19:17.07 | kram | i think it needs more discussion |
| 19:17.12 | kram | you might want to play things back |
| 19:17.15 | Moc_ | so you can call playback(pl,"12*#") and it will only play option 12*# |
| 19:17.19 | kram | you might want to call arbitrary routines |
| 19:17.32 | sjobeck3 | synco: http://tinyurl.com/5rjpt |
| 19:18.29 | synco | thanks man |
| 19:18.54 | Moc_ | kram, the idea at the base is just to save us from creating static recording for menu in meetme and voicemail |
| 19:20.00 | kram | but if you do'nt use static recordings for the menus they will sound very unnatural |
| 19:20.28 | kram | just try option '2" to switch folders and listen |
| 19:20.32 | kram | see how unnatural it sounds |
| 19:20.49 | Moc_ | that because our recording are not well made, and we have to big delay between them |
| 19:21.12 | *** join/#asterisk labo (~ariel@hankster.caliente.com.mx) |
| 19:21.17 | Moc_ | the big problem is the delay between each msg, it why Ive put a delay value |
| 19:22.58 | Moc_ | if you play back the seperate part correctly of a msg, it should sound perfect |
| 19:24.38 | *** part/#asterisk gbs (~GBS@adsl-5062c3cb.monradsl.monornet.hu) |
| 19:31.04 | Josh1 | ok a little progress |
| 19:31.04 | Josh1 | http://pastebin.com/229267 |
| 19:31.18 | Josh1 | it says "Unable to create channel type 'SIP'" |
| 19:31.25 | Josh1 | when i try to call locall |
| 19:31.29 | Josh1 | from 6001 or 6002 |
| 19:31.37 | *** join/#asterisk kuj (~kuj@c-67-165-241-16.client.comcast.net) |
| 19:32.24 | sjobeck3 | sorry for pedestrian question. does any one how if you can tell from outside the server case, which are FXS & which are FXO, on the card, are the jacks identical? any coloring? markings? etc? thx |
| 19:32.46 | ManxPower | Josh1: Are both 6001 and 2002 showing up with the correct IP address in sip show peers? |
| 19:32.55 | ManxPower | sjobeck3: You can't. |
| 19:32.59 | Nugget | sjobeck3: they're identical. |
| 19:33.09 | sjobeck3 | thank you (x2) |
| 19:33.22 | Josh1 | hmm |
| 19:33.23 | Josh1 | xlite is |
| 19:33.26 | ManxPower | well 6002, not 2002 |
| 19:33.26 | Josh1 | but 6001 |
| 19:33.27 | Josh1 | isnt |
| 19:33.37 | ManxPower | Josh1: if the ip address isn't showing then you can't call that device. |
| 19:34.05 | sjobeck3 | nugget: can I see the line ringing on that FXO channel at CLI? can you "see" the call coming in? debug? |
| 19:34.47 | markit | anyone using the new announced transfer provided by asterisk? patch 3241 |
| 19:34.56 | DrPete | whats the diff between iax and sip? can i still call sip phones for free if I am conneted to nufone using iax |
| 19:35.11 | ManxPower | sjobeck3: Yes, you can. |
| 19:35.26 | ManxPower | sjobeck3: If you plug a phone line into an FXS module you will blow it up |
| 19:35.36 | sjobeck3 | the module? the phone? |
| 19:35.56 | ManxPower | sjobeck3: If you plug a phone line into an FXS port you will blow up the port and have to have it replaced. |
| 19:36.26 | ManxPower | sjobeck3: You should see Starting Simple Switch on... when the line rings |
| 19:36.37 | sjobeck3 | manxpower: ut oh. I hope I havent done that already. doh. I'm a goober. i'll have to go check. damn! |
| 19:36.40 | Chuji | ring ring... poof... |
| 19:37.22 | DrPete | dam, eheh. with voip is there any point to normal extensions? |
| 19:37.36 | ManxPower | DrPete: "normal extensions" |
| 19:37.59 | DrPete | ManxPower: sorry the ones you plug into one of them tdm cards |
| 19:38.13 | ManxPower | DrPete: The phones are cheap. Also faxs. |
| 19:38.34 | ManxPower | You don't want to try to fax over VOICE over IP. The last person that tried took a gun and went to McDonalds. Not pretty. |
| 19:38.45 | DrPete | ManxPower: ahh right of course |
| 19:38.45 | DrPete | lol |
| 19:39.08 | DrPete | yeah, i was going to do it, but advised not too, i use fax to email now |
| 19:39.21 | file[laptop] | fax... over VoIP... I'd rather spend an evening with ManxPower then try it... |
| 19:39.36 | ManxPower | file[laptop]: I think I'm supposed to feel insulted. |
| 19:39.52 | DrPete | lol |
| 19:40.08 | ManxPower | file[laptop]: It WOULD be an evening you would NEVER forget. |
| 19:40.33 | DrPete | well i think i am gonna grab a bit of cedit with nufone, whats the diff between iax and sip, can i still call all the sip phones free? |
| 19:40.45 | ManxPower | DrPete: No, you cannot call SIP phones for free. |
| 19:40.55 | ManxPower | Hell you can't call any phone for free that is not connected to your provider. |
| 19:41.19 | sjobeck3 | manxpower & drpete: doesnt DUNDI address this? a bit? |
| 19:41.23 | DrPete | balls, i thohgt it was all like free |
| 19:41.50 | ManxPower | sjobeck3: I don't do DUNDi, but I doubt it. Not really. |
| 19:42.12 | ManxPower | In theory DUNDi is supposed to let providers interconnect, so eventually I guess it could help. |
| 19:42.36 | sjobeck3 | manxpower & drpete: or dialing by name (someone@servername.domain.com) addresses this? presuming you know the other person's sip addy? |
| 19:42.48 | DrPete | I see, but i assume I can make an accont with a suppiler and just use there sip call and dont pay them cash? |
| 19:42.54 | ManxPower | sjobeck3: And they allow guest calls. |
| 19:42.57 | *** join/#asterisk nullogic (~nullogic@66.240.64.121) |
| 19:43.04 | sjobeck3 | manx: right, right |
| 19:43.09 | sjobeck3 | manxpower: right, right |
| 19:43.25 | DrPete | can asterisk detect what network i am calling and use the correct provider |
| 19:43.27 | Damin | DrPete: Have you read ANY documentation on the Wiki? |
| 19:43.29 | sjobeck3 | manxpower: do folks typically allow guests to SIP to their * ? |
| 19:43.52 | sjobeck3 | manxpower: or IAX guests to their * server ? |
| 19:43.54 | Druken | sjobeck3, i do.... |
| 19:43.56 | ManxPower | sjobeck3: *I* don't normally. |
| 19:43.57 | DrPete | Damin: what wiki? lol yeah I am read some |
| 19:44.02 | Druken | i allow a guest sip only |
| 19:44.14 | ManxPower | I don't want random people calling me. |
| 19:44.31 | sjobeck3 | drunken: what context they dropped in to ? how (very broadly speaking) they handled? |
| 19:44.34 | ManxPower | It's bad enough they e-mail me for support on my AGI scripts in the abandoned-stuff directory. |
| 19:44.43 | Druken | my guest goes into my main menu for my business |
| 19:45.00 | sjobeck3 | druken: druken or drunken? :) |
| 19:45.06 | sjobeck3 | druken: sorry |
| 19:45.16 | *** join/#asterisk _tekati_ (~captain@cpe-66-75-215-63.bak.rr.com) |
| 19:45.16 | sjobeck3 | druken: that makes sense. |
| 19:45.45 | sjobeck3 | druken: you use DUNDI ? |
| 19:46.02 | Druken | i don't... |
| 19:47.03 | *** join/#asterisk kentster (~kentster@vpn1.ccstg.com) |
| 19:49.28 | *** join/#asterisk reinvented (~chatzilla@reinvented.isn.net) |
| 19:51.02 | *** part/#asterisk Josh1 (~Joshwah@adsl-8-175-108.mia.bellsouth.net) |
| 19:51.13 | *** join/#asterisk kentster (~kentster@vpn1.ccstg.com) |
| 19:53.28 | ckruetze | I've a problem with variables in the dialplan, can somebody help? |
| 19:54.45 | ckruetze | I somehow need to forward the variables together with the call from one Asterisk box to another one, can I do that? |
| 19:55.01 | Chuji | kind of |
| 19:55.06 | Chuji | how many variables? |
| 19:55.11 | Druken | with the use of a database :) |
| 19:56.17 | *** join/#asterisk oej (~oej@64-151-42-78-dhcp-kc.everestkc.net) |
| 19:56.33 | ckruetze | not sure howmany there will be, we'll create a voice menu that askes the user several questions and then it decides where to forward the call to, the new box needs to know the aswers. Probably 4 to 8 variables. |
| 19:56.57 | ckruetze | Yes, I know that I could do it with a database, but if possible I'd like to avoide that. |
| 19:57.21 | Druken | i can't see anyway around it |
| 19:58.18 | Chuji | iax2 needs to have a URL header like sip |
| 19:58.37 | Chuji | That way you could format a url with variables seperated by &'s |
| 19:59.23 | ckruetze | can't I just create header fields and somehow get them into the channel? |
| 20:00.22 | ManxPower | In theory you can set _ or __ at the beginning of the varible name for CVS-HEAD |
| 20:00.31 | ManxPower | I don't know if that work was ever finished. |
| 20:00.33 | Chuji | ckruetze : What I generally do is pad the CID with numbers that the receiving * box dialplan acts on |
| 20:01.12 | Chuji | ckruetze : I then strip those off keeping the callerID in tact and acting upon the first numbers accordingly |
| 20:01.24 | Chuji | ckruetze : Won't work in all cases, but I've managed OK doing that |
| 20:01.30 | sjobeck3 | manxpower: if my card looks just like this: |
| 20:01.31 | sjobeck3 | www.digium.com/index.php?menu=wildcard_tdm400p2 |
| 20:01.31 | sjobeck3 | which jacks are which? |
| 20:01.44 | ckruetze | Hm, can the CID be somthing like 60 to 100 digits long? |
| 20:02.00 | Chuji | uhh, I doubt it |
| 20:03.27 | ManxPower | Side A: Dial(IAX2/user@siteb/customer-1234@parse) |
| 20:03.34 | randu | Hello, What brand is good for converters that take analog phone system into digital so we can use regular phones with our VOIP? |
| 20:03.36 | ckruetze | Ok, then I guess I've to use a database, but how do I identify the call if the caller didn't provide a CID? |
| 20:04.02 | ManxPower | Side B: [parse] exten => _.,1,AGI(parse,agi) |
| 20:04.13 | ManxPower | where customer-1234 is an extension. |
| 20:04.22 | ckruetze | MaxPower: Sounds cool, but how long can [parse] be? |
| 20:04.23 | ManxPower | You can have almost any string as an extension |
| 20:04.23 | gambolputty | sipura |
| 20:04.28 | sjobeck3 | nugget: you out there? |
| 20:04.31 | Nugget | moo |
| 20:04.41 | ManxPower | ckruetze: I however long an extension can be. See the source. |
| 20:04.52 | ckruetze | ok |
| 20:05.14 | sjobeck3 | nugget: from this : http://www.digium.com/images/TDM22b.jpg, can you tell which ports are which, from the perspective of the jacks? |
| 20:05.20 | ManxPower | ckruetze: or ask on the mailing list the _ and __ variables are supposed to be passed |
| 20:05.29 | ckruetze | Thanks a lot. I'll check the source and give it a try, if it is long enough for me. |
| 20:05.46 | bkw_ | ok quick someone paypal me 3500 dollras |
| 20:05.48 | bkw_ | er dollars |
| 20:05.49 | bkw_ | hehe |
| 20:05.51 | Nugget | the sockets are numbered on the card and correspond to the numbered ports. |
| 20:05.54 | ManxPower | sjobeck3: I would have to go into my server closet dig thru the cables, crawl under the rack and look. No, you are not worth that pain. |
| 20:06.28 | sjobeck3 | manxpower: ah, cmon, where is the luv? |
| 20:06.31 | Chuji | bkw_ : Will .35 do? |
| 20:06.37 | sjobeck3 | nugget: i nust have missed them, I'll look again |
| 20:06.51 | ckruetze | bkw_ 3500 $, did the price for mini macs go up or did you find a new, cool toy? |
| 20:06.58 | Nugget | moreover, when the zaptel drivers initialize it tells you which are which. |
| 20:06.58 | sjobeck3 | nugget: where you guess those #;s are located? |
| 20:07.12 | Nugget | I don't remember. it's been months since I last layed eyes on a tdm400p |
| 20:07.17 | Qwell | sjobeck3: red are fxo ports |
| 20:07.21 | Qwell | assuming thats what you're asking |
| 20:07.21 | Nugget | I install them and then drink to wipe the memory |
| 20:07.21 | ManxPower | Nugget: He's trying to map ports on the back to channel numbers |
| 20:07.43 | sjobeck3 | nugget: oh, right, ok, I have that, what do I do with that output once I have it? which end to start from? |
| 20:07.44 | Nugget | I recall that it's totally unambiguous when holding the card in your hand |
| 20:07.59 | ManxPower | sjobeck3: It will not damage the port to plug a PHONE into an FXO port. |
| 20:08.16 | ManxPower | So if you get the ports configured and working and asterisk loaded you should hear dialtone on the FXS ports. |
| 20:08.19 | sjobeck3 | manxpower: I'm a goober, I dont want to blow up my house, my kids are home today |
| 20:08.37 | sung | mooo. |
| 20:08.43 | Nugget | moo |
| 20:08.43 | ManxPower | I would just remove the FXO modules until you have all the FXS working and loaded. |
| 20:08.51 | sjobeck3 | manxpower: or sparks shooting inside case if I choose wrong? |
| 20:09.48 | sjobeck3 | manxpower: what command spits out the ports again? not ztcfg. ....... ? |
| 20:09.57 | ManxPower | sjobeck3: If you plug a phone line into an FXs port and the line rings you generally throw out the FXO module and buy a new one. |
| 20:10.04 | ManxPower | ztcfg -hhh |
| 20:10.17 | ManxPower | ..er...ztcfg -vvv |
| 20:10.45 | sjobeck3 | innsbruck:/etc/asterisk # ztcfg -vvv |
| 20:10.45 | sjobeck3 | <PROTECTED> |
| 20:10.45 | sjobeck3 | Zaptel Configuration |
| 20:10.45 | sjobeck3 | ====================== |
| 20:10.45 | sjobeck3 | <PROTECTED> |
| 20:10.47 | sjobeck3 | Channel map: |
| 20:10.49 | sjobeck3 | <PROTECTED> |
| 20:10.51 | sjobeck3 | Channel 01: FXO Kewlstart (Default) (Slaves: 01) |
| 20:10.53 | sjobeck3 | Channel 02: FXO Kewlstart (Default) (Slaves: 02) |
| 20:10.56 | sjobeck3 | Channel 03: FXS Kewlstart (Default) (Slaves: 03) |
| 20:10.57 | sjobeck3 | Channel 04: FXS Kewlstart (Default) (Slaves: 04) |
| 20:10.58 | Chuji | DOOD! |
| 20:10.58 | DrPete | its www.nufone.net i hope i have not just been scammed by someone |
| 20:10.59 | sjobeck3 | <PROTECTED> |
| 20:11.01 | sjobeck3 | 4 channels configured. |
| 20:11.03 | sjobeck3 | <PROTECTED> |
| 20:11.05 | sjobeck3 | innsbruck:/etc/asterisk # |
| 20:11.08 | Chuji | ~pastebin |
| 20:11.12 | jbot | rumour has it, pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca |
| 20:11.19 | Chuji | sjobeck3 ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ |
| 20:11.27 | sjobeck3 | chuji & jbot: sorry, right, right, thx |
| 20:11.32 | sjobeck3 | pastebin.......right |
| 20:11.57 | *** join/#asterisk Mike_TK (~Mike_TK@213.180.245.62) |
| 20:11.57 | sjobeck3 | jbot: thx for education, right, right |
| 20:13.00 | sjobeck3 | druken: no, that's ok, I am a goober today, for some reason, biorythms maybe, ice storm maybe |
| 20:13.18 | Druken | Chuji: i'm awear |
| 20:13.32 | Chuji | why the "eye roll" then? |
| 20:14.14 | Chuji | heh |
| 20:14.15 | sjobeck3 | chuji & druken: boys, girls, boys, girls, it was me, it's over. |
| 20:14.42 | Druken | sjobeck3: are you european? |
| 20:15.57 | sjobeck3 | druken: two generations ago, but, no, now I/we are full on embarrased cowboy amerikan types kicking butt all over the planet to distract from problems here at home ...... feel me? |
| 20:16.13 | *** join/#asterisk SuPrSluG_ (~SuPrSluG@pool-70-16-40-16.buff.east.verizon.net) |
| 20:16.39 | SuPrSluG_ | hello |
| 20:16.43 | Druken | not exactly... i'm just getting a strong european vibe from ya... tis all :) |
| 20:16.52 | sjobeck3 | completly off the topic, apologies, but how does one recovery a nick on this channel. |
| 20:17.10 | sjobeck3 | druken: ok, well, that's fine, I dont shrink from it. thx for help so far, BTW, thx. |
| 20:17.23 | SuPrSluG_ | Brixius: if not goto ..... then what |
| 20:17.27 | Druken | did i help? |
| 20:17.29 | Druken | hehe |
| 20:18.17 | sjobeck3 | druken: 100% |
| 20:18.17 | sjobeck3 | (btw: i headed to basement now to re-jigger it; back in sec'.) |
| 20:18.27 | *** join/#asterisk humberto (~hav@201.128.177.84) |
| 20:18.37 | humberto | hello |
| 20:18.59 | Druken | I R Confuzzled.... |
| 20:19.18 | humberto | excuse me guys,, whats the difference between safe_asterisk startup and asterisk plain (voipinfo is very little info about this) |
| 20:19.37 | sjobeck3 | druken: i meant my FXO & FXS wires to my box. like I said, damn, i need some gingk otoday, I'm not making sense, my fault |
| 20:20.02 | Druken | ahh.. got ya |
| 20:20.21 | Druken | am i to assume you have a tdm card? |
| 20:21.00 | Druken | someone needs to make a cheap FXS single port card.. hehehe |
| 20:21.41 | *** join/#asterisk Mike (~mike@201.129.122.236) |
| 20:21.51 | flewid | sup |
| 20:22.03 | flewid | anyone here experimenting with the PA-168S chipset? |
| 20:22.08 | flewid | specifically with iax2 :) |
| 20:22.12 | sjobeck3 | druken: me? yes? 2 & 2 (of each). |
| 20:23.11 | Druken | ahh ok |
| 20:23.40 | sjobeck3 | druken & chuji & manxpower: just found numbers on back of card, starting from bottom of card, increasing toward top of card. Missed it before. |
| 20:23.44 | DrPete | anyone know the nufone server, i cant seem to find any docs |
| 20:23.51 | ManxPower | <PROTECTED> |
| 20:24.02 | sjobeck3 | manxpower: y |
| 20:24.14 | kFuQ | could someone give me some assistance on SetCIDName ? I am trying to set the CID name for my cell when i call in thru the POTS line and it isn't working correctly |
| 20:24.24 | sjobeck3 | drpete: switch-2.nufone.net |
| 20:24.57 | DrPete | sjobeck3: great thanks |
| 20:25.08 | kFuQ | and in the RTFM, it says something about setting the ANI with a flag, but i have not been able to find any documentation on ANI Flags |
| 20:25.09 | Druken | sjobeck3: what exactly are you rying todo ? |
| 20:25.10 | ManxPower | thanks |
| 20:25.24 | ManxPower | kFuQ: Unless you have a PRI it doesn't matter. |
| 20:26.01 | kFuQ | ManxPower: Have any ideas on why SetCIDName isn't functioning properly ? |
| 20:26.55 | flewid | kFuQ: what version of * |
| 20:26.55 | JerJer | anyone got a known good tiff image that works with TxFAX? |
| 20:26.55 | kFuQ | Spoofed:CID] |
| 20:26.55 | kFuQ | exten => 1234567890,1,SetCIDName(The Cell|<123>456-7890) |
| 20:26.55 | ManxPower | kFuQ: There are so many answers to that I don't know where to start. Details, dude, we need details. |
| 20:26.55 | flewid | JerJer: tiff image? |
| 20:26.56 | flewid | JerJer: i'm using tiff 3.7.1 |
| 20:26.59 | kFuQ | 1.x * |
| 20:27.01 | flewid | and it works for tx and rx |
| 20:27.14 | ManxPower | kFuQ: Caller*ID num NEVER has anything except numbers in it. |
| 20:27.16 | flewid | exten => s,1,SetCIDName(qh - sales) |
| 20:27.21 | JerJer | flewid: can you send me a tiff file? i'm playing with txfax right now |
| 20:27.28 | flewid | just any old tiff? |
| 20:27.31 | flewid | sure, sec |
| 20:27.33 | kFuQ | hmm |
| 20:27.34 | ManxPower | kFuQ: Ah, never mind you are setting the NAME, not the number. |
| 20:27.39 | kFuQ | yah |
| 20:27.41 | *** join/#asterisk Swiff (~swiff@fs-lnx1.flexshop.dk) |
| 20:27.46 | ManxPower | kFuQ: What interface is the call going out of? |
| 20:27.57 | kFuQ | incoming thru Zap/1 |
| 20:27.59 | JerJer | hopefully something you know works with TxFAX...i've tried a few different tiff's i've found online and my fax machine never seems to spit anything out |
| 20:28.08 | flewid | kFuQ: i couldn't get setcidname to work unless it was after cvs october 2004 |
| 20:28.19 | ManxPower | kFuQ: So the call comes into Zap/1, you want to set the name, then send the call to a local phone? |
| 20:28.24 | kFuQ | yah |
| 20:28.35 | Druken | that's easily done |
| 20:28.38 | ManxPower | kFuQ: Try not putting things like - <, etc./ |
| 20:28.40 | Druken | i do that with mine |
| 20:28.45 | Swiff | I can't figure out how to make music play, instead of the dialin tone? Ie, i want an incoming call to go directly to musiconhold until extension 101 answers - how is this done? |
| 20:28.45 | ManxPower | Set the same to something simple like The Cell 1234 |
| 20:28.47 | flewid | yeah i was gonna say remove the weird chars |
| 20:28.50 | kFuQ | so instead of cell coming up City,State -- make it say "the Cell" or some shit like that |
| 20:28.54 | humberto | guys ,, in astman I get a anoying msg of peerstatus....... |
| 20:28.58 | file[laptop] | Swiss: Dial does it... type show application dial |
| 20:29.02 | humberto | how do I remove it? |
| 20:29.06 | flewid | JerJer: hmm, how do you have yours set up? |
| 20:29.10 | Druken | kFuQ: SetCIDName(this is text) |
| 20:29.15 | Swiff | file[laptop]: that was for me? :) |
| 20:29.16 | flewid | JerJer: mine is if a fax comes in, converts it to pdf, and emails me |
| 20:29.23 | flewid | if i send out, i use a physical fax machine |
| 20:29.29 | file[laptop] | Swiff: yes |
| 20:29.32 | flewid | but i've also tested with an internel extension to make sure tx works |
| 20:29.34 | *** join/#asterisk hermie (~nick@24.236.167.53.bay.mi.chartermi.net) |
| 20:29.36 | humberto | 'Ignoring unknown event 'PeerStatus' |
| 20:29.40 | Swiff | file[laptop]: alright, thanks :) |
| 20:30.00 | kFuQ | hmm |
| 20:30.07 | kFuQ | still not coming up right |
| 20:30.09 | kFuQ | :-( |
| 20:30.14 | flewid | kFuQ: put it to like |
| 20:30.15 | Druken | what is your line? |
| 20:30.16 | flewid | 1234 |
| 20:30.20 | flewid | just something realllly simple |
| 20:30.28 | ManxPower | kFuQ: I assume you see the SetCIDName on the CLI? |
| 20:30.34 | ManxPower | kFuQ: What phone are you using? |
| 20:30.42 | kFuQ | <PROTECTED> |
| 20:30.42 | kFuQ | exten => 1234567890,1,SetCIDName(The Cell) |
| 20:30.56 | flewid | can you put colons in context defs? |
| 20:31.14 | kFuQ | i have colons in almost all my contexts and it works fine |
| 20:31.21 | flewid | neat i did not know that |
| 20:31.38 | flewid | what phone are you using? |
| 20:31.48 | file[laptop] | " |
| 20:31.53 | kFuQ | cell ----- landline ----- ata186 -- cordless |
| 20:31.57 | file[laptop] | try, SetCIDName("Testing 123") |
| 20:32.01 | Druken | well, i must ask the same question as ManxPower, are you seeing the SetCIDName being run in the CLI ? |
| 20:32.07 | JerJer | i am just trying to send a fax out with TxFAX |
| 20:32.08 | sjobeck3 | flewid: what software you usaing for fax->mail ? |
| 20:32.18 | flewid | sjobeck3: libtiff + ps2pdf |
| 20:32.25 | flewid | er tiff2pdf |
| 20:32.33 | kFuQ | Druken: no |
| 20:32.35 | sjobeck3 | flewid: tricky ? easy? |
| 20:32.43 | flewid | sjobeck3: took an afternoon |
| 20:32.44 | Druken | then that's your problem |
| 20:32.47 | file[laptop] | haha |
| 20:32.51 | sjobeck3 | right, right |
| 20:32.52 | JerJer | but rxfax still likes to segfault |
| 20:32.54 | kFuQ | i have that context included in all the rest |
| 20:32.59 | flewid | JerJer: what version of spandsp? |
| 20:33.15 | *** join/#asterisk [TripleF] (kvirc@HSE-QuebecCity-ppp3496601.sympatico.ca) |
| 20:33.16 | Druken | kFuQ: just because it's included doesn't mean it's going to be run |
| 20:33.17 | kFuQ | <PROTECTED> |
| 20:33.17 | kFuQ | <PROTECTED> |
| 20:33.18 | flewid | asterisk = cvs as of yesterday, spandsp 0.0.8, tiff 3.7.1 |
| 20:33.20 | kFuQ | tru |
| 20:33.32 | flewid | kFuQ: turn debugging on |
| 20:33.37 | flewid | and you should see it try and set it as a call comes in |
| 20:33.38 | kFuQ | which one ? |
| 20:33.39 | flewid | on the CLI |
| 20:33.40 | kFuQ | sip ? |
| 20:33.41 | ManxPower | kFuQ: We need to see the actual CLI output of a CALL. |
| 20:33.46 | ManxPower | put it on pastebin.ca |
| 20:33.48 | JerJer | flewid: where did you find that version? |
| 20:33.56 | flewid | JerJer: which version? the newer spandsp? |
| 20:33.58 | JerJer | spandsp-0.0.2pre6 |
| 20:34.01 | flewid | on the opencall site |
| 20:34.08 | flewid | it's up to pre8 |
| 20:34.11 | flewid | sorry not 0.0.8 |
| 20:34.12 | flewid | pre8 |
| 20:34.13 | flewid | :) |
| 20:34.17 | redder86 | pre8... hrmmm |
| 20:34.31 | flewid | gives a lot more output too when receiving/sending |
| 20:34.33 | flewid | it's nice for debugging |
| 20:34.34 | ManxPower | JerJer: using it with stable or head? |
| 20:34.39 | *** join/#asterisk T0tal (~t0tal@sonicwall.dcdi.net) |
| 20:34.54 | flewid | i was having weird errors with tiff 3.7.0, complaining about not being able to find jpeg |
| 20:34.57 | flewid | erm, jpeg_destroy |
| 20:35.01 | flewid | nfi why tho |
| 20:35.05 | flewid | so i upgraded to 3.7.1, and it works. |
| 20:35.13 | T0tal | I have a sipura spa2000 and I am trying to find a howto on hooking it up and seting up a asterisk box. |
| 20:35.23 | T0tal | I printed out the manual draft 2 |
| 20:35.29 | T0tal | but I am having a hard time following it |
| 20:35.30 | *** join/#asterisk Josh1 (~Joshwah@adsl-8-175-108.mia.bellsouth.net) |
| 20:35.42 | T0tal | I have a 4 port fx0 card in it. |
| 20:35.58 | ManxPower | I've reached my limit of n00b questions today. I'll be in #asterisk-stable |
| 20:35.58 | flewid | sjobeck3: the next thing is to setup email -> fax |
| 20:36.01 | *** part/#asterisk ManxPower (~eric@dsl-208-164-150-160.datasync.com) |
| 20:36.02 | Nugget | heh |
| 20:36.07 | flewid | i saw someone suggested setting up the mail server to handle like .fax |
| 20:36.11 | flewid | and the subject is the # to dial |
| 20:36.17 | flewid | and it generates a .call file or something, sounds interesting |
| 20:36.23 | Josh1 | How can I make it so that it asks the person which extension they want? "For sales, please press one. For support, please press two." |
| 20:36.39 | sjobeck3 | flewid: indeed. is this being detailed on wiki? or lists? etc? |
| 20:36.48 | flewid | Josh1: look on the wiki for 'IVR' or 'extensions.conf' |
| 20:36.54 | sjobeck3 | flewid: i'll put a watch on it |
| 20:37.05 | flewid | sjobeck3: i haven't seen anyone put anything on the wiki |
| 20:37.17 | flewid | we were talking about it the other day on here, and someone said to put the mail server to accept .fax |
| 20:37.20 | flewid | and that was the end of it |
| 20:37.30 | kFuQ | http://www.pastebin.com/229291 |
| 20:37.40 | sjobeck3 | flewid: voila, your big chance to be famous |
| 20:37.46 | *** join/#asterisk Nohair (~Jez@srscomp.demon.co.uk) |
| 20:37.46 | flewid | haha |
| 20:38.09 | sjobeck3 | flewid: throw it up there, it'd be huge! huge, i say |
| 20:38.15 | flewid | oh it would be |
| 20:38.19 | flewid | but it's already doable with hylafax |
| 20:38.25 | flewid | so i dunno if it's worth reinventing the wheel |
| 20:38.25 | sjobeck3 | flewid: no hyperbole intended |
| 20:38.33 | sjobeck3 | flewid: right, right |
| 20:38.42 | Nohair | Hi any one know what Didn't finish caller-id spill mean ? |
| 20:39.12 | flewid | kruq: you're not setting the CID |
| 20:39.20 | flewid | erm CNAM right before it dials the extensions |
| 20:39.45 | Nohair | its on a zaptel interface |
| 20:40.56 | Nohair | any 1 setup uk caller id |
| 20:41.19 | flewid | dammit outta smokes and coffee |
| 20:41.54 | Nohair | any 1 setup uk caller id |
| 20:42.01 | sjobeck3 | flewid: that stuff aint good for you. |
| 20:42.34 | flewid | neither is staring at a computer monitor for 20 hours a day |
| 20:42.36 | flewid | but we all do it :) |
| 20:43.22 | Jameno123 | argh! /me pulls my hair out :( |
| 20:43.25 | *** join/#asterisk nullogic (~nullogic@66.240.64.121) |
| 20:44.01 | Jameno123 | exten => s,6,Queue(queue_general|t,T|||300) |
| 20:44.04 | sjobeck3 | flewid: that hasnt been proven yet. (has it?) |
| 20:44.13 | Nohair | any 1 setup uk caller id |
| 20:44.19 | Jameno123 | why would that return in 15sec instead of 300 seconds? |
| 20:45.51 | Jameno123 | shouldnt the 300 override the queues.conf timeout= ? |
| 20:46.30 | Jameno123 | at least from my understanding from (page=Asterisk+cmd+Queue) on the wiki |
| 20:47.25 | *** join/#asterisk doughecka_ (~Doug@adsl-18-82-212.sdf.bellsouth.net) |
| 20:48.39 | Nohair | can any 1 help with uk callerid |
| 20:50.15 | Jameno123 | timeout = 15 retry = 5 ... hrm. |
| 20:51.13 | Swiff | I can't figure out how to use the SetMusicOnHold and Dial applications together? I want it to play music, instead of the dial-in tones |
| 20:51.51 | bkw_ | file |
| 20:51.54 | bkw_ | oh file |
| 20:51.55 | bkw_ | wake up |
| 20:52.39 | Jameno123 | Swiff, exten => s,6,Dial(${ARG2},15,tm) |
| 20:53.10 | Jameno123 | dial takes a 3rd paramter ;) see: http://www.voip-info.org/tiki-index.php?page=Asterisk+cmd+Dial |
| 20:53.11 | Swiff | Jameno123: and it will take the "default" class? |
| 20:53.25 | Nohair | can any 1 help with uk callerid |
| 20:54.20 | Jameno123 | hrmpf, ok, i just completely removed timeout=15 from my queues.conf, i set it to 5000 and it STILL drops my call @ 15sec |
| 20:54.22 | Jameno123 | wtf! ? |
| 20:56.07 | Swiff | Jameno123: Should i combine it with the MusicOnHold(), in that case, should the MusicOnHold() be before or after Dial? |
| 20:57.29 | *** join/#asterisk {^DaNi^} (~dcp@193.127.8.2) |
| 20:58.47 | Nohair | any 1 in the uk/? |
| 21:00.08 | Jameno123 | Swiff, i dunno about that, id think youd use SetMusicOnHold(class) |
| 21:00.21 | Swiff | yes, before the dial, and leave out the musiconhold() |
| 21:00.30 | Swiff | it's apparently better to try around :D |
| 21:01.06 | InfraRed | Nohair: yes |
| 21:01.08 | *** join/#asterisk shepherd (matt@pcp0011024458pcs.huntsv01.al.comcast.net) |
| 21:01.16 | InfraRed | plenty of people are in the uk |
| 21:01.17 | InfraRed | :) |
| 21:01.17 | Jameno123 | set the music on hold class, dial w/ paramter 'm' |
| 21:01.29 | Jameno123 | thats all you should need |
| 21:02.35 | Nohair | infared Do you use zaptel cards? |
| 21:02.43 | Druken | classical music should be outlawed on MOH |
| 21:02.59 | sjobeck3 | druken: here here |
| 21:03.23 | Druken | :) |
| 21:04.55 | samueltc | hey, anyone can suggest me switch to use for a ~100 phones implentation (no PoE) |
| 21:05.13 | samueltc | will be used for PCs too (!) |
| 21:05.57 | Druken | a switch ? |
| 21:06.03 | Druken | like a networking switch ? |
| 21:06.37 | samueltc | eh! yes.. ethernet switch :) |
| 21:06.57 | iMediax | one w/ vlan support at least |
| 21:07.40 | samueltc | why should I use vlan? |
| 21:07.56 | *** join/#asterisk rvhi (~rv@mail.o-matrix.org) |
| 21:08.02 | rvhi | hi |
| 21:08.12 | iMediax | seperate pc traffic from phone traffic |
| 21:08.13 | samueltc | each phone/pc will be on the same cable |
| 21:08.20 | iMediax | eww |
| 21:09.10 | Druken | the same cable? |
| 21:09.33 | iMediax | lol |
| 21:09.58 | samueltc | i mean, i'll run one cable for each pc/phone... no? |
| 21:10.06 | rvhi | anyone knows how to define multiple numbers in extension field of extension.conf? |
| 21:10.25 | iMediax | u want to seperate your phone trafiic from your pc traffic |
| 21:10.47 | rvhi | e.g. have 234, 258, 289 having the same action |
| 21:11.01 | samueltc | QoS should priorize the voip traffic... |
| 21:11.26 | DrPete | in the iax.conf what should I set the bandwidth too, its low by default I have a 768 dsl connection |
| 21:11.36 | iMediax | sure if you can use QoS on both ends |
| 21:11.54 | rvhi | something like? exten => _(234|258|289),1,Dosomething |
| 21:12.20 | Druken | on a setup that large, why not just run two seperate networks? |
| 21:12.57 | iMediax | thats what I would do |
| 21:13.19 | samueltc | hmm ok... |
| 21:14.04 | samueltc | i was thinking it would be 'cool' to have one cable, one switch array, ... |
| 21:14.04 | samueltc | eh |
| 21:14.12 | Druken | obvoiusly if you can buy 100 phones, your not buying shit, so why chince out on the network? |
| 21:14.23 | *** join/#asterisk aYCa\ (~UsaMan@mstr195175-30087.dial-in.ttnet.net.tr) |
| 21:14.55 | Druken | it'd look even more impressive with two isolated networks |
| 21:14.57 | Druken | hehehe |
| 21:15.26 | flewid | that's what i've setup |
| 21:15.34 | Yatta | if i get a status Unmonitored ... does it mean * is still working ? |
| 21:15.35 | flewid | one net for phones, one net for workers surfing |
| 21:15.48 | flewid | Yatta: means you don't have a 'qualify = 100' or whatever for that user |
| 21:16.46 | samueltc | hmm ok nice, i'll do that way.. ! |
| 21:17.38 | iMediax | CYA |
| 21:17.39 | *** join/#asterisk Qwell (~north@70-32-102-18.ontrca.adelphia.net) |
| 21:17.42 | Yatta | ok |
| 21:17.56 | samueltc | iMediax: i'm not leaving |
| 21:18.07 | flewid | anyone here use the PA168S chipset with iax? |
| 21:18.16 | Yatta | hey how can i make iaxphone use a differnent port?? |
| 21:19.04 | Yatta | i eman iaxcomm .. |
| 21:19.05 | iMediax | cover your ass |
| 21:19.21 | iMediax | on the 2 networks ... |
| 21:19.29 | samueltc | yes |
| 21:19.38 | samueltc | then I can use some 'cheap' switch |
| 21:20.26 | samueltc | some 300$ 24ports dlink switch |
| 21:21.05 | modulus_ | my gawd i setup a fbsd4.2 box for ssh access to database 4 years ago |
| 21:21.14 | modulus_ | and i rebooted it for the first time today |
| 21:21.52 | flewid | nice |
| 21:22.00 | Druken | why a reboot? |
| 21:22.11 | modulus_ | it went offline |
| 21:22.14 | Damin | msg shido6 Yeah.. I managed to fix it. Apparently, all calls from Zap channels are delivered to the "s" extension in the context of your choice, so I needed to add "s/2166192000,1,Hangup" to match. Now it works fine. |
| 21:22.49 | Jameno123 | who knows ANYthing about queues? i can not get mine to work properly... |
| 21:23.09 | flewid | so... nobody is testing with the pa-168? |
| 21:23.22 | Damin | Jameno123: Describe the problem, and post config files to pastebin.ca please.. |
| 21:23.27 | Druken | queues are fun |
| 21:24.06 | Druken | Damin: yeah, of course zap's are put to s, how are you going to get did info from a normal telco line? |
| 21:24.42 | Chuji | Any perlies around? |
| 21:24.49 | Chuji | why put a bang in front of a variable? |
| 21:24.50 | modulus_ | yup |
| 21:24.52 | Chuji | if (!$carddata->{number}) |
| 21:25.02 | modulus_ | returns true if eval is false |
| 21:25.05 | flewid | if not a number? |
| 21:25.07 | flewid | not sure |
| 21:25.13 | flewid | not a programmer :( |
| 21:25.20 | Damin | Druken: Callerid.. |
| 21:25.28 | modulus_ | or returns true if no return (NULL) |
| 21:25.35 | Damin | Druken: I guess I'm spoiled by PRI and IAX2 ;) |
| 21:25.36 | Chuji | modulus_ : OK, thanks |
| 21:25.46 | Druken | Damin: :) |
| 21:25.59 | Druken | i wish i was spoilt by pri |
| 21:26.08 | flewid | i wish i had a pri into my house |
| 21:26.19 | Chuji | modulus_ : And what if the bang wasn't there? |
| 21:26.26 | Damin | Druken: That's why this works.. |
| 21:26.26 | Damin | exten => s/2164104542,3,Playback(ed) |
| 21:26.27 | Damin | exten => s/2164104542,4,Playback(is-a-very-very-gay-boy) |
| 21:26.27 | Damin | exten => s/2164104542,5,Playback(moo2) |
| 21:26.56 | Damin | PRI to the house is totally overrated. Just get good quality broadband, and put the PRI in at the office. ;) |
| 21:27.10 | flewid | Damin: my house is my office |
| 21:27.11 | Druken | make donations to my pri fund, druken@cspf.org |
| 21:27.11 | flewid | :) |
| 21:27.23 | Druken | i take payments via paypal |
| 21:27.29 | modulus_ | chuji, then the if checks to see if there's a valu (not NULL) |
| 21:27.39 | flewid | i work from home, and our data centre is 3000km's away |
| 21:27.46 | flewid | it would be nice to test with a pri local to me |
| 21:27.53 | flewid | but i don't think my boss will pay for a t1 to my house |
| 21:27.54 | flewid | =) |
| 21:28.20 | Chuji | modulus_ : So it's just a "not"? |
| 21:28.41 | modulus_ | correct |
| 21:28.51 | flewid | wow i was right |
| 21:28.52 | flewid | :) |
| 21:29.02 | flewid | 18 years of looking at sourcecode at least i've learned something :) |
| 21:29.08 | flewid | Chuji: amen. |
| 21:29.14 | Jameno123 | Damin, http://pastebin.ca/4268 simply what happens is no matter which values i specify in timeout= for queue_general, it wont ovverride when i pass the timeout to cmd queue(). |
| 21:29.35 | Jameno123 | if i take out timeout= it assumes a default of 15, and drops my call back to the inbound context |
| 21:29.57 | Jameno123 | if i dont pass timeout, it keeps it in the queue 'forever' |
| 21:30.17 | Jameno123 | if i do pass a timeout, it doesnt abide by it, and simply ignores what ive passed. |
| 21:30.58 | *** join/#asterisk Dr-Linux (~sshah@202.125.141.6) |
| 21:31.04 | JerJer | flewid: no luck :( RxFax always causes an error with my fax machine and TxFax never seems to make my fax machine print anything out |
| 21:31.12 | flewid | hmm |
| 21:31.17 | flewid | sec i'll show you the output from my txfax |
| 21:33.03 | Dr-Linux | tzanger |
| 21:33.07 | Damin | Jamneo: Can you post some console debug showing the call progress? |
| 21:33.35 | Dr-Linux | SuPrSluG_ |
| 21:33.43 | SuPrSluG_ | yes |
| 21:33.53 | SuPrSluG_ | u get it to work? |
| 21:34.10 | flewid | http://pastebin.ca/4269 |
| 21:34.16 | flewid | JerJer: ^^^ |
| 21:34.27 | Damin | Jameno123: I've used this before |
| 21:34.28 | Damin | [queues] |
| 21:34.28 | Damin | exten => 1,1,Queue(operations) |
| 21:34.28 | Damin | exten => 2,1,Queue(accounting) |
| 21:34.35 | Yatta | me again.... I'm getting No such extension '393547846' in context 'local' when i try to dial 393547846 |
| 21:34.45 | Damin | But they stay in the queue forever that way.. |
| 21:35.14 | Jameno123 | Damin, ya :( same prob here, i cant leave them in the queue forever |
| 21:35.17 | *** join/#asterisk sivana (~richard@northbay-pppoe-211.vianet.ca) |
| 21:35.25 | Damin | Jameno123: I think what you are getting at is that you want them to pop OUT of the queue and go somewhere after a short period of time, right? I.E. You don't want them to be in the call queue for longer than 90 seconds? |
| 21:35.30 | flewid | ; this next line sets it to call techqueue1 |
| 21:35.30 | flewid | ; with no retries on timeout (n) |
| 21:35.30 | flewid | ; without allowing caller to hangup with * (H) |
| 21:35.30 | flewid | ; with allowing called user to transfer (t) |
| 21:35.31 | flewid | ; Queue(queuename|options|optionalurl|announceoverride|timeout) |
| 21:35.31 | flewid | ; example: Queue(dave|t|||45) |
| 21:35.33 | flewid | exten => s,9,Queue(techqueue1|nt|||20) |
| 21:35.46 | iMediax | Yatta: what is your extension under [local] |
| 21:36.07 | sivana | pastbin.ca |
| 21:36.09 | sivana | pastebin.ca |
| 21:36.13 | wankel | any idea why normal calls between 7960s work but conferencing gives "no path to translate"? |
| 21:36.22 | Druken | ~pastebin |
| 21:36.23 | jbot | i guess pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca |
| 21:36.24 | Damin | Jameno123: I may be mistaken, but I think the "timeout" parameter is the number of seconds for app_queue to try and deliver the call before it re-starts... |
| 21:37.30 | kpfleming | wankel: what codec are you using? |
| 21:37.38 | Jameno123 | Damin, http://pastebin.ca/4270 here's the status |
| 21:37.51 | Damin | Jameno123: Try adding an "n" to the Queue parameters: |
| 21:37.52 | Damin | 'n' -- no retries on the timeout; will exit this application and go to the next step. |
| 21:37.54 | Yatta | iMediax, i hade include => iaxtel700 |
| 21:38.27 | iMediax | ok extensions under that then |
| 21:38.34 | Jameno123 | hrmm, ok, but its not even timing out, is the prob :) |
| 21:38.40 | Jameno123 | until the timeout= is being reached |
| 21:39.09 | kpfleming | isn't that what it's supposed to do? |
| 21:39.38 | JerJer | flewid: yes i get that too.. but the tiff file has one row of the fax in it |
| 21:39.39 | Jameno123 | Damin, ie: if timeout=60, and i specify 30, should it not return @ 30 seconds and then drop it into voicemail as instructed? |
| 21:39.52 | flewid | JerJer: what versions? |
| 21:40.03 | JerJer | newest as of right this second |
| 21:40.09 | flewid | JerJer: are you running asterisk as a user or root? |
| 21:40.20 | kpfleming | Jameno123: you cannot set the master timeout shorter than the ringing timeout in queues.conf... it will always ring the agents for that long (at least) |
| 21:40.20 | Jameno123 | instead, if waits until 60 seconds, and then returns.. or for example if i pass 300 seconds, and timeout=15 it hangs up at 15 seconds, and returns to context incoming |
| 21:40.30 | JerJer | flewid: root |
| 21:40.35 | flewid | interesting |
| 21:40.41 | flewid | i had a similar problem but i was running as user |
| 21:40.50 | flewid | i had to own the /var/spool/asterisk-fax directory to asterisk:asterisk |
| 21:41.14 | flewid | kpfleming: was it you I was talking to about the PA-168 chipset a few days/week ago? |
| 21:41.16 | Jameno123 | kpfleming, ya, i assumed so also, however, it doesnt care :) if timeout=15 (which it should be), it never 'retries' until it gets to the 300 seconds i passed (if i pass 300 seconds) |
| 21:41.20 | kpfleming | flewid: no |
| 21:41.28 | flewid | oky |
| 21:41.55 | Jameno123 | like i tired Queue(queue_general.....300) w/ timeout=15 |
| 21:41.58 | kpfleming | Jameno123: what do you mean by "retry" then? if you set timeout=15, what is your "retry" set to? |
| 21:42.00 | Jameno123 | it hangs up @ 15 seconds. |
| 21:42.03 | flewid | JerJer: you using that 'mailfax' script too? |
| 21:42.06 | flewid | i can paste the one i'm using |
| 21:42.07 | JerJer | i'm saving to /tmp |
| 21:42.08 | Jameno123 | retry=5 |
| 21:42.09 | kpfleming | you have "n' in your queue params then, take it ou |
| 21:42.11 | flewid | i modified it a bit |
| 21:42.22 | Jameno123 | kpfleming, no i dont :) i have t,T |
| 21:42.48 | Jameno123 | lets say: exten => s,6,Queue(queue_general|t,T|||300) |
| 21:42.53 | Jameno123 | timout=15, retries=5 |
| 21:42.59 | flewid | http://pastebin.ca/4271 |
| 21:43.05 | kpfleming | why do you have comman between t and T? that's wrong |
| 21:43.16 | *** join/#asterisk nullogic (~nullogic@66.240.64.121) |
| 21:43.24 | kpfleming | it's ignoring your 300, because you have specified too many arguments to Queue |
| 21:43.24 | Jameno123 | kpfleming, tried to remove it, though the wiki says put it there |
| 21:43.31 | kpfleming | then it's wrong |
| 21:43.39 | Jameno123 | even so |
| 21:43.44 | Jameno123 | <PROTECTED> |
| 21:43.48 | Jameno123 | does the same thing. |
| 21:43.58 | Jameno123 | 'drops back to context=incoming, @ 15 seconds |
| 21:44.02 | Jameno123 | if timeout=15 |
| 21:44.12 | Damin | Does anyone have a recording of Allison saying the name "renee"? |
| 21:44.14 | Jameno123 | is that still too many args? |
| 21:44.14 | JerJer | Bad rows 29 |
| 21:44.26 | JerJer | Image size (bytes) 0 |
| 21:44.34 | sivana | has anyone written a user's manual for the voicemail? |
| 21:44.42 | flewid | sivana: search the mailing list |
| 21:44.46 | flewid | someone posted a pdf a few weeks ago |
| 21:44.59 | flewid | JerJer: weird |
| 21:45.10 | kpfleming | Jameno123: no, that is the correct number of args, a total of 4 | characters between them is correct |
| 21:45.21 | Jameno123 | from my understanding timeout= call/ring timeout on the phones. |
| 21:45.30 | Jameno123 | how long it should ring each agent/member, correct? |
| 21:45.48 | Jameno123 | retry's is how many times it restarts trying to retry the call to the agents |
| 21:46.04 | Jameno123 | timeout * retry has to be larger than the timeout i pass, otherwise, it will return too soon, no? |
| 21:46.09 | JerJer | flewid: can you send a tiff file that you have received with RxFax? i want something i know will work with TxFax |
| 21:46.10 | sivana | where is the mailling list search? |
| 21:46.14 | kpfleming | no, that is wrong |
| 21:46.22 | flewid | JerJer: sure sec |
| 21:46.22 | sivana | ~mailist |
| 21:46.24 | kpfleming | timeout= is the amount of time it will try to ring an agent/agents |
| 21:46.31 | kpfleming | retry= is the amount of time it will wait between retries |
| 21:46.45 | kpfleming | the timeout specified on the Queue command is the max time the caller will stay in the queue |
| 21:47.16 | Jameno123 | so timeout specified in queue() must be LARGER than timeout= ? |
| 21:47.31 | kpfleming | yes, absolutely, or the timeout= will take precedence |
| 21:47.44 | Jameno123 | thats what i thought, let me try this again :( give me a few mins |
| 21:47.47 | Jameno123 | lets see what happens |
| 21:47.49 | kpfleming | it will not check the Queue() timeout until after it has tried to ring an agent (it won't short circuit the ring) |
| 21:48.20 | Jameno123 | so i have |
| 21:48.26 | Jameno123 | exten => s,6,Queue(queue_general|tT|||60) |
| 21:48.56 | Jameno123 | [queue_general] && timeout=15 && retry=5 |
| 21:49.13 | Jameno123 | that seems sufficient? that should wait 60 seconds in the queue, correct? |
| 21:50.06 | gambolputty | Is there a way to get the text contents of a file into an * variable? |
| 21:51.05 | Jameno123 | show queues reporting 32 seconds :) nice |
| 21:51.09 | Jameno123 | lets see if it kills me @ 60 |
| 21:51.44 | Jameno123 | 1:18 :( |
| 21:51.46 | *** join/#asterisk implicit (~implicit@ip68-5-148-1.oc.oc.cox.net) |
| 21:52.02 | Jameno123 | 1:30 ... i guess thats good |
| 21:52.20 | Jameno123 | killed me at 90, hrm |
| 21:52.39 | flewid | i wanna get an fm card so i can make a neat little agi that lets people pick the type of music to listen to |
| 21:52.48 | flewid | or a tonne of categorized public domain mp3's :) |
| 21:52.53 | Jameno123 | opps, sec, :) heh, my mistake i accidently left queues timeout=90 |
| 21:53.01 | Jameno123 | changed to 15, as stated |
| 21:53.02 | sivana | flewid: where's the mailing list search facility? |
| 21:53.08 | flewid | sivana: google :) |
| 21:53.15 | flewid | sivana: site:lists.digium.com <query> |
| 21:55.17 | Jameno123 | kpfleming, Damin: awsome, it works now, thanks for your help... it dropped me back to incoming @ 1:02 :) perfect. |
| 21:55.24 | wankel | kpfleming: sorry, got a call. all of the 7960s are using g729a on all lines. |
| 21:55.27 | kpfleming | no problem |
| 21:55.39 | sivana | flewid: thanks |
| 21:55.40 | kpfleming | wankel: cisco phones will not conference unless you are using G.711 |
| 21:55.43 | flewid | sivana: np |
| 21:55.46 | Jameno123 | guess we should edit the wiki :) |
| 21:55.50 | kpfleming | yes, please do |
| 21:56.26 | wankel | kpfleming: thanks |
| 21:56.33 | kpfleming | welcome |
| 21:58.17 | Damin | Jameno123: No problem.. Just glad that I could contribute.. |
| 21:58.17 | Nugget | there's a trick to making the cisco phones conference g729, but I don't remember what it is |
| 21:58.42 | *** join/#asterisk ElecTro (turk_G_irL@mstr195175-8509.dial-in.ttnet.net.tr) |
| 21:58.59 | *** join/#asterisk anthm (~anthm@CPE-69-76-83-52.wi.rr.com) |
| 21:58.59 | *** mode/#asterisk [+o anthm] by ChanServ |
| 21:59.05 | *** join/#asterisk ckruetze (~ckruetze@i3ED63B25.versanet.de) |
| 22:00.38 | kpfleming | Nugget: would be good to find out, we were told by Cisco it won't happen |
| 22:01.00 | Jameno123 | pfft! oops, need to modify that page again :) it removed the [ ] around the context names |
| 22:02.03 | Darwin_35 | bkw you around ? |
| 22:02.08 | Jameno123 | hrm... i forget tikiwiki syntax to put the [ ] back in... is it just double [[ ? |
| 22:02.47 | Nugget | I know it's come up in the channel here. It's got something to do with tricking the phone into thinking the different lines are different servers, because the licensing is per account or something |
| 22:02.53 | Nugget | sorry I don't remember more |
| 22:04.04 | Damin | Is there a way to actually search Mantis for keywords and get a list of bugreports that match that keyword? |
| 22:05.38 | wankel | kpfleming: will asterisk rewrite the SIP and do transcoding for 7960 conferences from 711 to 729? |
| 22:06.00 | kpfleming | wankel: no, i don't think thats possible |
| 22:06.26 | flewid | in iax.conf should it be 'port=5036' or 'bindport=4569' |
| 22:06.26 | *** join/#asterisk NoCarrier (~NoCarrier@67.132.43.2) |
| 22:06.27 | flewid | ? |
| 22:06.37 | flewid | i was told to use port=5036, but everything else says 4569 |
| 22:06.49 | JerJer | 5036 is for iax version 1 |
| 22:06.51 | wankel | kpfleming: well, you'd have to proxy the rtp and transcode and stuff... yuck. |
| 22:06.54 | Darwin_35 | where is bkw when you need him |
| 22:06.56 | JerJer | 4569 is for the current iax version 2 |
| 22:07.00 | flewid | ah |
| 22:07.17 | Darwin_35 | where is the new fax ware |
| 22:07.22 | kpfleming | wankel: the thing is, asterisk does not know the phone is conferencing, it thinks it is just handling two calls |
| 22:07.23 | NoCarrier | . |
| 22:07.44 | JerJer | , |
| 22:07.51 | v-office | +++ |
| 22:08.05 | wankel | kpfleming: yeah, but it could transcode any rtp stream if it rewrites the SIP packets |
| 22:11.01 | file[laptop] | oh baby baby gimme gimme your love |
| 22:11.03 | file[laptop] | gimme gimme a smile |
| 22:11.09 | file[laptop] | I want you to gimme your body babe and stay with me for awhile |
| 22:11.13 | file[laptop] | so come on gimme gimme gimme your love |
| 22:11.27 | Dr-Linux | file[laptop]: are you okey ? :S |
| 22:11.38 | file[laptop] | I am perfectly fine |
| 22:12.38 | Elshar | How would I increase the gain/volume on an outgoing call that uses a sip phone to call an asterisk box which uses iax to forward to an iax provider? |
| 22:12.42 | iMediax | file is being his norm |
| 22:13.25 | flewid | <PROTECTED> |
| 22:13.25 | flewid | Jan 15 17:13:09 NOTICE[5083]: chan_iax2.c:6747 iax2_poke_noanswer: Peer 'mattipweb' is now UNREACHABLE! Time: 0 |
| 22:13.29 | file[laptop] | Elshar: asterisk doesn't have anything to do with that, muck about with your phone |
| 22:13.34 | flewid | anyone seen iax2 clients do that before? automagically go unreachable |
| 22:14.14 | Elshar | file: That's what I was afraid of. :P |
| 22:14.24 | JerJer | flewid: is there a firewall in the way? |
| 22:14.26 | Elshar | file: The sip adapter is one of those linksys pap2 phones. :/ |
| 22:14.30 | flewid | JerJer: nope |
| 22:14.35 | file[laptop] | Elshar: it's adjustable on there |
| 22:14.36 | *** join/#asterisk zotz (~zotz@24.244.133.136) |
| 22:14.41 | JerJer | what do have qualify set to? |
| 22:14.41 | flewid | JerJer: well, in that particular connection there is, but with my home phone there isn't. |
| 22:14.44 | Elshar | file: Yea, if I can find the damned setting. :) |
| 22:14.45 | flewid | works fine with iaxy's |
| 22:14.48 | flewid | but not this netweb |
| 22:14.58 | flewid | 800 |
| 22:15.16 | JerJer | i've never liked qualify |
| 22:15.17 | file[laptop] | Elshar: Admin Login -> Advanced View -> Regional |
| 22:15.29 | file[laptop] | Elshar: FXS Port Input Gain / FXS Port Output Gain |
| 22:15.57 | Elshar | Thanks :) |
| 22:16.34 | flewid | JerJer: i think it's this phone |
| 22:16.46 | flewid | this is the last time i buy a phone from eezeephone |
| 22:16.58 | flewid | i shoulda bought the ones from china, but i didn't wanna pay shipping on that |
| 22:19.19 | *** join/#asterisk aminorex (~tony@c66.191.69.132.dul.mn.charter.com) |
| 22:22.58 | flewid | JerJer: i'm out, lemme know how the faxing goes =) |
| 22:23.04 | flewid | time to make a customized * gentoo dist |
| 22:23.06 | flewid | adios |
| 22:24.00 | Moc_ | conf is quite |
| 22:24.06 | Moc_ | quiet |
| 22:27.00 | Damin | Moc_: I'm getting Polycom SUBSCRIBE debug output to see what changed... |
| 22:29.46 | *** join/#asterisk jtodd (~jtodd@216.64.26.251) |
| 22:37.33 | *** join/#asterisk JoeLlama (snork@wbar104.lax1-4.31.117.163.lax1.dsl-verizon.net) |
| 22:37.35 | JoeLlama | Hi :) |
| 22:37.41 | JoeLlama | my goodness... |
| 22:41.16 | *** join/#asterisk sivana (~richard@northbay-pppoe-211.vianet.ca) |
| 22:41.34 | Shido6 | ? |
| 22:42.27 | *** join/#asterisk Nohair (~Jez@srscomp.demon.co.uk) |
| 22:42.31 | *** join/#asterisk FryGuy (fryguy@c-24-2-50-122.client.comcast.net) |
| 22:43.14 | Nohair | Can any one help with this message ??? Didn't finish Caller-ID spill |
| 22:44.13 | *** join/#asterisk nullogic (~nullogic@66.240.64.121) |
| 22:44.18 | *** join/#asterisk clive- (~pirch@rrba-146-74-63.telkomadsl.co.za) |
| 22:44.23 | Nohair | Can any one help with this message ??? Didn't finish Caller-ID spill |
| 22:44.49 | Nohair | any 1 using callerid with zaptel cards??? |
| 22:44.54 | JoeLlama | so like how hard is it to set up asterisk on my PC? |
| 22:45.04 | sivana | Nohair: from Digium? |
| 22:45.14 | Nohair | sivana yes |
| 22:45.23 | sivana | which card |
| 22:45.24 | JoeLlama | and how expensive for a small phone system (just a few lines) |
| 22:45.35 | Nohair | silvana Tdm400 |
| 22:45.52 | *** part/#asterisk Yatta (~yatta@port0056-adsl11.cwjamaica.com) |
| 22:46.04 | sivana | do you have callerid turned on in zapata.conf? |
| 22:46.13 | JoeLlama | I must be -m :( oh well... |
| 22:46.41 | Nohair | sivana yes but i am getting a error Didn't finish Caller-ID spill |
| 22:49.16 | Nohair | Can any one help with this message ??? Didn't finish Caller-ID spill |
| 22:49.52 | sivana | did you search google? |
| 22:50.16 | hermie | google? What the heck is that? |
| 22:50.18 | JoeLlama | so like how hard is asterisk to set up and how expensive? |
| 22:50.28 | sivana | hehe |
| 22:50.42 | [TripleF] | google is a plugin ;) |
| 22:50.52 | Nohair | yes i have searched the net dry |
| 22:50.56 | [TripleF] | used to get information on many stuff.. all available on the internet ;) |
| 22:51.09 | hermie | Nohair: really? |
| 22:51.24 | JoeLlama | hello? {echo echo echo} |
| 22:51.41 | hermie | Nohair: because I just typed "Caller-ID spill" into google |
| 22:51.53 | Nohair | and |
| 22:52.12 | sivana | ~google Caller-ID spill |
| 22:52.50 | Nohair | yes read all of those articles and they are all asking the same question |
| 22:53.40 | JoeLlama | oh of course I forgot... I quafed a potion of invisibility. Guess I'll wait until it wears off. Silly me. |
| 22:54.06 | Merlin | i have the sudden urge to play moria |
| 22:54.18 | file[laptop] | JoeLlama: we can't answer that question - it depends what you want to do, how quick you can learn, etc - just go Google, read examples, etc |
| 22:54.29 | JoeLlama | I learn quick :) |
| 22:54.37 | file[laptop] | then Google, use voip-info.org |
| 22:54.40 | file[laptop] | ~useful asterisk docs |
| 22:54.41 | jbot | useful asterisk docs is, like, it has been said that useful asterisk docs is (BOOKMARK THEM!): http://www.digium.com/index.php?menu=documentation (look at the "Unnoficial Links") and http://www.voip-info.org/wiki-Asterisk (the Wiki), and http://www.fnords.org/~eric/asterisk (ManxPower's site), and http://asteriskdocs.org, also, read all files in /usr/src/asterisk/doc |
| 22:54.43 | JoeLlama | and I'm not sure what I wanna do... I gues I'll just to read up. |
| 22:54.44 | file[laptop] | tada |
| 22:55.18 | JoeLlama | www.voip-info.org okay :) |
| 22:55.50 | JoeLlama | I used skype and although I undertand they use some proprietary protocol it works really well. |
| 22:56.00 | JoeLlama | I talk from the USA to my client in Israel. |
| 22:56.48 | Merlin | i wouldn't talk to my clients without encryption |
| 22:57.00 | JoeLlama | I think it has it but not sure. |
| 22:57.05 | JoeLlama | and yeah encryption is good. |
| 22:57.12 | JoeLlama | although for this project it's no big deal. |
| 22:57.17 | wankel | merlin: hope you don't use any cell phones to talk to them in the us :) |
| 22:57.23 | JoeLlama | heh |
| 22:57.27 | wankel | or land lines |
| 22:57.27 | *** join/#asterisk mofu (~eric@pcp04232064pcs.plyntv01.mi.comcast.net) |
| 22:57.41 | *** join/#asterisk benno2 (~benno2@host252-116.pool80117.interbusiness.it) |
| 22:58.14 | Merlin | wankel: i don't use t-mobile |
| 22:58.22 | benno2 | hi, anyone that can recommend a good (fast) PDA, possibly pocketpc to use it as a voip terminal in a wlan ? |
| 22:58.29 | Nohair | Can any one help with this message ??? Didn't finish Caller-ID spill |
| 22:59.32 | benno2 | I've seen most PDAs have the mic and speakers placed in the wrong places. (eg speaker near mic, thus lots of echo, and I have yet to see a good softphone for pda that has good echo cancellation) |
| 23:00.18 | Nohair | Can any one help with this message ??? Didn't finish Caller-ID spill |
| 23:00.31 | JoeLlama | I checked on that caller ID spill thing: http://tinyurl.com/4zsnx you saw that too I'm sure. |
| 23:02.33 | Nohair | joelama not one article tells me how to fix it or what the problem is every body is asking the same question |
| 23:06.07 | bonbon-home | anyone know if there is a directive in extensions.conf which will split a string based on delimiter? |
| 23:06.17 | sivana | Nohair: what's the exact error message? |
| 23:06.25 | *** part/#asterisk humberto (~hav@201.128.177.84) |
| 23:07.14 | *** join/#asterisk chofmann (~10333717E@p50801C09.dip.t-dialin.net) |
| 23:07.50 | Nohair | chan_zap.c:3434 zt_handle_event: Didnt finish caller-id spill. Cancelling |
| 23:07.59 | sivana | are you in the UK? |
| 23:09.20 | Nohair | yes |
| 23:09.39 | hermie | ding ding ding - there is a solution on Google |
| 23:09.45 | hermie | http://lists.digium.com/pipermail/asterisk-users/2004-September/064556.html |
| 23:09.45 | blitzrage | hi! |
| 23:10.00 | *** join/#asterisk Yatta (~yatta@port0056-adsl11.cwjamaica.com) |
| 23:10.38 | sivana | hermie: you beat me to it :P) |
| 23:10.59 | Nohair | hold on i will try |
| 23:11.01 | sjobeck3 | blitzrage: thx. you party. |
| 23:11.10 | hermie | sivana: I found that when he first said he had searched google... because I found it |
| 23:11.25 | hermie | wow, that was circumlocution |
| 23:12.56 | Yatta | if i get "Unable to negotiate codec" does that mean soemthign in my iax.conf is messed up? |
| 23:13.13 | Yatta | i can dial and hear one ring.. but then i see that error msg |
| 23:13.15 | Nohair | Nope doesnt fix it and thats how my system is setup! |
| 23:13.27 | DrPete | if i call my xlite extension from my xlite phone, should i get an engaged tone? |
| 23:15.25 | RaYmAn-B1 | DrPete: no..at least the xlite I have installed has 3 'lines'..so if you call yourself it'll ring on a second/third line |
| 23:17.40 | DrPete | RaYmAn-B1: hmm, somethins is wrong then, it registers ok, but I dont think it has been assigned an exten number |
| 23:18.23 | *** join/#asterisk Leeky (~Leeky@home.leeky.org.uk) |
| 23:18.45 | Nohair | any 1 help with the following???? |
| 23:18.52 | RaYmAn-B1 | DrPete: have you actually setup an extension for it in extensions.conf? |
| 23:18.57 | Nohair | chan_zap.c:3434 zt_handle_event: Didnt finish caller-id spill. Cancelling |
| 23:19.19 | Leeky | Nohair - inbound or outbound? |
| 23:19.45 | Nohair | inbound |
| 23:20.02 | DrPete | RaYmAn-B1: errr, prolly not |
| 23:20.28 | file[laptop] | NoCarrier: Wait(2) before you do anything with the call in the dialplan |
| 23:20.43 | Hymie | file[laptop]: do you think we have time to just sit around and wait? |
| 23:20.50 | file[laptop] | er |
| 23:20.53 | file[laptop] | Nohair |
| 23:21.09 | file[laptop] | :p |
| 23:21.27 | Nohair | sorry file[laptop] what was that |
| 23:21.28 | Leeky | Nohair - if it's an analog line, it'll need a couple of rings to get all the CLI info... actually what file said! |
| 23:21.49 | Nohair | ok its a bt phone in the uk |
| 23:22.16 | Leeky | Nohair - Yeah, with a PSTN phone, the CLI is sent with the first (?) ring. |
| 23:22.24 | DrPete | I am in the UK :)P |
| 23:22.33 | Leeky | Nohair - If you just let it pick up, it'll never get the CLI info. |
| 23:22.42 | Nohair | Leeky in the UK sent before the phone rings |
| 23:23.02 | *** part/#asterisk benno2 (~benno2@host252-116.pool80117.interbusiness.it) |
| 23:23.27 | Leeky | Nohair - Still needs a Wait() before answering to get CLI :) |
| 23:24.36 | *** join/#asterisk ManxPower (~eric@dsl-208-164-150-160.datasync.com) |
| 23:24.55 | ManxPower | *grumble* This PC I'm working on fails memtest |
| 23:25.07 | file[laptop] | F! |
| 23:25.11 | *** join/#asterisk Gh0sty (~Ghosty@ip-81-11-138-173.dsl.scarlet.be) |
| 23:25.19 | mquin | heh |
| 23:25.24 | blitzrage | sjobeck3: no problem, glad I could help |
| 23:25.24 | ManxPower | I don't suppose anyone out there has 32MB EDO SIMMS they can let me have. I'll pay for overnight shipping to be sent out on monday. |
| 23:25.45 | blitzrage | ManxPower: eeeesh... ummmm, not here unfortunately |
| 23:25.53 | hermie | ManxPower: I might |
| 23:26.15 | mquin | the only EDO I've got has been on my keychain for three years, so I wouldn't expect it to work |
| 23:26.16 | ManxPower | blitzrage: These are 200Mhz PCs that will be NAT/Router/QoS/VPN boxes. |
| 23:26.28 | DrPete | lol |
| 23:26.30 | hermie | ManxPower: if I haven't thrown it out... |
| 23:26.30 | ManxPower | non-EDO will be OK too. |
| 23:27.13 | ManxPower | I think these are "72-pin SIMMS" |
| 23:27.22 | ManxPower | Not SDRAM or anything like that. |
| 23:27.39 | DrPete | I have some somewhere, but I am in the UK :( |
| 23:27.40 | file[laptop] | this season of Andromeda is so screwed up... |
| 23:28.07 | ManxPower | Aha! It might not have locked up afterall. |
| 23:28.15 | Leeky | phew |
| 23:28.20 | DrPete | lol |
| 23:28.23 | ManxPower | Still would not hurt to have 64MB in it rather than the current 48M |
| 23:30.54 | Nohair | any 1 help with the following???? |
| 23:31.01 | Nohair | chan_zap.c:3434 zt_handle_event: Didnt finish caller-id spill. Cancelling |
| 23:31.24 | Leeky | Nohair - the Wait(2) made no difference then? |
| 23:31.39 | Nohair | leeky nope |
| 23:32.03 | *** join/#asterisk bprice20 (~bprice20@alb-69-200-26-118.nycap.rr.com) |
| 23:32.07 | *** join/#asterisk dnc (~duncan@213.244.225.42) |
| 23:32.14 | hermie | Nohair: did you set usecallerid=yes? |
| 23:32.15 | bprice20 | thx for the heads up on simple telecom's free accounts |
| 23:32.26 | Nohair | hermie yes |
| 23:32.38 | bprice20 | they allowed me to shot the boss asterisk up and running albeit w/ our 1800 number @ work |
| 23:32.42 | *** join/#asterisk xachen (justin@gangsta.citelnetworks.com) |
| 23:32.51 | bprice20 | something I could not get working w/ fwd or iaxtel |
| 23:32.53 | xachen | Hey |
| 23:33.07 | xachen | WOuld somebody be able to help me with how to make a basic incoming call system? |
| 23:33.17 | xachen | we can go out. but when we go in it gives the busy tone |
| 23:33.18 | Leeky | xachen - what do you need? |
| 23:33.32 | xachen | well |
| 23:33.36 | xachen | we are using SIP completely |
| 23:33.39 | Leeky | *nods* |
| 23:33.40 | xachen | with broadvoicew |
| 23:33.45 | xachen | we can connect to broadvoice |
| 23:33.47 | xachen | and make calls out |
| 23:33.52 | xachen | but we can't get calls to come in |
| 23:33.55 | xachen | we get the busy signal |
| 23:33.59 | *** join/#asterisk Qwell (~north@70-32-102-18.ontrca.adelphia.net) |
| 23:34.06 | xachen | so just a system that they call in and wait for one of us to pick up |
| 23:34.34 | *** join/#asterisk chayewala (~trashbmxe@cs2416064-46.houston.rr.com) |
| 23:34.41 | chayewala | hello |
| 23:34.45 | Josh1 | :P |
| 23:34.57 | Leeky | have you tried monitoring the SIP connection from broadvoice - sip debug peer xxxx (from CLI) ? |
| 23:34.59 | chayewala | how can I play somethinging on a called out line |
| 23:35.11 | chayewala | I want to be able to play interactive msgs |
| 23:35.15 | chayewala | when dialing out |
| 23:35.27 | Nohair | any 1 help with the following???? |
| 23:35.29 | chayewala | similar to telemarketing |
| 23:35.31 | Nohair | chan_zap.c:3434 zt_handle_event: Didnt finish caller-id spill. Cancelling |
| 23:35.35 | xachen | I get |
| 23:35.35 | xachen | SIP Debugging Enabled for IP: 147.135.4.128:5060 |
| 23:35.59 | chayewala | hello |
| 23:36.15 | chayewala | any tips |
| 23:36.20 | bprice20 | chaywala |
| 23:36.23 | chayewala | yes |
| 23:36.24 | bprice20 | I'll hel you |
| 23:36.25 | JerJer | chayewala: make a custom dialer application |
| 23:36.29 | Leeky | xachen - Now make an inbound call and check that you get a whole load of debug stuff - that'll check that at least the call is reaching your * box. |
| 23:36.30 | chayewala | ok |
| 23:36.43 | samueltc | hey, for production environnment i'm planning to buy: box, tdm cards, ups, switch, voip phones. Did I miss something? |
| 23:36.47 | xachen | its reaching it |
| 23:37.03 | chofmann | @ManxPower: Have 4 SIMMS (72pin) here that I don't need any more. I think two are 32 MB each and 2 are 16 MB each but I can be wrong. |
| 23:37.34 | hermie | samueltc: a nice rack and some zip ties? |
| 23:37.46 | chofmann | Seems to be EDO and NOT SDRAM. |
| 23:37.52 | *** join/#asterisk BoRiS (~boris@S01060040ca1e5b54.wp.shawcable.net) |
| 23:38.05 | samueltc | hermie: hehe ok |
| 23:38.32 | Leeky | xachen - You should just set up a context in the sip.conf and create a section in extensions.conf with that name. |
| 23:38.49 | ManxPower | chofmann: Let me check another source a min |
| 23:38.53 | hermie | samueltc: but seriously, how about a 66 block |
| 23:39.01 | *** join/#asterisk SuPrSluG_ (~SuPrSluG@pool-70-16-40-16.buff.east.verizon.net) |
| 23:39.08 | Leeky | xachen - And then do a exten => s,1,Answer ... which will perform the actions whenever a call comes in on that context, I believe...? |
| 23:39.10 | xachen | http://pastebin.ca/4277 |
| 23:39.22 | xachen | ok |
| 23:39.25 | hermie | samueltc: and if you are serious about production, get a cheap channel bank and T1 card and not TDM |
| 23:39.36 | xachen | so just exten => s,1,Answer ?? |
| 23:39.44 | *** part/#asterisk markit (~marco@host119-245.pool8172.interbusiness.it) |
| 23:39.46 | hermie | samueltc: Digium T1 is better than TDM stuff |
| 23:39.52 | Leeky | xachen - And then ... Dial(whatever) etc... |
| 23:40.07 | samueltc | hermie: at this time we only need 6 or 7 lines |
| 23:40.30 | Leeky | xachen - I'm still relatively new to Asterisk, so I could be wrong... but I'm pretty confident that's right :) |
| 23:40.32 | samueltc | ok! chan back<-->t1<--ast |
| 23:40.48 | wankel | phones go on the left :) |
| 23:41.11 | samueltc | wankel: your left? |
| 23:41.54 | wankel | phones<-->channel bank<-->t1 card<-->asterisk |
| 23:42.13 | chayewala | is there a telemarketing application for asterisk ? |
| 23:42.29 | hermie | samueltc: yeah, the cost is a little more, but it is way worth it, since you can add any combo of FXS and FXO w/ the right bank |
| 23:42.37 | Druken | jbot burn chayewala |
| 23:42.39 | jbot | ibot pours gasoline all over chayewala, ignites the fire, and then enjoys some toasty marshmallows with the glorious blaze |
| 23:42.40 | *** join/#asterisk clive- (~pirch@rrba-146-74-63.telkomadsl.co.za) |
| 23:42.50 | wankel | you mean to be a telemarketer and annoy people or to block telemarketers so they don't annoy you? |
| 23:43.07 | chayewala | neither |
| 23:43.10 | Leeky | xachen - any luck with that? |
| 23:43.18 | chayewala | my requirement is very simple |
| 23:43.18 | samueltc | wankel: where is the pots? :) |
| 23:43.27 | Druken | hey kram |
| 23:43.27 | samueltc | hermie: suggestion? |
| 23:43.35 | *** join/#asterisk nullogic (~nullogic@66.240.64.121) |
| 23:43.38 | chayewala | I just want to play a msg after calling out from asterisk when called party answers |
| 23:43.39 | *** join/#asterisk shmaltz (~chatzilla@chaimfried.jnet.chl.net) |
| 23:43.44 | kram | hi druken |
| 23:43.46 | wankel | oh, s/phones/pots/ if you're doing sip phones and voip termination. |
| 23:43.54 | shmaltz | bkw_ u available? |
| 23:44.08 | samueltc | wankel: ah! |
| 23:44.17 | shmaltz | any bug marshals avaiable? |
| 23:44.18 | wankel | err, sip phones and pstn termination |
| 23:44.20 | Druken | kram: how ya doing? |
| 23:44.21 | Leeky | chayewala - A message to the person you're calling? I think I've seen an example for that around somewhere... |
| 23:44.32 | chayewala | thats all |
| 23:44.38 | chayewala | small msg on each outside call |
| 23:44.40 | Druken | it can be done, yes... |
| 23:44.45 | wankel | samuel: you're doing sip phones and analog phone lines? |
| 23:44.47 | mofu | sounds like the "annoy people" option to me . . . . |
| 23:44.50 | clive- | does anyone know whats the HL and HS options in the dial string mean ? |
| 23:44.53 | wankel | mofu: yeah, it does |
| 23:44.54 | chayewala | no no |
| 23:44.59 | xachen | er |
| 23:45.00 | xachen | nope |
| 23:45.05 | wankel | i'm always annoyed when something plays a message after i answer |
| 23:45.11 | Leeky | xachen - drat... what happens? |
| 23:45.21 | chayewala | those who are calling from my asterisk box. I want both parties to hear the little advertisement |
| 23:45.31 | samueltc | wankel: yes i'm using sip phones and 7 pstn |
| 23:45.31 | Yoda-BZH`ZzZ | une bonne nuit je vous souhaite ! / A good night I wish you |
| 23:45.54 | chayewala | wankle I wont call you :-) |
| 23:46.02 | wankel | samueltc: oh, okay. pstn<-->channel bank<-->t1 card<-->asterisk<-->switch<-->phones, then :) |
| 23:46.03 | chayewala | what is your number wanke btw |
| 23:46.11 | mofu | chayewala: advertising in exchange for telecom services? |
| 23:46.18 | chayewala | yes |
| 23:46.19 | wankel | 01178129167432 |
| 23:46.20 | Druken | i would never use a service that gives the person i'm calling a advertisment |
| 23:46.21 | wankel | be sure to call lots |
| 23:46.36 | chayewala | any way |
| 23:46.45 | chayewala | looks like there is no service :-) |
| 23:47.18 | chayewala | or agi script for this purpose |
| 23:47.27 | Druken | chayewala: what you want todo is possible... but i don't like the idea behind it... so i'm not going to think about how, but i know it can be done |
| 23:47.29 | shmaltz | twisted: you avaiable? |
| 23:47.32 | Nohair | any 1 help with the following???? |
| 23:47.34 | Nohair | chan_zap.c:3434 zt_handle_event: Didnt finish caller-id spill. Cancelling |
| 23:47.40 | wankel | druken: i built something like that... i dunno, i guess 4 years ago at this point. it actually got a fairly good response. all the people who buy the calling cards at the 7-11 loved it. they're happy to call their family across the US for free. we actually had ads every 5 minutes with a quiet beep preceeding them so they could stop talking. |
| 23:48.00 | wankel | druken: all hand-coded in C against dialog libraries at the time, of course. none of this wussy AGI stuff :) |
| 23:48.06 | wankel | s/dialog/dialogic/ |
| 23:48.26 | chayewala | surprising |
| 23:49.19 | ardor | chayewala why dont you put both parties on hold as soon as they connect play the message and then bridge the channles again... |
| 23:49.29 | scubasteve | Scuba Sam! |
| 23:49.35 | ScubaSam | :) |
| 23:49.46 | chayewala | ardor - how would that be done? |
| 23:50.01 | ardor | chayewala I dont know look into music on hold. |
| 23:50.05 | chayewala | ok |
| 23:50.16 | chayewala | well |
| 23:50.20 | chayewala | thanks fols |
| 23:50.23 | chayewala | folks |
| 23:50.30 | chayewala | only jerjer knows and he is quite |
| 23:50.30 | wankel | don't call us. we'll call you. |
| 23:50.34 | chayewala | quiet |
| 23:50.55 | shmaltz | any bug marshals avialable? |
| 23:51.08 | scubasteve | Oh god, here we go with the snuggling. |
| 23:51.27 | Leeky | hehe |
| 23:52.53 | scubasteve | Anyone got pics of a Sipura 841 screen? The phone is powered off in their stock photography. |
| 23:53.34 | ManxPower | I have the phone |
| 23:53.54 | shmaltz | twisted: you availabel? |
| 23:54.03 | scubasteve | ManxPower - What's your overall impression? |
| 23:54.14 | scubasteve | ManxPower - I've read the keys are sticky, some call it rubbery... |
| 23:54.20 | *** join/#asterisk chayewala (~trashbmxe@cs2416064-46.houston.rr.com) |
| 23:54.33 | ardor | chayewala how do i do music on hold? |
| 23:54.34 | chayewala | basically what I want is a wakeup call like setup |
| 23:54.51 | ManxPower | scubasteve: The keys ARE rubber. The phone seems to be the best <$100 phone out there |
| 23:55.14 | chayewala | I hae no idea ardor |
| 23:55.14 | ManxPower | The headset port seemed flakey on this one, a second one is arriving this coming week. |
| 23:55.14 | chayewala | I have not done it |
| 23:55.16 | shmaltz | ManxPower: I think for the fact with more than one line apperarance it is the best |
| 23:55.21 | scubasteve | ManxPower - I meant rubbery as in tactile response. Some on the users and biz list called it "sticky". |
| 23:55.23 | shmaltz | what about auto answer? |
| 23:55.30 | ManxPower | I can't figure out how to adjust the handset mic and speakerphone mic gain |
| 23:55.44 | ManxPower | scubasteve: It seems OK to me. |
| 23:55.47 | shmaltz | ManxPower does it support auto answer? |
| 23:55.53 | scubasteve | ManxPower - All of the features seem to work right with *? Transfer, indicator light, yada yada ? |
| 23:55.57 | ManxPower | I don't know if it supports autoanswer or not. |
| 23:56.09 | scubasteve | shmaltz - There's no manual "yet". |
| 23:56.11 | ManxPower | scubasteve: I've not tried much of the features. I got the phone yesterday. |
| 23:56.24 | ardor | those clipcom phones cool nice |
| 23:56.31 | ardor | look nice |
| 23:56.42 | shmaltz | ManxPower: overall would you say it makes a good phone in an office? |
| 23:56.56 | scubasteve | ManxPower - Gotcha. I'm looking for something inexpensive and simple for some family on the West coast.... haven't decided on the 101 or the 841 yet... |
| 23:57.14 | ManxPower | shmaltz: Once the manual comes out and I get the second phone in we'll strongly consider it for a 40 person deployment |
| 23:57.30 | ManxPower | scubasteve: The 101 is a piece of shit compared to this. |
| 23:57.35 | samueltc | hehehehe |
| 23:57.41 | scubasteve | Yeah, makes sense. Ok "employees" here's your phone. No manual. Have fun. |
| 23:57.49 | scubasteve | ManxPower: Damn. Ok, 841 it is. |
| 23:57.49 | shmaltz | ManxPower: is it wall mountable, although that without POE I don't think I will ever want to mount it on the wall |
| 23:58.07 | ManxPower | yes, it's wall mountable as far as I can tell. |
| 23:58.35 | samueltc | ManxPower: where did you buy your sipura? |
| 23:59.05 | shmaltz | b/c the GrandStream BT 102 is suppose to be wall mountable (it even has the wholes on the back to mount it) but it lacks a latch on the handset to make sure it doesn't slike out when mounted on the wall |
| 23:59.13 | scubasteve | ManxPower just made up my mind. The 841 will be my standard "low end" endpoint. |
| 23:59.40 | ManxPower | scubasteve: Get one and try it out first. |
| 23:59.56 | shmaltz | scubasteve: in my opinion it should be, its the only on around $100 that does more than one line |
