| 00:00.31 | *** join/#asterisk korcan (~johnynum5@ip65-44-169-66.z169-44-65.customer.algx.net) |
| 00:00.58 | *** join/#asterisk korcan (~johnynum5@ip65-44-169-66.z169-44-65.customer.algx.net) |
| 00:04.40 | gamedna | is there a way to find out at runtime what the speex default configuration is? |
| 00:07.11 | *** join/#asterisk Micc_ (~quassel@c-98-225-57-96.hsd1.wa.comcast.net) |
| 00:08.05 | Micc_ | Anyone know of good call center reporting/monitoring software that works with asterisk? |
| 00:09.01 | Micc_ | I'm looking in the budget of free to under $1k |
| 00:09.20 | Micc_ | or maybe something hosted but under $20 a seat. |
| 00:09.29 | Micc_ | per month. |
| 00:09.50 | Tukeke | O_O |
| 00:09.55 | Tukeke | capitalism |
| 00:09.55 | Tukeke | :P |
| 00:11.31 | carrar | vici dial? |
| 00:17.26 | Tukeke | está escuchando: Willie Colon & Ruben Blades - Siembra - Pedro Navaja - (0:21/7:25) |
| 00:17.57 | Micc_ | that looks like a little more than they need. |
| 00:18.19 | Tukeke | O_o |
| 00:18.23 | Micc_ | they just need something simple for about 10-20 agents |
| 00:18.40 | Micc_ | Tukeke, are you rolling your eyes at me? |
| 00:19.13 | Tukeke | Micc, no |
| 00:19.55 | Micc_ | Tukeke, what are your funny faces about then? O_O ? |
| 00:19.55 | bougyman | Micc_: how about orderlystats? |
| 00:19.59 | bougyman | you should be able to get it for that. |
| 00:20.04 | bougyman | not hosted. |
| 00:20.32 | Micc_ | I think I saw an ad for that on voip-info |
| 00:22.19 | Micc_ | Hmm, that looks like it might be the ticket. I wonder if it works on multi-tenant asterisk servers. |
| 00:22.41 | bougyman | yep |
| 00:22.47 | bougyman | multi-tenant is a tad more expensive. |
| 00:22.58 | bougyman | we got an unlimited license for $3000, single tenant. |
| 00:23.07 | bougyman | but 20 user was something like $595 |
| 00:23.17 | bougyman | so still well under your $1000 |
| 00:23.59 | Tukeke | está escuchando: Cada Loco con su Tema - no se - Si Nos Fueramos Venio - (2:45/4:47) |
| 00:24.28 | Tukeke | :P |
| 00:27.26 | xheliox | Tukeke: Knock that crap off, no one cares what song you're listening to. |
| 00:27.30 | leifmadsen | Not sure how much it is, but I like QueueMetrics |
| 00:27.46 | leifmadsen | Tukeke: you should turn that off or we'll have to mute you |
| 00:27.56 | Tukeke | xheliox, -.- |
| 00:34.23 | *** join/#asterisk gamedna (~gamedna@cpe-70-125-155-74.satx.res.rr.com) |
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| 00:41.27 | Tukeke | Yankee GO HOME !!! |
| 00:44.54 | gamedna | are there any advantages to using .conf vs .ael files? |
| 00:48.50 | gamedna | any negatives to using .ael vs conf? |
| 00:53.22 | gamedna | anyone? |
| 00:57.17 | leifmadsen | gamedna: well dialplan is better supported (code wise), but if AEL works for you, there is no disadvantage. If you find a bug however, you'll have to work around it (although I'm not saying it is likely) |
| 00:57.37 | leifmadsen | Personally, I prefer dialplan, but that's maybe because I've been using it since before AEL existed |
| 00:59.07 | gamedna | leifmadsen: Thanks for the feedback. is AEL still supported by the devs? |
| 01:00.39 | *** join/#asterisk andresmujica (~andresmuj@ubuntu/member/andresmujica) |
| 01:01.48 | leifmadsen | gamedna: it is to a certain extent, but the developer who initially built it has moved onto other projects |
| 01:03.21 | gamedna | leifmadsen: thanks, that is really a good point. |
| 01:03.55 | gamedna | i know you can use the two methods together, so i guess its probably best to learn both. |
| 01:05.23 | *** join/#asterisk mroe (~anon__@unaffiliated/roe) |
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| 01:06.27 | *** join/#asterisk coppice (~chatzilla@245.168.17.210.dyn.pacific.net.hk) |
| 01:07.22 | mroe | It seems that order matters when loading wctdm and wcte12x I thought listing them in explicit order in /etc/modules would be enough to load them in the correct order. Is there a better way to ensure these modules load in the proper order? |
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| 01:09.28 | *** join/#asterisk nicoAMG (~nicoamg@201.237.49.131) |
| 01:26.59 | exothermc | trying to compile asterisk right now, but when I do make menuconfig chan_dahdi is XXX. |
| 01:27.14 | exothermc | I have dahdi compiled and installed. |
| 01:39.34 | *** part/#asterisk nny (~Scott@cpe-071-076-058-253.sc.res.rr.com) |
| 01:40.39 | p3nguin_ | Would a one minute recording of no audio have a smaller file size than a one minute recording of excessive audio (blasting rock music, jet engine, etc.) both recorded in the same format? |
| 01:41.14 | exothermc | p3nguin_: Typically should be the same. |
| 01:41.31 | exothermc | but there maybe some silence suppression which would cause no rtp to flow |
| 01:42.19 | p3nguin_ | Not sure how that would come into my equation. |
| 01:42.53 | exothermc | p3nguin_: for your understanding you should assume they are the same. |
| 01:42.56 | p3nguin_ | I just want to know if no audio is the same size as some or lots of audio. |
| 01:43.34 | p3nguin_ | How about a larger sample, maybe an hour? |
| 01:44.03 | exothermc | Doesn't matter. |
| 01:44.26 | p3nguin_ | okay. |
| 01:44.44 | exothermc | There is a packet sent multiple times per second. That packet contains the audio for that sample, no matter what is in that actual sample, the packet is the same size. |
| 01:45.41 | *** join/#asterisk mindCrime (~chatzilla@cpe-075-189-213-049.nc.res.rr.com) |
| 01:45.41 | exothermc | Now if you record that into a format other than the raw format, you may or may not have size differences based on the compression codec you are using, but that is beyond the scope of asterisk. |
| 01:45.59 | p3nguin_ | I wasn't even talking about Asterisk at all. |
| 01:46.25 | exothermc | p3nguin_: Well that is completely dependent on the compression codec. |
| 01:47.29 | p3nguin_ | So the codec could compress voices or other sounds more than no noise at all, and could actually make a voice recording of a smaller file size? |
| 01:51.26 | exothermc | likely to be the complete opposite of what you describe that would be my guess. |
| 01:52.55 | p3nguin_ | But in a raw format, substance and emptiness would be the same size. |
| 01:54.25 | p3nguin_ | I guess that makes sense when I think about it and compare it to something like a text file. |
| 01:55.05 | p3nguin_ | A text file with 100 characters would be the same size if you change each of those 100 characters to other characters. |
| 02:01.00 | *** part/#asterisk mroe (~anon__@unaffiliated/roe) |
| 02:09.27 | *** join/#asterisk OrNix (~ornix@l49-0-149.cn.ru) |
| 02:23.07 | *** join/#asterisk thansen (~thansen@67.199.179.78) |
| 02:49.28 | *** join/#asterisk angavmx (~angav@189.140.220.7) |
| 02:50.09 | angavmx | Anyone using a SIP trunk with voicetrading? |
| 02:55.00 | p3nguin_ | ~trunk |
| 02:55.01 | infobot | methinks trunk is a word with varying definitions. In Asterisk, a trunk is a "stream of UDP packets containing IAX2 frames from more than 1 call"; in telecom, a trunk is a "single voice channel between two pieces of switching equipment."; in Ethernet a trunk carries more than one 802.1q VLAN. There is no such thing as a "SIP Trunk" -- Don't use the term. |
| 02:55.33 | gamedna | infobot: a trunk is also the nose of an elephant |
| 02:55.34 | infobot | gamedna: okay |
| 02:55.46 | p3nguin_ | hahaha |
| 02:55.56 | *** join/#asterisk b0gatyr (~b0gatyr@adsl-10-92-8.mia.bellsouth.net) |
| 02:56.29 | gamedna | angavmx: just looked at their site... they show "grey" "standard" and "premium" rates available |
| 02:56.56 | *** join/#asterisk raden (~J@66-168-15-100.dhcp.stpt.wi.charter.com) |
| 02:57.21 | gamedna | i wonder what kind of quality you would actually get w/ those grey rates |
| 02:59.15 | bougyman | YAY Fred from LOD Communications. |
| 03:00.44 | gamedna | p3nguin_: how do you use infobot? |
| 03:00.58 | bougyman | he fixed the dtmf issue with a canreinvite=no and rfc2833compensate=yes |
| 03:02.44 | p3nguin_ | gamedna: Ask it things, tell it things. |
| 03:02.56 | gamedna | infobot: what is asterisk? |
| 03:02.58 | infobot | gamedna: I think you lost me on that one |
| 03:03.09 | gamedna | infobot: what is the meaning of life? |
| 03:03.11 | infobot | gamedna: what are you talking about? |
| 03:03.17 | gamedna | infobot: asterisk |
| 03:03.18 | infobot | asterisk is probably an open source telephony toolkit, or #asterisk on irc.freenode.net, or http://www.asterisk.org, or just like a mini-mall |
| 03:03.25 | gamedna | infobot: help |
| 03:03.32 | p3nguin_ | oh no. |
| 03:03.45 | p3nguin_ | I hope it only replies to that in private. |
| 03:04.39 | drmessano | ~help @ gamedna |
| 03:04.49 | drmessano | lol |
| 03:04.55 | drmessano | ~shoot gamedna |
| 03:04.56 | infobot | ACTION shoots gamedna in the ear with a spitwad! |
| 03:04.56 | gamedna | yea, infobots usually only reply in private |
| 03:05.05 | gamedna | hahaha |
| 03:05.12 | drmessano | ~shoot gamedna |
| 03:05.13 | infobot | ACTION shoots gamedna in the eye with a glue gun! |
| 03:05.13 | gamedna | ~shoot drmessano |
| 03:05.14 | infobot | ACTION shoots drmessano in the ear with a frozen turkey cannon! |
| 03:05.22 | gamedna | hhaha |
| 03:05.23 | p3nguin_ | ~insult drmessano |
| 03:05.37 | drmessano | ~format gamedna |
| 03:05.47 | drmessano | ~fdisk gamedna |
| 03:05.52 | drmessano | aww |
| 03:06.06 | drmessano | ~sudo gamedna |
| 03:06.11 | drmessano | ~sudo |
| 03:06.13 | infobot | somebody said sudo was (SUperuser DO) better than su, according to talon.It is able to give limited super user privileges to specific users, or can allow you to do silly things like run X apps with root perms, or good in scripts with "username ALL = NOPASSWD: /some/program", or http://www.aplawrence.com/Basics/sudo.html, or good for ordering sandwiches, or not pseudo |
| 03:06.30 | drmessano | HAHAHAH |
| 03:06.37 | gamedna | Hehe! |
| 03:06.39 | drmessano | "good for ordering sandwiches" |
| 03:06.44 | gamedna | ~fsck drmessano |
| 03:06.45 | infobot | e2fsck /dev/drmessano : warning! filesystem contains morons! |
| 03:06.49 | drmessano | Make me a sandwich |
| 03:06.49 | gamedna | ;) |
| 03:06.52 | p3nguin_ | sudo make me a sandwich |
| 03:06.56 | drmessano | sudo make me a sandwich |
| 03:07.04 | gamedna | (poof, you are a sandwich) |
| 03:07.10 | drmessano | ~sudo make me a sandwich |
| 03:07.11 | infobot | okay |
| 03:07.14 | drmessano | HAHAHAH |
| 03:07.30 | drmessano | ~make me a sandwich |
| 03:07.30 | infobot | make: *** No rule to make target `me a sandwich'. Stop. |
| 03:07.40 | p3nguin_ | haha |
| 03:07.43 | drmessano | ~make poo |
| 03:07.44 | infobot | make: *** No rule to make target `poo'. Stop. |
| 03:07.44 | Maliuta | drmessano: I don't think make takes multiple targets as arguments |
| 03:07.52 | p3nguin_ | ~make love |
| 03:07.53 | infobot | make: *** No rule to make target `love'. Stop. |
| 03:07.55 | p3nguin_ | aww |
| 03:08.00 | drmessano | ~make war |
| 03:08.01 | infobot | make: *** No rule to make target `war'. Stop. |
| 03:08.07 | *** join/#asterisk yidiyuehan (~yidiyueha@bb121-7-242-73.singnet.com.sg) |
| 03:08.15 | drmessano | ~drop table infobot; |
| 03:10.20 | yidiyuehan | Is there an IRC channel for astmanproxy? |
| 03:13.00 | russellb | i didn't think anyone used that anymore |
| 03:13.09 | russellb | but to answer your question, no |
| 03:13.15 | russellb | this one would be as relevant as it gets |
| 03:17.33 | gamedna | what is astmanproxy? |
| 03:18.04 | russellb | http://lmgtfy.com/?q=astmanproxy |
| 03:18.06 | gamedna | nm, just looked it up on voip-info |
| 03:18.11 | russellb | heh |
| 03:18.18 | gamedna | russellb: yea, i deserved that |
| 03:19.06 | drmessano | Yes you did |
| 03:19.10 | drmessano | and don't you forget it |
| 03:19.15 | drmessano | As matter of fact |
| 03:19.22 | gamedna | (* sits in shame in the corner *) |
| 03:19.28 | drmessano | Store it in a hash table so you can reference it faster later |
| 03:19.41 | russellb | drmessano: now who said you got to show up and rub it in :-p |
| 03:19.44 | drmessano | Oh, forgot the !!!! |
| 03:19.53 | gamedna | laugh |
| 03:19.55 | drmessano | :( |
| 03:19.59 | russellb | pwnt |
| 03:20.05 | drmessano | (* sits in shame in the corner *) |
| 03:20.11 | russellb | wins |
| 03:20.49 | drmessano | Shouldn't you be busy fixing a bug or something? |
| 03:20.58 | drmessano | I hear there's bugs in Asterisk |
| 03:21.07 | russellb | shouldn't you be busy getting me a beer? |
| 03:21.10 | drmessano | Ones with BIG POINTY TEETH |
| 03:21.12 | drmessano | lol |
| 03:21.23 | drmessano | Yes sir |
| 03:21.25 | drmessano | Sorry, sir |
| 03:21.54 | russellb | ~hug drmessano |
| 03:21.55 | infobot | ACTION gets a running start and tackle-hugs drmessano |
| 03:22.04 | yidiyuehan | Wanted to develop an interface to control asterisk and update the event. |
| 03:22.19 | yidiyuehan | Wanted to use astmanproxy but it seems that not much docu available.... |
| 03:22.23 | drmessano | ~hug russellb |
| 03:22.24 | infobot | ACTION sneaks up on russellb and suddenly hugs russellb tightly |
| 03:22.27 | russellb | yidiyuehan: you can just connect directly to the manager interface. I'm not aware of any reason you need the proxy anymore. |
| 03:22.29 | drmessano | That's as good as it gets, bro |
| 03:23.36 | yidiyuehan | russellb, direct connection is not that easy as I need to build the library serving as an agent to talk with asterisk. |
| 03:23.41 | yidiyuehan | and astmanproxy has done that for me. |
| 03:23.54 | yidiyuehan | particularly I may need to control multiple * boxes. |
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| 03:28.25 | mroe | ok, gonna do my best: I have a T-1 connected to *. 6 channels of that T-1 are in a 'hunt group'. Other than the main number does each channel need to have DID assigned to it, like how analog hunt groups work? |
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| 03:55.58 | gamedna | anyone here use vitelity? |
| 04:00.58 | ChannelZ | I do |
| 04:01.27 | gamedna | ChannelZ: what has your experience been like? |
| 04:01.47 | ChannelZ | Fine so far. |
| 04:01.58 | ChannelZ | But their IAX doesn't work so use SIP |
| 04:02.10 | gamedna | How many channels do they give you? |
| 04:02.35 | ChannelZ | Good question. |
| 04:02.47 | gamedna | they dont really say on their site |
| 04:02.53 | ChannelZ | At least 2 incoming that I know of. |
| 04:03.09 | ChannelZ | I just use it for my home phone so I've never really cared |
| 04:03.18 | gamedna | ah ok |
| 04:03.28 | gamedna | right now im using voip.ms |
| 04:03.31 | gamedna | outbound, metered |
| 04:03.40 | ChannelZ | They have a 'virtual pri' service if you need lots but I don't know what the cutoff really is |
| 04:03.55 | gamedna | call quality has been good |
| 04:05.03 | TJNII | I never really had problems with their IAX service, but I dropped it several months ago as I never used it. |
| 04:05.12 | gamedna | im shopping around for a good, reliable inbound carrier w/ about 4 channels |
| 04:05.31 | gamedna | pay per min sounds ok from vitelity, 1.2c/min is not bad |
| 04:05.44 | ChannelZ | huh. I couldn't get it to work at all. I could register and it would maybe sometimes send 1 call to me, and after that it's like it forgot my registration and assumed I was unavailable |
| 04:05.46 | gamedna | flowroute has the same rate but unlim chan |
| 04:06.07 | gamedna | ChannelZ: Which provider? |
| 04:06.16 | ChannelZ | Vitelity. For IAX |
| 04:06.20 | TJNII | Well, maybe it was doing that for me and I didn't know it. I set it up as a backup, it was almost never used. |
| 04:06.35 | gamedna | ChannelZ: I am using IAX w/ voip.ms |
| 04:06.44 | TJNII | It always showed as resistered OK, though...... |
| 04:06.45 | ChannelZ | I went in circles with support, they kept having me change totally inconsequential things and then said "we don't recommend using IAX" |
| 04:07.27 | ChannelZ | I asked them why they put it on their page as a supported service then. They stopped replying. |
| 04:07.53 | ChannelZ | but SIP has been fine for my needs and I really don't care one way or the other, it just makes firewalling a little easier. |
| 04:08.13 | TJNII | Heh, yea. IAX was why I went with them. I wanted a backup that used a different protocol. |
| 04:08.32 | ChannelZ | Huh. I wonder what the deal with me is |
| 04:08.52 | ChannelZ | was anyway. I should try it again sometime, but meh.. |
| 04:08.57 | TJNII | Did they always show as registered, but the calls were dropped? |
| 04:10.09 | ChannelZ | Well from my side I couldn't tell that anything was wrong. I registered fine, it periodically poked them with no errors, asterisk seemed happy.. but I'd dial my own number and it would just give me an error message, and then send me a failure email like my connection was down or I wasn't registered and it didn't know where to send the calls. |
| 04:10.21 | ChannelZ | I even have a static IP. |
| 04:10.36 | TJNII | Then I could have had the same problem and not known it. |
| 04:10.48 | ChannelZ | Were you using it for outbound only? |
| 04:10.58 | ChannelZ | or I guess you said you weren't really using it at all much :) |
| 04:11.06 | TJNII | I never advertised the Vitelity DID, I only used it a couple of times to test my main DID through Broadvoice. |
| 04:12.02 | TJNII | I guess I didn't use it enough to get the "full experience," then. |
| 04:12.10 | ChannelZ | Ah. Yeah it was strange, it's like it accepted my registration but couldn't remember it longer than about 5 seconds. Or other servers weren't getting info from whichever one I registered to |
| 04:12.37 | gamedna | strange |
| 04:12.41 | gamedna | who do you use for DID now? |
| 04:12.49 | TJNII | I use Broadvoice. |
| 04:12.52 | ChannelZ | It was hard to track down really, it seemed like if I called within about 10-15 seconds of registering it would work and then stop, but I think the more I tested it was just totally random when calls would or wouldn't come in |
| 04:13.04 | TJNII | Huh. |
| 04:13.29 | TJNII | Eh, It worked when I used it, but in retrospect I used it very, very little. |
| 04:13.56 | TJNII | Heh, I'm still registered even though my account is closed. |
| 04:14.04 | ChannelZ | Like I said the SIP has been flawless but I was a little disapointed at the total IAX fail |
| 04:14.08 | TJNII | I should probably erase that from my iax.conf...... |
| 04:15.23 | TJNII | Yea, I got mine when I dropped my cell and was paranoid about being pure VoIP. My SIP account with broadvoice has been fine. |
| 04:15.33 | TJNII | The power and internet has been less reliable. |
| 04:16.27 | ChannelZ | heh |
| 04:16.49 | gamedna | i have lots of problems w/ broadvoice |
| 04:16.54 | TJNII | Speaking of which ... I need to yell at the city tomorrow for interrupting power without telling me... |
| 04:17.00 | TJNII | At least the UPS is getting used...... |
| 04:17.29 | TJNII | gamedna: Really? Like what? |
| 04:17.37 | gamedna | registration drops all the time |
| 04:18.06 | gamedna | so inbound calls are really iffy |
| 04:18.12 | gamedna | calls can be jittery at times |
| 04:18.21 | TJNII | I have seen that. Registrations will fail for about 15 minutes. It is rare and inconsistant for me, though. |
| 04:18.36 | gamedna | happens often enough for me |
| 04:18.59 | gamedna | have emailed them several times, and they keep blaming my ISP |
| 04:19.29 | gamedna | but then i ask them... why dont my other sip / IAX2 connections have the same problems |
| 04:20.00 | TJNII | I think I had issues like that with one of their proxies, my memory is fuzzy, though.... |
| 04:20.07 | gamedna | comcast, verizon fios, time warner, etc.. i have used it on all these ISP's and have had the same problems |
| 04:20.20 | gamedna | tried different proxies too |
| 04:21.07 | likwid|mac | what router are you using |
| 04:21.20 | likwid|mac | you may have already said i just got back |
| 04:21.33 | gamedna | TJNII: you are not the first person that has said that broadvoice works well for you... |
| 04:21.49 | TJNII | Huh. That hasn't been my experience. Sometimes I get jittery calls, but it hasn't been bad enough to be a concern.... |
| 04:21.59 | TJNII | They must just hate you. |
| 04:22.05 | gamedna | likwid|mac: different routers... DIR-655, linksys, and i have a couple IPCOP firewalls |
| 04:22.13 | likwid|mac | i have a pbx with the same problems you are describing |
| 04:22.19 | likwid|mac | it was the sonicwall |
| 04:22.21 | gamedna | TJNII: yea, that is what i mean... |
| 04:22.27 | likwid|mac | once i removed it everything cleared up |
| 04:22.38 | gamedna | right now im on a DIR-655 |
| 04:23.02 | gamedna | and i can talk all day long to voip.ms, sipgate, and my other PBX in boston (from texas) w/o a problem |
| 04:23.05 | likwid|mac | my problem as far as i could tell was the way sonicwall handled nat |
| 04:23.08 | gamedna | call out using broadvoice |
| 04:23.09 | gamedna | ugh |
| 04:23.22 | gamedna | likwid|mac: that makes sense |
| 04:23.47 | likwid|mac | i would sit and watch it and it would register and unregister over and over |
| 04:23.58 | likwid|mac | sometimes calls would fail sometimes they would go through |
| 04:24.02 | gamedna | not trying to bash broadvoice btw... they have been good to me over the years. |
| 04:24.52 | likwid|mac | im sure you already tried adding a qualify statement ot the trunk |
| 04:25.24 | gamedna | yes |
| 04:25.54 | gamedna | well... let me double check... to be sure |
| 04:26.09 | likwid|mac | i ended up moving mine out from behind the router |
| 04:26.16 | [TK]D-Fender | Qualify will not help them any... |
| 04:26.30 | likwid|mac | it will try to keep the connection alive? |
| 04:26.46 | [TK]D-Fender | And with SonicWALL you only have to disable the SIP NAT Traversal crap |
| 04:26.56 | likwid|mac | yea i tried it ever way |
| 04:27.03 | likwid|mac | i never could get a configuration that would work |
| 04:27.05 | gamedna | qualify is set to yes, as per broadvoice's support instructions |
| 04:27.11 | [TK]D-Fender | likwid|mac: First UDP is "stateless, and you don't need to keep THEM alive. |
| 04:27.21 | [TK]D-Fender | They live quite well without our help |
| 04:27.35 | likwid|mac | Second? |
| 04:27.54 | [TK]D-Fender | likwid|mac: that would be what followed the "and" |
| 04:28.09 | likwid|mac | ;) |
| 04:28.14 | [TK]D-Fender | likwid|mac: Also why there is a comma there |
| 04:28.27 | gamedna | <[TK]D-Fender w/o qualify=yes, the registration drops much more raplidly |
| 04:28.33 | gamedna | er.. rapidly |
| 04:28.48 | [TK]D-Fender | gamedna: that has absolutely NOTHING to do with your register statement |
| 04:28.53 | likwid|mac | it made a difference on mine too... |
| 04:29.02 | [TK]D-Fender | Delusionally perhaps |
| 04:29.14 | gamedna | <[TK]D-Fender ... i am aware, but that is the case according to them |
| 04:29.28 | [TK]D-Fender | gamedna: Fever dreams, nothing more |
| 04:29.35 | gamedna | do you have a broadvoice account? |
| 04:30.00 | likwid|mac | how much is broadvoice charging |
| 04:30.21 | gamedna | likwid|mac: they have different plans, byod light is 9 a month, |
| 04:30.23 | *** join/#asterisk pinoyskull (~pinoyskul@122.55.80.205) |
| 04:30.38 | likwid|mac | unlimited? |
| 04:30.40 | gamedna | no |
| 04:30.50 | gamedna | likwid|mac: im planning on dropping them once i find a better inbound provider |
| 04:31.01 | gamedna | just need to port my number to them |
| 04:31.14 | likwid|mac | what does your orig cost? |
| 04:31.26 | gamedna | i dont use them for outbound |
| 04:32.05 | [TK]D-Fender | Origination is INBOUND |
| 04:32.22 | gamedna | 9 /month unlimited inbound |
| 04:32.22 | likwid|mac | termination is outbound and origination is inbound |
| 04:32.29 | likwid|mac | oh wow |
| 04:32.32 | likwid|mac | thats cheap |
| 04:32.52 | gamedna | http://www.broadvoice.com/rateplans_byod.html |
| 04:32.52 | *** join/#asterisk coppice (~chatzilla@m121-203-239-112.smartone-vodafone.com) |
| 04:33.05 | gamedna | actually 5.95, but ends up being 9.XX a month |
| 04:33.10 | gamedna | after taxes and fees |
| 04:33.11 | *** join/#asterisk adolfomaltez (~taro@190.87.103.192) |
| 04:33.14 | [TK]D-Fender | Les.net = $4/mo |
| 04:33.57 | gamedna | <[TK]D-Fender do you use les? |
| 04:34.02 | likwid|mac | i wonder if broadvoice has limits lol |
| 04:34.13 | gamedna | likwid|mac: i have never hit them |
| 04:34.17 | [TK]D-Fender | gamedna: As of recently yes, as do several of my customers. |
| 04:34.21 | [TK]D-Fender | gamedna: All happy |
| 04:34.23 | likwid|mac | would they be suspicious if i ported a few hundred dids |
| 04:35.02 | gamedna | <[TK]D-Fender how many channels per DID? |
| 04:35.27 | likwid|mac | i bet one if its unlimited |
| 04:35.31 | likwid|mac | maybe 2 |
| 04:36.47 | gamedna | likwid|mac: they say IP Trunks unlimited on their site... but sometimes there is fine print |
| 04:37.47 | [TK]D-Fender | 2 normally. Other setups is up to 5 |
| 04:38.35 | likwid|mac | how do they provide service so cheap and make money |
| 04:38.39 | gamedna | <[TK]D-Fender might give them a try and see how it goes... |
| 04:39.25 | likwid|mac | Per-Minute DIDs support four-concurrent channels. |
| 04:39.32 | likwid|mac | Flat-Rate DIDs support two-concurrent channels. |
| 04:39.40 | likwid|mac | Multiple Concurrent Channels (Virtual PRI) available |
| 04:39.47 | likwid|mac | so there area few options |
| 04:39.55 | gamedna | where did you see that? |
| 04:40.23 | likwid|mac | http://les.net/products/product_ipdidusa.php |
| 04:40.54 | gamedna | still not bad... |
| 04:41.05 | gamedna | will work well for some of my customers |
| 04:41.08 | likwid|mac | depends on what your doing with it |
| 04:41.45 | likwid|mac | why dont you just colo a switch and buy wholesale and resell it to customers |
| 04:41.51 | likwid|mac | do you have the traffic? |
| 04:42.12 | ChannelZ | Hmm. My MOH no worky. |
| 04:42.13 | [TK]D-Fender | I don't actually use them for VoIP technically... I have them originate, and then retermintate a # to a sales guy's cell phone. Basic telcos were dumb-fucks whos aid they couldn't do it. |
| 04:43.01 | likwid|mac | thats pretty popular in my area |
| 04:43.26 | likwid|mac | we have alot of snow birds that want a local number but want to keep their cell phone from up north |
| 04:43.31 | likwid|mac | so the grandkids can call |
| 04:47.15 | likwid|mac | btw side not on moh i really like the new music in the latest asterisknow release |
| 04:50.28 | likwid|mac | the number you have dialed is a party on your line please hang up and allow sufficient time for your party to answer |
| 04:50.33 | gamedna | likwid|mac: were you asking me or tk about the traffic? |
| 04:50.54 | likwid|mac | you |
| 04:51.06 | gamedna | i am thinking about it |
| 04:51.22 | likwid|mac | level 3 has lowered their minimums |
| 04:51.37 | gamedna | likwid|mac: to what? |
| 04:51.44 | likwid|mac | 10000 |
| 04:52.12 | gamedna | i actually know someone @ level3 from childhood |
| 04:52.25 | likwid|mac | cool |
| 04:52.28 | *** join/#asterisk Ayatolla (~Ayatolla@112.207.210.90) |
| 04:52.29 | gamedna | may be able to get that waived |
| 04:52.42 | likwid|mac | first time i called they told me a million mins |
| 04:52.46 | likwid|mac | than 250000 |
| 04:52.50 | likwid|mac | now 10000 |
| 04:52.51 | gamedna | haha |
| 04:52.53 | likwid|mac | so who knows |
| 04:53.01 | gamedna | what is their rate for 10k? |
| 04:53.23 | gamedna | you can pm if you like |
| 04:53.28 | likwid|mac | idk the salesman just called yesterday and is going to call back with a proposal |
| 04:53.48 | likwid|mac | im sure you know that i cant tell the rates anyway... |
| 04:54.19 | likwid|mac | every carrier i have delt with uses ndas |
| 04:54.27 | [TK]D-Fender | checkout time, later all |
| 04:55.05 | gamedna | likwid|mac: true dat... but @10k i figured they would not do that |
| 04:55.14 | likwid|mac | im not at 10k |
| 04:55.26 | likwid|mac | thats just what the min is |
| 04:55.26 | Ayatolla | hi folks, i used to play around with astersik for some years ago.. at that time the was something called fwdOut.. any similar site/asterisk community now a days? |
| 04:56.00 | likwid|mac | asterisk is not a toy |
| 04:56.06 | ChannelZ | ahh interesting, I think MOH has issues when channels are bridged |
| 04:56.07 | likwid|mac | but it is fun to play with |
| 04:56.46 | likwid|mac | i think apple pissed of adobe and now they are messing with my flash player |
| 04:57.07 | *** join/#asterisk GameGamer43|Mac (~GameGamer@CPE-65-27-76-78.new.res.rr.com) |
| 04:57.14 | gamedna | im working on writing an asterisk synthesizer |
| 04:57.43 | gamedna | i definitely find asterisk fun to play with |
| 04:57.50 | likwid|mac | the the the the the number you have ddddddddialed is not in in in service |
| 04:57.55 | Ayatolla | asterisk is fun to play with |
| 04:58.02 | DogBoy | fun? |
| 04:58.24 | gamedna | anyone ever make a loopback dialplan? |
| 04:58.36 | likwid|mac | not on purpose |
| 04:58.38 | Ayatolla | haha |
| 04:58.39 | gamedna | where it keeps making sip connections to itself. |
| 04:58.46 | gamedna | that is fun. |
| 04:58.51 | *** join/#asterisk SiNGLer (~singler@81-7-123-162.static.zebra.lt) |
| 04:58.55 | likwid|mac | i did it earlier today |
| 04:59.12 | gamedna | how long b/f your * box died? |
| 04:59.32 | likwid|mac | hum |
| 04:59.35 | likwid|mac | it didnt |
| 04:59.44 | gamedna | mine died after 45 misn |
| 04:59.50 | likwid|mac | i didnt let it go long enough |
| 04:59.59 | likwid|mac | it was a production system that was in service |
| 05:00.02 | gamedna | sip show channels was LONG!!! |
| 05:00.12 | likwid|mac | thats cool |
| 05:00.22 | likwid|mac | i should do it in my sandbox |
| 05:00.29 | gamedna | that is where i did it |
| 05:00.42 | gamedna | i wanted to figure out how to deal with that problem in case i did that by accident |
| 05:00.43 | likwid|mac | oh no do it on a primary switch |
| 05:00.46 | Ayatolla | and the purpose? |
| 05:01.03 | gamedna | and how long I had before the system would di |
| 05:01.04 | gamedna | die |
| 05:02.34 | gamedna | Ayatolla: just basically stress testing.. |
| 05:04.04 | gamedna | likwid|mac: other than level3, what other providers do you use? |
| 05:04.43 | likwid|mac | global crossing |
| 05:06.24 | gamedna | r u in a datacenter? |
| 05:06.31 | *** join/#asterisk roe (~roe___@unaffiliated/roe) |
| 05:06.33 | likwid|mac | no im in my bedroom |
| 05:06.38 | gamedna | hahaha |
| 05:07.08 | likwid|mac | but yes |
| 05:09.18 | gamedna | having global crossing in your bedroom must ruin your sex life |
| 05:09.57 | likwid|mac | as i only serve a small area i am in negations with the ILEC to interconnect |
| 05:10.21 | likwid|mac | having global crossings in the bedroom only helps |
| 05:11.08 | gamedna | v. kewl |
| 05:11.12 | likwid|mac | ironicly one on my maserplans is to run fiber from the co to my garage lol |
| 05:11.14 | gamedna | likwid|mac: what area do you serve? |
| 05:11.23 | likwid|mac | its only 1800 feet |
| 05:11.29 | likwid|mac | is that lame |
| 05:11.35 | likwid|mac | it feels weird saying it |
| 05:11.36 | gamedna | likwid|mac: no... that is awesome |
| 05:11.43 | gamedna | hahaha |
| 05:11.48 | gamedna | im 26 miles from the CO |
| 05:11.49 | likwid|mac | but for now im in the datacenter |
| 05:11.52 | likwid|mac | oh |
| 05:11.58 | likwid|mac | im in a very small rural area |
| 05:12.05 | likwid|mac | tier 4 all the way |
| 05:12.07 | likwid|mac | woohoo |
| 05:12.13 | gamedna | (damn) |
| 05:13.27 | likwid|mac | the local underground cable contractor wants 20000 dollars to make the run |
| 05:13.47 | likwid|mac | so i am seriously considering doing it with a shovel and a garden hose |
| 05:13.55 | likwid|mac | ok more than a garden hose maybe a fire hose |
| 05:14.12 | likwid|mac | i feel like the concept is sound though |
| 05:14.48 | likwid|mac | im more excited about the idea of having a fiber connection in my house than anything else |
| 05:15.43 | gamedna | your world of warcraft will run awesome! |
| 05:15.57 | likwid|mac | i actually dont have any games lol |
| 05:15.59 | gamedna | frag some noobs w/ that l33t sht |
| 05:16.20 | gamedna | had to go there though. |
| 05:16.42 | likwid|mac | but the local radio station is interested in me streaming their signal over the internet |
| 05:16.45 | likwid|mac | so that might be cool |
| 05:17.16 | likwid|mac | than you all can listen to the oldies all daya nd night long |
| 05:17.33 | gamedna | you should give them an asterisk server where people can call in and listen |
| 05:17.56 | likwid|mac | hum |
| 05:18.03 | likwid|mac | might work |
| 05:18.07 | likwid|mac | not a bad idea |
| 05:18.14 | likwid|mac | not sure who would call in and listen though |
| 05:18.33 | gamedna | hehe |
| 05:20.00 | AliRezaTaleghani | L-) can i have good, wiki or guide about SIP trunk, the matter is that one side is behind NAT |
| 05:23.25 | likwid|mac | http://www.voip-info.org/wiki/view/NAT+and+VOIP |
| 05:23.39 | WIMPy | ~sipnat |
| 05:24.42 | infobot | i guess sipnat is Quick guide on configuring Asterisk + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 05:24.44 | likwid|mac | By far the best way to deal with the issue of VoIP NAT Traversal is to avoid the cause of the problem in the first place: |
| 05:24.44 | likwid|mac | â¢Do not use NAT and obtain public IP addresses for all VoIP devices |
| 05:24.44 | likwid|mac | â¢If you cannot avoid NAT, use IP Tunneling between VoIP devices on different LANs |
| 05:24.45 | likwid|mac | â¢Use a public service. Eg sign up both sides to FWD and call from one to the other. Look at user authentication page for ways to control who has access to your internal lines |
| 05:24.45 | likwid|mac | â¢Use servers that implement IETF's http://tools.ietf.org/html/draft-ietf-sipping-nat-scenarios. One of those is http://www.voip-info.org/wiki/view/YATE |
| 05:25.43 | gamedna | FWD = free world dialup? |
| 05:29.08 | bougyman | i've found #2 to be the most reliable. |
| 05:29.17 | bougyman | it's even better than #1 in many cases. |
| 05:29.34 | bougyman | where the public IP may be dynamic, the tunnelled (openVPN for me) IP will always be the same |
| 05:29.46 | bougyman | that helps a lot on network failures/disconnects/resets. |
| 05:29.54 | gamedna | i use #2 as well, but recently switched to using IAX2 |
| 05:30.02 | gamedna | IAX2 works well over nat |
| 05:30.23 | likwid|mac | does iax2 require registration or can they be static |
| 05:30.24 | bougyman | dunno, i stopped using a couple years ago |
| 05:30.46 | bougyman | iax2 with too many calls over a connection (about 20-25) led to crosstalk on our boxes. |
| 05:31.00 | gamedna | i use multiple trunks |
| 05:31.00 | likwid|mac | really |
| 05:31.08 | gamedna | and limit to 20 calls per trunk |
| 05:31.17 | likwid|mac | oh |
| 05:31.23 | gamedna | for IAX |
| 05:31.23 | bougyman | i haven't had the prob with sip |
| 05:31.37 | gamedna | but that is b/c i have an inbound and an outbound trunk |
| 05:31.42 | gamedna | so i can monitor them differently |
| 05:31.47 | gamedna | er seperately |
| 05:32.20 | likwid|mac | ? |
| 05:32.34 | gamedna | Office A <---> Office B |
| 05:32.41 | gamedna | Trunk 1 ... allows calls from A to B |
| 05:32.48 | gamedna | trunk 2 allows calls from B to A |
| 05:32.54 | bougyman | on the same hardware, same box, we were getting crosstalk at 25 calls consistently. |
| 05:33.08 | bougyman | changed to sip and we've run as high as 96 calls with no issues. |
| 05:33.15 | gamedna | bougyman: what kind of crosstalk? |
| 05:33.23 | bougyman | gamedna: callers hearing other calls. |
| 05:34.41 | gamedna | complete calls getting crossed... ?? |
| 05:35.00 | gamedna | or just frames here an thtere |
| 05:36.39 | gamedna | bougyman: what ver of *? |
| 05:36.44 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 05:37.01 | bougyman | gamedna: old, 1.4.22 |
| 05:37.09 | gamedna | ah. |
| 05:37.12 | bougyman | gamedna: complete calls getting crossed. |
| 05:45.08 | gamedna | bougyman: very strange... |
| 05:46.12 | bougyman | gamedna: not from a google search. seems quite common. |
| 05:47.33 | gamedna | bougyman: not seeing anything for 1.6 |
| 05:47.42 | gamedna | lots for 1.2 and 1.4.. |
| 05:49.32 | gamedna | hmmm, good to know though, will look out for it if i expand the trunks to 40 chan |
| 05:50.14 | likwid|mac | could just use sip |
| 05:50.19 | likwid|mac | is iax better? |
| 05:50.39 | bougyman | it should be. |
| 05:50.47 | bougyman | but hasn't proved to be, in our environment. |
| 05:51.10 | likwid|mac | i set up all my customers to use iax2 |
| 05:51.18 | likwid|mac | but they are all like 4 or 5 channels |
| 05:51.23 | likwid|mac | so not been an issua |
| 05:51.24 | *** join/#asterisk russ (foobar@ip70-176-251-1.ph.ph.cox.net) |
| 05:52.11 | gamedna | local lan i find sip to be better |
| 05:52.32 | *** join/#asterisk nix8n82 (~nate@63.162.27.14) |
| 05:52.37 | gamedna | in my experience i find IAX to be better over the net |
| 05:52.46 | gamedna | but those are < 20 channels |
| 05:55.21 | gamedna | still find it amazing that people get such different results |
| 05:55.39 | *** join/#asterisk jeffgus (~jeffgus@2002:ad33:b504::1) |
| 05:58.13 | *** join/#asterisk upb (cmpxchg@preteam.org) |
| 06:00.34 | *** join/#asterisk aidinb (~Aidin@unaffiliated/aidinb) |
| 06:00.57 | likwid|mac | does asterisk need anything special to use 722 |
| 06:05.53 | bougyman | a license? |
| 06:11.41 | *** join/#asterisk [netman] (~netman@83.54.35.15) |
| 06:13.15 | likwid|mac | who sells them |
| 06:14.03 | likwid|mac | G.722 patents have expired, so it is freely available. |
| 06:14.25 | *** join/#asterisk mpe (~mpe@gate.ipvision.dk) |
| 06:14.30 | *** join/#asterisk [netman] (~netman@83.54.35.15) |
| 06:15.58 | SiNGLer | likwid|mac: if I remember correctly 1.6 supports 722 out nativelly |
| 06:16.02 | *** join/#asterisk shamelessn00b (~chatzilla@58-65-172-114.nayatel.pk) |
| 06:16.19 | likwid|mac | 1.4? |
| 06:16.25 | SiNGLer | dunno |
| 06:16.32 | likwid|mac | hm |
| 06:16.41 | likwid|mac | i only have one 1.6 switch |
| 06:16.50 | likwid|mac | the rest are 1.4 and 1.2 |
| 06:17.47 | SiNGLer | quick googling shows this: http://www.voip-info.org/wiki/view/Asterisk+codecs |
| 06:18.15 | *** join/#asterisk [netman] (~netman@83.54.35.15) |
| 06:19.04 | likwid|mac | i need 100 g729 licenses...for educational purposes |
| 06:20.08 | gamedna | likwid|mac: is that true? |
| 06:20.25 | likwid|mac | no |
| 06:20.27 | gamedna | likwid|mac: nm, misread... 722, not 729 .. i almost crapped |
| 06:20.40 | likwid|mac | there for commercial purposes but it was wortha try |
| 06:21.14 | gamedna | i need 1000 g729 licenses for..... philanthropic purposes. |
| 06:21.42 | likwid|mac | your quite the philanthropist |
| 06:21.50 | bougyman | <PROTECTED> |
| 06:22.11 | bougyman | they can all go to hell, we built out with extreme switches so we didn't have to worry about bandwidth or throughput |
| 06:23.23 | gamedna | which switches do you have? Summits? |
| 06:23.37 | bougyman | Extreme |
| 06:23.46 | gamedna | bougyman: which line of extreme? |
| 06:23.46 | fenrus | *shrug* |
| 06:24.18 | bougyman | oh, yeah. |
| 06:24.23 | boodu | someone has already compiled mISDN with oslec patch ? |
| 06:24.23 | bougyman | X650 is the core |
| 06:24.58 | gamedna | v.nice |
| 06:25.30 | bougyman | i just like that they're all the same os, from the tiniest extreme to the largest. |
| 06:26.03 | AliRezaTaleghani | hello |
| 06:26.23 | AliRezaTaleghani | can i have a simple problem? (ofcourse for u) |
| 06:26.36 | AliRezaTaleghani | how can i setup trunk |
| 06:26.44 | bougyman | what kind of trunk? |
| 06:26.54 | gamedna | ~trunk |
| 06:27.16 | infobot | well, trunk is a word with varying definitions. In Asterisk, a trunk is a "stream of UDP packets containing IAX2 frames from more than 1 call"; in telecom, a trunk is a "single voice channel between two pieces of switching equipment."; in Ethernet a trunk carries more than one 802.1q VLAN. There is no such thing as a "SIP Trunk" -- Don't use the term. the nose of an elephant |
| 06:27.16 | AliRezaTaleghani | on side is a asterisk with valid IP |
| 06:27.16 | AliRezaTaleghani | the other is behind NAT |
| 06:27.26 | AliRezaTaleghani | a wiki, guide or some what step by step |
| 06:27.32 | AliRezaTaleghani | SIP trunk i mean |
| 06:28.07 | bougyman | set nat=auto on it in sip.conf |
| 06:28.12 | bougyman | taht's all the advice I have. |
| 06:28.20 | AliRezaTaleghani | how about host=? |
| 06:28.24 | AliRezaTaleghani | set to dynamic? |
| 06:28.28 | bougyman | http://www.google.com/url?sa=t&source=web&cd=1&ved=0CBkQFjAA&url=http%3A%2F%2Fwww.panoramisk.com%2F90%2Fsip-trunk-with-asterisk%2Fen%2F&ei=3kJiTP_rEMT48AbP_cDaCQ&usg=AFQjCNGxgCjWPao-f4WTTcbE127f01KKYg |
| 06:28.32 | bougyman | perhaps that? |
| 06:28.33 | gamedna | hahaha, nose of an elephant got added to infobot |
| 06:28.35 | gamedna | nice |
| 06:28.38 | AliRezaTaleghani | this should be done about the NAT ed side? |
| 06:29.05 | AliRezaTaleghani | k. let me try more .. will be back soon |
| 06:30.11 | *** join/#asterisk Tim_Toady (~moi@77.49.3.102.dsl.dyn.forthnet.gr) |
| 06:36.48 | likwid|mac | i just paid the same bill twice |
| 06:36.50 | likwid|mac | :( |
| 06:37.31 | gamedna | likwid|mac: whats your address, im gonna send you an invoice |
| 06:37.39 | likwid|mac | lol |
| 06:37.48 | Maliuta | can we all send him one? |
| 06:37.53 | likwid|mac | 123 west fake street |
| 06:38.13 | likwid|mac | anytown,anystate 12345 |
| 06:38.19 | likwid|mac | just mail it there |
| 06:38.27 | gamedna | OH!!! .. WEST fake street... |
| 06:38.30 | gamedna | no wonder why it didnt get there |
| 06:38.38 | likwid|mac | or you can just just give it the post man and tell him to take it to chris |
| 06:40.09 | gamedna | phew... thanks for reminding me actually, i almost forgot to pay a bill |
| 06:40.34 | likwid|mac | the number y ou have dialed is not in service |
| 06:40.42 | likwid|mac | please check the number and dial again |
| 06:41.25 | gamedna | fortunately, its not that bill |
| 06:41.34 | *** join/#asterisk c0rnoTa (~c0rnoTa@109.188.47.151) |
| 06:42.54 | gamedna | i have two remote astersik servers that are now joined by a single IAX trunk. Unfortunately they both have extensions in the same range 200-299 |
| 06:43.37 | gamedna | when dialing from one to the other, the callerid gets reported as the source extension. Is it possible to modify the callerid to show something like 5201, where 5 is the outbound route to the other remote office? |
| 06:44.07 | *** part/#asterisk c0rnoTa (~c0rnoTa@109.188.47.151) |
| 06:44.33 | gamedna | any ideas? |
| 06:46.49 | likwid|mac | caller i d module |
| 06:47.23 | likwid|mac | and dial plan injection module |
| 06:47.31 | likwid|mac | if you wanna use freepbx |
| 06:47.35 | likwid|mac | hum |
| 06:47.45 | gamedna | both boxes are trixbox, so basically freepbx |
| 06:47.45 | likwid|mac | yea |
| 06:48.02 | likwid|mac | add the 5 to the outbound trunk |
| 06:48.19 | likwid|mac | and create dialplan injections with the 5 |
| 06:48.34 | likwid|mac | might be an easier way not sure |
| 06:49.22 | gamedna | are you refering to the CID lookup? |
| 06:49.25 | gamedna | module? |
| 06:49.28 | likwid|mac | no |
| 06:49.35 | likwid|mac | set caller id moduke |
| 06:49.58 | likwid|mac | your going to have to upload it |
| 06:50.18 | gamedna | this one? |
| 06:50.19 | gamedna | http://www.freepbx.org/support/documentation/module-documentation/third-party-unsupported-modules/set-callerid |
| 06:50.35 | likwid|mac | yea |
| 06:51.28 | likwid|mac | there is probably a macro than can be written in the dialplan if you know more than i do |
| 06:51.43 | gamedna | k |
| 06:51.49 | gamedna | let me try the cid thing |
| 06:51.58 | gamedna | i would rather set it up with the UI |
| 06:52.20 | likwid|mac | im working on a puzzle here too |
| 06:53.37 | *** join/#asterisk pinoyskull (~pinoyskul@124.6.182.55) |
| 06:54.09 | gamedna | whats that? |
| 06:54.43 | likwid|mac | so i have a main switch that just routes calls |
| 06:54.54 | likwid|mac | several pbx's out in offices |
| 06:55.22 | gamedna | ah |
| 06:55.31 | likwid|mac | when a call comes in from a carrier it comes to the main switch than goes to the remote switch |
| 06:55.50 | gamedna | not using something like opensips for that? |
| 06:56.11 | likwid|mac | when a remote switch places a call it goes to the main switch for routing either to another switch in house or outside to the carrier |
| 06:56.53 | gamedna | http://www.opensips.org/ |
| 06:57.00 | gamedna | are you familiar w/ that? |
| 06:57.15 | likwid|mac | my concern is what happens when the carrier sends a call to the main switch and it doesnt match one of the did's in the main switch for some reason it will route it back to the carrier |
| 06:57.21 | likwid|mac | creating a loop |
| 06:57.29 | likwid|mac | a perhaps costly loop |
| 06:57.44 | likwid|mac | im not sure how to pevent it |
| 06:58.12 | gamedna | can you make 2 contexts |
| 06:58.33 | gamedna | where the remote switches incoming calls go into a different context than the carrier |
| 06:59.08 | gamedna | inbound calls from the carrier are only allowed to the DIDs |
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| 06:59.28 | likwid|mac | but i need the remote switches to access the dids |
| 06:59.52 | gamedna | not sure what you mean by that |
| 07:00.05 | gamedna | the DID info is only on the remote switches? |
| 07:00.12 | likwid|mac | the remote switches are phone customers |
| 07:00.49 | *** join/#asterisk qvsqvs (~anonymous@196.214.133.227) |
| 07:00.56 | likwid|mac | so if one customer calls another i need the main switch to route the call instead of sending it out to the carrier |
| 07:01.24 | SiNGLer | likwid|mac: you can set channel variable, and before routing to carrier check value or something like that |
| 07:01.51 | likwid|mac | hm |
| 07:02.47 | likwid|mac | its only going to be a problem if i happen to make a mustake and dont have one of my dids in my main switch |
| 07:02.52 | likwid|mac | but i make alot of mistakes |
| 07:03.15 | gamedna | can you make a script to query the DID's from each remote server |
| 07:03.21 | gamedna | and update your main switch accordingly |
| 07:03.46 | likwid|mac | hum |
| 07:03.52 | gamedna | that way you dont have to worry about making mistakes |
| 07:04.07 | gamedna | <PROTECTED> |
| 07:04.17 | gamedna | ( you can read that as <<< is lazy ) |
| 07:05.19 | likwid|mac | it also kinda sucks cause i wont be able to just blindle add did's to my carrier account and have them go to ss-noservice |
| 07:05.46 | likwid|mac | since the any any will just route bck to the carrier |
| 07:06.06 | gamedna | carrier provide an API? |
| 07:06.24 | likwid|mac | yes |
| 07:06.25 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 07:06.31 | likwid|mac | lol more automation |
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| 07:06.46 | gamedna | are you API savy? |
| 07:06.56 | likwid|mac | not really |
| 07:07.13 | gamedna | i can help you w/ the API stuff if you like |
| 07:07.23 | likwid|mac | maybe i should just buy a commercial softswitch |
| 07:07.35 | gamedna | i may be a complete n00b when it comes to asterisk, but i am a bonafied software engineer |
| 07:07.45 | gamedna | more like bonafried, but that is another thing |
| 07:07.55 | likwid|mac | hm |
| 07:08.14 | gamedna | from my perspective, it looks like you need a web interface to manage your DID's |
| 07:08.18 | likwid|mac | where you located> |
| 07:08.24 | gamedna | san antonio, tx, USA |
| 07:08.25 | gamedna | u? |
| 07:08.41 | likwid|mac | orlando, fl |
| 07:08.56 | gamedna | mmmm... disneyworld |
| 07:09.16 | likwid|mac | yea im actually an audio engineer that landed in telecom |
| 07:09.25 | likwid|mac | i used to work for the disney company |
| 07:09.39 | gamedna | ah... what division? My brother is an Imagineer |
| 07:09.52 | likwid|mac | nice |
| 07:09.57 | likwid|mac | not imaginnering lol |
| 07:10.12 | likwid|mac | just the tech operations |
| 07:10.12 | gamedna | (he is in Pittsburgh... doing robotics research ) |
| 07:10.16 | gamedna | still fun |
| 07:10.23 | likwid|mac | cool |
| 07:10.25 | gamedna | disney has some crazy stuff |
| 07:10.39 | likwid|mac | they do alot more than people know about |
| 07:10.47 | gamedna | yeap |
| 07:11.11 | gamedna | bro is researching autonomous humanoid robots for disney. |
| 07:11.40 | gamedna | basically disney wants to have free walking pirates of the carabean |
| 07:12.39 | likwid|mac | they will im sure |
| 07:12.48 | gamedna | its pretty far off.. |
| 07:12.53 | gamedna | probably 20 years |
| 07:15.26 | likwid|mac | actually im not sure this loop a mistake would create would generate a bill cause the channel is unanswered |
| 07:15.36 | likwid|mac | its just router back to the carrier |
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| 07:15.50 | gamedna | yea, but it uses bandwidth |
| 07:16.15 | likwid|mac | yea but is it self sustaining or does it end when the caller ends the call |
| 07:16.49 | likwid|mac | alot of things have to line up for this katastrophe to become a reality |
| 07:18.14 | likwid|mac | i think im going to set it up durring off peak hours and see what happens |
| 07:18.23 | likwid|mac | maybe it will colapse on its own |
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| 07:21.59 | likwid|mac | interesting |
| 07:22.08 | likwid|mac | it loops 3 times than dies on its own |
| 07:22.28 | gamedna | to me it sounds ok for one call |
| 07:22.34 | gamedna | but may be bad if there are 100 |
| 07:22.36 | gamedna | or 20 |
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| 07:27.21 | likwid|mac | oh well im off to bed |
| 07:28.34 | gamedna | likwid|mac: nite.. |
| 07:37.23 | russ | its so beautiful |
| 07:37.25 | russ | http://pastebin.com/8Sfr5Cey |
| 07:37.45 | ChannelZ | Double rainbow, OH MY GOD! |
| 07:38.14 | russ | needs a little more work, but I'm getting really close |
| 07:38.28 | russ | I really wish the ITU was some type of publishing body |
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| 07:45.04 | russ | hmm...is 7, 10 digits the same as 7 to 10 digits |
| 07:46.29 | russ | probably not, 7 to 10 is probably 7, 8, 9, 10 |
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| 07:48.20 | WIMPy | What a bloat |
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| 07:55.26 | WIMPy | More like doubleloat |
| 08:00.29 | tzafrir | WIMPy, hi |
| 08:02.34 | WIMPy | Hi tzafrir: I have to admit I didn't really get your question. Do other cards change their behaviour regarding to jumper settings? |
| 08:03.22 | tzafrir | WIMPy, the qozap drivers do not seem to have such a switch. Rather, they detect the settings from the jumpers |
| 08:03.53 | tzafrir | Now, if you have to both change jumpers *and* change modle parameters, this is plain silly |
| 08:04.33 | WIMPy | Agree, but the te jumpers are only between the line drivers and the socket and in no way connected to the chip. |
| 08:05.22 | WIMPy | That's why I don't use them. It's easier to put a X-Over adaptor in beween, especially if the card is built in already. |
| 08:06.19 | tzafrir | I also don't like the idea of using a module parameter for that |
| 08:06.33 | tzafrir | as I want to allow a system of more than one card |
| 08:06.41 | tzafrir | Even though those are not common |
| 08:06.54 | WIMPy | It's the only way, unless you add a fifth jumper per port. |
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| 08:08.13 | WIMPy | With mISDN2/LCR you don't configure NT/TE mode in the driver. It's only done in the application. |
| 08:09.12 | tzafrir | well, you do configure nt/te in the application . You don't configure nt/te ATM in the driver |
| 08:09.22 | tzafrir | But you suggest to configure it in the driver |
| 08:10.15 | WIMPy | That wasn't my idea. Acually I was under the impression that it was neccessary to tell the driver in order to be able to configure NT mode at a higher level. |
| 08:10.21 | tzafrir | IIRC with the HFC card you have to know if the card is NT or TE even for the low-level operations. Not exactly sure |
| 08:10.53 | tzafrir | Something related to the HDLC decoding? |
| 08:11.17 | WIMPy | I didn't dig that deep, but that sounds unlikely to me. |
| 08:13.22 | WIMPy | At the lowest level it could be as simple as a bit telling the hardware to transmit the E-channel instead of receiving it. |
| 08:17.02 | WIMPy | It's been quite some years since I looked at the low level stuff. |
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| 08:36.12 | WIMPy | tzafrir: BTW: There is no support for the USB versions, is there? |
| 08:36.30 | tzafrir | WIMPy, noone wrote it yet, I guess |
| 08:37.13 | WIMPy | Would have made testing easier. I usually use the netbook for testing. |
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| 09:06.55 | russ | now that's starting to look like useful data http://pastebin.com/GPd3yQzH |
| 09:07.07 | russ | dear ITU, please publish data like the above |
| 09:08.23 | russ | autogenerated from http://www.itu.int/dms_pub/itu-t/opb/sp/T-SP-E.164C-2010-MSW-E.doc |
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| 09:09.19 | gamedna | russ: what are you working on? |
| 09:09.40 | russ | I'm trying to make a e164 parser and dial as you go AGI piece |
| 09:10.14 | russ | it takes as an argument the phone number you are dialing from |
| 09:10.58 | russ | so if you are dialing from +359... (bulgaria), then it knows that if you dial 00, you are trying to dial internationally |
| 09:11.15 | gamedna | neat |
| 09:11.22 | gamedna | 011 for usa, etc.. |
| 09:11.58 | russ | then if you pass it a number, it can also fill in the iso3166 alpha2 code |
| 09:12.54 | russ | then I'll need to make stuff to pull additional data from nanpa |
| 09:13.02 | russ | other people can fill in other countries |
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| 09:13.15 | gamedna | so you are just doing the framework for this |
| 09:13.25 | WIMPy | Sounds interesting. |
| 09:13.35 | gamedna | great stuff |
| 09:13.45 | russ | and a set of scripts to pull and process data from sources like ITU and nanpa and put it in either xml, or maybe postgres |
| 09:14.43 | *** join/#asterisk bn-7bc (bjarne@2001:470:dc32:0:c62c:3ff:fe2d:b7db) |
| 09:14.59 | gamedna | well, im off to bed... nite all. |
| 09:15.46 | russ | I saw all the area codes hardcoded in the callerid superfecta scripts and wanted to vomit |
| 09:16.07 | *** part/#asterisk gamedna (~gamedna@cpe-70-125-155-74.satx.res.rr.com) |
| 09:16.20 | WIMPy | Wasn't that the same list as before? |
| 09:16.27 | russ | ? |
| 09:16.40 | WIMPy | The pastebin |
| 09:17.31 | hrhrhr_ | anyone use dundi |
| 09:17.33 | russ | It is updated to have the iso3166 code, and also to parse the other fields (eg, if international_prefix is 001,007, there are now two international_prefix tags) |
| 09:17.37 | WIMPy | No, it's not the same, just looks similar. |
| 09:17.54 | WIMPy | IC |
| 09:19.03 | hrhrhr_ | or enum |
| 09:19.06 | hrhrhr_ | enum sounds cool |
| 09:19.07 | hrhrhr_ | ish |
| 09:19.53 | hrhrhr_ | is enum a way of determining whether the company you're calling has a voip compatible pbx |
| 09:20.25 | russ | I really wish http://www.localcallingguide.com/ would share their work |
| 09:21.01 | hrhrhr_ | whassat |
| 09:21.21 | hrhrhr_ | online fonebook? |
| 09:21.34 | russ | unrelated to your query |
| 09:22.00 | russ | it tells you if a call is a local call or not and if you need to dial a 1, etc |
| 09:22.17 | russ | I HATE the bell message, you do not need to dial a 1 when dialing this number |
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| 09:22.39 | hetii | #j C |
| 09:23.50 | WIMPy | "It has never been more complicated to use a telephone, than now." |
| 09:24.23 | russ | also an annoyance, many voip providers don't accept e164 |
| 09:37.23 | bn-7bc | russ: rwell IMHO the reaosn for that is with ei64 tere is no provifer the can charge termination fees to so no revenue is created |
| 09:43.36 | henk | can anyone recommend a good reference to asterisk, sip.conf especially? i need accurate and uptodate information on how to use register-statements and peer/user blocks. voip-info.org seems pretty old. |
| 09:46.08 | *** join/#asterisk TimeRider (steve@109.224.131.68) |
| 09:46.32 | hrhrhr_ | russ |
| 09:46.37 | hrhrhr_ | the whole e164 thing |
| 09:46.49 | hrhrhr_ | we have a * box |
| 09:47.04 | hrhrhr_ | i bet there's loads of ppl all over the world with similarly capable systems |
| 09:47.37 | hrhrhr_ | but opening the initial dialogue with them you'd like dial their pstn/pri alternative and never know it could be a free call |
| 09:47.50 | hrhrhr_ | is e164 the solution to that? or am i barking up the wrong tree |
| 09:50.13 | DennisG | hi, is here someone who knows how to get a correct FIFO way for multiple queues? |
| 09:51.02 | DennisG | i have a client with multiple queues and the FIFO is very odd.. if there are 2 people on place 1 in the queue then they will be assigned random to the same agent |
| 09:52.36 | DennisG | is there a way to have FIFO for ALL queues? just based on the summary of all available queues |
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| 10:15.57 | hrhrhr_ | the answer to my own question above, is a resounding 'yes' |
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| 10:21.18 | tzafrir | hrhrhr_, you were barking at the wrong tree? |
| 10:28.41 | hrhrhr_ | no |
| 10:28.48 | hrhrhr_ | e164 is exactly what i thought it was |
| 10:28.53 | hrhrhr_ | and it looks pretty cool |
| 10:29.13 | hrhrhr_ | im just trying to find out who got control of 4.4.e164.arpa. |
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| 10:59.03 | AliRezaTaleghani | L-) can someone lead me about AGI? |
| 10:59.27 | *** join/#asterisk coppice (~chatzilla@m121-203-205-157.smartone-vodafone.com) |
| 10:59.30 | AliRezaTaleghani | for example, i have an script, which check the users, input in some conditions, |
| 10:59.41 | AliRezaTaleghani | and finally decide to change the user, queue |
| 10:59.42 | *** join/#asterisk TimeRider (~steve@109.224.131.68) |
| 10:59.53 | AliRezaTaleghani | how can i set user queue, by AGI |
| 11:00.05 | AliRezaTaleghani | i need just the last part, how to set queue |
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| 11:18.11 | Pouet78 | Hi all |
| 11:19.01 | Pouet78 | I have a problem with my asterisk 1.4.17 (on Ubuntu 8.4 LTS) on Voicemail |
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| 11:20.12 | Pouet78 | I have the prompt (in fact no prompt just the beep) but when I leave a message the audio files are empty and statistics are always 0 sec duration :( |
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| 11:35.06 | *** join/#asterisk hmmwhatsthat (hmmwhatsth@217.27.169.158) |
| 11:36.51 | hmmwhatsthat | Im trying to verify that a user actually has Record()ed something. Can it be done without executing shell commands from the dialplan? |
| 11:43.13 | DennisG | yes. with the System() command |
| 11:45.24 | *** part/#asterisk AliRezaTaleghani (~AliRezaTa@unaffiliated/AliRezaTaleghani) |
| 11:46.07 | hmmwhatsthat | DennisG: That is esentially executing shell commands, I was hoping for a variable or similar |
| 11:47.46 | Pouet78 | Why all my messages are empty? |
| 11:48.05 | Pouet78 | (0 duration time) |
| 11:48.22 | tzafrir | Pouet78, empty? |
| 11:48.27 | tzafrir | Or don't exist? |
| 11:48.33 | tzafrir | If empty: no disk space? |
| 11:48.50 | Pouet78 | no disk space problem |
| 11:49.12 | Pouet78 | wav are 44 bytes (only the header) |
| 11:49.28 | shamelessn00b | anyone used sangoma cards? |
| 11:50.01 | Pouet78 | ; ; Message Information file ; [message] origmailbox=1234 context=default macrocontext= exten=1235 priority=1 callerchan=SIP/0912345003-081dd968 callerid=0912345003 origdate=Wed Aug 11 01:06:19 PM CEST 2010 origtime=1281524779 category= duration=0 |
| 11:50.17 | Pouet78 | any idea? |
| 11:50.39 | *** join/#asterisk defswork (~andy@mx2.3gcomms.co.uk) |
| 11:51.46 | hmmwhatsthat | Pouet78: Have you tried debugging this? i.e. "core set debug 10" and "core set verbose 10" or some such and see what * says? |
| 11:54.33 | Pouet78 | nothing really special : |
| 11:54.37 | Pouet78 | <PROTECTED> |
| 11:56.19 | hmmwhatsthat | Pouet78: Just a shot in the dark here but have you loaded all the necessary codecs for transcoding? |
| 11:57.16 | hmmwhatsthat | Pouet78: codecs and formats |
| 11:57.21 | *** part/#asterisk hrhrhr_ (~c1@213.1.224.2) |
| 11:57.35 | hmmwhatsthat | g729 -> wav |
| 11:57.46 | Pouet78 | I think so |
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| 11:58.42 | Pouet78 | in fact for the moment I don't care the format so I asked to record in g729|gsm|wav49|wav |
| 11:59.20 | hmmwhatsthat | Pouet78: try setting it to only use wav |
| 12:01.19 | Pouet78 | this was the case before |
| 12:01.31 | Pouet78 | I added the other to try... |
| 12:02.35 | hmmwhatsthat | Pouet78: Ok. Might be a missing lib since only the header is written but Im not familiar enough with the source code to say for sure |
| 12:02.44 | Pouet78 | in translations, I only don't have ilbc and g722 which should not be used. |
| 12:04.34 | Pouet78 | I don't have sound card configured (disabled in BIOS) so no alsa or oss in my system, is this a problem? |
| 12:04.48 | hmmwhatsthat | Pouet78: no |
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| 12:09.35 | Pouet78 | so you have an idea of the possible missing lib? |
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| 12:10.52 | hmmwhatsthat | Pouet78: Not really, but Id check what audio libs are actually linked. On another note. Is the call successful if the call is answered instead of forwarded to vm? |
| 12:11.15 | hmmwhatsthat | i.e. answered by another phone |
| 12:12.20 | Pouet78 | Yes the call is received by asterisk because I correctly the beep prompt |
| 12:12.58 | hmmwhatsthat | Pouet78: Yes but can both parties hear eachother? |
| 12:13.43 | Pouet78 | Oh when I call one phone to an other, no problem... |
| 12:14.22 | Pouet78 | I only need one voicemail (direct) |
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| 12:17.32 | hmmwhatsthat | Pouet78: Strange problem indeed. Are the phones and asterisk in the same subnet? |
| 12:19.05 | Pouet78 | yes (in a little complicated as I use ADSL RGW with FXS connected through ADSL + PPP ...) |
| 12:19.13 | henk | which soundformat is recommended for PlayBack()? and how to record, via asterisk, sox, whatever? |
| 12:21.51 | hmmwhatsthat | Pouet78: Well since theyre all in the same subnet you can easily use wireshark or tcpdump to sniff the RTP streams going via asterisk |
| 12:21.58 | Pouet78 | henk: For me the best is to have many formats available, then asterisk will chose the best |
| 12:22.23 | henk | Pouet78: ok, how do you record? how do you convert? |
| 12:23.47 | hmmwhatsthat | henk: record > http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Record convert > http://www.voip-info.org/tiki-index.php?page=Convert+WAV+audio+files+for+use+in+Asterisk |
| 12:23.54 | Pouet78 | I have made the dumps from the asterisk. Seems that all packets are received |
| 12:24.45 | henk | hmmwhatsthat: yeah, thanks, i know that site... it's just awfully out of date in several places. |
| 12:25.08 | hmmwhatsthat | Pouet78: Ok, besides the RTP streams you see SIP signalling, right? Who sends the SIP BYE? |
| 12:25.11 | henk | is that recording and certing stuff still valid? nothing "better" or "easier" came around in the meantime? |
| 12:25.38 | hmmwhatsthat | henk: you might want to try the google ;-) |
| 12:25.52 | Pouet78 | Phone |
| 12:26.23 | henk | hmmwhatsthat: i know the google but that turns up hundreds of year old results whereas here i can get the latest info that's actually still used... |
| 12:26.27 | hmmwhatsthat | Pouet78: Ok, does the phone have some kind of logging capability which would allow you to see the reason for hanging up |
| 12:26.57 | henk | but nevermind, i'll just use the old way and pretend someone said "that's still up to date and the way to do it" |
| 12:27.09 | hmmwhatsthat | henk: :-) |
| 12:27.15 | Pouet78 | No I don't have access to phone logs |
| 12:27.42 | hmmwhatsthat | Pouet78: Bummer, thatd really help |
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| 12:39.03 | jamko | IRC newb here, so please excuse me if my etiquette is bad. I am having an issue with how asterisk is handling T.38 re-invites. Anyone game? |
| 12:41.09 | *** join/#asterisk hemantvoip (~hemant.vo@122.177.46.163) |
| 12:41.42 | hemantvoip | Hi All |
| 12:41.57 | jamko | hey |
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| 12:43.18 | hemantvoip | Hi jamko |
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| 12:49.20 | jamko | Hell all |
| 12:49.36 | hemantvoip | hows the things out there jamko? |
| 12:49.43 | hemantvoip | looks like you got kicked out |
| 12:50.11 | henk | i have some comprehension problems and hope someone can help. we have a cisco call manager (C1700) here and it maps '99' to our asterisk via ipv4. i just upgraded asterisk to 1.6 for several reasons. i did and do _not_ have a register statement in my sip.conf an yet asterisk is and was able to receive calls to '99'. how come? |
| 12:50.15 | jamko | I need some help with a T.38 re-invite issue. Late night last night hemantvoip. Yourself? |
| 12:50.28 | *** join/#asterisk Arsenick (~y@fedora/Arsenick) |
| 12:51.02 | hemantvoip | i am just doing *nothing* |
| 12:51.36 | jamko | henk - just a guess here but, allowguest=no ? |
| 12:51.53 | hemantvoip | you are talking about this issue: https://issues.asterisk.org/view.php?id=8677 |
| 12:53.10 | [TK]D-Fender | henk: You don't need to register to receive calls |
| 12:53.49 | [TK]D-Fender | henk: REGISTER tells the other server WHERE to send calls to. If they already know your address then you have no need for registering |
| 12:55.18 | jamko | hemantvoip: I saw that issue last night, and I don't think it is the same thing. I will tell you a bit about it. |
| 12:57.27 | hemantvoip | need to see some logs |
| 12:58.39 | jamko | Running 1.6.2.10 - If the mediatrix ATA initiates the T.38 Reinvite, all works fine.. If my provider (gafachi) initiates the T.38 Reinvite, asterisk will try routing the call back to the incoming context of extensions.conf, to the callerid set for the mediatrix peer. Here is what comes across the console: -- Now forwarding SIP/mediatrix1-0000002a to 'Local/5555555555@incoming_calls' |
| 12:58.39 | jamko | (thanks to SIP/my_service_provider_4-0000002b) |
| 12:58.39 | jamko | [Aug 10 11:59:17] NOTICE[23691]: chan_local.c:534 local_call: No such extension/context 5555555555@incoming_calls while calling Local channel |
| 12:58.39 | jamko | [Aug 10 11:59:17] NOTICE[23691]: app_dial.c:789 do_forward: Failed to dial on local channel for call forward to '5555555555@incoming_calls' |
| 13:00.22 | hemantvoip | you need to verify if the extension is defined |
| 13:01.41 | jamko | It's not defined. This issue is when terminating from asterisk. After the T.38 Reinvite, Asterisk attempts to route the call back to the incoming call context? It makes no sense. |
| 13:01.57 | hemantvoip | hmmmmm |
| 13:02.14 | jamko | Do you want the debug output from /var/logs/asterisk, pcap tcpdump capture, or sip degub output from the console? |
| 13:02.30 | hemantvoip | mail me at hemant.voip@gmail.com |
| 13:02.39 | hemantvoip | or if you have pastbin |
| 13:06.02 | jamko | I will e-mail you. Thanks!! |
| 13:06.21 | henk | [TK]D-Fender: makes sense. thank you |
| 13:07.56 | *** join/#asterisk sysreq (~sysreq@unaffiliated/sysreq) |
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| 13:19.00 | bougyman | [TK]D-Fender: rfc2833compensate and turning off reinvites was the solution to my dtmf issue. |
| 13:19.13 | bougyman | thanks for your assistance debugging yesterday. |
| 13:22.46 | *** join/#asterisk tomarch (~tom@fbx1.reseau-concept.net) |
| 13:24.06 | [TK]D-Fender | bougyman: Glad you found it |
| 13:24.23 | bougyman | well, Fred from LOD Communications found it. |
| 13:24.48 | bougyman | but I had all the traces and stuff he needed to do so, and had taken enough steps that he got there pretty quick. |
| 13:25.08 | bougyman | thumbs up to LOD Communications support. |
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| 13:38.02 | m0t3jl | Hi, is there a phone out there that would support both Skype and SIP? ;) |
| 13:38.42 | henk | after upgrading to 1.6 my recordings are played a bit strange. hard to explain: a bit too fast and skipping over some 'frames' or something. any idea how to debug that or what might cause it? |
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| 13:41.59 | [TK]D-Fender | m0t3jl: An Android handset with both clients. |
| 13:42.23 | m0t3jl | [TK]D-Fender, pretty fancy and expensive, huh? ;) |
| 13:42.48 | [TK]D-Fender | m0t3jl: depends on your idea of "expensive". |
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| 13:52.01 | henk | where is the path to sound files configured? there are several paths in asterisk.conf, but none for sounds specifically... where should i put my own sounds? |
| 13:52.02 | *** join/#asterisk Da-Geek (~Da-Geek@nat/redhat/x-smfmakavhqlnawlt) |
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| 13:53.41 | [TK]D-Fender | henk: Wherever you want |
| 13:55.04 | henk | [TK]D-Fender: ok, where will asterisk look for them if i give a relative path? |
| 13:57.23 | likwid|mac | ./var/lib/asterisk/custom |
| 13:57.53 | henk | likwid|mac: ok, thanks. |
| 13:58.03 | henk | is that path hardcoded? |
| 13:58.08 | likwid|mac | i have no idea |
| 13:58.29 | likwid|mac | ./var/lib/asterisk is the default |
| 13:58.46 | likwid|mac | you can always add a different lang set and put it in a new folder |
| 13:59.02 | likwid|mac | than add the language=nameofnew folder string |
| 13:59.20 | likwid|mac | in the sip.conf and aix.conf |
| 13:59.24 | likwid|mac | iax.conf |
| 13:59.33 | henk | likwid|mac: ok, thanks |
| 13:59.45 | likwid|mac | are you using freepbx? |
| 14:00.23 | likwid|mac | don't forget to have them at 8000 mono if they are wav files or they won't play |
| 14:00.57 | bougyman | henk: what codec? |
| 14:01.02 | bougyman | we had that problem with gsm. |
| 14:01.19 | likwid|mac | i cant seem to make gsm sound good |
| 14:01.30 | [TK]D-Fender | henk: asterisk.conf ----> astvarlibdir |
| 14:01.34 | [TK]D-Fender | (sounds) |
| 14:02.15 | henk | likwid|mac: nope, plain asterisk |
| 14:02.32 | likwid|mac | oic |
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| 14:02.44 | henk | bougyman: not quite sure, i think it's gsm... |
| 14:02.55 | henk | [TK]D-Fender: ah, great thanks :) |
| 14:02.55 | wcselby | o/ |
| 14:03.07 | likwid|mac | i've never tried to compile it myself |
| 14:03.28 | likwid|mac | probably some combination of laziness and ignorance |
| 14:03.45 | henk | likwid|mac: oh, sorry... it's precompiled from debian testing. but it's plain debian, no specialized asterisk distribution. |
| 14:04.03 | henk | bougyman: what did you do about it? |
| 14:05.48 | bougyman | henk: dropped gsm |
| 14:05.50 | bougyman | we're all ulaw now. |
| 14:06.40 | likwid|mac | i wasnt sure what to do when i got pat fleet's voice files in ulaw thought they wouldnt work but they work fine |
| 14:06.48 | michael-i | Is there a negative consequence to putting the pattern indicator "_" in front of extensions in extensions.conf which are not patterns? |
| 14:06.50 | likwid|mac | better in fact ;) |
| 14:07.20 | leifmadsen | michael-i: yes there is |
| 14:07.34 | michael-i | leifmadsen: speed? messes up the look-up tables? |
| 14:07.41 | leifmadsen | _nancy,1,NoOp() |
| 14:07.46 | leifmadsen | that will not do what you expect |
| 14:07.53 | henk | [TK]D-Fender: astvarlibdir => /var/lib/asterisk and my sounds are in /var/lib/asterisk/sounds/custom/*.gsm. i tried PlayBack('custom/file') and 'sounds/custom/file' but asterisk keeps saying "file not found"... any idea? |
| 14:08.00 | leifmadsen | the letter 'n' has special meaning in pattern matches |
| 14:08.03 | henk | bougyman: ok, did you convert your old filed or just rerecord? |
| 14:08.07 | michael-i | and _1234,1,NoOp() ? |
| 14:08.09 | bougyman | henk: neither. |
| 14:08.20 | bougyman | all our sounds come from fs, it transcodes on the fly. |
| 14:08.28 | michael-i | so far, I do not allow any string extensions. They're numeric or patterns |
| 14:08.29 | leifmadsen | michael-i: that would be fine, but is totally unnecessary and dangerous if you change it to have anything with letters in it |
| 14:08.34 | leifmadsen | "so far" |
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| 14:08.40 | henk | bougyman: and what format are they stored in? mp3, wav, WAV? |
| 14:08.50 | bougyman | henk: mostly wav. |
| 14:09.05 | bougyman | but we have some mp3 and some shoutcast streams for hold music and other stuffs. |
| 14:09.07 | michael-i | leifmadsen: so far is right... gotcha. I'm trying to avoid some really crappy rewriting in my gui to handle both cases now |
| 14:09.08 | henk | bougyman: ok, thanks. |
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| 14:10.08 | michael-i | leifmadsen: my extensions.conf generator needs an overhaul :) thanks for the info |
| 14:10.30 | *** part/#asterisk Mhaddog (~Mhaddog@99-207-100-183.pools.spcsdns.net) |
| 14:10.52 | [TK]D-Fender | MicIs your product released to the public? |
| 14:10.57 | [TK]D-Fender | michael-i: Is your product released to the public? |
| 14:11.00 | leifmadsen | just be aware that if you use a pattern match, and want certain letters literally matched (like X, N, Z or x, n, z) then you have to do something like: exten => _[n]a[n]cy,1,NoOp() |
| 14:11.17 | *** join/#asterisk moy (~moy@UNVLON55-1176057127.sdsl.bell.ca) |
| 14:11.23 | michael-i | [TK]D-Fender: yes : http://www.askozia.com |
| 14:12.00 | leifmadsen | snap |
| 14:13.08 | michael-i | working on 2.1 right now and expanding the applications to also be able to use patterns instead of static extensions...it's snowballing out of control |
| 14:13.31 | leifmadsen | heh |
| 14:13.52 | [TK]D-Fender | michael-i: Is the GUI downloadable separately? |
| 14:13.56 | michael-i | "I'll refactor this in the next release" has finally caught up with me |
| 14:14.14 | michael-i | [TK]D-Fender: no, it's a solid firmware with everything pretty tightly coupled together |
| 14:14.20 | [TK]D-Fender | :/ |
| 14:14.31 | michael-i | I know :) I get that request about once a month |
| 14:14.39 | Naikrovek | that's all? |
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| 14:15.00 | leifmadsen | michael-i: FAQ? :) |
| 14:15.06 | hrhrhr_ | so |
| 14:15.13 | hrhrhr_ | is anyone actually using enum |
| 14:15.17 | hrhrhr_ | forums suggest it's a bit... shit |
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| 14:15.40 | leifmadsen | I've used it a bit, and it works sometimes. We have a chapter in the new book that suggests Freenum.org is a better alternative |
| 14:15.43 | hrhrhr_ | i can't even find a uk number on enumquery |
| 14:16.08 | leifmadsen | does the UK have ENUM provisioned? |
| 14:16.13 | hrhrhr_ | yup |
| 14:16.24 | hrhrhr_ | nominet appear to be looking after it |
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| 14:17.17 | hrhrhr_ | and i guess i would have to open the box up to the internet |
| 14:17.25 | hrhrhr_ | which fills me with a sense of dread |
| 14:19.40 | leifmadsen | hrhrhr_: well you obviously need internet access to be able to place a SIP call.... |
| 14:19.44 | leifmadsen | and to do the lookup |
| 14:19.52 | leifmadsen | there is no getting around that |
| 14:19.56 | hrhrhr_ | i meant for incoming calls |
| 14:20.00 | hrhrhr_ | is isn adoption likely to be any better than enum? |
| 14:20.07 | leifmadsen | probably not |
| 14:20.12 | leifmadsen | depends who you're calling and such |
| 14:20.20 | leifmadsen | and for ENUM you don't have to accept calls to use it |
| 14:20.29 | hrhrhr_ | of course |
| 14:20.41 | leifmadsen | I've set it up to do an ENUM lookup first, and if no response or number, call out SIP provider directly |
| 14:20.51 | hrhrhr_ | but if everyone adopted that mentality there would be no enum i guess |
| 14:21.10 | hrhrhr_ | what's the lookup delay etc like? |
| 14:21.36 | hrhrhr_ | i've tried to look amongst major uk retailers to see if anyone is supporting it and i'm not having much luck |
| 14:21.38 | henk | ok, does anyone have any definitive information on where asterisk looks for sounds? mine are in /var/lib/asterisk/sounds/custom, which is underneath astvarlibdir (/var/lib/asterisk) in asterisk.conf. nevertheless asterisk can neither find 'sounds/custom/myfile' nor 'custom/myfile' in a PlayBack() even though those files do exist with a .gsm suffix. any idea what could be wrong? |
| 14:21.44 | hrhrhr_ | it seems like another great idea fallen |
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| 14:22.00 | michael-i | leifmadsen: just updated my faq with it in there...just takes a nudge for some tasks I guess |
| 14:22.56 | leifmadsen | michael-i: amen on that -- I finally spent time getting authentication and directory control on my subversion server for a couple of clients I was using it for along with multiple mediawiki's using the same code base but separate tables |
| 14:23.08 | leifmadsen | needed a nudge from someone |
| 14:26.50 | patrick^ | is there a way to specify a sip template to use when using asterisk realtime database ? |
| 14:28.26 | patrick^ | in other words can the "name" field in the database contain also a template to use |
| 14:29.18 | patrick^ | as in [1234567890](template1) |
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| 14:55.37 | jamko | Anyone real savvy with reinvites, and T.38?? Have a challenging one for you if you are. |
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| 14:59.38 | anonymouz666 | oh my gosh |
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| 15:05.35 | kalimc | I have a virtual PRI (from http://www.canadids.com) working with my asterisk system, everything is working as expected, I can make and receive calls over my VOIP phone. What I'd like to do is connect my home lines (previously installed by bell) to my asterisk box, without the need for SIP phones. What methods, or hardware would I need to make the demarkation point for my home phone lines connect into my asterisk b |
| 15:06.15 | mroe | ok, gonna do my best: I have a T-1 connected to *. 6 channels of that T-1 are in a 'hunt group'. Other than the main number does each channel need to have DID assigned to it, like how analog hunt groups work? |
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| 15:06.40 | kalimc | Sorry, was disconnected, my question was: |
| 15:06.43 | kalimc | I have a virtual PRI (from http://www.canadids.com) working with my asterisk system, everything is working as expected, I can make and receive calls over my VOIP phone. What I'd like to do is connect my home lines (previously installed by bell) to my asterisk box, without the need for SIP phones. What methods, or hardware would I need to make the demarkation point for my home phone lines connect into my asterisk b |
| 15:06.46 | mroe | kalimc I believe you are asking about analog cards |
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| 15:07.27 | kalimc | Ok, so I would need an FXO card to bridge my phone lines, to the PRI? |
| 15:08.02 | bougyman | FXS |
| 15:08.02 | kalimc | Regular Phone ---> FXO Card ---> Asterisk ---> PRI |
| 15:08.06 | kalimc | ah ok |
| 15:08.08 | kalimc | FXS |
| 15:08.10 | bougyman | S = station |
| 15:08.13 | bougyman | O = office |
| 15:08.22 | bougyman | you want stations connected to asterisk. |
| 15:08.41 | ChannelZ | no it sounds like he wants POTS lines connected to it. |
| 15:08.50 | Qwell | ChannelZ: "home lines" |
| 15:09.00 | Qwell | he wants to connect to the dmarc so he can plug phones in the existing outlets |
| 15:09.02 | Qwell | ie; FXS |
| 15:09.04 | kalimc | Well, I have 2 DID's (via PRI) |
| 15:09.06 | Qwell | BUT! |
| 15:09.16 | kalimc | Qwell, yes |
| 15:09.32 | kalimc | Rogers does this with their phone/modem |
| 15:09.39 | Qwell | obviously, you'll only be able to make one call at a time via those outlets |
| 15:09.51 | Qwell | since, afterall, they are all physically connected |
| 15:09.54 | kalimc | Yes, they gave me a free second channel |
| 15:09.54 | keith4 | oh, you want to replicate something like Vonage? |
| 15:09.56 | bougyman | you could split the wiring, but yeh. |
| 15:10.09 | Qwell | bougyman: yeah that was my next comment. it's a pain depending |
| 15:10.10 | kalimc | I really only need the one |
| 15:10.28 | keith4 | you need whatever the green modules are, from digium ;-) |
| 15:10.37 | bougyman | kalimc: use one for voip and one for your rj-12 lines. |
| 15:10.44 | kalimc | ah ok |
| 15:10.48 | Qwell | it was neat at the last apt I was in. I replaced an outlet *inside*, and because of how the wiring was, it propagated through |
| 15:11.10 | kalimc | so the physical lines, would I connect an RJ-45 cable from the De-mark into the FXS card? |
| 15:11.15 | *** join/#asterisk rafael-ec (~rafael@200.110.234.162) |
| 15:11.21 | bougyman | kalimc: likely rj-12 |
| 15:11.26 | *** join/#asterisk russ (foobar@ip70-176-251-1.ph.ph.cox.net) |
| 15:11.43 | bougyman | and probably a usb single-FXS adapter. |
| 15:11.48 | Qwell | kalimc: RJ-11(12?), really |
| 15:12.06 | [TK]D-Fender | kalYou just want to use a regular phone? |
| 15:12.09 | *** join/#asterisk n3hxs (~HAMming@63.68.135.4) |
| 15:12.11 | kalimc | Yes |
| 15:12.32 | kalimc | I have 3-4 phones in my place here, that all go down to the de-mark in the basement |
| 15:12.38 | [TK]D-Fender | kalimc: Then get an ATA instead |
| 15:12.44 | bougyman | kalimc: just get an ATA |
| 15:12.57 | keith4 | much cheaper |
| 15:12.58 | kalimc | from that demark, I want to be routed to my asterisk that has a pri w/did |
| 15:13.04 | bougyman | http://www.voip-info.org/wiki/view/Analog+Telephone+Adapters |
| 15:13.09 | kalimc | ah ok |
| 15:13.18 | kalimc | I'll look into that |
| 15:13.30 | bougyman | stay away from the linksys/sipura/cisco |
| 15:13.50 | keith4 | mroe: are you asking if you can have multiple channels for a single DID via PRI, like you can with an ITSP? |
| 15:13.51 | kalimc | can anyone recommend a good ATA? |
| 15:14.16 | kalimc | (not too overboard in pricing) |
| 15:14.17 | mroe | keith4, yes. |
| 15:14.21 | russ | zaptel tdm400 |
| 15:14.34 | Qwell | russ: welcome to 2010 :p dahdi, tdm410 |
| 15:14.44 | ruyo | Anyone knows of any problem of a third call comming in in PTP BRI interface? |
| 15:14.47 | kalimc | lol |
| 15:14.53 | ruyo | (With mISDN 1.1.8) |
| 15:15.02 | keith4 | mroe: oh, then i have no idea |
| 15:15.11 | Qwell | ruyo: wouldn't the telco just reject it? |
| 15:15.36 | ruyo | Apparently depends, if you have call forward or hold enabled, it doesn't. |
| 15:16.13 | [TK]D-Fender | kalimc: Linksys PAP2T-NA, or Linksys SPA-2102 |
| 15:16.27 | ruyo | The problem is that Asterisk isn't replying "busy", so I get a line unavailable kind of tone |
| 15:16.28 | *** part/#asterisk rafael-ec (~rafael@200.110.234.162) |
| 15:16.31 | ruyo | Like congestion. |
| 15:16.31 | kalimc | thankyou TK, I will look at them now |
| 15:16.31 | mroe | thanks for the question clarification, it will probably help get my question answered |
| 15:17.16 | *** join/#asterisk sgimeno (~chatzilla@163.117.211.10) |
| 15:19.38 | kalimc | Just to clarify, if I were to get either ( Linksys PAP2T-NA, or Linksys SPA-2102) would that remove the need for asterisk (plug in settings to router) or would I still be able to use asterisk with it? |
| 15:19.54 | ruyo | I get this on mISDN debug when I place a third call to asterisk --> http://pastebin.com/wJrDu5vb |
| 15:20.05 | kalimc | I'd like to get the benefits of voicemail, and all the asterisk goodies. |
| 15:20.07 | bougyman | keith4: i like http://www.yealink.com/en/view.asp?ClassLayer=76&t_ENName=IP%20Phone&p_Number=SIP-GW3CM |
| 15:20.16 | [TK]D-Fender | kalimc: the ATA jsut lets you use analog phones as SIP devices. That doesn't mean they LEAD anywhere |
| 15:20.25 | *** join/#asterisk Ad-Hoc (~nimbus@62.1.222.111.dsl.dyn.forthnet.gr) |
| 15:20.27 | keith4 | bougyman: good to know ;-) |
| 15:20.30 | [TK]D-Fender | kalimc: So what would be processing the CALLS made using it? |
| 15:20.42 | [TK]D-Fender | [11:06]<mroe>ok, gonna do my best: I have a T-1 connected to *. 6 channels of that T-1 are in a 'hunt group'. Other than the main number does each channel need to have DID assigned to it, like how analog hunt groups work? <- Whcat kind of T1? |
| 15:20.46 | kalimc | I see, so I could then connect the sip devices through asterisk! :) |
| 15:21.05 | kalimc | Correct? |
| 15:21.49 | bougyman | correct.. |
| 15:21.57 | keith4 | [TK]D-Fender: he said "yes" when I assumed he was referring to PRI... |
| 15:22.07 | [TK]D-Fender | keith4: I assume as little as possible |
| 15:22.11 | bougyman | with the yealink i posted you wouldn't need a phone switch, though, unless you wanted it. |
| 15:22.13 | mroe | [TK]D-Fender, yes that was my question |
| 15:22.14 | keith4 | bougyman: what's so great about the yealink? |
| 15:22.27 | mroe | [TK]D-Fender, keith4 summed it up a bit better for me |
| 15:22.34 | bougyman | keith4: just low price and nice quality, it's a value judgement. |
| 15:22.37 | kalimc | ok perfect. thank you for your help. I am sure this is commonly done these days, so I will google for instructions. |
| 15:22.51 | [TK]D-Fender | mroe: Normally PRI's don't have subset hunt groups. All DID's pointed at them can land on any free channel. |
| 15:23.07 | [TK]D-Fender | mroe: I've seen some very rare exceptions to this however |
| 15:23.09 | keith4 | bougyman: where could I buy one? |
| 15:23.27 | bougyman | http://yealinkstore.com/ |
| 15:23.39 | mroe | [TK]D-Fender, sorry, I don't really understand your answer |
| 15:24.01 | [TK]D-Fender | keith4: Yealink, GAH. What do you actually need? |
| 15:24.19 | keith4 | [TK]D-Fender: nothing. i just wanted to investigate his value claims |
| 15:24.33 | keith4 | i've never actually used an ATA, but i keep meaning to get one to play with |
| 15:24.34 | bougyman | [TK]D-Fender: he wants an ata, he's been recommended a 410 , 400, and the yealink |
| 15:24.43 | [TK]D-Fender | mroe: Channels are not normally assigned DID's. All DID's aimed at your PRI can fall on any open channel normally. You don't assign "groups", etc |
| 15:24.45 | keith4 | bougyman: actually, that was kalimc |
| 15:24.50 | bougyman | oh, heh. |
| 15:25.01 | [TK]D-Fender | keith4: PAP2T-NA for you then as you[re UK |
| 15:25.07 | keith4 | am not! |
| 15:25.16 | keith4 | is insulted |
| 15:25.51 | [TK]D-Fender | [11:24]<bougyman>[TK]D-Fender: he wants an ata, he's been recommended a 410 , 400, and the yealink <- First two aren't even ATA's, and the third is a cheap-shit Chinese device no regular here would ever recommend. |
| 15:26.03 | [TK]D-Fender | keith4: Weren't you? |
| 15:26.18 | [TK]D-Fender | keith4: Perhaps I've mixed you up along the way... whereabouts? |
| 15:26.32 | bougyman | keith4: the yealink outperforms the sipuras, i haven't tried the 400 or 410 cards. |
| 15:27.22 | bougyman | er [TK]D-Fender. |
| 15:27.23 | mroe | [TK]D-Fender, thanks, so I can yell at our telco for being dumb |
| 15:27.44 | [TK]D-Fender | bougyman: Outperforms in what way? |
| 15:27.51 | bougyman | i've tried the zorcoms, sipuras, yealink, audiocodes, and a few others. |
| 15:28.00 | bougyman | [TK]D-Fender: works. |
| 15:28.14 | bougyman | the sipuras have more than a few critical bugs. |
| 15:28.17 | [TK]D-Fender | bougyman: So do all the Linksys models |
| 15:28.31 | [TK]D-Fender | bougyman: And Sipura hasn't existed for YEARS |
| 15:28.32 | bougyman | yes, i mean sipura, cisco, and the linksys stuff. |
| 15:28.52 | bougyman | they all share at least one of these bugs. |
| 15:29.03 | [TK]D-Fender | [11:28]<bougyman>the sipuras have more than a few critical bugs. <- Please do share in detail as they are what most of us here use, recommend, and are happy with |
| 15:29.04 | bougyman | and cisco refuses to fix it, i told them about it over a year ago. |
| 15:29.23 | bougyman | inbound caller id, if it has a comma, is not quoted in the sip header. |
| 15:29.39 | bougyman | that breaks all sorts of stuff. |
| 15:29.57 | keith4 | [TK]D-Fender: US |
| 15:30.10 | [TK]D-Fender | keith4: www.telephonydepot.com |
| 15:30.32 | [TK]D-Fender | keith4: How many ports do you need? |
| 15:30.37 | keith4 | [TK]D-Fender: yah, that's my usual vendor. they don't have the yealink ATAs, that I can see. (not that I want to get one, necessarily) |
| 15:30.55 | keith4 | i would've defaulted to the linksys pap2, probably. that's sufficient for testing, no? |
| 15:31.07 | bougyman | sure. |
| 15:31.29 | keith4 | ... which is good, because they don't have much for ATAs, other than cisco/linksys |
| 15:31.39 | keith4 | (or grandstream. ick) |
| 15:32.04 | [TK]D-Fender | keith4: If you only need 1 port, then get your money's worth and get the SPA-3102 instead as that'll also give you an FXO interface, and integrated router (which you don't have to use) |
| 15:32.09 | [TK]D-Fender | ~gs |
| 15:32.09 | infobot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
| 15:32.11 | [TK]D-Fender | ^^^ |
| 15:32.22 | keith4 | ha. *extreme* prejudice |
| 15:32.32 | bougyman | the spa3102's are the ones that gave us the most headache. |
| 15:32.46 | bougyman | had to write http://gitorious.org/spa3102-invite-packet-scrubber just to be able to use them at all. |
| 15:34.22 | keith4 | interesting |
| 15:34.25 | bougyman | if you want a spa3102 i have boxes of them. |
| 15:34.38 | keith4 | just found one on ebay for $9.99. can you beat that? |
| 15:34.41 | bougyman | paperweights, you pay shipping i'll send you onw. |
| 15:34.43 | bougyman | :) |
| 15:34.52 | jamko | T.38 reinvite issue. Anyone? |
| 15:35.46 | [TK]D-Fender | bougyman: I do see one consistent element if your claims about problems with these devices. YOU :p |
| 15:36.09 | bougyman | [TK]D-Fender: I'm responsible for the firmware not quoting From headers? |
| 15:36.16 | bougyman | i hadn't looked at it like that before. |
| 15:36.36 | keith4 | heh |
| 15:37.51 | *** part/#asterisk mheadd (~mheadd@c-68-45-252-237.hsd1.de.comcast.net) |
| 15:38.42 | ruyo | Did ISDN die everywhere, including Europe? |
| 15:39.08 | Qwell | ruyo: ...no |
| 15:39.42 | ruyo | * ISDN BRI |
| 15:40.01 | Qwell | ruyo: no |
| 15:40.34 | ruyo | Ok. |
| 15:40.35 | *** join/#asterisk imox1234 (~imox1234@p4FC5C507.dip0.t-ipconnect.de) |
| 15:41.08 | ruyo | Is just that I read a lot people complaining about the future of the development of mISDNv2 |
| 15:41.51 | *** join/#asterisk ariel_ (~chatzilla@c-24-127-196-248.hsd1.fl.comcast.net) |
| 15:42.18 | *** join/#asterisk CunningPike (~CunningPi@204.239.8.157) |
| 15:43.09 | [TK]D-Fender | ruyo: That's not the only driver youe know... |
| 15:43.44 | keith4 | bougyman: so where would you be shipping these 3102s from? ;-) |
| 15:43.46 | *** join/#asterisk shadey_ (~blah@213.1.224.2) |
| 15:44.54 | ruyo | [TK]D-Fender, yeah, but it's the better implemented so far, no? |
| 15:46.59 | bougyman | keith4: dallas |
| 15:47.09 | bougyman | you want two? |
| 15:48.31 | *** join/#asterisk Z_God (~julius@2001:610:1908:8000:21e:ecff:fe5d:679e) |
| 15:55.41 | *** join/#asterisk Gugge (~gugge@vlan2.dlxhosting.dk) |
| 15:56.03 | *** join/#asterisk nix8n82 (~nathan@63.162.27.14) |
| 15:56.47 | *** join/#asterisk lost_soul (~noymfb@cpe-67-249-141-227.twcny.res.rr.com) |
| 15:58.33 | *** join/#asterisk w9sh (~chatzilla@adsl-068-209-117-205.sip.asm.bellsouth.net) |
| 15:58.35 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
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| 16:06.41 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 16:09.53 | carrar | Y*A*W*N |
| 16:15.50 | *** join/#asterisk b11d` (~no@234-200-29-134.hcc.mnscu.edu) |
| 16:15.53 | b11d` | hey everyone |
| 16:16.23 | b11d` | just curious.. on 1.8, if I have a PRI configured, but not plugged in, I should be seeing a "No D-Channel available, using Channel 24 as default anyway" warning, correct? |
| 16:16.33 | b11d` | Im just trying to understand if I'm seeing correct behavior here or not |
| 16:17.44 | *** join/#asterisk Wimd (~Wim_ctx@62-213-207-59.colo.kangaroot.net) |
| 16:18.30 | Wimd | hey, i have some weird problem with an URI dial. it works on a test trixbox, but on the plain asterisk server i actually need it on it wont work |
| 16:19.02 | Wimd | the thing is that i do dial(sip/num@host) and i get a sip response from a totally diffrent host |
| 16:19.33 | Wimd | when i do it on the other asterisk the sip invite just keeps retransmitting |
| 16:20.04 | Wimd | ive been looking around and all i can see for uri calls is that srvlookup=yes needs to be set, wich is the case |
| 16:20.37 | *** join/#asterisk Shaaan (~Un1x@CPE000024cccb7c-CM0014045acc3c.cpe.net.cable.rogers.com) |
| 16:25.35 | Naikrovek | anyone know when Polycom SIP firmware 3.3 is going GA |
| 16:26.40 | mroe | is there anything interesting in the release? |
| 16:27.34 | *** join/#asterisk QubeZ (~nkasu@64.128.254.34) |
| 16:27.40 | QubeZ | hello all |
| 16:27.49 | QubeZ | is there a way to use chanspy but not have it cycle through extensions, but rather allow the user to enter it? |
| 16:28.14 | *** join/#asterisk devmod (~devmod@c-76-100-208-204.hsd1.md.comcast.net) |
| 16:29.16 | [TK]D-Fender | Wimd: Trixbox uses * so your comparison means nothing |
| 16:29.25 | [TK]D-Fender | Wimd: and we don't see configs & debug |
| 16:29.58 | [TK]D-Fender | QubeZ: Do your own READ() first |
| 16:30.20 | QubeZ | [TK]D-Fender: ok |
| 16:33.15 | jamko | Hello, need some help with a T.38 config issue on asterisk 1.6.2.10.. Just want to pass the traffic through from sip ata, to sip provider.. Both have T.38 capability, but call fails if reinvite comes from provider. Works fine if reinvite comes from mediatrix ata. Anyone? I have full debug dumps, and pcap logs ready and waiting |
| 16:37.37 | *** join/#asterisk grEvenX (~even@cC0FD00C3.dhcp.bluecom.no) |
| 16:39.06 | Wimd | [TK]D-Fender, http://pastebin.com/8ndSKQJs |
| 16:39.42 | Wimd | first part is a working invite |
| 16:40.12 | [TK]D-Fender | Wimd: Contact: <sip:547@192.168.17.248> <------- you clearly haven't set your * up to work properly from behind NAT |
| 16:40.40 | Wimd | well, that one is actually the one that does work |
| 16:40.51 | [TK]D-Fender | Wimd: It's wrong. |
| 16:41.27 | Wimd | i can dial number@ip from the server with 192.168... thats behind nat |
| 16:41.32 | [TK]D-Fender | Wimd: And your second gets no answer and you've provided no details, and masked numbers |
| 16:41.47 | [TK]D-Fender | Wimd: We don't have a clear description of where each server is, etc |
| 16:41.49 | Wimd | the first and second dial ar the same uri |
| 16:42.21 | [TK]D-Fender | Wimd: Doesn't matter who you call when youa re configued in a way that you can even get an ANSWER |
| 16:42.27 | [TK]D-Fender | can't |
| 16:42.56 | [TK]D-Fender | Wimd: It's not the callee that is the problem, its the caller |
| 16:44.30 | Wimd | so, if i have an asterisk, directly on the internet so no nat and i try to call a number@host and hot get any answer |
| 16:44.53 | Wimd | and then i have another server, natted, and when i do the same thing i do get an answer |
| 16:45.13 | Wimd | where should i be looking? ive temporarely disabled iptables on the public server, no diffrence |
| 16:45.33 | QubeZ | [TK]D-Fender: would you take a quick peek at this and offer thoughts? http://pastebin.com/XVMNXqYC |
| 16:45.51 | bougyman | have you tried canreinvite=no in your sip.conf, Wimd? |
| 16:46.29 | Qwell | canreinvite is for rtp. you'd get one-way audio, not call failure |
| 16:46.36 | *** join/#asterisk devmod (~devmod@c-76-100-208-204.hsd1.md.comcast.net) |
| 16:46.59 | [TK]D-Fender | Wimd: You have masked important info and I have no proof any of what you did is legit, or info on your networking, etc. |
| 16:47.09 | [TK]D-Fender | Wimd: You offer nothing for us to debug |
| 16:47.41 | jamko | Hello, need some help with a T.38 config issue on asterisk 1.6.2.10.. Just want to pass the traffic through from sip ata, to sip provider.. Both have T.38 capability, but call fails if reinvite comes from provider. Works fine if reinvite comes from mediatrix ata. Anyone? I have full debug dumps, and pcap logs ready and waiting |
| 16:48.37 | Wimd | well, [TK]D-Fender, if you need to know the axact uri, its 0103004410@77.73.226.25 |
| 16:49.06 | Naikrovek | we need debug logs, some error messages from a sip debug or the asterisk log |
| 16:49.07 | Naikrovek | something |
| 16:49.08 | [TK]D-Fender | Wimd: No, we need to see the FULL packets, and I we can't see that IP's are at all right. No firewall settings, nothing. |
| 16:49.24 | [TK]D-Fender | Wimd: Play this game and you'll lose right out of the gate. |
| 16:58.55 | *** join/#asterisk Shane-S (~chatzilla@c-68-32-239-184.hsd1.nj.comcast.net) |
| 16:59.02 | *** join/#asterisk mathslinux (~root@120.42.46.126) |
| 16:59.26 | *** join/#asterisk bklang (~bklang@tesla.alkaloid.net) |
| 17:00.11 | *** join/#asterisk Qwell (~north@pdpc/sponsor/digium/Qwell) |
| 17:00.11 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 17:00.13 | *** join/#asterisk TheDavidFactor (~chatzilla@nc-71-52-237-137.dhcp.embarqhsd.net) |
| 17:01.06 | bklang | Hello all. I'm trying to use AMI to detect DTMF for calls in-progress. I have it working fine for some calls, but not others. The difference seems to be which SIP peers are involved. But I want to verify my basic assumption, that Asterisk should generate DTMF AMI events even when two channels are bridged? (I am using Asterisk 1.6.1 and 1.8.0-beta3 and canreinvite=no to keep the media local) |
| 17:01.17 | Naikrovek | nice |
| 17:01.51 | bougyman | bklang: using rfc2833compensate or no? |
| 17:02.34 | bklang | no, but dtmfmode=rfc2833 |
| 17:03.44 | bklang | oddly, the DTMF detection works fine before and after the call is bridged to an outbound channel |
| 17:03.53 | bklang | it only seems to fail during the period the call is bridged |
| 17:04.00 | bklang | and I get events to support that |
| 17:04.19 | bklang | Enabling RTP debugging shows that I can see the rfc2833 information being passed through Asterisk |
| 17:04.23 | bklang | and the DTMF shows up on the far end as well |
| 17:04.29 | bklang | it's only the AMI events that appear to be missing |
| 17:08.14 | TheDavidFactor | I've got a client using sangoma PRI cards that's having problems with DTMF being lost. The cards do not have hardware EC so no hardware DTMF. They're using a really old asterisk (1.4.2) and libpri (1.4.4) we've already added relaxdtmf=yes; I would assume that PRI's are sending DTMF out of band so it should be pretty reliable. Before they spend money on HWEC cards can anyone tell me if it... |
| 17:08.16 | TheDavidFactor | ...would be worth upgrading to the latest libpri? if so, which version of asterisk is required to use the latest libpri? |
| 17:09.13 | *** join/#asterisk jshriver (~jshriver@cblmdm24-53-177-197.buckeyecom.net) |
| 17:09.17 | jshriver | Greetings |
| 17:09.32 | jshriver | anyone know how to test callerid functionality on an asterisk box? |
| 17:10.31 | Shane-S | Hi, work asked me to look into phone systems, I am 100% newbie to phones. We current have 4 POTS connecting to a phone server, that then has a unit powering digital phones, but they use cat3/rj11 connectors, I assume asterisk can run them, but what is needed to communicate with them |
| 17:10.38 | *** join/#asterisk xpot-mobile (~james@c-98-202-72-167.hsd1.ut.comcast.net) |
| 17:10.59 | jshriver | Computer running asterisk, and a digium card with FXO chips |
| 17:11.30 | Qwell | Shane-S: Asterisk does not work with "digital" phones. |
| 17:11.44 | Qwell | unless somebody makes (usually very expensive) hardware for them. |
| 17:11.45 | bklang | Shane-S: digital phones are almost always completely proprietary. You may be able to get Asterisk to talk to your digital phone system, but it will be over a standard interface (such as T-1, SIP or analog) |
| 17:11.49 | Naikrovek | Shane-S: hardware is available to interface with those POTS lines, but you'll need to get some new phones |
| 17:12.18 | Naikrovek | Shane-S: the phones are cheap and good |
| 17:12.35 | Naikrovek | i was thinking about this the other day |
| 17:12.51 | jshriver | aye you'll need sip phones |
| 17:12.56 | Naikrovek | people go barking "I WANT VOIP" then they see what a cisco solution costs then they go around barking "VOIP IS DEAD" |
| 17:13.09 | Shane-S | thanks |
| 17:13.17 | Naikrovek | but an Asterisk & Polycom solution is pretty damn cheap |
| 17:13.24 | jshriver | and save yourself some headache and buy a premade box with service plan. Running an el-cheapo one here and it's a flippin nightmare. FXO chips burn out every storm, instable.. just a nightmare |
| 17:13.30 | Naikrovek | for what you get it's a bargain and a half |
| 17:14.39 | Naikrovek | Shane-S: I was in a similar situation to you just 2 years ago |
| 17:14.39 | xuser | jshriver: do you ground the lines? |
| 17:14.45 | jshriver | I have wondered though why does asterisk itself not have a # sip phones limit, but if you buy a box from say Digium you're only allowed n phones. Dont see why the # of phones has anything to do with pricing |
| 17:14.48 | Shane-S | Does the digium site have phone prices, so I can at least show work what a solution might cost compared to the company we have wanting to bring in a phone server...that umm...is a pentium III desktop...which scared me |
| 17:15.07 | jshriver | xuser: yup, each line is on a $100 phone line protector that's grounded. |
| 17:15.08 | Naikrovek | Shane-S: how many digital phones do you have |
| 17:15.25 | jshriver | Elk-955 |
| 17:15.56 | Shane-S | about 40, but 5 phones are the advanced with an LCD and/or phone bank light/button to monitor who is on the lines |
| 17:16.03 | Naikrovek | okay |
| 17:16.04 | Naikrovek | easy |
| 17:16.23 | Shane-S | the rest just have number pade and the outgoing/voicemail/transfer buttons |
| 17:16.27 | Naikrovek | okay |
| 17:16.35 | Naikrovek | i'll PM you some details to get started |
| 17:16.42 | Shane-S | thanks |
| 17:17.49 | *** join/#asterisk Crashvr (~arthur@200-170-196-78.core01.spo.ifx.net.br) |
| 17:19.24 | [TK]D-Fender | Shane-S: What model of phones exactly? |
| 17:19.49 | Shane-S | I am not at work...hang on let me text the maintenance guy |
| 17:23.09 | *** join/#asterisk gamedna (~gamedna@cpe-70-125-155-74.satx.res.rr.com) |
| 17:25.14 | gamedna | Shaaan: are you around? |
| 17:34.45 | *** join/#asterisk Sargun (~Sargun@atarack/Staff/Sargun) |
| 17:35.17 | *** join/#asterisk rgsteele (~rgsteele@apo.aweber.com) |
| 17:37.23 | *** join/#asterisk mayfield (~mayfaild@76-250-152-224.lightspeed.snantx.sbcglobal.net) |
| 17:40.52 | Sargun | What SIP/IAX providers do you guys use for termination and origination in the UK? |
| 17:42.55 | keith4 | ~itsp |
| 17:42.56 | infobot | [~itsp] An ITSP is an Internet Telephony Service Provider (or VoIP telephone company). They allow you to either SEND calls to the PSTN (this is called termination), RECEIVE calls from the PSTN (called origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs. |
| 17:43.12 | Chainsaw | ~itsplist-uk |
| 17:43.13 | infobot | hmm... itsplist-uk is UK based ITSps include http://www.voiptalk.org/ http://www.voipon.co.uk/ http://www.gradwell.com/ and a few other tinpot companies you can dig up with google. |
| 17:43.13 | keith4 | guess there's not a ~itsplist-uk? |
| 17:43.18 | keith4 | ah, there is |
| 17:43.33 | Chainsaw | Of course there is. |
| 17:43.46 | keith4 | i vote for the tinpot companies |
| 17:50.14 | TheDavidFactor | is there a PRI expert who can tell me how to tell if DTMF is being sent inbound or out-of-band on a PRI? |
| 17:50.33 | Shane-S | [TK]D-Fender: well the maintenance guy is gonna be awhile, I will tell you phones when I go in tomorrow :D |
| 17:51.05 | [TK]D-Fender | Shane-S: What overall system is it? |
| 17:51.07 | jshriver | quit |
| 17:52.24 | Shane-S | I want to say vodavi...but that is wrong. I have only seen the voicemail server, so I will have to check the PBX, it is in a closet I never go into |
| 17:52.42 | Shane-S | let me call into work and see if the secreatary can tell me the phone brand name at least |
| 17:53.12 | *** join/#asterisk bkruse1 (~bkruse@75.76.105.124) |
| 17:53.12 | *** mode/#asterisk [+o bkruse1] by ChanServ |
| 17:53.29 | *** join/#asterisk crazybyte (~crzp@unaffiliated/crazypenguin/x-000001) |
| 17:54.14 | Shane-S | okay phones are branded with infinite |
| 17:54.28 | Crashvr | hello guys, someone will explain to me an efficient way to make calls using SIP timeout in time for the return of the ring, and not by time to answer. |
| 17:54.33 | *** join/#asterisk paulc (~paulc@unaffiliated/paulc) |
| 17:54.36 | Naikrovek | Shane-S: http://www.buymebuyme.com/retail/customer/product.php?productid=46421&cat=387&partner=froogle ? |
| 17:54.56 | Naikrovek | looks like vodavi may be right |
| 17:55.07 | Shane-S | hehe yeah I was right :P |
| 17:55.16 | Naikrovek | http://www.buymebuyme.com/retail/skin1/prodimages/46421.jpg |
| 17:55.23 | Shane-S | now the PBX...I never look at long enough to know the name :P |
| 17:55.32 | Naikrovek | probably vodavi as well |
| 17:55.59 | mroe | buymebuyme.com? |
| 17:56.07 | Naikrovek | i dunno i found it on google images |
| 17:56.52 | Crashvr | if anyone knows please send me message! |
| 17:57.00 | [TK]D-Fender | Shane-S: those phones are all but certainly unusable with * |
| 17:57.05 | Naikrovek | yeah |
| 17:58.06 | [TK]D-Fender | Crashvr: Not happening. That would require large rwrites to chan_sip, app_dial, and their underpinnings |
| 17:58.26 | Shane-S | alright, can * communicate/integrate with the PBX, and handle the POTS/Voicemail/Message parts, or is that all PBX...sorry totally new to this |
| 17:59.55 | fauxalliance | infobot Naikrovek++ |
| 18:00.34 | Shane-S | I only ask to know if there is a middle of the road "upgrade" price to offer, or if I am just going to report back its $6,000+ no matter what we pick |
| 18:00.37 | Naikrovek | :) |
| 18:01.01 | *** join/#asterisk qvsqvs (~anonymous@41.31.119.242) |
| 18:06.31 | *** join/#asterisk Micc_ (~quassel@c-98-225-57-96.hsd1.wa.comcast.net) |
| 18:07.30 | *** join/#asterisk imox1234 (~imox1234@p4FC5C507.dip0.t-ipconnect.de) |
| 18:08.13 | *** join/#asterisk andresmujica (~andresmuj@ubuntu/member/andresmujica) |
| 18:10.20 | bklang | Alright, I've narrowed down my DTMF problem. When using alaw as the codec, DTMF events are not generated when bridged. When using ulaw, DTMF events are generated when bridged. DTMF events work fine on both codecs when the call is not bridged between two channels |
| 18:10.24 | bklang | And this is using rfc2833, not inband |
| 18:11.04 | bklang | I would not have thought the codec would make a difference. Isn't that the point of rfc2833? |
| 18:11.24 | Naikrovek | well it may be a bug |
| 18:11.33 | bklang | Yeah, guess I'll open a ticket |
| 18:11.34 | Naikrovek | but yes you're right, that's the point of out-of-band DTMF |
| 18:12.02 | bklang | strange that I see the same behavior in both Asterisk 1.6.1 and 1.8. If it is a bug, it's a longstanding one. I'm kinda surprised I'd the be the first to hit it |
| 18:12.11 | Naikrovek | well |
| 18:12.19 | Naikrovek | in the US, ulaw rules the roost |
| 18:12.40 | Naikrovek | europe uses ulaw but maybe they don't use asterisk or ... hell i dunno |
| 18:12.45 | bklang | heh yeah |
| 18:12.50 | Naikrovek | europe uses alaw* |
| 18:13.26 | *** join/#asterisk RobH (~robh@wikimedia/RobH) |
| 18:15.51 | WIMPy | Hmm. Seems to work for me, but I'm not sure what the phone uses, apart from that it's not in band. |
| 18:15.52 | [TK]D-Fender | Codec doesn't dertermine RFC2833 |
| 18:16.54 | *** join/#asterisk chuckp (~chuckp@c-76-106-198-76.hsd1.fl.comcast.net) |
| 18:18.04 | Naikrovek | yeah that's what we mean |
| 18:18.16 | Naikrovek | rfc2833 shoudl work independent of any codec |
| 18:20.42 | rustyclarkson | Which media player does asterisk use to playback MOH? (in mode=files) |
| 18:20.44 | rustyclarkson | aplay? |
| 18:24.01 | Naikrovek | i always thought asterisk played them itself |
| 18:24.05 | Naikrovek | but i could be wrong |
| 18:24.41 | gamedna | bklang: i just fired up asterisk 1.8 w/ wireshark and zoiper configured for alaw |
| 18:24.56 | Naikrovek | conclusion? |
| 18:25.20 | bklang | Keep in mind that DTMF is working with ALAW. It's only in a specific circumstance I have an issue |
| 18:25.30 | gamedna | im geting rtpevents out of band |
| 18:25.54 | bklang | It's only when the incoming call (using alaw) is bridged with an outbound call. Even then DTMF works (it is properly sent to the far end) but AMI "DTMF" events are not triggered |
| 18:25.59 | bklang | that's really my issue, I need those DTMF events |
| 18:26.09 | gamedna | RFC 2833 RTP EVENT, EventID: DTMF Four (4) |
| 18:26.14 | bklang | the oddity is that the events work fine with GSM and ULAW |
| 18:26.39 | bklang | Right, I get the events when the call is accepted by Asterisk, it's only after I place an outgoing call and the channels are bridged that the events stop |
| 18:26.44 | gamedna | let me try w/ a Bridge() in th callplan |
| 18:26.53 | bklang | just try a Dial() in the dialplan |
| 18:27.03 | bklang | wait for the call to be accepted by the far end and then see if you get DTMF events |
| 18:27.20 | gamedna | oh, that i did |
| 18:27.30 | gamedna | yes, im getting the events on the far side |
| 18:27.39 | bklang | I'm not talking about RTP events, I'm talking about AMI events |
| 18:27.49 | bklang | RTP works end-to-end, it's the AMI events that go missing |
| 18:28.24 | gamedna | so, RTP event goes from phone to asterisk, then when you dial out to another endpoint the event dissapears |
| 18:28.49 | bklang | no, the RTP events work fine. When DTMF is detected by Asterisk it should write an AMI event to all AMI listeners who have events enabled. |
| 18:29.19 | gamedna | well, this is my setup... |
| 18:29.33 | gamedna | zoiper -> Asterisk box 1 -> IAX TRUNK -> Asterisk Box 2 |
| 18:29.46 | gamedna | dial in to box 1 |
| 18:29.52 | gamedna | press 1 to dial to other box |
| 18:29.58 | *** join/#asterisk grolloj (~chatzilla@h-68-166-73-162.nycmny83.static.covad.net) |
| 18:29.59 | gamedna | calls out to the other box |
| 18:30.03 | gamedna | press 1 do play monkeys |
| 18:30.12 | gamedna | and monkeys play |
| 18:30.15 | bklang | Yes, that works fine for me |
| 18:30.20 | gamedna | ok |
| 18:30.28 | gamedna | i guess im not really following then |
| 18:30.32 | bklang | To see what I'm talking about you have to open a connection to the Asterisk Manager Interface and request taht events be sent to you on that socket |
| 18:30.51 | bklang | you will see the DTMF events when you press "1" to connect to Asterisk Box 2 |
| 18:31.02 | bklang | but you probably won't see the DTMF event when you press "1" to hear monkeys |
| 18:31.15 | gamedna | but that works under ulaw |
| 18:31.54 | bklang | you're seeing events on the AMI connection? |
| 18:34.17 | gamedna | bklang, one moment let me set that all up |
| 18:35.19 | *** join/#asterisk philipp64|laptop (~chatzilla@75-92-150-245.war.clearwire-wmx.net) |
| 18:39.13 | gamedna | bklang: what is the command you issue to get the events? |
| 18:39.22 | gamedna | im logged into my AMI on both boxes |
| 18:39.45 | bklang | Action: login |
| 18:39.46 | bklang | Events: on |
| 18:40.05 | bklang | you can also do |
| 18:40.10 | gamedna | im already logged in |
| 18:40.11 | bklang | Action: Events |
| 18:40.14 | gamedna | k |
| 18:40.23 | bklang | Eventmask: On |
| 18:40.40 | gamedna | ok |
| 18:40.48 | bklang | brb 10 minutes, picking up my daughter from the bus |
| 18:41.46 | gamedna | bklang: confirmed its working on box one, not receving on box 2 |
| 18:42.09 | gamedna | bklang: switching to ulaw and trying again |
| 18:42.54 | imox1234 | hey, i was wondering if someone could help me with a problem i have... I already posted it in the issues forum of asterisk but after some talk there is no more response https://issues.asterisk.org/view.php?id=17601 |
| 18:43.22 | gamedna | bklang: you are right... |
| 18:43.33 | gamedna | DTMF shows up on both for ulaw and gsm |
| 18:43.58 | Naikrovek | but not alaw? |
| 18:44.01 | imox1234 | i already tried updating to 1.6.2.11 |
| 18:44.02 | gamedna | bklang: but DTMF does not show up on the second box under AMI when running alaw |
| 18:44.08 | gamedna | Naikrovek: correct |
| 18:44.11 | Naikrovek | bug! |
| 18:44.14 | gamedna | Naikrovek: im running 1.8 |
| 18:44.27 | Naikrovek | well crap we gotta get that fixed before release brah |
| 18:44.48 | gamedna | Naikrovek: Asterisk SVN-trunk-r280910 built by nick @ NickMAC.local on a i386 running Darwin on 2010-08-05 09:02:54 UTC |
| 18:44.51 | Naikrovek | and by "we" i mean "not me because i don't work there" |
| 18:45.18 | *** join/#asterisk SaiSoma (~chatzilla@client105.jdcc.edu) |
| 18:45.28 | gamedna | that is the first box |
| 18:45.32 | gamedna | second box is... |
| 18:46.09 | gamedna | Asterisk 1.6.0.26-FONCORE-r78 built by root @ revisor.trixbox.com on a i686 running Linux on 2010-06-08 22:01:27 UTC |
| 18:46.18 | gamedna | let me reverse and see if it changes |
| 18:46.22 | gamedna | calling from 1.6 to 1.8 |
| 18:46.23 | Naikrovek | trixbox?! |
| 18:46.30 | gamedna | so? |
| 18:46.34 | Naikrovek | ick |
| 18:47.00 | gamedna | its just one of my testing boxes |
| 18:47.09 | Naikrovek | i'm just kiddin anyway |
| 18:47.13 | gamedna | haha |
| 18:47.14 | Naikrovek | i run it in production |
| 18:47.22 | Naikrovek | can't wait to get rid of it tho |
| 18:47.31 | gamedna | same here, just love their phone provisioning |
| 18:47.41 | Naikrovek | heh i don't even use that part |
| 18:48.05 | gamedna | love that i can provision 60 phones in under 30 mins |
| 18:48.14 | Naikrovek | yeah |
| 18:48.20 | Naikrovek | i wrote a perl script for that |
| 18:48.22 | anonymouz666 | what phones? |
| 18:48.34 | gamedna | polycom, grandstream, and aastra |
| 18:49.20 | gamedna | polycoms take a while to boot up, but that is not my problem really |
| 18:49.30 | gamedna | i can bulk add all my extentions and just hand out phones |
| 18:49.48 | Naikrovek | yeah |
| 18:49.53 | Naikrovek | that's what i do |
| 18:49.56 | gamedna | freepbx supposedly has a module now for this,b ut i have not really tested it |
| 18:49.56 | Naikrovek | handy dandy i tell ya |
| 18:50.03 | gamedna | Naikrovek: oh yea |
| 18:50.05 | *** join/#asterisk JimVanM (~jimvanm@bas1-toronto06-2925209635.dsl.bell.ca) |
| 18:50.30 | gamedna | bklang: i reversed the scenario... asterisk 1.6 -> 1.8 box and the same thing happens |
| 18:50.39 | gamedna | bklang: ulaw works, and alaw does not work |
| 18:50.53 | Naikrovek | time to head to #asterisk-dev and let them know |
| 18:50.56 | Naikrovek | or file bug rept |
| 18:50.59 | Naikrovek | report |
| 18:51.02 | gamedna | ill let bklang handle that |
| 18:51.07 | gamedna | ;) |
| 18:51.13 | gamedna | since i dont use ami much |
| 18:55.08 | gamedna | crazy |
| 18:56.00 | gamedna | bklang: i THINK the packets are being sent from box 1 to box 2 but they are not being interpreted by box 2 |
| 18:59.37 | *** join/#asterisk qvsqvs (~anonymous@41.28.117.117) |
| 19:05.25 | *** join/#asterisk nix8n82 (~nathan@63.162.27.14) |
| 19:10.45 | bklang | gamedna: yep, ulaw and gsm work, alaw does not |
| 19:10.50 | bklang | https://issues.asterisk.org/view.php?id=17843 |
| 19:10.52 | gamedna | k |
| 19:11.40 | gamedna | well i confirm it |
| 19:11.55 | gamedna | its pretty strange |
| 19:12.14 | bklang | yeah, it is strange |
| 19:12.14 | bklang | I |
| 19:12.33 | bklang | I get the warnings when I use a codec other than ulaw or gsm about inband DTMF detection, I suspect it's an errant check along those lines |
| 19:14.01 | imox1234 | have nobody any idea? |
| 19:14.11 | imox1234 | https://issues.asterisk.org/bug_view_advanced_page.php?bug_id=17601 |
| 19:15.08 | *** join/#asterisk ManxPower (~manxpower@216.186.151.147) |
| 19:15.14 | ManxPower | ~answers |
| 19:15.15 | infobot | [~answers] Asterisk Book: http://astbook.asteriskdocs.org & SIP w/ NAT: http://www.aocomputing.net/?p=3 & Wiki: http://www.voip-info.org & Docs: /path/to/src/asterisk/doc & 1.4 .vs. 1.6.0 .vs. 1.6.1: /path/to/src/asterisk/UPGRADE*.txt |
| 19:17.16 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 19:17.16 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 19:17.48 | bklang | gamedna: would you mind commenting on that ticket that you can reproduce the issue? |
| 19:21.03 | ecrane | any ideas on why asterisk thinks a sip peer is dynamic even though I specify the host's IP in sip.cfg? http://pastebin.org/473088 |
| 19:21.41 | [TK]D-Fender | ecrane: You are backwards |
| 19:21.56 | [TK]D-Fender | ecrane: I NEEDS to be dynamic in order to be ALLOWED to register |
| 19:21.59 | [TK]D-Fender | It* |
| 19:22.20 | ecrane | doh... thanks. |
| 19:22.28 | *** join/#asterisk megalomano (~samus@38.124.169.126) |
| 19:25.19 | gamedna | k |
| 19:25.23 | gamedna | bklang: will do |
| 19:27.19 | bklang | gamedna: THanks. Thanks also for your help in troubleshooting/verifying the issue |
| 19:27.49 | gamedna | bklang: you are welcome... Glad i could help out. I find that diving into problems are the best way to learn |
| 19:28.18 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 19:28.36 | *** join/#asterisk uqlev (~yuriy@91.184.221.31) |
| 19:43.25 | gamedna | bklang: done |
| 19:43.30 | bklang | Thanks again |
| 19:43.42 | *** join/#asterisk shido6 (~shido6@nat/yahoo/x-qgzhakzcbbdjzxuz) |
| 19:43.47 | gamedna | took me a sec, b/c i had to finish my lunch |
| 19:43.47 | gamedna | ;) |
| 19:44.09 | gamedna | take a look at it and let me know if you want me to add anything else? |
| 19:44.41 | bklang | I think between our two comments it describes the issue. I'm reading through chan_sip.c and rtp_engine.c right now... |
| 19:44.43 | gamedna | i would have uploaded the capture file, but i had already closed out of wireshark |
| 19:52.52 | gamedna | bklang: any insight? |
| 19:53.12 | bklang | I've found the function that creates the AMI event. Now tracing that backward ... |
| 19:56.47 | bklang | seems to only emanate from either ast_write() or ast_read() |
| 20:04.49 | gamedna | bklang: im not really familiar with AMi, can you Dial out using AMI? |
| 20:05.26 | bklang | Kinda. You can "originate" a call, which opens a channel and directs it either to another channel (such as a SIP peer) or into the dialplan (context,exten,prio) |
| 20:06.33 | gamedna | whats the command for that? |
| 20:06.41 | gamedna | i only see Bridge, but not Dial or something similar |
| 20:06.48 | gamedna | oh, nevermidn |
| 20:06.52 | gamedna | i see originate |
| 20:07.08 | bklang | Originate :) |
| 20:07.28 | bklang | it's not Dial like you'd think of with AGI because there is no corresponding channel with AMI |
| 20:07.32 | gamedna | wish they sorted the command list |
| 20:07.35 | bklang | it's a unilateral action |
| 20:07.49 | gamedna | right |
| 20:08.00 | gamedna | so what happens when you originate? |
| 20:08.07 | gamedna | do you have to bridge that call to another channel? |
| 20:08.24 | bklang | no, you have to specify a destination |
| 20:08.29 | bklang | the destination can be another channel or it can be a dialplan location |
| 20:08.43 | bklang | http://www.voip-info.org/wiki/view/Asterisk+Manager+API+Action+Originate |
| 20:08.56 | gamedna | i get that, but in order for the call to be meaningful, you need some type of person on the other side... |
| 20:09.03 | bklang | person or dialplan, yeah |
| 20:10.11 | gamedna | ah, so you set Channel as the destination |
| 20:10.12 | *** join/#asterisk bn-7bc (bjarne@2001:470:dc32:0:6233:4bff:fe0e:bc1) |
| 20:10.18 | gamedna | and Exten to the other side |
| 20:10.52 | [TK]D-Fender | Channel is who you cll. They get dumped into the DIALPLAn or into a specific app directly |
| 20:11.04 | [TK]D-Fender | That is all. Never another device directly |
| 20:11.07 | gamedna | i see that now |
| 20:11.14 | bklang | yeah, I misspoke above. You can either drop them into the dialplan or execute a specific app |
| 20:11.21 | bklang | if you want to directly bridge two channels you can use the Dial app |
| 20:11.24 | gamedna | Sequence of events: first the Channel is rung. Then, when that answers, the Extension is dialled within the Context to initiate the other end of the call. Note that the Timeout only applies to the initial connection to the Channel; any timeout for the other end can be specified, for instance in a Dial command in the definition of the Context. |
| 20:11.40 | bklang | Application: Dial \n Data: SIP/otherguy,30 |
| 20:12.24 | gamedna | this is pretty powerful |
| 20:12.35 | bklang | Yep |
| 20:12.55 | gamedna | so, here is another thing, using ami i could dial 2 external numbers and then bridge them together |
| 20:13.19 | bklang | Yes, though you'd need to specify a holding place after they connect but before they bridge |
| 20:13.24 | bklang | maybe a parking lot, for example |
| 20:13.25 | gamedna | make a call, play an announcement, then make another call, play an announcement.. Then bridge them toegehter |
| 20:13.31 | bklang | yep |
| 20:13.35 | bklang | or a conference room would work |
| 20:13.41 | gamedna | true... |
| 20:13.47 | bklang | If you get into programming with AMI and AGI, I might recommend you check out the Adhearsion project. It provides a nice API on top of AGI/AMI |
| 20:13.56 | [TK]D-Fender | [16:11]<bklang>if you want to directly bridge two channels you can use the Dial app <- this does not directly bridge 2 channels. This spawns a new call to bridge to. Yours implied existing |
| 20:13.58 | gamedna | i have actually been reading up on that |
| 20:21.04 | *** join/#asterisk mac-mini- (~mac-mini@unaffiliated/macmini/x-648924) |
| 20:23.06 | *** join/#asterisk RobH (~robh@wikimedia/RobH) |
| 20:23.28 | *** part/#asterisk RobH (~robh@wikimedia/RobH) |
| 20:25.04 | *** join/#asterisk JerJer (~PhatJ@asterisk/original-h323-guy/JerJer) |
| 20:25.29 | JerJer | Could someone here send me a fax? Trying T.38 on 1.6 |
| 20:26.16 | anonymouz666 | haha |
| 20:26.18 | anonymouz666 | this is JerJer |
| 20:26.34 | JerJer | allegedly |
| 20:26.35 | anonymouz666 | I would send you an e-mail with some file attached |
| 20:26.39 | anonymouz666 | it is easier and WILL work |
| 20:26.59 | anonymouz666 | :P |
| 20:27.00 | JerJer | lawyers deal with official documents so fax it is |
| 20:27.31 | JerJer | i don't have a POTS line and testing T.38 inbound and outbound at the same time is kinda not 'testing' :) |
| 20:27.55 | Qwell | HP has a fax callback thingie |
| 20:28.09 | JerJer | yeah, it never calls me back |
| 20:28.19 | nix8n82 | there are several online fax services, A couple of them offer a trial and I think there may be one that allows you to send one or two a day free. either way you will have to google it |
| 20:34.54 | *** part/#asterisk ManxPower (~manxpower@216.186.151.147) |
| 20:35.03 | JerJer | nix8n82: thanks |
| 20:35.40 | boodu | hello |
| 20:35.54 | JerJer | boodu: howdie |
| 20:36.16 | nix8n82 | JerJer, your welcome, you did the work |
| 20:36.44 | JerJer | heh so-called 'work' :) |
| 20:36.57 | *** join/#asterisk hardwire (~spencersr@69-161-26-211.static.acsalaska.net) |
| 20:37.05 | exothermc | When doing SIP to TDM, are there some settings I can change to lower the load? |
| 20:37.28 | exothermc | I'm running on an older box, but even with 20 calls I'm spiking the CPU. |
| 20:37.47 | exothermc | I'm just wondering if there are some RTP intensive modules that can be unloaded etc. |
| 20:38.37 | JerJer | exothermc: are you transcoding ? |
| 20:38.49 | exothermc | JerJer: Nope ulaw only. |
| 20:39.13 | *** join/#asterisk DogBoy (~john@unaffiliated/dogboy) |
| 20:40.10 | moy | exothermc: how big the "spike" and is it really just a spike or the load stays for the duration of the calls? |
| 20:40.34 | exothermc | pretty much always high load, but you can see it max the CPU. spike is the wrong word. |
| 20:40.46 | JerJer | load as in cpu usage or 'load average' ? |
| 20:41.50 | exothermc | JerJer: single CPU box with a load average of 4 sustained and CPU hits 99.9 a lot, and stays above 85. |
| 20:42.31 | exothermc | The majority of the CPU time is spent on hardware interupts. |
| 20:43.23 | JerJer | hmmm something isn't right with that then |
| 20:43.26 | moy | exothermc: how are you checking that? .... is there high cpu usage only when there is calls? ... there is some thread spinning most likely .... use top or ps to find the offending thread, then pstack to have a hint of which module launched that thread |
| 20:43.42 | JerJer | i've done way more than 20 calls SIP -> TDM without transcoding on a single box |
| 20:43.47 | *** join/#asterisk andresmujica (~andresmuj@ubuntu/member/andresmujica) |
| 20:43.53 | JerJer | exothermc: what CPU? |
| 20:44.05 | JerJer | which* |
| 20:44.27 | exothermc | JerJer: Intel(R) Celeron(R) CPU 2.40GHz |
| 20:44.41 | JerJer | kinda lower end, but should do way more than 20 |
| 20:44.46 | exothermc | JerJer: cpu family 15 model 4 |
| 20:45.16 | exothermc | moy: ok it is asterisk that is spinning up btw, let me play withy pstack |
| 20:45.16 | *** join/#asterisk bse-r (~bse-r@190.242.6.138) |
| 20:45.18 | ecrane | <--------- noob user of ekiga 2.0 + asterisk = fail |
| 20:45.35 | bse-r | i created a dial plan called [ext-local-custom] to monitor zap trunk with BLF exten => line1,hint,Zap/2 exten => line,1,Dial(Zap/2), the problem is, if the person is using the aastra 9143i phone and press the "blf programmed key" the person open the zap trunk and asterisk is ignoring PIN SETS |
| 20:45.46 | bse-r | so the person can dial ignoring PIN because it is opening the ZAP trunk directly |
| 20:46.09 | gamedna | bklang: just went though adhearsion's docs. Wonder if it could be used to manage a cluster of Aseterisk boxes |
| 20:46.22 | gamedna | interesting concept actually |
| 20:46.28 | bklang | gamedna: It definitely could, with the caveat being that AMI is specific to one Asterisk host |
| 20:46.41 | bklang | but it defintely could process calls on AGI from multiple Asterisk hosts |
| 20:46.43 | moy | exothermc: yes asterisk, but asterisk spawns many threads, you need to find which one is being a cpu hog, top -H -p `pidof asterisk`, shows per thread info |
| 20:47.07 | JerJer | moy: that is a very nice man |
| 20:47.11 | JerJer | er command |
| 20:47.11 | exothermc | moy ok doing that now. |
| 20:47.45 | JerJer | moy: then strace the pid that is cpu bound? |
| 20:47.49 | gamedna | can adhearsion initiate the AMI connection? |
| 20:47.57 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 20:48.05 | gamedna | all the examples show them connecting when someone calls |
| 20:48.44 | moy | JerJer: yes strace, or "pstack `pidof asterisk` > file" and find the stack corresponding to the PID/TID of the offending thread, which typically is some loop |
| 20:48.51 | JerJer | gamedna: you mean like a persistent AMI connection? |
| 20:49.00 | exothermc | moy: Looks like it is spread pretty evenly across 15 or so threads. |
| 20:49.01 | gamedna | gamedna: yea... |
| 20:49.02 | JerJer | moy: very good to know |
| 20:49.05 | JerJer | takes notes |
| 20:49.13 | gamedna | JerJer: yea |
| 20:49.14 | exothermc | moy: No one thread really taking the cake here. |
| 20:49.33 | moy | exothermc: pastebin the output |
| 20:50.06 | exothermc | moy http://pastebin.com/ramn2MN6 |
| 20:50.29 | moy | exothermc: so now do |
| 20:50.41 | moy | pstack `pidof asterisk` > /tmp/asterisk-stack |
| 20:50.52 | moy | and pastebin the file |
| 20:51.17 | gamedna | JerJer: bklang: i know that thinks like FOP and such maintain connections.. Just wondering if Adhearsion can do it |
| 20:51.19 | bklang | gamedna: yes, it can initiate AMI connections, but only to one Asterisk host at a time |
| 20:51.42 | moy | several threads above 6% is not very good sign |
| 20:51.45 | bklang | part of the power in Adhearsion comes from being able to use AGI and AMI in one programming framework |
| 20:51.56 | bklang | but it works just fine using either by itself |
| 20:52.16 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 20:52.49 | *** join/#asterisk [Outcast] (~anonymous@64.202.62.5) |
| 20:53.09 | exothermc | moy: ok waiting to install pstack (yum takes a very long to time to do anything at this load) |
| 20:53.18 | bklang | Adhearsion runs as a daemon so it maintains the AMI connection (for things like events) as long as the process is running |
| 20:53.46 | gamedna | it keeps one thread for each connection right? |
| 20:54.25 | gamedna | so if that daemon gets 5 calls, each call is handled in a seperate thread |
| 20:55.51 | moy | exothermc: ok ... although as you said, the biggest chunk seems to come from hardware interrupts ... is that load kept even after hanging up all calls? |
| 20:56.32 | moy | which hardware are you using? |
| 20:57.06 | exothermc | moy: sangoma A104d |
| 20:57.28 | exothermc | moy: I'll pull this leg out of route and let the calls die down, but I don't want to just drop them all. |
| 20:57.36 | moy | driver version and signaling being used? PRI? |
| 20:57.54 | exothermc | PRI and the latest driver |
| 20:58.14 | *** join/#asterisk shido6_ (~shido6@nat/yahoo/x-rhwqwutzkwbfmakq) |
| 20:58.17 | exothermc | actually I'm using 3.5.11 |
| 20:58.29 | moy | so not latest :) |
| 20:58.32 | *** part/#asterisk bse-r (~bse-r@190.242.6.138) |
| 20:59.00 | exothermc | moy: ya so many different installs a little hard to keep track off the top of the head, had to look. |
| 20:59.26 | moy | exothermc: pastebin also your wanpipe1 -> wanpipe4 configurations ... and wanrouter hwprobe verbose output |
| 21:00.20 | bklang | gamedna: one thread for each AGI connection, yes. |
| 21:00.39 | bklang | It has two global threads for AMI, shared among all calls. One thread for events and one thread for actions |
| 21:00.53 | bklang | If you're interested there's an active channel at #adhearsion. I don't want to add too much noise here |
| 21:03.06 | *** join/#asterisk QubeZ (~nkasu@64.128.254.34) |
| 21:03.08 | QubeZ | hello all |
| 21:03.20 | *** join/#asterisk TimeRider (~steve@109.224.131.68) |
| 21:03.55 | exothermc | moy: http://pastebin.com/baRxy8zF http://pastebin.com/fT7TFxt9 http://pastebin.com/WRYu0Y3a |
| 21:04.09 | QubeZ | can anyone help with a chanspy issue? I'm trying to figure out how to validate its an extension in the spygroup: http://pastebin.com/XVMNXqYC |
| 21:04.18 | *** join/#asterisk devmod (~devmod@c-76-100-208-204.hsd1.md.comcast.net) |
| 21:04.56 | paulc | Anyone here had any experience with PortaOne (PortaSIP, PortaBilling, etc)? |
| 21:05.04 | exothermc | moy: Ok got gdb installed, running pstack now |
| 21:05.05 | moy | exothermc: is that the output of wanrouter hwprobe verbose? I thought verbose gave a little more output |
| 21:05.42 | exothermc | moy: http://pastebin.com/sexEjKna |
| 21:06.16 | exothermc | moy: and here is the stack: http://pastebin.com/BnBSL3FD |
| 21:07.20 | [TK]D-Fender | QubeZ: That is a broke up sample, please post it complete |
| 21:07.28 | *** join/#asterisk jeffgus (~jeffgus@2002:ad33:b504::1) |
| 21:07.41 | [TK]D-Fender | QubeZ: ${vSPYGROUP} <-- and nowhere is this being set that we can see. |
| 21:08.33 | moy | exothermc: you need to take both the top output and the pstack close enough in time, the top you pasted mentions threads that no longer exist in the pstack (because probably some calls were hangup or whatever) |
| 21:08.52 | exothermc | moy: ok jas |
| 21:08.56 | moy | I also notice you have quite old hardware, isn't it? did you get that card on ebay or something? |
| 21:09.34 | [Outcast] | I all ask again so does any one know how to change the via link ? |
| 21:09.53 | exothermc | moy: ya |
| 21:10.33 | exothermc | moy: Ok so now the call load is lowered down to 5 calls and no threads are over 3%. Do I have to abuse the users to get proper debug? |
| 21:12.42 | [TK]D-Fender | exothermc: No, but I'm sure its a lot of fun :) |
| 21:15.25 | rustyclarkson | Can anyone tell me if asterisk uses an external player to playback MOH, when mode=files? |
| 21:16.30 | exothermc | ok ramping up calls again. |
| 21:16.34 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 21:17.33 | ChannelZ | rustyclarkson: no.. files are files |
| 21:17.44 | exothermc | moy: You want the threads over what before I dump the pstack? |
| 21:18.01 | exothermc | moy: I have some at 3.6 now. |
| 21:18.28 | moy | exothermc: yes, run top, copy the snapshot showing the load, then immediately afterwards run pstack |
| 21:18.57 | exothermc | even though my load average is still 0.00? |
| 21:19.19 | rustyclarkson | ChannelZ: thanks for your feedback |
| 21:19.48 | exothermc | it is like it will take some time to build it up higher. |
| 21:19.56 | exothermc | oh there we go. |
| 21:20.09 | *** join/#asterisk TimeRider (~steve@109.224.131.68) |
| 21:21.08 | *** join/#asterisk nix8n82 (~nate@63.162.27.14) |
| 21:24.55 | exothermc | moy: http://pastebin.com/by5pxwuu http://pastebin.com/6hVp0Qyn |
| 21:25.13 | *** join/#asterisk cdahmedeh (~cdahmedeh@CPE001cdfab341a-CM001225409602.cpe.net.cable.rogers.com) |
| 21:26.53 | moy | exothermc: so it seems the channel bridges are the ones ... but seems like a driver/hardware problem, see my pm |
| 21:26.56 | JerJer | very odd: res_fax.c:1041 receivefax_t38_init: error reading frame while generating CED tone (v1.2.0) |
| 21:27.15 | exothermc | moy: Ya emailing Marc now. |
| 21:27.19 | nix8n82 | bougyman, you have spa3102 that in theory work? |
| 21:27.52 | nova911 | Landline call's over opevpn from remote office is getting dropped after 20 seconds well as normal extension call's working fine |
| 21:27.56 | exothermc | moy: So you guys don't want to put NFAS in the libpri stack of freeTDM ? :) |
| 21:29.40 | moy | not that we don't want ... we're busy as hell, but patches are welcomed :) |
| 21:30.02 | exothermc | moy: lol ya I hear you, I couldn't code my way out of a wet paper bag though. |
| 21:32.19 | bougyman | nix8n82: if you don't need caller id inbound from them, they work fine. |
| 21:32.29 | *** join/#asterisk riddlebox (~james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 21:33.06 | bougyman | if you do, you need something like http://gitorious.org/spa3102-invite-packet-scrubber |
| 21:33.27 | bougyman | other than that I had no severe issues, just a lot of tweaking of gains to get the echo to go away |
| 21:33.37 | bougyman | their echo cancel is awful, support will tell you to turn it off. |
| 21:34.06 | bougyman | i stopped using them because support told me they didn't care about fixing the inbound callerid issue, they recommended I turned off inbound caller id. |
| 21:34.17 | bougyman | that wasn't very acceptable for my clients. |
| 21:34.41 | bougyman | i made that packet scrubber just to get me by til I could get FXO cards to replace them. |
| 21:34.43 | nix8n82 | yeah I want caller id info too |
| 21:34.55 | *** join/#asterisk lost_soul (~noymfb@cpe-67-249-141-227.twcny.res.rr.com) |
| 21:34.56 | bougyman | anything with commas in it will break most sip stacks. |
| 21:35.06 | bougyman | never tried against asterisk, it might or might not care. |
| 21:35.24 | nova911 | anyone for helpâ¦. |
| 21:35.25 | bougyman | against kamalio, two commercial devices, and freeswitch, the packet was rejected for a malformed sip header. |
| 21:36.03 | bougyman | i think i tried yate, too, and it was broken. |
| 21:39.01 | nix8n82 | how much do you want for one? |
| 21:41.33 | *** join/#asterisk sysreq (~sysreq@unaffiliated/sysreq) |
| 21:41.52 | *** join/#asterisk voxter (~hardcore@macpro.daytonhome.voxter.net) |
| 21:43.25 | voxter | i realize that 1.2 is way the hell out of focus now, but im curious if any of you have experienced two phones that connect from different locations, one receives early media via SIP and one simply does not? I'm debugging a weird case where some users dont get early "ringing" media, and registering my own phone to it (same phone firmware and all) works fine |
| 21:43.43 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 21:43.43 | voxter | tcpdump shows that the asterisk box in question forwards early media rtp to me, but never tries to send it to the other user. |
| 21:43.54 | *** join/#asterisk Woody2143 (~Woody2143@machine76.Level3.com) |
| 21:47.01 | QubeZ | [TK]D-Fender: the spygroup is being set in the extensions.incl file like standard: exten => 880,n,Set(__vSPYGROUP='fl-tc01') in the extensions.conf file |
| 21:47.13 | *** join/#asterisk mweichert_ (~mweichert@216.16.254.34) |
| 21:47.32 | mweichert_ | Hello |
| 21:48.08 | [TK]D-Fender | QubeZ: Stop putting quotes around variables <- |
| 21:48.20 | [TK]D-Fender | QubeZ: this isn't a real programming language |
| 21:48.23 | QubeZ | [TK]D-Fender: ok |
| 21:48.29 | mweichert_ | When I use follow me to forward calls to a cell phone, once the call has been established, is the call reliant at all yet on the PBX? If the PBX goes down, will it affect the call on the cell phone? |
| 21:48.32 | [TK]D-Fender | queAndd your sample was diced and incomplete |
| 21:48.54 | [TK]D-Fender | mweichert_: What is "forward"? |
| 21:48.56 | QubeZ | [TK]D-Fender: i left it incomplete because im not sure how to complete it |
| 21:49.15 | [TK]D-Fender | QubeZ: Try actually doing something and highlight the point of failure |
| 21:49.31 | mweichert_ | [TK]D-Fender, follow me |
| 21:51.26 | *** join/#asterisk puzzled (~foobar@puzzled.xs4all.nl) |
| 21:51.33 | [TK]D-Fender | mweichert_:Since when is the call NOT bridged by *? |
| 22:05.57 | *** join/#asterisk philipp64|laptop (~chatzilla@75-92-150-245.war.clearwire-wmx.net) |
| 22:05.57 | *** join/#asterisk boodu (~antoine@host-115-126-167-240.adsl.nautile.nc) |
| 22:05.57 | *** join/#asterisk rdahlin_1 (~rdahlin_1@78-73-19-238-no168.tbcn.telia.com) |
| 22:10.33 | *** join/#asterisk boodu (~antoine@host-115-126-167-240.adsl.nautile.nc) |
| 22:11.41 | *** join/#asterisk philipp64|laptop (~chatzilla@75-92-150-245.war.clearwire-wmx.net) |
| 22:11.41 | *** join/#asterisk rdahlin_1 (~rdahlin_1@78-73-19-238-no168.tbcn.telia.com) |
| 22:20.49 | *** join/#asterisk Beltechs (~Beltechs@208.127.3.20) |
| 22:22.33 | Beltechs | Hello Im using asterisk 1.6 one of my extensions is all of sudden unregistering its on the same lan and this just started today |
| 22:22.52 | Beltechs | any ideas what might be causing this extension to fall off? |
| 22:23.07 | leifmadsen | Beltechs: not enough information provided |
| 22:24.09 | troy42 | if you can only provide that, paypal leif $150 to fix it :-) |
| 22:24.13 | leifmadsen | someone refresh my memory what the variables are for either setting or getting the filename of what was recorded with Monitor()? tex/channelvariables.tex is failing me |
| 22:24.38 | leifmadsen | I'm busy working on automated testing scripts... |
| 22:24.42 | Beltechs | the extension sits on the same lan as the PBX, when I reboot the phone it registers and then after a half hour it show greyed out in fop |
| 22:24.54 | Beltechs | but maybe you can help me and I can donate |
| 22:25.01 | troy42 | Record uses RECORDED_FILE, i think it's MIXMONITOR_FILE for mixmonitor |
| 22:25.07 | troy42 | er _FILENAME |
| 22:25.27 | troy42 | google says yup |
| 22:25.54 | troy42 | actually, i've only used mixmonitor, not monitor |
| 22:25.56 | Beltechs | Im using polycom sp550 |
| 22:26.15 | leifmadsen | troy42: I'm using Monitor() as mentioned |
| 22:26.21 | Beltechs | there are 47 other polycoms on the same lan and only 1 is doing this |
| 22:26.34 | leifmadsen | Beltechs: perhaps it needs to be RMA'd? |
| 22:26.45 | leifmadsen | Beltechs: perhaps the port is bad? perhaps the cable is bad? perhaps the port on the switch is bad? |
| 22:27.05 | leifmadsen | Beltechs: always check layer 1 (Physical layer) first and verify there is no problem with it |
| 22:27.09 | leifmadsen | then move up the layers |
| 22:27.16 | Beltechs | got ya |
| 22:27.21 | leifmadsen | perhaps it has nothing to do with layer 7 |
| 22:27.26 | troy42 | i dunno, if 47 work and 1 don't, i'd blame something that's common to all 48 |
| 22:27.27 | troy42 | ducks |
| 22:28.22 | *** join/#asterisk devmod (~devmod@c-76-100-208-204.hsd1.md.comcast.net) |
| 22:29.11 | Beltechs | troy42 i dont know what that means but the phone files were automatically generated through an endpoint manager |
| 22:29.23 | Beltechs | if it makes a difference |
| 22:29.40 | troy42 | that's just me chuckling at problem diagnosis |
| 22:29.41 | leifmadsen | probably doesn't |
| 22:29.50 | leifmadsen | Beltechs: he was being sarcastic |
| 22:29.58 | *** join/#asterisk tzafrir (~tzafrir@bzq-218-155-147.cablep.bezeqint.net) |
| 22:29.59 | leifmadsen | go check your connections |
| 22:30.14 | leifmadsen | try moving the phone to a different port on the network and see if it still does it |
| 22:30.33 | leifmadsen | that's a good test to see if it's the phone or the port -- try another phone on the same port as the dead phone and see if it does it too |
| 22:30.48 | leifmadsen | basically I'm saying there is probably little chance it's actually asterisk |
| 22:30.54 | leifmadsen | and if it is -- you need to provide a sip trace showing what is going on |
| 22:35.58 | Beltechs | got ya, I will have to do that onsite tomorrow. |
| 22:36.19 | *** join/#asterisk philipp64|laptop (~chatzilla@75-92-150-245.war.clearwire-wmx.net) |
| 22:37.17 | Beltechs | I also have 3 extension at a remote site and 1 or another is always unregistering. |
| 22:37.46 | Beltechs | leif should I use like sip debug ? |
| 22:37.57 | leifmadsen | Beltechs: you should like yes |
| 22:38.09 | Beltechs | lol |
| 22:38.28 | Beltechs | cool let me find the correct command Its different on 1.6 |
| 22:39.12 | Beltechs | sip set debug on ?? |
| 22:39.35 | Beltechs | how do I reference the extension? |
| 22:39.46 | Beltechs | sip set debug on XXXX ?? |
| 22:41.38 | bougyman | Beltechs: help sip set |
| 22:41.45 | bougyman | you can do it by ip or peer |
| 22:41.58 | Beltechs | like my extension right? |
| 22:42.11 | Beltechs | sip set debug on :6000?? |
| 22:42.11 | bougyman | i dunno, i always use IP. |
| 22:42.31 | Beltechs | got ya let me see if I can get the ip |
| 22:47.26 | *** join/#asterisk boodu (~antoine@host-115-126-167-240.adsl.nautile.nc) |
| 22:54.07 | *** join/#asterisk citrus2 (~citrus2@72.215.183.28) |
| 22:54.51 | citrus2 | whats a cheap and easy card for asterisk that will support isdn/t1? |
| 23:02.04 | [TK]D-Fender | ~cheap |
| 23:02.05 | infobot | i guess cheap is a bad idea. If you're setting up an Asterisk system, one of the wisest pieces of advice you can take is this: DON'T BE A CHEAPSKATE. |
| 23:02.32 | [TK]D-Fender | citrus2: a single T1? Any expectation for expanded needs? |
| 23:03.23 | troy42 | roll 256MB VM with 10 hertz timing, save coin |
| 23:04.07 | Micc_ | Are there any plans for asterisk to support 100rel PRACK? Or is that not something that is useful for asterisk to support? |
| 23:09.09 | russellb | Micc_: no immediate plans |
| 23:09.17 | russellb | it would be _VERY_ hard to implement |
| 23:09.32 | russellb | and you can likely work around it if you need it by putting a proxy in front of Asterisk that does support it |
| 23:11.55 | *** join/#asterisk pabelanger-lap (~pabelange@microsolve5.ontera.net) |
| 23:12.39 | jamko | Anyone ever experience a sip.conf file corrupting, to where a UA will get "wrong password" errors, eventhough the password is correct? Then in order to fix the issue, you have to add or delete some other lines elsewhere in the sip.conf file? |
| 23:13.10 | jamko | I fixed it once before by retyping the entire file, but it was a nightmare. Copy and paste would not work. |
| 23:14.42 | Micc_ | Does OpenSER and SIPS support 100rel/prack? |
| 23:16.27 | *** join/#asterisk andresmujica (~andresmuj@ubuntu/member/andresmujica) |
| 23:18.38 | bougyman | Micc_: if you tell it to |
| 23:18.49 | Micc_ | right right. |
| 23:18.57 | bougyman | openser (kamalio) is like a closed firewall. |
| 23:19.05 | bougyman | it will only do what you tell it to, explicitly, nothing more. |
| 23:19.10 | *** join/#asterisk cdahmedeh (~cdahmedeh@CPE001cdfab341a-CM001225409602.cpe.net.cable.rogers.com) |
| 23:20.27 | bougyman | the cluecon talk suggested the codebase is shared again. |
| 23:20.41 | bougyman | with openser/kamalio, that is. |
| 23:26.41 | *** join/#asterisk bkruse (~bkruse@75.76.105.124) |
| 23:26.41 | *** mode/#asterisk [+o bkruse] by ChanServ |
| 23:29.55 | Micc_ | so which one is better sips or openser or ser? For simple load balancing to asterisk servers. |
| 23:30.02 | *** join/#asterisk devmod (~devmod@c-76-100-208-204.hsd1.md.comcast.net) |
| 23:30.15 | Micc_ | I'm not convinced its even needed when most devices support a backup registrar. |
| 23:37.39 | Beltechs | sip set debug on 192.168.0.XXX:5060 6000 is this command correct? |
| 23:38.48 | bougyman | Micc_: kamalio seemed to get the most buzz, but all 3 seem adequate, in my experience. |
| 23:39.05 | bougyman | heck kamalio and ser are just diff frontends to the same code these days. |
| 23:39.37 | bougyman | Micc_: load balancing != failover, which a backup registrar can indeed handle. |
| 23:41.02 | bougyman | sip set debug 192.168.132.20:5060 |
| 23:41.43 | *** join/#asterisk tzafrir_laptop (~tzafrir@212.179.75.202) |
| 23:42.13 | bougyman | if you do sip set debug peer 3122 it would just look up the ip that peer is registered to and use it. |
| 23:42.21 | bougyman | so that's a shortcut if it shows up in sip show peers |
| 23:42.40 | Beltechs | no such command |
| 23:42.57 | bougyman | hrm. |
| 23:43.01 | bougyman | sorry, i'm on a very old box here. |
| 23:43.07 | Beltechs | ahh |
| 23:43.14 | bougyman | maybe someone on a newer version can show the equivalent. |
| 23:43.34 | bougyman | red*CLI> sip set debug peer 3122 |
| 23:43.35 | bougyman | SIP Debugging Enabled for IP: 192.168.137.254:2048 |
| 23:43.38 | bougyman | that's how I do it here. |
| 23:43.57 | Beltechs | let me try |
| 23:44.47 | Beltechs | ok that gave me something like what you posted |
| 23:46.58 | pabelanger-lap | Anybody recently played with GNUdialer? |
| 23:47.59 | bougyman | only if recently means october of 09 |
| 23:48.22 | bougyman | after using a horrible commercial dialer and testing every open source one I could find, i decided to write one |
| 23:48.37 | bougyman | vici has actually worked for us, though it's almost amazing that it does. |
| 23:49.43 | *** join/#asterisk ruben23 (~ITadmin@125.212.40.2) |
| 23:50.05 | ruben23 | hi is voice traffic or voip consider as broadcast..? |
| 23:50.58 | bougyman | not unless you're using a broadcast voip protocol. |
| 23:51.04 | bougyman | so almost all of the time, No. |
| 23:51.29 | bougyman | do you mean broadcast from a networking standpoint? |
| 23:52.09 | ruben23 | bougyman: yes on a networking standpoint layer 2 broadcast |
| 23:52.37 | bougyman | then it's generally unicast, not multicast/broadcast. |
| 23:53.21 | ruben23 | bougyman:ok thanks. |
| 23:59.08 | Beltechs | hello I have poste my peer debug http://pastebin.com/nnqrYLXQ The phone is able to make outbound calls. During this debug the phone just rang in my ear but never rang at the extension. |
