| 00:00.08 | brc_ | using # in the dialplan doesn't work, it says invalid extension on the console |
| 00:00.11 | [TK]D-Fender | Loggin off in a little more complicated.... I think you can macro it into working faster. |
| 00:00.21 | brc_ | how's that |
| 00:00.22 | [TK]D-Fender | or call a CLI feature for that. |
| 00:00.29 | [TK]D-Fender | Saw it somewhere |
| 00:00.34 | brc_ | k |
| 00:00.37 | brc_ | any idea where?.. |
| 00:02.41 | [TK]D-Fender | looking now. There is "agent logoff" in the cli that I'm reading up on. |
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| 00:03.24 | marcus2 | hrm |
| 00:03.58 | [TK]D-Fender | http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20AgentCallbackLogin |
| 00:04.04 | marcus2 | so i have meetme workin great with zap channels, but it sounds like ass with sip channels.... very stuttery/gravely, unusable call quality |
| 00:04.25 | [TK]D-Fender | check out the "logging off the queue manually" |
| 00:04.36 | marcus2 | is there something obvious that i should be looking at? the sip clients are all on the same ethernet, and calls made with the sip clients are otherwise excellent |
| 00:04.45 | marcus2 | (ie. between sip clients, or sip->zap, or zap->sip) |
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| 00:05.59 | Juxt | i am trying to originate calls via .call file |
| 00:06.11 | Juxt | and it works great except i can't read the DIALSTATUS |
| 00:06.13 | IronHelix | dont telemarket, thanks |
| 00:06.18 | IronHelix | :) |
| 00:06.24 | marcus2 | heh |
| 00:06.26 | Juxt | ha i am not a telemarketer |
| 00:06.30 | h3x | shit |
| 00:06.42 | h3x | var/spool/asterisk/outgoing tisnt good enough for telemarketing |
| 00:06.44 | *** part/#asterisk Sedorox (i=brandon@smartserv/cna/Sedorox) |
| 00:06.49 | Juxt | exactly |
| 00:06.55 | h3x | haha |
| 00:06.59 | h3x | its for ... wake up calls! |
| 00:07.21 | Juxt | well something similar |
| 00:07.23 | Dr_Ray | wardialing |
| 00:07.30 | h3x | yes! |
| 00:07.42 | Juxt | well the point is, i can't read the damn dialstatus |
| 00:08.03 | Juxt | so i assume it sends the call to the context only if it's been answered |
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| 00:12.14 | many | moc: what does the LOCAL_USER stuff do? backtracking how users inside a module having active channels? how many users are registered? |
| 00:14.17 | JunK-Y | many: this is all explain in the macro definition, no? |
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| 00:35.47 | Juxt | is there an easy way to determine the status of the call via the manager api? |
| 00:38.27 | tzafrir_laptop | Juxt, how do you know what call? |
| 00:38.30 | kippi | is there a easy way see check what version of asterisk u are runing? |
| 00:38.44 | Sedorox | kippi: show version |
| 00:38.47 | Sedorox | on the console |
| 00:38.47 | tzafrir_laptop | do you assume that there is only one call? e.g: through a zap channel? |
| 00:38.53 | Juxt | tzafrir_laptop: i use originate |
| 00:38.59 | Juxt | command |
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| 00:39.03 | Juxt | and than track by uniqueid |
| 00:39.04 | theblue | Hi all. |
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| 00:48.26 | TheCops | I've got this warning when I compile 1.2.0: include/asterisk/utils.h:173: warning: `always_inline' attribute directive ignored, someone know why ? |
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| 00:52.05 | CRCC | hi |
| 00:52.36 | CRCC | has anybody had problem with outbound calls with capi |
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| 00:54.20 | CRCC | @denon |
| 00:54.26 | CRCC | !help |
| 00:54.41 | CRCC | newbee help |
| 00:54.43 | file | O.o |
| 00:56.22 | file | CRCC: don't do that. |
| 00:56.53 | CRCC | sorry can you tell me how to get info here |
| 00:56.58 | CRCC | i am new to this |
| 00:57.43 | ginvent | Anyone around to help me debug my fax issue? |
| 01:00.28 | Nugget | plug the fax into a landline. problem solved. |
| 01:02.19 | ginvent | Thanks. |
| 01:02.29 | ginvent | I am trying to use SpanDSP with asterisk |
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| 01:04.41 | ginvent | ugh |
| 01:04.41 | IronHelix | do this- put a piece of tape on your fax machine that says SPANDSP on it |
| 01:04.41 | IronHelix | :) |
| 01:04.41 | [TK]D-Fender | b00m |
| 01:05.28 | {zombie} | ginvent: what you trying to do? |
| 01:05.28 | ginvent | Ah... technical help... gotta love it. |
| 01:05.28 | {zombie} | yeah everyone's a comedian this morning |
| 01:05.28 | Nugget | fax + voip == endless hemmorhoids. |
| 01:05.28 | ginvent | Not voip. |
| 01:05.28 | h3x | foip! |
| 01:05.28 | ginvent | Asterisk--->spandsp---->X100P |
| 01:06.03 | {zombie} | receiving faxes, or sending? |
| 01:06.13 | ginvent | both... but receiving is easier to test. |
| 01:06.34 | ginvent | It dials, beeps, starts to connect... then poop... hangup zap/1 |
| 01:06.39 | {zombie} | I've not yet managed to get sending to work, but I can receive fine |
| 01:06.51 | ginvent | I don't get any debug messages either. |
| 01:06.56 | [TK]D-Fender | Anytime * gets its hands anywhere neaer my faxes they die horribly |
| 01:07.08 | {zombie} | first thing to check, are you losing any interrupts on your zaptel card? |
| 01:07.16 | *** join/#asterisk BleedingMe (n=Bleeding@ppp-71-137-216-107.dsl.scrm01.pacbell.net) |
| 01:07.20 | ginvent | How do I check that? |
| 01:07.26 | ginvent | I bet I am. |
| 01:07.51 | {zombie} | have you got zttool? |
| 01:07.53 | ginvent | I do. |
| 01:08.19 | {zombie} | that should tell you |
| 01:08.23 | ginvent | hmmm |
| 01:08.31 | ginvent | It says 0 |
| 01:08.32 | {zombie} | or check /proc/zaptel/* |
| 01:08.36 | ginvent | ? |
| 01:08.37 | *** join/#asterisk Neter66 (n=neter66@216.191.72.153) |
| 01:09.16 | {zombie} | also, to see txfax/rxfax debug messages you will need to run asterisk and attach to its console directly, rather than via -r |
| 01:09.28 | ginvent | How do I do that... |
| 01:09.40 | {zombie} | shutdown asterisk, then /usr/sbin/asterisk -vvvvvc |
| 01:09.45 | ginvent | K |
| 01:10.02 | {zombie} | are you running the latest spandsp/[rx]xfax? |
| 01:10.15 | ginvent | I am. |
| 01:10.17 | Neter66 | question that i was hoping someone here could answer... i have an iax2 channel going to my provider, .conf file has 'qualify=yes' in it, i can see the response times with iax2 show peers, all works well... except... |
| 01:10.19 | ginvent | I just started asterisk. |
| 01:10.25 | *** join/#asterisk Druken (n=druken@CPE00121716da99-CM000e5cde4ca2.cpe.net.cable.rogers.com) |
| 01:10.30 | Neter66 | if my upstream router reboots for any reason, i get "UNREACHABLE" |
| 01:10.34 | Neter66 | and it never goes away |
| 01:10.35 | ginvent | should I send a fax? |
| 01:10.42 | Neter66 | i have to reboot the box |
| 01:10.44 | Neter66 | any ideas? |
| 01:10.50 | {zombie} | ginvent: try it |
| 01:11.19 | {zombie} | you should see things like: |
| 01:11.19 | {zombie} | DCS with final frame tag |
| 01:11.19 | {zombie} | In state 9 |
| 01:11.37 | {zombie} | don't ask me how to actually interpret those messages, but at least it gives you something to work with :) |
| 01:11.38 | ginvent | Ok, I am waiting for a fax. |
| 01:12.15 | BleedingMe | Neter66: it will stay unreachable until you restart the asterisk box? |
| 01:12.37 | Druken | is it possible to use spandsp to receive the faxes, and then send them to a real fax machine ? |
| 01:12.38 | ginvent | I don't see that. |
| 01:12.40 | ginvent | I get this: |
| 01:13.04 | ginvent | -- Executing Macro("Zap/1-1", "faxreceive") in new stack |
| 01:13.04 | ginvent | <PROTECTED> |
| 01:13.07 | ginvent | Things like that. |
| 01:13.12 | ginvent | No debug... |
| 01:13.13 | Neter66 | BleedingMe: ya, it's wierd. I have tried to restart asterisk, restart the network, and only restarting the box solves the issue |
| 01:13.38 | {zombie} | Druken: you're better off just sending them to the real fax machine directly, and not using spandsp |
| 01:13.43 | ginvent | ah core debug. |
| 01:13.44 | ginvent | hang on. |
| 01:13.46 | *** join/#asterisk digime (n=digime@ip68-111-235-172.sd.sd.cox.net) |
| 01:13.47 | *** join/#asterisk shido6 (n=shido6@d221-68-216.commercial.cgocable.net) |
| 01:13.52 | {zombie} | ginvent: does it actually start rxfax? |
| 01:14.01 | Dr_Ray | Druken |
| 01:14.04 | ginvent | yup |
| 01:14.04 | *** join/#asterisk bjohnson (n=bjohnson@i216-58-60-57.cybersurf.com) |
| 01:14.11 | Druken | Dr_Ray: yes sir? |
| 01:14.18 | Dr_Ray | you were looking for me? |
| 01:14.21 | *** part/#asterisk digime (n=digime@ip68-111-235-172.sd.sd.cox.net) |
| 01:14.23 | Pikoro | morning everyone |
| 01:14.26 | BleedingMe | Neter66: and the connection to your provider is dead? |
| 01:14.46 | Druken | yep... but damned if i can remember why now... hehehe probably to see if you found a control board for me... |
| 01:14.47 | Neter66 | BleedingMe: ya, i get "all circuits busy" error until i reboot |
| 01:14.58 | Dr_Ray | I have an ernest ETX board you can have |
| 01:15.10 | Dr_Ray | I dunno what condition it is in |
| 01:15.19 | Dr_Ray | but you are welcome to it |
| 01:15.29 | Druken | how much ? |
| 01:15.31 | Neter66 | what I would like to do is monitor the status, and reboot the box if the "unreachable" ever shows up for more than 1 minute. any thoughts? |
| 01:15.31 | {zombie} | Neter66: does your upstream router change IP addresses (dynamic IP) when it reboots? |
| 01:15.32 | Dr_Ray | 0 |
| 01:15.39 | Druken | i like that price... |
| 01:15.48 | BleedingMe | Neter66: and basically it's just loosing internet access, as far as the asterisk box is concerned right? |
| 01:15.51 | Dr_Ray | if you end up fixing it and selling it, slide me something |
| 01:16.03 | Dr_Ray | but , you'll need relays and a hopper |
| 01:16.07 | Neter66 | zombie: it might, i don't know. but i have IP address in iax.conf instaed of dns name (as i heard there was an issue with this) |
| 01:16.13 | Druken | sell it? no.. it'll be going into my phone |
| 01:16.22 | {zombie} | Neter66: and is that router running Linux with ip_conntrack perhaps? |
| 01:16.23 | Dr_Ray | I think you underestimate how much work it will be |
| 01:16.28 | {zombie} | it's just that I'm battling a similar problem |
| 01:16.39 | Dr_Ray | send me your address petscii@gmail.com |
| 01:16.50 | Neter66 | BleedingMe: that's what it thinks - but i can ping external |
| 01:17.04 | {zombie} | conntrack is "remembering" the old IP address |
| 01:17.11 | {zombie} | but only for that session |
| 01:17.14 | Druken | Dr_Ray: i've already purchased a new hookswitch and handset for it... i was ready to buy a new control board for the bitch too |
| 01:17.16 | BleedingMe | Neter66: you shouldn't have to reboot just because you lose internet access... what version of asterisk are you running? |
| 01:17.35 | {zombie} | I think rebooting just stops asterisk for long enough so that the conntrack entry times out |
| 01:17.35 | Dr_Ray | ko |
| 01:17.43 | Neter66 | BleedingMe: latest build of @home |
| 01:18.00 | BleedingMe | Neter66: that's not a version of Asterisk :) |
| 01:18.21 | Dr_Ray | Druken - I assume yo ulive in american? |
| 01:18.23 | Neter66 | BleedingMe: i know, i know, i just answered that while i logged onto the box to get build number |
| 01:18.25 | Dr_Ray | er, america |
| 01:18.26 | Neter66 | :) |
| 01:19.05 | Druken | Dr_Ray: you assume incorrectly... canadian... |
| 01:19.18 | Dr_Ray | that's fine, I'm not shipping to buttcrack africa |
| 01:19.22 | Dr_Ray | :) |
| 01:19.29 | Druken | i wouldn't either :) |
| 01:19.49 | Dr_Ray | do you have canadian coinmechs? |
| 01:20.05 | BleedingMe | buttcrack africa? don't think i'll ever be visiting there |
| 01:20.06 | *** join/#asterisk Neter66 (n=neter66@CPE0012170db24c-CM0012c9db4f34.cpe.net.cable.rogers.com) |
| 01:20.09 | Druken | nope.. i was just going to rip the magnets out of the american one... |
| 01:20.17 | Neter66 | BleedingMe: version 1.0.9 |
| 01:20.47 | Druken | Dr_Ray: http://www.abss.ca/gallery/album04 |
| 01:21.05 | Druken | that's what i wanted ya for... i took pictures :) |
| 01:21.08 | Dr_Ray | you kn ow it's by weight |
| 01:21.21 | Neter66 | zombie: is the conntimer a provider timer, or one locally within asterisk? |
| 01:21.46 | Druken | Dr_Ray: nah... canadian coins are magnetic, so they don't pass threw the coin mech, american coins do |
| 01:21.58 | Druken | american coins are magnetic |
| 01:22.00 | Dr_Ray | oh |
| 01:22.02 | Druken | er.. aren't |
| 01:22.19 | BleedingMe | Neter66: only thing i could think of trying is maybe setting your qualify manually to something like 100 .. just to see what happens |
| 01:22.39 | {zombie} | Neter66: it's a linux thing |
| 01:22.56 | Neter66 | BleedingMe: 100 being milliseconds as the response time from the ping? |
| 01:23.02 | BleedingMe | yah |
| 01:23.16 | BleedingMe | i think the default is 300 when you just say yes |
| 01:23.41 | Neter66 | i think i remember reading that it was 2000, but i may be wrong |
| 01:23.56 | BleedingMe | that's two seconds!? i don't think so :) |
| 01:24.00 | Druken | Dr_Ray: those pics help to know if the control board ya got might work ? |
| 01:24.21 | Dr_Ray | it'll work |
| 01:24.31 | *** join/#asterisk [hC] (n=hardcore@c-24-127-192-210.hsd1.fl.comcast.net) |
| 01:24.37 | Dr_Ray | the problem is that control boards need parts that are wired for them |
| 01:24.55 | Neter66 | zombie: any thoughts on how to debug if that might be the problem? |
| 01:25.08 | Druken | well, my coinco doesn't have any wires... it's totally mechanical |
| 01:25.09 | Dr_Ray | like ernest boards won't work with protel relays |
| 01:25.16 | TheCops | someone have installed 1.2.0 on fedora core 3 ? |
| 01:25.28 | {zombie} | Neter66: shut down asterisk for a few minutes then restart it |
| 01:25.31 | {zombie} | rather than rebooting the whole machine |
| 01:25.38 | {zombie} | or |
| 01:25.41 | BleedingMe | i got 1.2.0 on Enterprise 3.. and I can't get it to start.. :) |
| 01:25.48 | {zombie} | temporarily add a firewall rule blocking access to your provider |
| 01:26.03 | {zombie} | I just did that to "solve" mine |
| 01:26.12 | TheCops | BleedingMe same here |
| 01:26.15 | {zombie} | iptables -I OUTPUT -d 202.72.135.6 -j DROP |
| 01:26.17 | TheCops | on fedora core 3 |
| 01:26.33 | BleedingMe | TheCops: what happens when you try to start it? |
| 01:26.37 | {zombie} | then removed that 2 minutes later |
| 01:26.39 | {zombie} | and it all came good |
| 01:26.45 | TheCops | BleedingMe, it can't compile at all |
| 01:26.50 | BleedingMe | ew |
| 01:26.58 | BleedingMe | Beta 2? |
| 01:27.11 | Dr_Ray | Druken , I run one kind of payphone, saves hassles for repairs |
| 01:27.54 | Druken | Dr_Ray: this is my first payphone... my only .. hehe for now |
| 01:28.04 | TheCops | BleedingMe nope, 1.2.0 |
| 01:28.14 | Dr_Ray | yeah, you'll need to figure out what works for you.. but I hope this helps |
| 01:28.25 | BleedingMe | right.. there 1.2.0 Beta 1 and Beta 2 (or RC2) ? |
| 01:28.33 | JunK-Y | tips of the day: dont go shopping x-mas gift with ur gf or u will buy something like that: http://junky.homelinux.org/MVI_4742.AVI |
| 01:28.36 | JunK-Y | hehe |
| 01:28.55 | TheCops | BleedingMe, 1.2.0 stable has been release on the website, eh? |
| 01:28.58 | Neter66 | zombie: thanks. i'll give that a try. any way to script it from wihtin asterisk? or am i just going to have to asterisk -r -x 'iax2 show peers' from cron every minute? |
| 01:29.07 | file[laptop] | JunK-Y: what are you doing to that poor horse?!? |
| 01:29.11 | file[laptop] | nasty. |
| 01:29.21 | BleedingMe | holy crap.. i just downloaded beta 2 yesterday! |
| 01:29.34 | Druken | Dr_Ray: yeah, my too... otherwise i'm throwing an additional $150 us into this phone |
| 01:29.36 | TheCops | lol |
| 01:29.44 | TheCops | BleedingMe, read the topic |
| 01:29.52 | BleedingMe | hehe.. duhhhh |
| 01:30.20 | BleedingMe | guess i'll have to download it |
| 01:30.38 | JunK-Y | hehee |
| 01:30.45 | *** join/#asterisk mod_ssl (n=user20sk@203.125.17.226) |
| 01:30.49 | mod_ssl | hello all |
| 01:31.48 | mod_ssl | have a quick question about zap trunks |
| 01:31.51 | *** join/#asterisk AmyTek (i=yyz@host181-218.pool8258.interbusiness.it) |
| 01:32.48 | JunK-Y | 42 |
| 01:33.02 | {zombie} | mod_ssl: don't ask if you can ask, just ask |
| 01:33.40 | mod_ssl | :) |
| 01:34.03 | mod_ssl | i have 2 fxo cards |
| 01:34.07 | mod_ssl | in my asterisk box |
| 01:34.48 | mod_ssl | and its all in group 0 |
| 01:35.12 | mod_ssl | channel fxsks=1 and fxsks=2 |
| 01:35.41 | mod_ssl | 1st is group=0 channel => 1 |
| 01:35.51 | mod_ssl | 2nd is group=0 channel => 2 |
| 01:35.59 | mod_ssl | when i address them in asterisk |
| 01:36.11 | mod_ssl | do i go Zap/g0-1/ |
| 01:36.19 | mod_ssl | ? |
| 01:36.22 | file[laptop] | g0 to address them as a group |
| 01:36.28 | file[laptop] | 1 or 2 to address them as individual channels |
| 01:36.41 | *** join/#asterisk AgiNamu (n=AgiNamu@8.7.80.194) |
| 01:36.55 | mod_ssl | so this would work Zap/g0-1/ |
| 01:36.56 | mod_ssl | ? |
| 01:37.02 | file[laptop] | did I say that? no |
| 01:37.08 | file[laptop] | g0 for a group, 1 or 2 for individual |
| 01:37.13 | file[laptop] | ie: Zap/g0/ Zap/1/ Zap/2/ |
| 01:37.36 | Druken | ok, what is the cmd to restart when convient in a crontab ? |
| 01:37.45 | JunK-Y | file[laptop]: if u want more vids, let me know, that poney is waiting me! |
| 01:37.48 | tzanger | Druken: don't do that |
| 01:38.11 | Druken | tzanger: explain ? |
| 01:38.11 | *** join/#asterisk obsidian-studios (n=obsidian@c-66-177-188-197.hsd1.fl.comcast.net) |
| 01:38.25 | mod_ssl | file: sorry dude:P i am just confused with the zaps:P |
| 01:38.42 | tzanger | Druken: FIX the underlying problem, don't band-aid it with a stupid restart every night/week/month |
| 01:39.03 | tzanger | you never get to the bottom of it, we can never get a proper fix in and you help asterisk decline into shitty windows-dom. |
| 01:39.31 | mod_ssl | file: can i pm u? |
| 01:39.56 | Druken | tzanger: well... the problem would be a single sipura 486 screwing things up, and i get ghost channels that don't go away without a restart |
| 01:40.13 | tzanger | Druken: sounds like you need a new ATA |
| 01:40.40 | Druken | tzanger: perhaps... but i've seen one of my pap2's do it once.. |
| 01:41.27 | Druken | tzanger: basically if the call is in progress, and the net connection goes down, and the phone is hung up, the call is dropped by the ata, but asterisk doens't know about it |
| 01:41.29 | tzanger | Druken: sounds like a deeper issue that needs to be fixed, even if it's worked around in * |
| 01:41.33 | Druken | so it stays there |
| 01:42.22 | Druken | it's just the fact that asterisk doesn't monitor the rtp to see if a call is still active or not... |
| 01:42.50 | tzanger | asterisk really can't |
| 01:42.55 | tzanger | because you could have had a silent frame |
| 01:43.14 | tzanger | now obviously if an hour passes without audio it's probably a pretty good indication that you missed the SIP goodbye... but still |
| 01:43.17 | Druken | silent frames for 12 hours? |
| 01:44.11 | tzanger | Druken: as I said... :-) |
| 01:44.17 | Druken | :) |
| 01:44.48 | Druken | i would say even 5 mins... now i dunno if asterisk can watch for no audio for a time of 5 mins... but it would save alot of headaches |
| 01:45.00 | tzanger | it coudl be a timer sure |
| 01:45.05 | tzanger | but the real issue is what is causing it |
| 01:45.15 | tzanger | is the ATA not sending the proper control message? |
| 01:45.20 | tzanger | or is * misresponding |
| 01:45.31 | Druken | not sure... |
| 01:46.26 | Druken | tzanger: it's one of those, how to you track it... hehehe |
| 01:46.41 | Druken | s/to/do |
| 01:46.49 | tzanger | with ethereal of course |
| 01:46.59 | tzanger | ethereal's sip support is actually quite good |
| 01:47.12 | [hC] | uhh, i thought that sip.conf had a setting for hanging up if it doesnt see rtp data for x amount of time already |
| 01:48.36 | Druken | [hC]: doubt it.. because the problem is.. with invites, you can not have asterisk in the path |
| 01:49.02 | *** join/#asterisk tengulre (n=tengulre@222.90.66.4) |
| 01:51.26 | {zombie} | if asterisk is not in the path, then asterisk won't show it as an active channel anyway |
| 01:51.50 | file[laptop] | Druken: there's an option in sip.conf fyi, to drop the call if there's no RTP traffic after so many seconds |
| 01:51.54 | Druken | {zombie}: very true |
| 01:51.55 | *** join/#asterisk flenders (n=chatzill@b03DE.static.pacific.net.au) |
| 01:52.40 | Druken | file[laptop]: oh... well i guess i'll have to look that up then |
| 01:52.41 | *** part/#asterisk obsidian-studios (n=obsidian@c-66-177-188-197.hsd1.fl.comcast.net) |
| 01:52.53 | *** join/#asterisk aiki (n=aiki@pool-64-222-121-213.burl.east.verizon.net) |
| 01:53.17 | {zombie} | rtptimeout |
| 01:53.24 | {zombie} | Terminate call if x seconds of no RTP activity when we're not on hold |
| 01:53.28 | aiki | any have a soft phone connecting to asterisk through a openvpn connection? |
| 01:53.30 | {zombie} | handy |
| 01:53.43 | {zombie} | aiki: yup |
| 01:53.45 | {zombie} | hardphones too |
| 01:54.01 | aiki | cool.. |
| 01:54.17 | *** join/#asterisk Igbothom (n=HiltonT@static-84.217.240.220.dsl.comindico.com.au) |
| 01:54.22 | aiki | so i have the softphone registering (so SIP is up) but rtp doesn't work.. |
| 01:54.50 | *** part/#asterisk MasterYoda (n=mnichols@pdpc/supporter/sustaining/MasterYoda) |
| 01:54.53 | aiki | can't seem to see anything (rtp packets) if I do tcpdump -i tun0 on the vpn box |
| 01:55.33 | {zombie} | make sure the soft phone is using the right IP address for the rtp stream |
| 01:55.45 | {zombie} | and that it isn't picking its "world" IP via stun etc |
| 01:56.05 | aiki | nope stun isn't on.. |
| 01:56.20 | aiki | wait i need to configure STUN on the softphone right? |
| 01:56.43 | {zombie} | no, not if it has a direct connection to the asterisk server via the tunnel |
| 01:57.23 | aiki | it does.. |
| 01:57.53 | aiki | basically i can ping the asterisk server, (and do https, 5060 etc) but just no RTP |
| 01:58.08 | {zombie} | is your default route via the openvpn link? |
| 01:58.33 | Druken | {zombie}, file[laptop], tzanger: thanks for that.. hehe don't i feel stupid now... :) |
| 01:58.50 | {zombie} | Druken: np, I learned something too :) |
| 01:59.03 | [hC] | sure... dont trust me, or anything! |
| 01:59.20 | Druken | oh yes... [hC] as well... sorry bout that |
| 01:59.22 | [hC] | :) |
| 01:59.23 | aiki | nope default gateway shows my dsl router's ip.. |
| 01:59.24 | [hC] | np |
| 01:59.48 | {zombie} | aiki: and are you able to run a packet sniffer on the machine with the softphone, and make sure the rtp packets aren't going out your DSL link? |
| 02:00.12 | aiki | lemme check.. |
| 02:02.13 | Druken | [hC]: you know about flash operator panel?? ehehe |
| 02:02.22 | [hC] | Druken: A little bit :) |
| 02:02.41 | Druken | can't remember how to do the dynamic sip button :) |
| 02:02.49 | aiki | ah.. i think i see the problem.. my dsl's internet network is the same 192.168.1.x as the network inside my vpn.. |
| 02:02.59 | [hC] | you can use regex |
| 02:03.09 | [hC] | so something like [SIP:/*] should work |
| 02:03.18 | [hC] | theres an example in the current default config file that comes w/ the tarball |
| 02:03.19 | *** join/#asterisk test34 (i=1000@unaffiliated/test34) |
| 02:03.34 | {zombie} | aiki: that shouldn't matter, so long as the machine with the VPN isn't on that network |
| 02:03.48 | aiki | it isn't |
| 02:04.39 | aiki | hmm have a problem installing winpcap on the m/c with the softphone... |
| 02:05.27 | test34 | I dont need zttool right ? (I can't find libnewt-devel source) |
| 02:05.46 | *** join/#asterisk SwK (n=SwK@12-219-151-128.client.mchsi.com) |
| 02:06.02 | *** join/#asterisk AgiNamu (n=AgiNamu@8.7.80.194) |
| 02:06.22 | tzafrir_laptop | test34, why do you need the newt-devel source? just install the -devel package |
| 02:06.29 | AgiNamu | Hey there.... if I want to dial an IAX2 peer, but i dont wanna timeout too long, should I just use the timeout param in Dial? |
| 02:06.38 | AgiNamu | i.e., if the host is offline |
| 02:06.41 | test34 | tzafrir_laptop, I didn't find any package for slackware |
| 02:07.04 | aiki | sorry this is a little offtopic.. but know any good openvpn client's for linux? |
| 02:07.13 | {zombie} | AgiNamu: use qualify in your iax2.conf to check the IAX peer is reachable |
| 02:07.24 | tzafrir_laptop | maybe for slackware you just need "libnewt" or something. Though you can do fine without it |
| 02:07.28 | {zombie} | that way the Dial will automatically fail when it's not |
| 02:07.33 | AgiNamu | oh cool |
| 02:07.38 | AgiNamu | and I can still use a timeout |
| 02:07.43 | tzafrir_laptop | aiki, what about openvpn? |
| 02:07.46 | {zombie} | yeah the timeout is how long it rings for |
| 02:07.54 | AgiNamu | in case internally the peer has issues (i.e., I IAX2 to ServerB and ServerB tries to SIP somewhere else) |
| 02:08.24 | AgiNamu | thanks zombie |
| 02:08.52 | aiki | tzafrir_laptop, was hoping for a nice gui package.. |
| 02:09.08 | *** join/#asterisk santiago (n=santiago@208.195.215.124) |
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| 02:33.39 | [TK]D-Fender | . |
| 02:33.45 | [TK]D-Fender | *crickets* |
| 02:37.07 | *** join/#asterisk ronaldl79 (n=chatzill@c-24-8-54-203.hsd1.co.comcast.net) |
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| 02:37.57 | xbmodder_lappy | hey |
| 02:38.25 | ronaldl79 | G'Day. |
| 02:38.39 | ronaldl79 | Everyone happy to have their hands on 1.2? |
| 02:39.40 | IronHelix | hell no |
| 02:39.44 | IronHelix | we shoulda stuck with 0.8 |
| 02:39.52 | ronaldl79 | lol |
| 02:39.54 | IronHelix | :) |
| 02:40.04 | ronaldl79 | Iron, are you using HEAD or 1.2? |
| 02:40.25 | IronHelix | for my personal box, 1.0.9 but thats only cuz its a POS and will take half an horu to compile |
| 02:40.38 | ronaldl79 | rofl |
| 02:41.04 | ronaldl79 | I wonder if the fixes in 1.2 are now in HEAD? Hmmmm... |
| 02:41.17 | Dr_Ray | I'm running whatever I was running 4 months ago |
| 02:43.43 | Pikoro | if it works, don't fix it |
| 02:43.44 | Pikoro | :D |
| 02:44.18 | Pikoro | oh IronHelix, it was an irq conflict |
| 02:45.07 | IronHelix | ahhh |
| 02:45.10 | IronHelix | that'll do it |
| 02:51.59 | *** join/#asterisk mog_home (n=mogorman@user-24-236-84-48.knology.net) |
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| 02:56.25 | justinu | anyone setup DNS NAPTR records? |
| 03:01.58 | *** join/#asterisk docelm0 (n=docelmo@static-71-251-95-2.tampfl.fios.verizon.net) |
| 03:03.06 | docelm0 | Say does anyone know how to pull variables set by the dialplan to a shared object? |
| 03:03.30 | docelm0 | or what function I can use? |
| 03:03.39 | IronHelix | you could use System() to get them out of *-land |
| 03:04.13 | mog_home | what d o you want docelm0 |
| 03:04.59 | docelm0 | I am trying to create a shared object module to run in asterisk and I am curious to know how to pull variables set by the dial plan |
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| 03:05.19 | file[laptop] | pbx_builtin_getvar_helper |
| 03:06.03 | file[laptop] | http://www.asterisk.org/doxygen/pbx_8c.html#a192 |
| 03:06.39 | mover | anyone know about this error? app_rxfax.c:150: error: structure has no member named `ast_chan' |
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| 03:16.30 | Corydon76-home | file[laptop]: pbx_builtin_pushvar_helper() |
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| 03:21.15 | ronaldl79 | Can anyone confirm if 1.2 code was patched to CVS HEAD? |
| 03:21.47 | ronaldl79 | I'm looking through the bug tracker now, but I'll spend all day trying to find an answer. |
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| 03:42.33 | JunK-Y | ronaldl79: specific question? |
| 03:45.06 | ronaldl79 | JunK-Y.... |
| 03:45.21 | ronaldl79 | I'm trying to confirm if HEAD is patched with the fixes from 1.2 |
| 03:45.32 | ronaldl79 | I guess everyone's asleep. It's a fairly simple question. |
| 03:45.48 | Pikoro | not sure |
| 03:45.56 | Pikoro | i'm using HEAD and it seems to work fine though |
| 03:46.14 | *** join/#asterisk mog_home (n=mogorman@user-24-236-84-48.knology.net) |
| 03:46.20 | JunK-Y | ronaldl79: it depends what exactly, but it should be for almost everything yeah. |
| 03:46.21 | ronaldl79 | Yeah, I just don't want to miss out on anything, otherwise...I'll just download 1.2 until HEAD is patched.... |
| 03:46.46 | JunK-Y | its probably the reverse. |
| 03:46.58 | *** join/#asterisk SwK (n=SwK@12-219-151-128.client.mchsi.com) |
| 03:47.00 | ronaldl79 | Ahh |
| 03:47.33 | *** join/#asterisk diamon (n=diamon@c-66-176-91-189.hsd1.fl.comcast.net) |
| 03:47.50 | docelm0 | Thanks file |
| 03:48.58 | diamon | Is there any special reason I'd want the G.729 codec except for the lower bandwidth needed? |
| 03:49.09 | IronHelix | no the bandwidth is pretty much it |
| 03:49.20 | IronHelix | 729 is about as bandwidth friendly as you can get |
| 03:49.22 | diamon | No real difference in sound quality? |
| 03:49.29 | IronHelix | its also a bit easier on the cpu than ilbc |
| 03:49.40 | IronHelix | it doesnt soudn AS good as ulaw but it sounds pretty good |
| 03:49.59 | Juxt | hard to hear a difference |
| 03:50.09 | Juxt | until you stumble on musiconhold or something |
| 03:50.10 | diamon | Ah, ok, that's another plus. And so long as it's pretty reasonable quality, that'd be all I'm looking for. |
| 03:50.24 | IronHelix | for voice its pretty good but 729 misses some of the nuances that ulaw catches |
| 03:50.32 | *** join/#asterisk juice (n=juice@mo-69-69-117-72.sta.sprint-hsd.net) |
| 03:50.41 | IronHelix | if you dont want to pay per channel, try GSM |
| 03:50.43 | Juxt | i've had luck trunking 90+ calls in g729 over a t1 |
| 03:50.46 | diamon | Isn't ulaw the next-best thing to uncompressed? |
| 03:50.53 | IronHelix | yeah pretty much |
| 03:50.54 | Juxt | ulaw is uncompressed |
| 03:51.01 | IronHelix | ulaw is 64kbit/sec |
| 03:51.06 | IronHelix | gsm is about 15kbit/sec |
| 03:51.11 | diamon | Ah, I thought it had RLE compression or something simplistic in it. |
| 03:51.17 | IronHelix | ilbc is 12 or 13 as i recall |
| 03:51.20 | IronHelix | ulaw is 8kbit/sec |
| 03:51.28 | IronHelix | multiply all these by two |
| 03:51.31 | IronHelix | for full duplex |
| 03:51.35 | IronHelix | and add overead |
| 03:52.01 | IronHelix | eh not ulaw, 729 is 8kbit/sec |
| 03:52.01 | *** join/#asterisk bmg505 (n=leon@rndf-146-52-95.telkomadsl.co.za) |
| 03:52.15 | IronHelix | so if you have more cpu you can get away with ilbc, or a bit more bandwidth gsm works |
| 03:52.17 | diamon | So, my cable pipe is supposed to be 768/6Mbit, so if a ulaw call takes about 64kbit, 64/768 would be about 12 calls over that at once (ideally and all, I'm sure). |
| 03:52.27 | Flauto | 8kb is not bad |
| 03:52.28 | diamon | Oh yeah, so 6 then. |
| 03:52.46 | Juxt | yeah provided you have mad QoS that's possible |
| 03:52.49 | JonR800 | if you need lower bandwidth than ulaw, i like g726. it's free and i think the sound quality is just under ulaw compared to the rest. |
| 03:52.50 | IronHelix | somewhere between 6 and 12 |
| 03:52.56 | IronHelix | 64kbit in each direction |
| 03:53.09 | IronHelix | so one 64 would come out of your 768k, the other 64 would come out of your 6meg |
| 03:53.25 | diamon | And I did set up QoS, only Terminal Services (work stuff) and VoiP have High priority. |
| 03:53.40 | IronHelix | terminal services should have above average, not high |
| 03:53.40 | *** join/#asterisk kks (n=kks@202.73.8.130) |
| 03:53.42 | diamon | Iron: Wow, I'm slightly off on a lot of things today... |
| 03:53.58 | IronHelix | voip > terminal services > everything > p2p |
| 03:54.07 | IronHelix | imho at least thats a good traffic shaping profile |
| 03:54.24 | Juxt | yeah it is |
| 03:54.28 | diamon | Iron: TS is too important in my job, really. I could do without the phone so long as I can TS anywhere. :) |
| 03:54.35 | Flauto | diamon, how to setup qoS? |
| 03:54.36 | Juxt | must be tricky to shape p2p cause it flies thru so many ports |
| 03:54.38 | IronHelix | you can still USE ts |
| 03:54.40 | Flauto | would anyone teach me? |
| 03:55.02 | Juxt | Flauto: http://lartc.org/ |
| 03:55.16 | IronHelix | but with ts, it doesnt matter if it takes an extra 1/10th of a second to move the mosue |
| 03:55.17 | Pikoro | wierd.. the ringer sounds wierd on the customer's end |
| 03:55.18 | IronHelix | voip it does matter |
| 03:55.26 | Pikoro | they keep hanging up cause it sounds like a busy signal |
| 03:55.30 | IronHelix | thus voip should be higher than TS, ts can still be higher than default |
| 03:55.54 | IronHelix | pikoro that could be in indications.conf |
| 03:55.56 | *** part/#asterisk mychi (n=murphyx@pool-151-197-185-19.phil.east.verizon.net) |
| 03:56.02 | IronHelix | asterisk is sending what it thinks is japanese ring tone |
| 03:56.06 | IronHelix | which might not be what you want |
| 03:56.10 | Pikoro | no it's not |
| 03:56.11 | Pikoro | haha |
| 03:56.28 | Pikoro | says country=us in indications.conf |
| 03:56.30 | Pikoro | hmm |
| 03:56.35 | Pikoro | but zap is set to jp |
| 03:56.40 | diamon | Juxt: I don't list P2P stuff, that way it defaults to the lowest priority. |
| 03:56.40 | Qwell | doesn't matter... |
| 03:56.43 | Flauto | if i want to use qos, do i have to use my linux machine as a router? |
| 03:56.45 | Qwell | indications trumps all :p |
| 03:56.48 | Flauto | or i don't |
| 03:56.56 | *** join/#asterisk simprix (n=simprixn@24-231-248-225.static.aldl.mi.charter.com) |
| 03:56.57 | Pikoro | doesn't sound like a US ringer either |
| 03:56.58 | IronHelix | flauto- check out linksys wrt54g router with sveasoft firmware |
| 03:56.58 | Qwell | Flauto: no, of course not |
| 03:57.03 | IronHelix | pretty simple and easy to setup |
| 03:57.18 | Flauto | great |
| 03:57.22 | Pikoro | hmm |
| 03:57.25 | IronHelix | keep in mind tho the new wrt v5 wotn work with sveasoft (what the hell is wrong with linksys? what do they expect to gain by switching to vxworks?) |
| 03:57.30 | Pikoro | i gotta find some ring definitians for JP |
| 03:57.31 | Flauto | i am loading the page now |
| 03:57.49 | *** join/#asterisk newmember (n=newmembe@70.72.189.149) |
| 03:57.55 | simprix | I am trying to setup asterisk with a sip xten client over voicepulse and when i call someone it sounds like im in a cave or wind tunnel but when they call me it sounds fine. Where can I look to fix this |
| 03:58.01 | JonR800 | IronHelix: lower memory footprint? cheaper boxes? |
| 03:58.09 | diamon | For QoS, I used ipcop as a linux-based dedicated firewall. Fairly straightforward, and modifiable enough for my needs. |
| 03:58.15 | JonR800 | that's just a guess, i don't actually know |
| 03:58.16 | justinu | simprix; that's odd |
| 03:58.21 | *** join/#asterisk emakris2 (n=emakris@c-24-128-56-2.hsd1.ma.comcast.net) |
| 03:58.23 | Flauto | i downloaded the chinese pdf file and i can not see anything i can understand |
| 03:58.23 | IronHelix | perhaps but from what i hear the new vxworks version doesnt have many of the features the old one did |
| 03:58.27 | IronHelix | causing them tons of CS problems |
| 03:58.29 | Flauto | hehe |
| 03:58.30 | simprix | does xten have problems |
| 03:58.30 | Flauto | okay |
| 03:58.36 | Flauto | i will read in english |
| 03:58.37 | Pikoro | gotta stop my customers from hanging up cause of the ring tone |
| 03:58.37 | IronHelix | simprix- could be a codec problem |
| 03:58.37 | Pikoro | heh |
| 03:58.56 | IronHelix | simp= in sip.conf set your codec order to be disallow=all allow=ulaw allow=alaw |
| 03:58.58 | IronHelix | see if that helps |
| 03:59.31 | JonR800 | IronHelix: it will affect sales in the long run, but for now they can produce them with less memory it seems. linksys has always been retarded. |
| 03:59.44 | diamon | Which of the *law codecs was it that faxing might work with again? |
| 03:59.49 | IronHelix | ulaw |
| 03:59.51 | justinu | g711 |
| 03:59.53 | IronHelix | haha |
| 04:00.00 | IronHelix | yeah thats true jon is uppose |
| 04:00.16 | IronHelix | they also are having a 'wrt54gl' which is just a wrt54g v4 with linux |
| 04:00.24 | IronHelix | it seems like a dumb idea to me |
| 04:00.51 | IronHelix | personally, i've bought at least 5 wrt boxes for myself and clients, and installed sveasoft on others that already have them |
| 04:00.59 | IronHelix | part of the appeal is that they are so readily avialabvle |
| 04:01.01 | ronaldl79 | Does anyone have mixed feelings about native MOH playback vs MPG123? |
| 04:01.05 | diamon | Thankfully, my local computer store overstocked on wrt54g's v2, so I just keep sending folks there to get real wireless routers. :) |
| 04:01.32 | Flauto | iron, i need to download that thing for qos? |
| 04:01.52 | JonR800 | IronHelix: personally i swore off their products after my 4th failed linksys switch. I have zero luck with their products. |
| 04:02.13 | IronHelix | i recommend it, its possible to do qos with the latest stock firmware, but i recommend sveasoft, more features |
| 04:02.14 | IronHelix | keep in mind |
| 04:02.19 | SkramX | Is everything in 1.2 backwards compatible? |
| 04:02.30 | konfuzed | hey is there a good chat room to send VoIP end users that are not computer phobes but not inclined to deal with a PBX or actually need to work out any thing for asterisk |
| 04:02.31 | IronHelix | you MUST reset your router to factory default using the web interface (NOT the button) after you upgrade |
| 04:02.41 | konfuzed | i dont want to send such end users here |
| 04:02.45 | IronHelix | #asterisk-n00bs? |
| 04:02.52 | JonR800 | SkramX: there is a lot that has been deprecated from 1.0.. i would be careful :) |
| 04:03.09 | simprix | IronHelix: that is already in my sip.conf |
| 04:03.11 | konfuzed | they just want a phorum to figure out using their softphones curiousities about voip |
| 04:03.22 | IronHelix | konfused- dont send them to chat, charge them $150/hr for consulting |
| 04:03.26 | IronHelix | and explain it all to them |
| 04:03.27 | SkramX | Ill stick with my 1.0.9 and wont change customers over |
| 04:03.28 | IronHelix | every bit of it |
| 04:03.29 | Flauto | juxt, do i need to download that 2.4 thing for qos? |
| 04:03.31 | IronHelix | in great detail |
| 04:03.39 | konfuzed | if they had money that would be a good idea |
| 04:03.39 | Juxt | absolutely. |
| 04:03.43 | Juxt | better 2.6 |
| 04:04.21 | *** join/#asterisk stkn_ (i=nobody@gentoo/developer/pdpc.active.stkn) |
| 04:04.21 | IronHelix | flauto remember you only have to install stuff on the computer if you do it on a linux level |
| 04:04.29 | konfuzed | instead they prefer to find free software and get free on site training long winded discusions about figuring out what those companies charge |
| 04:04.31 | IronHelix | which only is useful if you use linux as your interent gateway |
| 04:04.32 | konfuzed | for |
| 04:04.50 | IronHelix | lol |
| 04:04.58 | diamon | So, as far as VoIP providers go, AFAICT I just need to research the firms and choose one with a price and feature set I want pretty much, yes? Anything special I need to know or anyone special I should use to allow my * server to use both VoiP and POTS via an X100P card? I'm just about to start loading Asterisk to a server here and want a provider that lets me call real phones for testing and such... |
| 04:05.19 | konfuzed | and then want to know how they can use voip without paying any one anything ever again |
| 04:05.21 | diamon | Voicepulse seems like my lead contender so far. |
| 04:05.32 | IronHelix | hahhaa nice konfused |
| 04:05.40 | IronHelix | tell them to get FWD or something |
| 04:05.41 | simprix | Does the xten have problems making outgoing calls sounding like im in a cave |
| 04:05.43 | diamon | konfuzed: You know, there's some clients out there that just aren't worth the money they bring in. :P |
| 04:05.55 | IronHelix | very true diamon |
| 04:06.02 | Juxt | guh voicepulse is a contender? |
| 04:06.05 | konfuzed | diamon: uh they dont have any money |
| 04:06.07 | Flauto | ironhelix, i do need to have linux as my gateway, means all the internet trafic is through my linux, |
| 04:06.18 | Pikoro | ok.. anyone know where i can find a indications.conf with a japan definition? |
| 04:06.23 | IronHelix | yeah then you dont need the wrt at all |
| 04:06.25 | konfuzed | still though just some simple people in low income housing |
| 04:06.29 | diamon | IronHelix: Yah, a hard-learned lesson I got long ago running my own PC tech shop. I don't service bottom-feeders any more. |
| 04:06.48 | IronHelix | i have a pretty simple strategy |
| 04:06.50 | IronHelix | i dont advertise |
| 04:06.51 | IronHelix | ever |
| 04:06.56 | Juxt | voicepulse is the reason i got into this business - they sucked so badly for me |
| 04:07.01 | diamon | konfuzed: That'd be different, in a social-responsibility sort of way. |
| 04:07.07 | IronHelix | but when people ask for my card, i give them 2 or 3. |
| 04:07.10 | IronHelix | works like a charm |
| 04:07.19 | IronHelix | that plus treat my clients well = new clients |
| 04:07.22 | diamon | Iron: Nice idea, actually. |
| 04:07.25 | konfuzed | i cant help but wanting to help them figure it out and get around the telcos and such but usually it is all over their head |
| 04:07.44 | IronHelix | this way the morons dont read advertisements and figure they can call me to fix their coffee cup holder |
| 04:07.47 | diamon | Iron: And that's a rule I've always followed. "If I don't treat my clients well, someone else will be glad to do so." |
| 04:07.54 | IronHelix | hehe |
| 04:08.33 | IronHelix | anyway diamon- for providers voicepulse is good, but when dealing with voip, esp the low price wholesale stuff, i suggest put a deposit of $10 or so on file with a few different providers, and setup your dialplan to failover |
| 04:08.51 | konfuzed | Im trying to seperate myself from having to speak to supposed "clients" or tease-clients |
| 04:09.18 | diamon | Iron: Arrgh, I used to think the Coffe Cup Holder thing was a joke. Now, I know it's not. For anyone interested, cheap styrofoam coffee cups fit perfectly in the inner track, and 'standard' coffee mugs fit perfectly in the inner 'mini-CD' ring... |
| 04:09.28 | justinu | lol |
| 04:09.34 | diamon | Iron: A deposit with a few providers? Eh? |
| 04:09.34 | ronaldl79 | Hmmm..consultants are charging $150/hour for *? |
| 04:09.39 | IronHelix | voicepulse is good, nufone is pretty popular here, docelm0 was working for a company called plainvoip... asterlink is run by some guys from here.. bkw_ maybe i think, voxee is also decent |
| 04:09.39 | konfuzed | i know somebody out there enjoys talking to them endlessly about their problems instead of their error messages but |
| 04:09.42 | konfuzed | it isnt me |
| 04:09.44 | konfuzed | ;^) |
| 04:10.01 | IronHelix | diamon it depends on how you buy service |
| 04:10.13 | IronHelix | if you buy a 'line' ie pay $xx/mo for unlimited in/out then dont bother |
| 04:10.20 | IronHelix | but if you buy wholesale, ie pay for minutes |
| 04:10.21 | konfuzed | ronaldl79: scale and scope of customer determines their willingness to pay $xxx/hr |
| 04:10.29 | docelm0 | ronald I only charge $80 |
| 04:10.42 | konfuzed | bsiness with a pbx thats costing them a fortune will defintely pay |
| 04:10.48 | konfuzed | law firms defintely pay |
| 04:10.56 | IronHelix | get 2 or 3 companies lined up, put the one you like first, and setup * to go to #2 or #3 if the first one is down |
| 04:11.00 | diamon | Iron: Ah, I get it. The 'least-cost routing' type stuff. |
| 04:11.00 | IronHelix | for greater reliability |
| 04:11.03 | ronaldl79 | Well, I guess I'll try and figure out a medium for my consulting rates for *. |
| 04:11.09 | IronHelix | not LCR, failover routing |
| 04:11.20 | docelm0 | Iron who wants minutes? |
| 04:11.20 | diamon | Right, but a similar way to go about it. |
| 04:11.25 | docelm0 | I sell minutes.. :) |
| 04:11.31 | Juxt | actually my LCR script sorts the carriers in the increasing cost order |
| 04:11.33 | konfuzed | people at home trying to figure out how not to have to pay a phone bill usually dont have squat to part with in the first place |
| 04:11.37 | IronHelix | i was wondering if you'd pop outta the woodwork |
| 04:11.40 | Juxt | so it does failover lcr |
| 04:11.56 | docelm0 | ronaldl79 where you from whats your background with asterisk? |
| 04:12.12 | IronHelix | very true konfused... if they arent willing to pay at least a little, they should stfu and get skype or fwd |
| 04:12.25 | diamon | Mmmm, skype... |
| 04:12.26 | Pikoro | seems zap's configs are used for zap trunks instead of indications.conf |
| 04:12.29 | docelm0 | I wrote a LCR.. :) |
| 04:12.30 | jtodd | question for the ether: Is there any way to get the IP address of the _last_ proxy sending a SIP INVITE, without specifying a specific SIP PEER entry? I know about $SIPURI, but that is the orginating UA. |
| 04:12.50 | konfuzed | my plan is to convert housing co-ops in to bulk semi-corporate customers |
| 04:13.05 | konfuzed | with the network in place that becomes a captured marketplace |
| 04:13.06 | IronHelix | thats not a bad idea |
| 04:13.09 | konfuzed | no competition |
| 04:13.09 | alephcom | Is anybody here using netlogic? |
| 04:13.14 | mog_home | ~thebook |
| 04:13.20 | jbot | [thebook] Asterisk: The Future of Telephony, released under the Creative Commons license and available at http://www.asteriskdocs.org << Read the book online! |
| 04:13.26 | diamon | Any ATA's strongly recommended around here? Preferrably anything I can get from CompUSA, as they are local to me and have a nice return policy. :) |
| 04:13.44 | ronaldl79 | docelm0: I've been in Denver for a few months now, I deploy Asterisk for businesses and non-profits...been involved with VoIP for over 10 years now. |
| 04:14.32 | ronaldl79 | docelm0: Feel free to visit sometime...http://www.riverscapecorp.com |
| 04:14.32 | IronHelix | diamon- NO NO NO NO NO |
| 04:14.34 | IronHelix | no compusa |
| 04:14.36 | docelm0 | nice.. I have a friend doing that in denver.. Makes fairly good money doing it. I am doing it in tampa |
| 04:14.45 | konfuzed | then all their whiny calls about how they cant read the manual or how they need free support for xyz under the sun that in some way uses the internet can just be forwarded to the Service department |
| 04:14.50 | IronHelix | what you buy at compusa will be LOCKED to a voip provider and will be almost certainly useless to you |
| 04:14.52 | diamon | Iron: Bah, that'd make my life easier. Oh well. Who then? |
| 04:15.03 | ronaldl79 | doc: What's your friend's name? I just met a group of * folks about two weeks ago.... |
| 04:15.08 | *** join/#asterisk YoMama (n=tchen@pcp02689850pcs.roylok01.mi.comcast.net) |
| 04:15.10 | IronHelix | check http://www.voipsupply.com i've had good luck wtih them. anything by Sipura is good |
| 04:15.10 | konfuzed | the nice service rep will explain to them how they can buy Tickets to waste our tech reps time\ |
| 04:15.16 | docelm0 | Mike Dewey |
| 04:15.24 | YoMama | IronHelix: my GXP is on its way |
| 04:15.39 | IronHelix | congrats dude! |
| 04:15.45 | ronaldl79 | Oh, I don't know him....he's deploying * in Denver? I wouldn't mind having lunch with him or something. It's always great to meet people using/deploying *. |
| 04:15.50 | ronaldl79 | doh |
| 04:16.05 | IronHelix | you got the gxp-2200 right? ;) |
| 04:16.10 | YoMama | IronHelix: ha...yeah |
| 04:16.29 | YoMama | so...i'm sure this is a hot button for a lot of you..but is there a way to make * do failover? |
| 04:16.35 | IronHelix | sure there is |
| 04:16.46 | *** join/#asterisk BhaalWK (i=bhaal@freenode/staff/bhaal) |
| 04:16.48 | YoMama | yeah? gimme the really short version of how it works |
| 04:16.53 | IronHelix | exten => whatever,1,Dial(sip/provider1/${EXTEN}) |
| 04:16.59 | IronHelix | exten => whatever,2,Dial(sip/provider2/${EXTEN}) |
| 04:17.03 | konfuzed | hey somebody gave me a Motoralla ATA that they got with a vonage service. |
| 04:17.14 | YoMama | IronHelix: eh...that's not what i meant :-P |
| 04:17.20 | IronHelix | how are you liking the paperweight konfused? |
| 04:17.34 | IronHelix | you mean failing asterisk itself over? |
| 04:17.52 | YoMama | IronHelix: you're registered to ServerA...ServerA's disk dies |
| 04:18.01 | YoMama | whaddya do |
| 04:18.16 | IronHelix | have a live backup copy of the conf files on serverb |
| 04:18.32 | konfuzed | IronHelix: well I told him its probably useless but I would see if there was some way to unlock it or flash the bios or reload different firmware or something |
| 04:18.32 | IronHelix | pop in server b, change its ip or update your DNS, total outage 5min tops |
| 04:19.16 | IronHelix | konfused- as i understand it, the tftp server is flashed into the thing, and it will only accept an update of that or any tftp file that is digitally signed by the provider |
| 04:19.40 | YoMama | IronHelix: bah...that requires manual effort |
| 04:19.46 | IronHelix | its worth a shot i guess, but imho not worth it. |
| 04:20.29 | konfuzed | IronHelix: it might be worthwhile if there was a guide to do it vs creating a research lab for the purpose |
| 04:20.39 | IronHelix | true |
| 04:20.39 | YoMama | IronHelix: realtime can read it's crap from mysql db right? |
| 04:20.58 | IronHelix | yomama- thats option #2, have several * servers in roundrobin dns that all read their config from a mysql db |
| 04:21.17 | IronHelix | i dont know that much about mysql but i think there is a way to cluster it redundantly |
| 04:21.18 | konfuzed | YoMama: do you have a particular reason to be more concerned about drive failure |
| 04:21.27 | YoMama | IronHelix: i know u can cluster mysql |
| 04:21.35 | IronHelix | er you wouldnt want roundrobin dns |
| 04:21.38 | IronHelix | that would be bad |
| 04:21.52 | IronHelix | but you might want static dns with a VERY short TTL |
| 04:22.11 | IronHelix | then setup a script, if *box1 stops responding, swap out the dns to point to *2 |
| 04:23.02 | YoMama | konfused: not disk failure in particular...just wanna be able to argue why * makes more sense over Cisco AVVID |
| 04:23.17 | konfuzed | oh well that changes the whole picture |
| 04:23.27 | db48x | I think he wants a cluster of * servers |
| 04:23.35 | YoMama | and so far...to me..that's the biggest draw for AVVID..u can cluster the CMs and the Unity VM |
| 04:23.42 | JonR800 | db48x: who doesn't. |
| 04:23.46 | db48x | if one fails, the calls are picked up by other servers :) |
| 04:23.48 | db48x | magically :) |
| 04:23.52 | konfuzed | cisco uses Asterisk servers to get around their cisco firewall voip problems |
| 04:24.06 | YoMama | db48x: dude..i've set that crap up before..it ain't magical at all |
| 04:24.16 | konfuzed | cisco will charge you thousands more and still uses asterisk |
| 04:24.21 | Pikoro | i get a wierd 8 second pause in this config |
| 04:24.30 | Pikoro | like, someone dials in, and they get the prompt... |
| 04:24.37 | Pikoro | says wait on the line to be connected... |
| 04:24.50 | Pikoro | then it pauses 8 seconds for some reason before the internal phones start ringing |
| 04:24.55 | YoMama | konfused: maybe...but when u go work for most companies..they want commerical support and failover |
| 04:24.58 | Pikoro | this is from a zap trunk |
| 04:25.17 | JonR800 | Pikoro: is there a wait in the extension logic ? ;) |
| 04:25.18 | YoMama | people moan when they hear email or the network is down..u tell 'em the phone system is down..they'll go home |
| 04:25.22 | konfuzed | YoMama: list out the cisco pre mail box license fees and then list the asterisk ones |
| 04:25.30 | Pikoro | there is one, but it is Wait(1) |
| 04:25.31 | db48x | YoMama: not magical for http perhaps, but that's different. with http someone requests a file and then the connection closes |
| 04:25.39 | YoMama | konfuzed: ARGH..dude..I know that * costs near nothing in comparison |
| 04:25.45 | Pikoro | and that is so the beginning of the announcement doesn(t get cut off |
| 04:25.53 | YoMama | that's not the point |
| 04:26.02 | konfuzed | well asterisk not only has commercail support but at less than half the cost of cisco support which in reality sucks shit |
| 04:26.35 | YoMama | konfused: actually...i'll disagree with u there..Cisco actually has relatively good support |
| 04:26.39 | JonR800 | Pikoro: hmm.. does it just sit there playing the sound then go? maybe there's 8 seconds of silence at the end of the prompt? (/me is just trying to think of the obvious first) |
| 04:26.50 | Pikoro | it's a built in prompt |
| 04:26.54 | konfuzed | YoMama: you've been in the corporate environment too long |
| 04:26.57 | Pikoro | privacy-please-stay-on-line-to-be-connected |
| 04:26.58 | YoMama | konfused: i've called their TAC plenty of times...u acutually speak to a real engineer |
| 04:27.08 | Pikoro | let me pastebin the relevant section |
| 04:27.20 | YoMama | konfused: if u want to make any decent money..that's where u go work..and if asterisk ever wants the respect it wants..it'll haveta address these questions |
| 04:27.35 | Pikoro | http://pastebin.ca/29096 |
| 04:27.52 | konfuzed | they cant fix it on the phone so they charge you several hundred dollars to bring you a complete replacement and they have nothing to offer for restoring what was lost |
| 04:28.11 | konfuzed | at least that was my experience workign as an admin in a large webhost data cneter |
| 04:28.31 | YoMama | konfuzed: it's an issue of risk mitigation |
| 04:29.09 | YoMama | no one wants a phone systme to go down...Cisco supports failover out of the box...i am NOT saying Cisco is better...the fact i've been dicking with * for the past two weeks is proof of that |
| 04:29.19 | Pikoro | haha |
| 04:29.24 | konfuzed | the only reason we had cisco gear was because the operators wanted to entice investors |
| 04:29.33 | YoMama | i see a lot of potential...there's just a few things that can be done to make it a world class solution..or maybe it's alreayd there and i jus tknow about it |
| 04:29.46 | konfuzed | we only actually used one piece of cisco gear |
| 04:29.49 | mog_home | you can get rollover yomama |
| 04:29.58 | mog_home | it just costs more, but get two machines |
| 04:30.03 | mog_home | and set it up if one goes down |
| 04:30.06 | mog_home | to roll over |
| 04:30.12 | konfuzed | the traffic manager load balance I forget what its called |
| 04:30.14 | mog_home | asterisk will do 0 for you out of the box |
| 04:30.25 | JonR800 | YoMama: you roll your own. |
| 04:30.33 | Pikoro | is there a 1 second pause for dialing an extension? |
| 04:30.36 | JonR800 | as im finding out :) |
| 04:30.39 | YoMama | mog_work: is there a way to make * servers failover automatically? |
| 04:30.44 | mog_home | yeah |
| 04:30.54 | mog_home | you will lose all calls that were up at that moment |
| 04:31.06 | mog_home | but you will have down time of few if any seconds |
| 04:31.16 | YoMama | well, that's true of Cisco's stuff too |
| 04:31.20 | *** join/#asterisk BladeRunner05 (n=gianni@adsl-ull-110-64.44-151.net24.it) |
| 04:31.22 | mog_home | thats what i thought |
| 04:31.27 | mog_home | but there is some telco grade stuff |
| 04:31.30 | db48x | heh |
| 04:31.31 | mog_home | that no calls drop |
| 04:31.31 | YoMama | if you're registered to server A and it kicks the can...you will lose the call and the phone has to re-register with server B |
| 04:31.31 | mog_home | ever |
| 04:31.37 | mog_home | even if you light it on fire |
| 04:31.40 | mog_home | yeah you can do that |
| 04:32.10 | YoMama | mog_work: yeah..that stuff is that self-healing realtime crap |
| 04:32.13 | YoMama | $$$$$$$ |
| 04:32.18 | mog_home | yup |
| 04:32.28 | YoMama | most businesses don't want that...only telco companies bother with that shit |
| 04:32.49 | YoMama | let's say i wanna set up a 24/7 call center and the main * server kicks the can at 3 am |
| 04:32.59 | YoMama | what if i don't want a call at 3 am to swap a box out? |
| 04:33.03 | konfuzed | YoMama: <-------- wants his cake and pie all at the same time |
| 04:33.04 | YoMama | u can't ALWAYS have someone on site who knows how to do that |
| 04:33.11 | YoMama | konfuzed: shit yeah i do |
| 04:33.11 | mog_home | yeah |
| 04:33.33 | db48x | have the second server watch a heartbeat from the first |
| 04:33.36 | BladeRunner05 | I get error compiling chan_capi 0.3.5 or chan_capi 0.4.0-PRE1 with asterisk 1.2.0 |
| 04:33.50 | db48x | if the heartbeat stops, server b takes over, resets the dns to point to itself, etc |
| 04:33.54 | mog_home | bingo |
| 04:34.00 | mog_home | db48x gets a silver star |
| 04:34.04 | Qwell | heartbeats...meh. two boxes that share memory :p |
| 04:34.10 | db48x | when you bring the first box up the next day, it becomes server b |
| 04:34.20 | konfuzed | YoMama: pay someone else to do it on contract and it wont be your responsibility when it goes awry and you'll have the commercial support level that cisco cant deliver |
| 04:34.23 | db48x | Qwell: well, or it just uses ping |
| 04:34.41 | mog_home | one day qwell |
| 04:34.43 | mog_home | one day |
| 04:35.02 | orlock | YoMama: many businesses _want_ it.. they just dont want to pay :) |
| 04:35.37 | YoMama | mog_work: if two servers are sharing the same realtime database...wouldn't that work too? |
| 04:35.49 | konfuzed | the real question is how to deliver that solution in the most affordable real way |
| 04:35.52 | orlock | but to do it properly... |
| 04:35.52 | mog_home | yeah that is one way to keep things in sync easily |
| 04:35.56 | YoMama | what if you put an IP sprayer in front of the boxes? |
| 04:35.57 | mog_home | its still real affordable |
| 04:35.58 | orlock | redundan switches.. ethernet.. etc |
| 04:36.02 | mog_home | having 2 boxes |
| 04:36.13 | db48x | YoMama: you don't want them to both be operating at once |
| 04:36.23 | db48x | YoMama: because then people will be registered on different servers |
| 04:36.25 | konfuzed | YoMama: get a couple ah sharp guys here to subcontract to and most of the nightmares will go away |
| 04:36.43 | konfuzed | uh not me of course |
| 04:36.50 | YoMama | db48x: so? can't u set it up so if the ext can't be found on one server..it tries the others? |
| 04:37.02 | mog_home | yeah you can do that too |
| 04:37.07 | JonR800 | eh |
| 04:37.13 | db48x | YoMama: I haven't seen anything to that effect |
| 04:37.17 | JonR800 | but don't you run into issues if one goes down? |
| 04:37.24 | YoMama | db48x: i have...IAX2 will let you do that |
| 04:37.33 | db48x | well, then that's cool |
| 04:37.39 | *** join/#asterisk insomni (n=insomni@x1-6-00-e0-18-6f-34-ad.k455.webspeed.dk) |
| 04:37.51 | konfuzed | ~eeks |
| 04:37.52 | jbot | i heard eeks is the Eeks eeks run for the hills IAX2 is here to stay |
| 04:38.11 | archimedes_xyz | ok, trying to upgrade to 1.2 and getting the following error in /var/log/asterisk/messages: "...../chan_zap.so: undefined symmbol: pri_restart" any ideas where I went wrong? |
| 04:38.14 | db48x | YoMama: in that case you only lose the calls that were on that server |
| 04:38.14 | konfuzed | im so pleased that I get to do all IAX2 from the start |
| 04:38.27 | YoMama | db48x: correct |
| 04:38.32 | konfuzed | screw that headache SIP and others |
| 04:38.55 | Pikoro | hmm |
| 04:39.37 | justinu | sip is where it's at |
| 04:39.41 | YoMama | db48x: that way, you could register with either * box and no one would be the wiser |
| 04:40.08 | db48x | well, either way the people who had their call dropped will know something happened |
| 04:40.30 | db48x | but since it should work just fine the next time the dial, it'll just be a glitch |
| 04:40.32 | YoMama | the question is..when one kicks the can..and the phones lose their registration...how will they know to connet to the unbroken box? |
| 04:40.33 | *** join/#asterisk d-tech (n=dtc@node-423a1ebb.cle.onnet.us.uu.net) |
| 04:40.48 | mog_home | well the way you really do it yomama is you have 3 machines |
| 04:40.55 | db48x | YoMama: if the dns has a low ttl, they'll have to look it up again |
| 04:41.02 | mog_home | you have machine one that routes to the two boxes |
| 04:41.09 | db48x | as long as you're not giving out the dead ip, they'll get a box that's up |
| 04:41.10 | mog_home | and it routes to the machine that is alive |
| 04:41.13 | mog_home | when it goes down |
| 04:41.13 | YoMama | db48x: i dunno...that doesn't seem entirely elegant |
| 04:41.15 | mog_home | route to the other |
| 04:41.16 | JonR800 | im waiting for the required ser recommendation. |
| 04:41.20 | mog_home | its pretty clean |
| 04:41.27 | orlock | There is also drdb |
| 04:41.37 | orlock | and the linux-HA stuff |
| 04:41.39 | YoMama | mog_home: three * boxes? |
| 04:41.43 | mog_home | no 2 |
| 04:41.47 | mog_home | the other one is just a router |
| 04:41.56 | orlock | both systems have their own IP, pluss a "fake" ip, so whichever system is up as the fake IP |
| 04:41.58 | YoMama | mog_home: right..or get one of those webfarm IP sprayer boxes |
| 04:42.01 | archimedes_xyz | what if the router goes down? |
| 04:42.14 | mog_home | if the router goes down you have big problems |
| 04:42.20 | archimedes_xyz | :) |
| 04:42.25 | mog_home | what if electricity starts going backwards |
| 04:42.29 | JonR800 | you get out of bed at 3am and drive to the call center. |
| 04:42.29 | mog_home | or gravity stops... |
| 04:42.40 | mog_home | in the end |
| 04:42.41 | orlock | archimedes_xyz: the big hassle is that if you only have one incoming physical connection, you cannot easily run it inot two routers |
| 04:42.46 | Qwell | if gravity stops, I have a backup plan |
| 04:42.48 | orlock | having redundant routers isnt the biggy |
| 04:42.50 | mog_home | there is always gonna be some single point of faliure |
| 04:42.52 | mog_home | always |
| 04:42.52 | Qwell | don't ask |
| 04:42.54 | orlock | its two connections to plug it into |
| 04:42.54 | db48x | what if protons start decaying really fast |
| 04:43.00 | mog_home | i know what it is qwell |
| 04:43.04 | Qwell | do not |
| 04:43.08 | mog_home | i saw real geniuses |
| 04:43.14 | archimedes_xyz | we setup a network for a project once for a group of banks, and we actually setup 2 Firewalls and redudant servers using the heartbeat tech |
| 04:43.15 | Qwell | never heard of it |
| 04:43.23 | mog_home | oh the proposed a plan for it |
| 04:43.26 | mog_home | nudity |
| 04:43.27 | Pikoro | Set(TIMEOUT(response)=1) |
| 04:43.27 | Qwell | haha...man...at my work |
| 04:43.29 | Pikoro | that fixed it |
| 04:43.41 | orlock | mog_home: i know, got fibre here AND an E1 with RSTP running over both of them, so we have ~3 seconds of fibre downtime till the RSTP brings up the E1 |
| 04:43.46 | Qwell | one day, our net connectivity just died...corp wide |
| 04:43.56 | Qwell | or, a large part of it anyhow |
| 04:43.59 | mog_home | lol |
| 04:44.09 | Qwell | well...they had two firewalls |
| 04:44.31 | Qwell | both were in the same datacenter, but they had hot spares for each, dual redundant power, all that kind of good shit |
| 04:44.44 | Qwell | it seems...the dipshit contractor that connected it... |
| 04:44.46 | archimedes_xyz | yup.. |
| 04:44.52 | orlock | Qwell: ahhhhhh |
| 04:44.55 | Qwell | put all 12 cables...onto the same power line |
| 04:44.58 | orlock | Qwell: one power source? |
| 04:44.59 | orlock | hahahah |
| 04:45.00 | archimedes_xyz | haha |
| 04:45.11 | archimedes_xyz | nice... slick... |
| 04:45.26 | orlock | we have UPS and generator backed lines |
| 04:45.31 | konfuzed | thats probably what his work order said |
| 04:45.36 | orlock | so one cable on UPS, one on line with generator |
| 04:45.42 | JonR800 | im sure that was a fun outage report to write. |
| 04:45.42 | YoMama | orlock: drdb is block-level replication |
| 04:45.46 | archimedes_xyz | did it go down at 11:00 each night? |
| 04:45.50 | orlock | YoMama: yup |
| 04:45.57 | distortion | about as good as our electician contractor plug our datacenter feed into commercial power after we had just spent 80k on a generator :) |
| 04:46.01 | YoMama | orlock: this would work..except failover from node1 to node2 would have to be at the app level..not the system level |
| 04:46.10 | YoMama | orlock: atleast..as far as * is concerned |
| 04:46.30 | orlock | YoMama: yeah, having things at the app level fail over and transparently keep working is a biggie |
| 04:46.36 | YoMama | orlock: plus the second box would be pure standby...u couldn't register to it |
| 04:46.46 | orlock | yeah |
| 04:47.02 | YoMama | orlock: if * died on the first box...drdb would still think node1 is ok because it'd still have heartbeat |
| 04:47.10 | YoMama | they need to develop app-side plugins for this shit |
| 04:47.18 | orlock | YoMama: yes, but should * ever die? |
| 04:47.29 | orlock | YoMama: i use nagios as well for monitoring that sort of stuff |
| 04:47.42 | mog_home | * will die |
| 04:47.46 | mog_home | its a fact of life still |
| 04:47.51 | mog_home | its only 5 years old |
| 04:47.52 | YoMama | orlock: sure..what if the HD started going bad..and it corrupted the bin on the box so it keptcoring |
| 04:47.54 | mog_home | and growing fast |
| 04:47.54 | orlock | YoMama: or you have the whole STONITH setup, where if one system notices that even a service isnt running right on the other.. BLAM.. off with its power! |
| 04:47.59 | mog_home | but it still will die |
| 04:48.25 | orlock | i've never used * though |
| 04:48.35 | orlock | installed it twice, really need to get my head around it |
| 04:49.05 | YoMama | there's gotta be a more elegant solution...the DNS TTL round-robin solution is a decent one |
| 04:50.23 | orlock | hmm.. |
| 04:50.23 | orlock | nahh |
| 04:50.23 | archimedes_xyz | ok, trying to upgrade to 1.2 and getting the following error in /var/log/asterisk/messages: "...../chan_zap.so: undefined symmbol: pri_restart" any ideas where I went wrong? |
| 04:50.23 | orlock | thats like.. hosting your website on a dyndns.org account with a dynamic IP :) |
| 04:50.23 | archimedes_xyz | hehe... |
| 04:50.23 | konfuzed | what sort of load does it take to make asterisk choke and die? |
| 04:50.23 | orlock | YoMama: something has to check and modify the DS table |
| 04:50.23 | orlock | YoMama: ever used Nagios? |
| 04:50.23 | YoMama | orlock: no...because you're gonna be doing it internally to your company so u don't have to worry about propogation...those DNS entires are only used locally |
| 04:50.30 | orlock | ahh yes |
| 04:50.30 | YoMama | orlock: haven't used it..heard of it |
| 04:50.38 | orlock | YoMama: good, worth looking at |
| 04:50.52 | orlock | YoMama: generates a network map, has a pile of plugins, etc |
| 04:50.56 | orlock | kickass network monitor |
| 04:51.24 | konfuzed | nadios is my choice aswell for network monitoring |
| 04:51.32 | konfuzed | haha nagios |
| 04:51.32 | diamon | konfuzed: Depends on the hardware under it. I've heard of a 500 simultaneous users 8-way Xeon server before. |
| 04:51.39 | YoMama | orlock: OOOOH...I used to use it back when it was called NetSaint :) |
| 04:51.51 | konfuzed | yeah |
| 04:52.08 | YoMama | orlock: yeah..it's pretty sweet...requires an assload of tweaking though |
| 04:52.24 | JonR800 | konfuzed: also depends what those users are doing. :) |
| 04:52.31 | orlock | ahh |
| 04:52.32 | orlock | heh |
| 04:52.34 | orlock | yeah, it does |
| 04:52.36 | konfuzed | well to handle up to 75 simul calls for 200 phones |
| 04:52.55 | diamon | JonR: That too, and the codec used as well I suppose. |
| 04:53.00 | YoMama | orlock: it's easy enough monitoring when node1 goes to shit..the question is...is there a way to make it failover without paying your sysadmin overtime and deal with frustrated people while he drives 30 minutes into work to switch the box over |
| 04:53.17 | diamon | Number of phones doesn't matter, max simultaneous calls does. |
| 04:53.20 | konfuzed | say 100 units averaging 2 part time phone users |
| 04:53.21 | orlock | YoMama: yeah. shell script on one box watching the other |
| 04:53.30 | konfuzed | not much in the day but mostly in the evening |
| 04:53.40 | orlock | but that comes down to a heartbeat i guess |
| 04:53.54 | diamon | konfuzed: That'd probably need a fairly hefty machine and a good bit of RAM. |
| 04:54.08 | JonR800 | konfuzed: http://www.voip-info.org/tiki-index.php?page=Asterisk+dimensioning |
| 04:54.16 | konfuzed | i cant really imagine toomany occurrences of plus 75 simultaneous |
| 04:54.29 | *** join/#asterisk alephcom (n=Miranda@207.34.97.130) |
| 04:54.44 | konfuzed | I figure I would have to put 2 to 3 grand on a box to handle it |
| 04:54.57 | diamon | JonR: Thanks, I was trying to remember the wording for that... |
| 04:55.06 | konfuzed | for 100 residential apartments that is |
| 04:55.13 | JonR800 | np |
| 04:55.37 | diamon | konfuzed: Sounds possible. Just remember a good UPS for it, and an EXCELLENT surge surpressor if it's to be installed local to the complex. |
| 04:56.07 | konfuzed | hhhmmm has there been any solid comparison of asterisk in amd64 vs i386 |
| 04:56.24 | diamon | konfuzed: And be sure about the laws regarding e911 service or the equivalent in your area. These folks would be dependent on that phone system for everything, the way you're describing it. |
| 04:56.35 | konfuzed | yeah I would put an * box on the property |
| 04:56.57 | konfuzed | I would alwys do failover PSTN lines |
| 04:57.21 | diamon | I favor APC UPS's myself, but any UPS beats nothing, in most cases. |
| 04:57.33 | diamon | konfuzed: So long as you have some way to do that. |
| 04:57.51 | konfuzed | and calls piped though my dsl provider / did provider all have back ended 911 service |
| 04:58.17 | konfuzed | oh sure I can have almost any service provisioned |
| 04:58.24 | konfuzed | north america wide even |
| 04:58.32 | konfuzed | I know how to make it happen but |
| 04:58.45 | konfuzed | Im having a hard time selling the concept across north america |
| 04:58.46 | konfuzed | ;^) |
| 04:59.02 | *** join/#asterisk YoMama (n=tchen@pcp02689850pcs.roylok01.mi.comcast.net) |
| 04:59.03 | YoMama | dammit |
| 05:00.03 | mog_home | ? |
| 05:00.04 | konfuzed | if a telco or ISP ( and soon to be TV) provides a service I can get it from them at wholesale white branded and bill the end user directly |
| 05:00.20 | diamon | konfuzed: Get one installed as a proof-of-concept. I find folks listen better when you show them it works. |
| 05:00.26 | konfuzed | uh only worth while on a larger scale though |
| 05:00.30 | YoMama | mog_home: my wireless at home sucks...it's a super duper old d-link access point that's barely 802.11b compliant |
| 05:00.44 | diamon | YoMama: Time for an upgrade! |
| 05:00.54 | konfuzed | I had one running in a shared office and they were to stupid to use it |
| 05:01.04 | YoMama | diamon: haha..yeah i know...but i rarely use it..my house is cabled |
| 05:01.09 | BladeRunner05 | How can I record an incoming or outgoing call ? |
| 05:01.11 | diamon | mog_home: That makes two of us. I like the Linksys WRT54G/GS myself for WiFi 11g. |
| 05:01.18 | mog_home | yeah i love those things |
| 05:01.19 | YoMama | BladeRunner05: CDR..it's built in |
| 05:01.28 | mog_home | is only good thing linksys makes |
| 05:01.37 | mog_home | that and that badass phone they just came out with |
| 05:01.46 | BladeRunner05 | Yomama: wich is the proper syntax ? |
| 05:01.46 | diamon | mog: Badass phone? |
| 05:02.01 | *** join/#asterisk BhaalWK (i=bhaal@freenode/staff/bhaal) |
| 05:02.03 | mog_home | they have a phone |
| 05:02.03 | Qwell | probably a rebranded sipura :p |
| 05:02.07 | mog_home | that looks like 7940 |
| 05:02.11 | konfuzed | its only gonna be worth while on a commercial agreement basis for multiple users |
| 05:02.12 | mog_home | its hott |
| 05:02.18 | YoMama | mog_home: the only reason i ask is i see huge potential for * for call center applications..i know people are already using it..but what about call centers with 500 people? u can't have 500 people cease working 'cause the * server burped |
| 05:02.18 | mog_home | whatever qwell |
| 05:02.20 | mog_home | it looks hott |
| 05:02.24 | Qwell | haven't seen it. |
| 05:02.34 | YoMama | BladeRunner05: look in /var/log/asterisk i think |
| 05:02.34 | mog_home | they had it astericon |
| 05:02.38 | YoMama | BladeRunner05: it's a freakin' CSV file |
| 05:02.47 | mog_home | agreed yomama |
| 05:02.56 | mog_home | but we arent there yet |
| 05:03.05 | BladeRunner05 | Yomama:ok I do |
| 05:03.10 | mog_home | if you need hardware failover |
| 05:03.15 | mog_home | we just dont have it yet |
| 05:03.16 | YoMama | mog_home: u know what kinda sick money people pay for call center software? ICM will make your nose bleed if i gave you starting prices |
| 05:03.17 | diamon | YoMama: That's why you use good-quality hardware with redundant features, so the server doesn't burp... |
| 05:03.25 | mog_home | ib et |
| 05:03.28 | konfuzed | appropriate use can be so hard to pin down |
| 05:03.31 | mog_home | we are almost there |
| 05:03.31 | *** join/#asterisk BhaalWK (i=bhaal@freenode/staff/bhaal) |
| 05:03.36 | mog_home | but we dont have total hw failover |
| 05:03.39 | Qwell | eh...realtime queues |
| 05:03.46 | YoMama | mog_home: and there's like 10 people in the US who know how to set up ICM and manage it properly |
| 05:03.59 | diamon | Konfuzed: That's far too true. |
| 05:04.03 | mog_home | and i am 8 of them |
| 05:04.04 | mog_home | mwahahaha |
| 05:04.07 | mog_home | ^_^ |
| 05:04.28 | YoMama | mog_home: the fact that you can dynamically allocate extensions and stuff like that means u guys are getting close |
| 05:04.40 | mog_home | oh we have been able to do that for a awhile |
| 05:04.51 | BladeRunner05 | Yomama: I need to record the calls not as a log information but their contents |
| 05:04.51 | YoMama | at some point...* development will need to shift more toward bulletproofing the app over adding new bells and whistles |
| 05:04.59 | mog_home | its already doing that |
| 05:05.01 | YoMama | i think the thing can already do more than most voip phone systems |
| 05:05.09 | Qwell | most? psh |
| 05:05.09 | mog_home | but few people have money to pay for that kinda dev |
| 05:05.21 | YoMama | BladeRunner05: look up the Monitor() command |
| 05:05.21 | diamon | YoMama: Yeah, on the mailing list you can see folks shifting focus to making it stronger, not prettier recently. |
| 05:05.22 | mog_home | we can do a lot qwell |
| 05:05.28 | YoMama | BladeRunner05: don't get arrested |
| 05:05.30 | mog_home | but there are some people that have features we dont |
| 05:05.33 | mog_home | and it pisses me off |
| 05:05.39 | konfuzed | YoMama: should * develop its own kernel and OS too |
| 05:05.42 | mog_home | is why i did jabber |
| 05:05.44 | mog_home | lol |
| 05:05.45 | mog_home | no |
| 05:05.50 | YoMama | konfuzed: no |
| 05:05.53 | mog_home | linux is good enough for asterisk |
| 05:05.55 | mog_home | and vice versa |
| 05:05.58 | BladeRunner05 | Yomama: ;-) |
| 05:06.00 | diamon | konfuzed: Phbbtp! :P |
| 05:06.09 | konfuzed | how much of this fail overis the hardware kernel and os responsibility |
| 05:06.16 | YoMama | konfuzed: they settle on some "certified" kernel revisions and distributions..and that'll be fine |
| 05:06.50 | YoMama | mog_home: i did a REALLY extensive VoIP comparison between Cisco, Nortel, Avaya, and * a year ago |
| 05:06.58 | diamon | So, anyone suggest a fairly decent softphone I can use to test my VoIP fun stuff? Win or linux, either or both. |
| 05:06.59 | mog_home | howd it go? |
| 05:07.15 | YoMama | mog_home: we picked Cisco because of the failover capabilities |
| 05:07.25 | mog_home | should have called me ^_^ |
| 05:07.30 | mog_home | but yeah i understand |
| 05:07.31 | konfuzed | the OS and Kernel have to master the hardware failover parts mostly. Asterisk can only be expected to fail over to another asterisk presuming a clean environment |
| 05:07.34 | diamon | I'd bet Nortel did very well, then cisco for toughness but less features, and on down. |
| 05:07.39 | YoMama | mog_home: and the fact that you can go hire a cisco VoIP consultant..u don't see a lot of people adveritisng themselves as * geeks |
| 05:07.49 | mog_home | well yeah you do |
| 05:07.56 | mog_home | but we work at digium ^_^ |
| 05:08.02 | YoMama | diamon: believe it or not...cisco was cheaper than all except * |
| 05:08.15 | mog_home | yeah cisco is cheaper than a shoretel |
| 05:08.20 | YoMama | Avaya was a ripoff |
| 05:08.24 | YoMama | that was the worst |
| 05:08.27 | mog_home | yeah they are bad |
| 05:08.28 | diamon | YoMama: Really? You're right, I wouldn't have believed that, though I'm sure Nortel was priciest. |
| 05:09.15 | konfuzed | see here in canada most business have some kind of nortel inplace that was over sold |
| 05:09.26 | YoMama | Avaya has two main VoIP lines...they got their SOHO/branch stuff...which gives u lots of bang for the buck...but it's not centrally manageable and i wasn't about to make some poor admin have to log into 30 different of those boxes |
| 05:09.27 | konfuzed | its easy to convert nortel customers |
| 05:09.37 | YoMama | the other line starts at like $200,000 |
| 05:09.53 | diamon | YoMama: Wow, 30 diff servers/PBX's, that'd be unkind for sure... |
| 05:09.55 | YoMama | centrally managed...but the interface to manage it all is still very CLI-driven...it's ugly |
| 05:09.56 | konfuzed | but im not in sales |
| 05:10.06 | YoMama | diamon: one for each branch office |
| 05:10.48 | YoMama | Cisco had the best of all of those worlds with their web interface and centralized management...plus the fact that the routers act as backup VoIP gateways was slick... |
| 05:11.06 | konfuzed | YoMama: are trying to go with an asterisk only solution for your whole setup |
| 05:11.25 | diamon | Omigawd, AOL has a VoIP offering? I can't even begin to imagine... |
| 05:11.39 | YoMama | konfuzed: well, i left my old job 'cause i couldn't take it there anymore..but i wanna evaluate it for the next company i end up at |
| 05:11.56 | *** join/#asterisk tartar (n=tartar@CPE0004e27b716e-CM014370001917.cpe.net.cable.rogers.com) |
| 05:12.16 | konfuzed | so youre talking about a development project then |
| 05:12.30 | YoMama | that's another thing too..that cisco's got an edge on...we could buy one Cisco 1760 for a branch office and it'd be the DS1 router, the voice gateway, and the failover CM all in one box |
| 05:12.31 | konfuzed | see context always throws me off |
| 05:12.41 | konfuzed | rather the lack of context is what throws me off |
| 05:13.21 | Corydon76-home | Uck, -fomit-frame-pointer is enabled under 'make dont-optimize' |
| 05:13.24 | YoMama | konfuzed: i'm just looking at it from a high-level..if * wants true adoption..u gotta get business-minded CIOs to buy into it..and given I've been a suit-wearing management asshole for the past five years..i have been programmed to think that way |
| 05:13.52 | YoMama | lucky for me i still know how to log into a *nix box :) |
| 05:14.05 | konfuzed | ok so you dont have an immediate call center on hand that you need to complete an implemented solution for by monday |
| 05:14.15 | YoMama | konfuzed: hell no |
| 05:14.32 | konfuzed | well thats entirely different |
| 05:14.33 | YoMama | but most of the companies i'd go work for probably run some kinda call center or another |
| 05:14.40 | mog_home | well i have to head out for a bit |
| 05:14.48 | mog_home | be back later |
| 05:14.49 | diamon | YoMama: And yet you're not talking about win-win synergy and whatnot... Maybe there *is* hope for managers! |
| 05:14.49 | *** join/#asterisk BhaalWK (i=bhaal@freenode/staff/bhaal) |
| 05:14.50 | YoMama | almost every single customer i've ever worked with that had a call center hated the software |
| 05:15.07 | YoMama | diamon: i'm not your typical suit..i actually understand technology |
| 05:15.10 | YoMama | :) |
| 05:15.29 | diamon | YoMama: Ah, the exception that proves the rule. Ok, my worldview is safe then. |
| 05:15.32 | konfuzed | YoMama: thats why you hire someone to deal withthe software and no body else has to think about it |
| 05:15.52 | YoMama | konfuzed: yeah...but those companies RAPE you man |
| 05:16.04 | konfuzed | you mean cisco? |
| 05:16.08 | konfuzed | or nortel? |
| 05:16.10 | YoMama | all of them |
| 05:16.13 | konfuzed | or lucent? |
| 05:16.15 | YoMama | especially call center software |
| 05:16.17 | YoMama | ALL of them |
| 05:16.23 | konfuzed | great |
| 05:16.33 | konfuzed | so the only place you dont get raped is with asterisk then |
| 05:16.34 | YoMama | as soon as u tell them u want a call center...you're looking at spending $200-300k on up |
| 05:16.48 | konfuzed | why tell them its a call center |
| 05:16.54 | diamon | So, does Packet8 work with Asterisk or no? |
| 05:17.00 | konfuzed | just buy their gear and over use it |
| 05:17.01 | YoMama | 'cause then u don't get all the complex call routing features |
| 05:17.15 | YoMama | u don't get IVR capability |
| 05:17.19 | YoMama | it's crippled |
| 05:17.31 | YoMama | there's 100 billion things u don't get if u don't tell them call center |
| 05:17.44 | diamon | YoMama: Does Asterisk do IVR now? I haven't looked in a long time... |
| 05:17.57 | konfuzed | so then what it takes to satisfy your PERCIEVED needs is $200+ solutions |
| 05:18.03 | YoMama | cisco specifically has a pieces of software that overlay their CM to give it call center capabilities |
| 05:18.25 | YoMama | diamon: it plugs into a piece of software that does..and according to reports..it really works..just eats a lot of processor |
| 05:18.43 | konfuzed | s/$200+/$200g+/ |
| 05:18.44 | YoMama | but think about it..who gives a shit? if Nortel wants $300,000 to set you up a 200 seat call center |
| 05:18.57 | YoMama | u could probably do the same call center using * for $30,000 |
| 05:19.10 | konfuzed | and spedn the money on geeks to do magic |
| 05:19.18 | YoMama | well, lemme think...200 phones * $100/piece..that's $20,000 right there |
| 05:19.38 | YoMama | then u gotta buy probably four or five servers |
| 05:19.55 | konfuzed | well first you nead a real IP network and then |
| 05:19.59 | YoMama | but u could have one doing nothing but auto attendant and call queuing and another doing nothing but IVR |
| 05:20.01 | konfuzed | add a phone silution on top |
| 05:20.15 | diamon | YoMama: I wanted to use IVR just for the crap of it at my house, see if I could eventually use it for something real later. :) |
| 05:20.17 | YoMama | konfuzed: eh...that gear is cheap in comparison |
| 05:20.33 | konfuzed | exactly |
| 05:20.48 | diamon | Any VoIP providers allow ipv6 that anyone can think of? |
| 05:20.53 | konfuzed | why run a whole second network when your existing network will do fine |
| 05:21.00 | diamon | Just a 'for the hell of it' question, really. |
| 05:21.04 | YoMama | diamon: hehe..in my next house..i'm gonna build it so it's got X10-capable switches..and set it up so i can call in...tone in a password..and then issue verbal commands |
| 05:21.15 | YoMama | how phat would that be? |
| 05:21.22 | diamon | YoMama: Yeah, that'd be spiffy! |
| 05:21.26 | YoMama | "Lights On" |
| 05:21.28 | YoMama | "What time?" |
| 05:21.35 | YoMama | "6 AM" |
| 05:21.38 | YoMama | haha...or even better |
| 05:21.46 | YoMama | "AC on" |
| 05:21.50 | diamon | YoMama: Put a mike in the house too, do the Star Trek thing. :) |
| 05:21.57 | YoMama | that way..the house will be nice and cool when u get home |
| 05:22.14 | YoMama | eh..that's too big brother |
| 05:22.18 | diamon | YoMama: Too many computers here and it's too hot in FL to not have the AC on all day. :P |
| 05:22.29 | YoMama | mount a weather-hardened webcam at the front door and the backyard though |
| 05:22.34 | diamon | YoMama: Nah, it'd be my own little brother. :P |
| 05:23.00 | YoMama | i might be a suit..but i'm still a geek at heart :) |
| 05:23.29 | YoMama | put it this way: i have a relay rack mounted in my basement |
| 05:24.18 | *** join/#asterisk digime (n=digime@ip68-111-235-172.sd.sd.cox.net) |
| 05:24.36 | *** join/#asterisk SplasPood (i=nobody@paravolve.net) |
| 05:24.43 | *** part/#asterisk digime (n=digime@ip68-111-235-172.sd.sd.cox.net) |
| 05:24.49 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 05:24.49 | *** mode/#asterisk [+o denon] by ChanServ |
| 05:24.57 | Damin | YoMama: That's nothing. I have a 7 foot tall Telco cabinet w/ home-run wiring to 24 jacks scattered throughout the house and a gigabit switch at the core! ;) |
| 05:25.00 | diamon | And I thought I was bad with my UPS control board... |
| 05:25.12 | YoMama | Damin: haha..that's hardcore |
| 05:25.25 | Damin | YoMama: I also have an ASN and BGP setup just for my house! ;) |
| 05:25.30 | YoMama | Damin: i live in the city..my house isn't big enough for 24 jacks |
| 05:25.51 | YoMama | Damin: u have an AS # for yourself? |
| 05:25.56 | YoMama | umm..that's beyond hardcore |
| 05:26.03 | YoMama | Damin: private or public AS? |
| 05:26.08 | Damin | YoMama: Public.. |
| 05:26.25 | YoMama | so..you've got true multi-homed internet access at home? how many providers u got running in there? |
| 05:26.26 | Damin | YoMama: Over the years, I've bought several ISPs! :) |
| 05:26.42 | YoMama | Damin: oh..so you run your own ISP out of your basement |
| 05:26.53 | YoMama | well, that's different..i ain't makin' any money off the crap whirring in my basement |
| 05:27.43 | Damin | YoMama: I have Cox Cable and SBC DSL. I establish IPSEC tunnels from a central core router in my house accross both Cable and DSL, and then peer directly w/ our Core router downtown. ;) |
| 05:28.01 | YoMama | ah |
| 05:28.16 | YoMama | so u got your core router serving as peer neighbors |
| 05:28.21 | Damin | diamon: I don't make any money off the crap in my basement either. ;) I'm just a geek. |
| 05:28.29 | Damin | YoMama: Yep. |
| 05:28.39 | diamon | Damin: Yeah, but I don't have a basement to store my toys in. :) |
| 05:29.15 | Damin | YoMama: Since everything is aggregated up through our ASN 6954, it is just for redundancy purposes.. :) |
| 05:29.21 | Damin | Alright.. I'm going to bed.. |
| 05:29.25 | YoMama | Damin: but when you're connected to your home network..and surfing the web..does it go over the IPSec tunnel or straight out? |
| 05:29.30 | diamon | So, opinion time. Is Slackware a decent distro to use for Asterisk? Is anything a 'suggested' distro? |
| 05:29.38 | Damin | YoMama: over the tunnel. |
| 05:30.03 | Damin | YoMama: Unless I specifically change my workstations to point to different gateways.. |
| 05:30.10 | YoMama | Damin: doesn't that slow it down a little? IPSec adds a little overhead..u end up getting some ip fragmentation and spending a little time reassmembling packets |
| 05:30.51 | diamon | Oooh, that reminded me, I still need to set my network to a 9k MTU and test that... Foo. |
| 05:30.54 | Damin | YoMama: I've got 6 meg x 1 meg DSL and 5 meg x 1 meg cable service. ;) |
| 05:31.01 | YoMama | Damin: and aren't u being a bit greedy by peering with the core router rather than setting up an edge router? IPSec eats processor |
| 05:31.38 | Damin | YoMama: The router doesn't actually do the IPSec termination.. that's handled by an Astaro Linux box.. |
| 05:31.49 | YoMama | aaah |
| 05:32.07 | YoMama | Damin: well, this is just my two cents..but rather than piping all of your outbound traffic to your work... |
| 05:32.17 | Damin | YoMama: And from my core at work, I'm not sending full routes to my basement. ;) Just default routes. |
| 05:32.43 | YoMama | why not set it up so any box where you're just doing desktop/workstation stuff...uses a different VLAN which doesn't use your tunnels |
| 05:33.00 | Damin | YoMama: That is a different approach to it.. :) |
| 05:33.33 | Damin | YoMama: Basically I'm willing to sacrafice some bandwith for high availability and redundancy. |
| 05:33.38 | YoMama | Damin: if u got any boxes you're running at your house with public IP addresses..obvoiusly those need to go over your IPSec tunnel multi-homed BGP and all that good stuff |
| 05:33.56 | Damin | YoMama: Over both the DSL and the Cable, I've got less than 10 MS to my downtown core.. |
| 05:34.07 | Damin | YoMama: So it's not an issue. ;) |
| 05:34.08 | YoMama | huh...not bad |
| 05:34.12 | YoMama | sheesh |
| 05:34.19 | Damin | Alright.. |
| 05:34.24 | Damin | I really do have to go to sleep! |
| 05:34.27 | Damin | See ya later.. |
| 05:34.28 | YoMama | goodnight |
| 05:34.59 | *** join/#asterisk oldbrat (n=daiviet@203.210.217.15) |
| 05:43.24 | *** join/#asterisk Insanity5 (n=feaw@68.111.5.23) |
| 05:43.41 | Insanity5 | How cna I place a three way call with a sipura adapter? |
| 05:43.45 | Insanity5 | and asterisk? |
| 05:43.49 | Pikoro | how do i update teh firmware on a gpx-2000? |
| 05:43.53 | Insanity5 | I don' |
| 05:44.04 | Insanity5 | I don't see it on any of the "vertical service activation codes" |
| 05:44.15 | *** join/#asterisk simon_ca (i=sedgett@S01060004e23873e1.vc.shawcable.net) |
| 05:44.27 | Insanity5 | Anyone? please? I have someone on hold :) |
| 05:45.26 | Pikoro | sorry Insanity5.. no idea |
| 05:45.44 | Insanity5 | I see every last damn code but three way |
| 05:45.45 | Insanity5 | arg |
| 05:45.48 | simon_ca | does the digium x100p use chan_modem? I see from 1.2 UPGRADE.txt that chan_modem disappears in 1.4.... |
| 05:45.51 | Insanity5 | maybe it's just flash-dial-flash |
| 05:46.14 | Qwell | simon_ca: no |
| 05:46.17 | oldbrat | simon_ca:no |
| 05:46.43 | simon_ca | thanks |
| 05:47.17 | Insanity5 | It it was flash dial flash :) |
| 05:47.33 | oldbrat | just run into one res_perl problem. try to run the WebServer.pm and it crashed * immediately. Has anyone have same problem? |
| 05:48.21 | *** join/#asterisk BhaalWK (i=bhaal@freenode/staff/bhaal) |
| 05:49.19 | oldbrat | i see that the bug is in perl_thread function |
| 05:51.04 | oldbrat | or run inside CLI : perl thread yourfunc |
| 05:51.15 | oldbrat | and see you * crash |
| 05:51.51 | mover | any asterisk cluster ppl here? |
| 05:52.19 | oldbrat | mover: what u want to do? |
| 05:53.04 | oldbrat | mover: i have about 20 boxes running |
| 05:53.30 | *** join/#asterisk BhaalWK (i=bhaal@freenode/staff/bhaal) |
| 05:53.36 | oldbrat | share same db, call routing rules |
| 05:53.53 | mover | i have 2 * setup and on the same realtime db. if i call a phone that is regisered on another the box find the user in realtime db but didnt add the called phone correct in the ast memory |
| 05:54.32 | mover | host and port will not be set so no route an build |
| 05:54.51 | mover | what switches i need to set? :-) |
| 05:55.07 | mover | rtcluser=yes or something? |
| 05:55.11 | oldbrat | mover: i don't use realtime, just flat files and some scripts |
| 05:55.14 | mover | i user on booth 1.2.0 |
| 05:55.37 | mover | yes but its the same with rtcache on |
| 05:55.39 | oldbrat | try to move to realtime |
| 05:55.51 | mover | i am on realtime :-) |
| 05:56.11 | mover | a box standalone work fine |
| 05:56.12 | oldbrat | upgrade 20boxes require a lot of time |
| 05:56.42 | mishehu | hmmm to update my kernel to 2.6.14.2 or not... that is the question at hand... |
| 05:58.07 | mover | oldbrat : what happen is a phone on box a called phone on box b box a get ipadd and port if it set from rtdb and add it in the SIP/registry and the bridgr the call |
| 05:58.11 | oldbrat | mover: in my setup |
| 05:58.12 | Pikoro | any idea how to upgrade one of these grandstream phones' firmware |
| 05:58.39 | oldbrat | mover: i separates user on each box |
| 05:58.45 | Pikoro | nevermind |
| 05:59.13 | oldbrat | mover: some thing like exten=1XXX,1,Dial(${EXTEN}@box1) |
| 05:59.30 | mover | all work the phone on box b will be shown in sip show peers on box a but have not set the ipaddr and port |
| 06:00.16 | mover | oldbrat thats the old way... work for me to but its not generic and have many admin overhead |
| 06:00.51 | oldbrat | mover: Cisco use same tactic, right? called directory number or something |
| 06:00.55 | mover | i guess you can gimme a hint where the damn ipaddr will be set in SIP/registry |
| 06:01.46 | mover | oldbrat i dunno but its the tactic i will use to build a hughe network of * boxes |
| 06:02.19 | oldbrat | mover: and i have 2K users right now :) |
| 06:03.40 | oldbrat | mover: regarding ipaddr, is sip friend and host=dynamic enough? |
| 06:03.41 | mover | thats nice |
| 06:04.51 | mover | oldbrat no you need to read the issue 4832 |
| 06:06.11 | oldbrat | mover: thanks, I do not use rt, so missed it |
| 06:07.46 | mover | its a real great idea |
| 06:07.52 | oldbrat | mover: that tactics not works for me |
| 06:08.10 | oldbrat | mover: cause sharing 1 central DB is impossible for me |
| 06:08.31 | drumkilla | *cough* DUNDi *cough* |
| 06:09.21 | mover | yea it have any traps like the voicemail if its stored on hdd on each box |
| 06:10.12 | mover | but voicemail is now able to put the messages into the db that can solve this issue |
| 06:10.50 | oldbrat | mover: haha, we have different requirements |
| 06:11.10 | oldbrat | each box in our network is separate, say billing |
| 06:11.25 | oldbrat | and they do share only limited routes with outside box |
| 06:11.37 | mover | yes ok |
| 06:11.59 | oldbrat | each box owner have to pay for thier PSTN routes right? |
| 06:12.34 | mover | the owner from my boxes are always the same:-) |
| 06:12.59 | oldbrat | mover: you are lucky |
| 06:13.29 | oldbrat | mover: i do have to make balances between those box at the end of month. Sig |
| 06:13.50 | oldbrat | but this is common situation here |
| 06:14.05 | mover | the only cause i want to get the service stable |
| 06:14.26 | marc324 | mover-- you use odbcstorage? |
| 06:14.38 | mover | no native mysql |
| 06:14.55 | marc324 | voicemessages are stored in db? |
| 06:15.02 | mover | yes |
| 06:15.24 | mover | on 1.2.0 i read something yesterday |
| 06:15.25 | marc324 | works? |
| 06:15.29 | mover | i dunno |
| 06:16.04 | oldbrat | heh, your db will have to work hard |
| 06:16.09 | mover | i need to get the rt-memory build to work ... than i look to the next issue |
| 06:16.33 | marc324 | oldbrat-- how do you handle voice messages? |
| 06:16.41 | mover | oldbrat yes thats the way why db are existe :-) to work hard ... i iguess |
| 06:17.10 | oldbrat | marc323: just as normal, in files |
| 06:17.41 | oldbrat | marc324: i have a little web frontend to allow local users download/play voicemessages |
| 06:17.57 | marc324 | oldbrat-- how do you access those files from a remote server? |
| 06:18.18 | marc324 | is the web frontend on the same server? |
| 06:18.47 | oldbrat | macr323: between our boxes, we have openvpn running |
| 06:19.17 | oldbrat | macr324: so it's possible for users contact thier own web server |
| 06:19.40 | oldbrat | marc324: yes, on same server |
| 06:21.05 | marc324 | oldbrat-- http server and * voicemail on the same server? |
| 06:21.30 | oldbrat | marc324: yes, any problem? |
| 06:23.31 | *** join/#asterisk santiago (n=santiago@208.195.215.124) |
| 06:23.31 | marc324 | i'm interested in running on accessing (read/write/delete) voice messages from a remote http server |
| 06:23.46 | Qwell | marc324: odbcstorage |
| 06:23.58 | Qwell | playing with the files directly is silly |
| 06:25.17 | oldbrat | Qwell: why? i dont see the reason to put more works on db |
| 06:25.48 | Qwell | thats like... |
| 06:25.51 | marc324 | having http on the same server is more work. |
| 06:25.53 | Qwell | the point...of a database |
| 06:26.19 | oldbrat | marc324: i agree |
| 06:27.18 | oldbrat | macr324: but sometimes u can not say to those stupid pigs in administrations that u need 1 more server |
| 06:27.28 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 06:27.32 | oldbrat | for db, web, whatever |
| 06:28.28 | oldbrat | thanksfully, my config works. And I want to leave them as is |
| 06:28.53 | *** join/#asterisk Jabroni (n=Hercules@red-corp-200.76.249.142.telnor.net) |
| 06:29.35 | *** part/#asterisk oldbrat (n=daiviet@203.210.217.15) |
| 06:29.41 | marc324 | can you set a max voicemessages kb / mailboxuser in odbcstorage? |
| 06:30.32 | rajiv | hmm. tos=0x68 seems to work in sip.conf but not iax.conf. |
| 06:31.02 | Qwell | marc324: odbcstorage doesn't have anything to do with settings |
| 06:31.10 | Qwell | just the storage |
| 06:32.09 | *** join/#asterisk oldbrat (n=daiviet@203.210.217.15) |
| 06:33.54 | Qwell | ^ thats what happens when you have a web voicemail using files |
| 06:34.20 | marc324 | how do you limit the amount of voicemail / mailboxuser? |
| 06:34.20 | *** join/#asterisk newmember (n=newmembe@70.72.189.149) |
| 06:34.33 | Qwell | voicemail.conf |
| 06:35.44 | marc324 | maxmsg in voicemail does max message /context.... |
| 06:35.47 | *** join/#asterisk oldbrat (n=daiviet@203.210.217.15) |
| 06:36.06 | marc324 | i need maxmsg / mailboxuser |
| 06:45.15 | *** join/#asterisk oej (n=Olle@apollo.webway.se) |
| 06:45.41 | *** join/#asterisk _DAW (n=bob@adsl-222-51-184.msy.bellsouth.net) |
| 06:51.22 | *** join/#asterisk newsmafia (n=newsmafi@wsip-68-15-19-142.sd.sd.cox.net) |
| 06:57.13 | *** join/#asterisk santiago (n=santiago@208.195.215.124) |
| 06:58.11 | *** part/#asterisk newsmafia (n=newsmafi@wsip-68-15-19-142.sd.sd.cox.net) |
| 07:01.30 | Qwell | pop quiz! |
| 07:01.36 | Qwell | How is full-duplex capability typically achieved in fiber-optic cable? |
| 07:03.39 | xbmodder | when a sysadmin gets pissed off enough to make it? |
| 07:04.02 | Pikoro | can anyone give me a quick exten string for dialing out 0120X. to a zap trunk? |
| 07:04.33 | Pikoro | exten => _90120x.,4,Dial(ZAP/g0/${EXTEN:1}) ? |
| 07:05.35 | Qwell | Pikoro: cap X |
| 07:05.44 | Pikoro | yah, but otherwise does that look ok? |
| 07:06.02 | Qwell | sure |
| 07:06.12 | Pikoro | ok, i'll give it a shot :D |
| 07:07.21 | YoMama | Qwell: wavelength |
| 07:07.36 | Qwell | YoMama: two fibers :p |
| 07:07.42 | YoMama | haha |
| 07:08.29 | Qwell | marc324: no, maxmsg is per box |
| 07:13.48 | *** join/#asterisk jcath (n=skycat@159.226.21.127) |
| 07:13.57 | jcath | good afternoon, every one:) |
| 07:14.41 | *** join/#asterisk krisguy (n=krisguy@h216-170-039-057.adsl.navix.net) |
| 07:15.31 | oldbrat | has anyone play with astjab? |
| 07:17.43 | YoMama | Qwell: u know why you'd hav some sound quality issues using x-lite? |
| 07:17.51 | jcath | hey, from some articles and in theory, we can take a voice modem(external/internal) as FXO, right? |
| 07:17.55 | YoMama | Qwell: x-lite -> * -> Sip proxy |
| 07:18.14 | YoMama | jcath: not in theory...winmodems can be used as FXO ports |
| 07:18.30 | *** join/#asterisk Chotaire (i=chotaire@chotaire.net) |
| 07:19.03 | Qwell | IF you write a zap driver for it |
| 07:19.07 | Qwell | which isn't an easy task |
| 07:19.12 | jcath | so, I can buy a pci modem adapter to install as a fxo port, right? |
| 07:19.16 | Qwell | no |
| 07:19.18 | Pikoro | yay.. i like asterisk now that it works |
| 07:19.19 | Pikoro | heh |
| 07:20.03 | Chotaire | greetings all.. I have one urgent question... I have one SIP provider who provided me with a SIP box that will do CLIR (when checked) and yes.. it works... however, I dont intend to use that box so I want to do CLIR with asterisk.. is there any common string to use CLIR on outgoing SIP calls? |
| 07:20.14 | Pikoro | jcath: appearantly there are only like 2 specific chipsets of voice modems you can use like that but from what I understand, the quality isn't up to production useage |
| 07:20.15 | Chotaire | I have not found any documentations for CLIR on 1und1.de (that's the SIP provider) |
| 07:21.04 | jcath | but... FXO acts as a voice modem, doesn't it? |
| 07:21.15 | Qwell | not always |
| 07:22.19 | jcath | ... I just want to buy a modem adapter or use my notebook's internal modem to test asterisk or yate |
| 07:22.33 | Qwell | not gonna happen |
| 07:22.45 | Qwell | buy an fxo |
| 07:23.51 | mishehu | gah. codec_gsm.so: undefined symbol: Short_term_analysis_filteringx |
| 07:24.51 | YoMama | Qwell: yeah u do |
| 07:25.26 | jcath | I am in China and someone told me that the fxo adapter from digium is not suitable to use here |
| 07:25.51 | jcath | have to hack the driver to meet the local phone line |
| 07:26.22 | YoMama | given i have no idea what teh specs are on a chinese phone line...good thing i can't answer that question |
| 07:26.31 | oldbrat | jcath: nope, work right away. I have 1 box in Peking |
| 07:26.41 | YoMama | Beijing |
| 07:26.54 | YoMama | the only time u use the word "Peking" is when you're tlaking about the duck :-P |
| 07:26.58 | jcath | oh |
| 07:27.01 | jcath | I am in Beijing |
| 07:27.11 | YoMama | jcath: Peking = Beijing |
| 07:27.14 | oldbrat | jcath: it was a TDM400 thing, not digital |
| 07:27.19 | jcath | I know |
| 07:27.29 | oldbrat | YoMama: thanks. Yes, Beijing |
| 07:27.44 | YoMama | oldbrat: i'm just feeling like a smartass right now |
| 07:27.50 | YoMama | but i wish someone would help me tune my box |
| 07:28.04 | YoMama | haha...that's a statement that you'd wanna hear from a girl too |
| 07:28.08 | jcath | r u in China? oldbrat |
| 07:28.11 | Chotaire | I got it.. SetCallerid,Anonymous did the trick ;) |
| 07:28.18 | Chotaire | nice |
| 07:28.43 | oldbrat | jcath: no. Just have a chance to install a box there. In last year. |
| 07:29.40 | jcath | oh, see. a voip box based on asterisk? |
| 07:30.38 | oldbrat | jcat: yes, for one of our offices abroad |
| 07:30.49 | jcath | good |
| 07:31.41 | *** join/#asterisk lme (n=lme@m136.net85-169-255.noos.fr) |
| 07:31.59 | jcath | fxo adapter is expensive to me :( |
| 07:32.09 | jcath | just want to try voip personally |
| 07:32.15 | *** join/#asterisk implicit (n=implicit@ip68-7-152-192.sd.sd.cox.net) |
| 07:32.35 | implicit | whatats up |
| 07:32.36 | oldbrat | Jcat: here u go: http://www.voip-info.org/wiki/view/X100P+clone |
| 07:32.57 | oldbrat | jcat: but I never use them myself. |
| 07:33.16 | jcath | then what u use? |
| 07:33.50 | jcath | yeah, I read that article yesterday |
| 07:33.53 | oldbrat | jcath: digium things |
| 07:35.07 | jcath | oh, see |
| 07:35.09 | oldbrat | jcath: in the past, others got that working, but not sure with current zaptel drivers. Maybe need a little tweak |
| 07:36.12 | jcath | so, it is a soft modem chipset? |
| 07:36.35 | oldbrat | jcath: X100P use same chipset |
| 07:37.39 | jcath | ok, i will reboot my notebook and enable to modem function to see what modem chip it is |
| 07:37.41 | jcath | :P |
| 07:38.33 | YoMama | blah |
| 07:38.34 | YoMama | b/ey |
| 07:44.46 | *** join/#asterisk jcath (n=skycat@159.226.21.127) |
| 07:47.11 | *** join/#asterisk Jestie (n=chatzill@mtngprs7.mtn.co.za) |
| 07:48.54 | ronaldl79 | Anyone download HEAD last night or this morning? |
| 07:49.18 | jcath | 0000:00:1f.6 Modem: Intel Corp. 82801DB/DBL/DBM (ICH4/ICH4-L/ICH4-M) AC'97 Modem Controller (rev 01) |
| 07:49.23 | jcath | does this work? |
| 07:49.25 | jcath | :P |
| 07:53.45 | *** join/#asterisk pr0 (n=LANmower@ndn-165-129-159.telkomadsl.co.za) |
| 07:53.49 | pr0 | lo all |
| 07:54.06 | pr0 | anyone here from South Africa (have a little location-centric question) |
| 07:54.23 | *** join/#asterisk BhaalWK (i=bhaal@freenode/staff/bhaal) |
| 07:54.24 | Vhata | pr0: me |
| 07:54.41 | pr0 | Vhata, any experience with the tdm400's? |
| 07:54.51 | Vhata | err, no. |
| 07:55.19 | *** join/#asterisk vexorg (n=vexorg@CPE000021ded913-CM001225419164.cpe.net.cable.rogers.com) |
| 07:55.23 | pr0 | I am getting hangup problems, and someone here told me that they sorted it in 1.2.0 (wich was cvs back then)... I loaded 1.2.0 but I think his configs were different than mine you see... |
| 07:58.36 | pr0 | oh well, I guess I tried, I'll ask again in a few days, maybe lucky enough to catch the same guy. |
| 07:58.45 | *** join/#asterisk BhaalWK (i=bhaal@freenode/staff/bhaal) |
| 07:58.47 | Jestie | pr0 i am from SA |
| 07:59.07 | pr0 | ever used the tdm range from digium? |
| 07:59.19 | Jestie | yep |
| 07:59.32 | Jestie | whazzip ? |
| 07:59.40 | MacDeath | Jestie : Hi |
| 07:59.51 | Jestie | hello everyone |
| 08:00.03 | MacDeath | Jestie <--- sa too :P |
| 08:00.10 | Jestie | I must say .. I have not been in IRC channels for zonks ... |
| 08:00.19 | Jestie | Realy impressed with how it grew .. sheeshh |
| 08:00.39 | MacDeath | this is a lot bigger than it was in december last year |
| 08:01.07 | MacDeath | you havent been on irc for ages and you chose to use GPRS? |
| 08:01.08 | MacDeath | lol |
| 08:01.10 | Jestie | hehe ... won;t know .. I think the last time i was in a channel was some 6 years ago |
| 08:01.11 | Jestie | hehe |
| 08:01.23 | Jestie | yeah hell .. |
| 08:01.33 | pr0 | Jestie, dont you wanna send me over the configs for your tdm- cards? I am getting hangup problems here... |
| 08:01.35 | Jestie | Telkom .. failed to install DSL at the new house ... |
| 08:01.40 | pr0 | it doesnt detect hangups |
| 08:01.43 | MacDeath | pr0 : Im also from SA |
| 08:01.43 | Jestie | monopolies ... hate tehm |
| 08:01.54 | pr0 | MacDeath, perhaps you too? |
| 08:02.08 | MacDeath | i use both tdm cards and isdn cards |
| 08:02.13 | pr0 | pastebin or dcc is fine... |
| 08:02.27 | MacDeath | what is it not doing? |
| 08:02.41 | Jestie | pr0 would love to .. but all is at the office .. i'm at home .. R&D ... |
| 08:03.04 | MacDeath | I have a problem with the tdm cards in that calls to outside the telkom network hangups are not detected |
| 08:03.04 | Jestie | i'll see if i can "slip" into a clients box ..and suck it from there ... |
| 08:03.27 | MacDeath | so if a call is made to a cellphone and the person put the call down |
| 08:03.36 | MacDeath | * still thinks the call is connected |
| 08:03.39 | pr0 | MacDeath, any telkom based calls doesnt hang up, I have heard someone got them working on 1.2.0, I just wanna see the configs that gets used that works... |
| 08:03.53 | pr0 | mainly for incomming calls... |
| 08:04.32 | jojo | I would like to get in touch with someone who has experience of large-scale call-recording (in my case it will be approx ~90 calls at peak). Anyone here or who knows someone/some business? |
| 08:04.34 | MozartsGhost | are u talking about SA Telkom ? |
| 08:04.43 | Jestie | pr0 / MacDeath : my appologies .. what hardware you running it on, and how bussy are these servers ? |
| 08:05.12 | pr0 | Jestie, completely not busy, tdm400's |
| 08:05.21 | pr0 | various pc systems. |
| 08:05.42 | MacDeath | Jestie : mine is also a tdm400 |
| 08:05.50 | MozartsGhost | I have a TDM04P running on 4 telkom lines |
| 08:05.57 | FuriousGeorge | whats digiums stance on tdm400p when it comes to having multiple cards? 2 is ok? |
| 08:06.00 | MozartsGhost | a few acctually |
| 08:06.02 | Jestie | jojo .. I know that our partner has done a 60-70 call centre recently .. if the recording excist I will find out .. our rule of thumb is to do the recording on a seperate machine ... |
| 08:06.07 | pr0 | MozartsGhost, got it to detect line hangups? |
| 08:06.09 | MozartsGhost | and HFC cards on a few isdn lines. |
| 08:06.16 | MozartsGhost | pr0, yeah works fine |
| 08:06.36 | pr0 | plz send me your zapata.conf and zaptel.conf from a working one... PLZ!?! |
| 08:06.43 | MozartsGhost | ok, sec. |
| 08:07.18 | MozartsGhost | what modules do u have on your card ? |
| 08:07.43 | pr0 | fxs |
| 08:07.52 | MacDeath | MozartsGhost : I am using a TDM400 |
| 08:08.01 | MacDeath | with 4 x FXo |
| 08:08.04 | MacDeath | mostly works |
| 08:08.07 | MozartsGhost | zok |
| 08:08.09 | pr0 | incomming calls get stuck in voicemail etc... hangups never gets picked up. |
| 08:08.12 | MacDeath | just some hangups are not connected |
| 08:08.17 | MacDeath | pr0 : YES |
| 08:08.25 | MacDeath | my calls in the voicemaiols never came out |
| 08:08.26 | MacDeath | ever |
| 08:08.27 | MacDeath | LOL |
| 08:08.46 | pr0 | I have managed to work around it a little but I'll never get it just right |
| 08:11.26 | MacDeath | we can blame telkom on that one |
| 08:12.28 | *** join/#asterisk delox99 (n=delox99@206-248-149-59.dsl.teksavvy.com) |
| 08:12.38 | pr0 | anyway thx guys, I gotta run! |
| 08:12.39 | delox99 | hi all |
| 08:12.52 | *** join/#asterisk SuperB (i=SuperB@206.80.108.124) |
| 08:13.07 | MozartsGhost | hmm, Damn. having a strange prob on one of my boxes. |
| 08:13.13 | MozartsGhost | works fine for about 20 hours or so. |
| 08:13.15 | MozartsGhost | then I get NOTICE[3448]: Unable to create channel of type 'Zap' |
| 08:13.35 | delox99 | we have a samsung pbx |
| 08:13.43 | MozartsGhost | thaught it was FD's. as someone suggested, but added Hangups after all ext's. no luck... |
| 08:13.59 | delox99 | i want to connect it to an asterisk server using an ata |
| 08:14.16 | SuperB | Quick Question - is anyone still having problems with chan_modem not being created even after downloading updated 1.2 tar.gz? This is on a REdHat 8.0 system |
| 08:14.22 | delox99 | how would i do that? |
| 08:15.04 | delox99 | all FXS ports are samsung codec based |
| 08:15.36 | delox99 | but can i connect an fxs to tht ports that are usually connected to te pstn lines? |
| 08:16.02 | delox99 | an ata should i say |
| 08:16.48 | *** join/#asterisk CleanerX (n=nix@nat-ph3-wh.rz.uni-karlsruhe.de) |
| 08:17.03 | delox99 | so that when we have long distance id route the call to the port that is connected to the ata then go through Asterisk |
| 08:17.30 | JonR800 | does ael support compound conditionals? |
| 08:19.30 | *** join/#asterisk webmind (n=webmind@feather.perl6.nl) |
| 08:19.58 | webmind | lovely.. auto join doesnt work on this network :( |
| 08:19.59 | webmind | re |
| 08:26.46 | Jestie | chat later ... |
| 08:27.50 | *** join/#asterisk fiftyCal (n=b@69-160-145-156.ontrca.adelphia.net) |
| 08:31.22 | *** join/#asterisk pif (n=ldm@zenon.apartia.fr) |
| 08:34.04 | shanky | good morning |
| 08:38.35 | Chotaire | dudes, I have this one SIP provider where I cannot get asterisk to take incoming calls. when calling that number, it's always busy and there's not one sign within asterisk why this happens. when using that damn fritzbox, everything will work. |
| 08:39.08 | *** join/#asterisk BladeRunner05 (n=feelme@adsl-ull-205-65.44-151.net24.it) |
| 08:40.05 | pif | hi, my sip phone (gxp-2000) send digits one by one to *, I'd like * to wait a few seconds before matching the extension, how? |
| 08:44.48 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 08:49.13 | *** join/#asterisk chapeaurouge (n=chap@85.201.80.249) |
| 08:50.09 | Chotaire | anyone using 1und1 sip provider with asterisk!? |
| 08:54.18 | *** join/#asterisk steff (n=steff@80.125.254.220) |
| 08:54.51 | *** join/#asterisk razu_ (n=razu@217-159-242-106-dsl.est.estpak.ee) |
| 09:00.09 | *** join/#asterisk Aze` (n=aze@85-18-136-114.ip.fastwebnet.it) |
| 09:04.58 | *** join/#asterisk zoa (n=kkk@pirus.securax.be) |
| 09:08.03 | zoa | haha |
| 09:08.08 | zoa | asterisk is on digg.com |
| 09:08.15 | zoa | and voipcharges.com is too |
| 09:08.22 | mog_home | but no slashdot... |
| 09:08.26 | zoa | not yet |
| 09:08.41 | zoa | will come maybe, but we should talk them into adding an asterisk section |
| 09:08.44 | zoa | like they do for apache |
| 09:08.50 | mog_home | heh |
| 09:08.53 | mog_home | not yet |
| 09:08.55 | mog_home | one day though |
| 09:10.32 | Chotaire | anyone know a reason why "sip show registry" wouldn't even listen a sent request although the configs are about the same with a different provider? |
| 09:11.20 | Chotaire | no matter what I try, I can't get asterisk to even try register with 1und1 it seems. |
| 09:12.22 | Aze` | Anyone can help me with fxo (tdm400) + splitter + dsl ? |
| 09:15.05 | _GiGi_ | hello |
| 09:15.21 | mog_home | hiya |
| 09:15.24 | mog_home | whats up aze |
| 09:15.33 | _GiGi_ | how i can send HOLD to another pbx in ZAP ? :> |
| 09:15.39 | _GiGi_ | in HDSL frame :) |
| 09:15.54 | Aze` | ;> |
| 09:16.00 | mog_home | what do you mean hold? |
| 09:16.03 | *** join/#asterisk A-Tuin|work (n=A-Tuin@nat.office.legend.net.uk) |
| 09:16.19 | *** join/#asterisk Antonie (n=Antonie@a213-84-26-148.adsl.xs4all.nl) |
| 09:16.42 | Aze` | _GiGi_ > Analog Line + splitter + adsl => fxo port (tdm400) <> Asterisk <> Sip adapter <> Fax-Machine |
| 09:17.05 | Aze` | I cant receive fax.. error connection, bad quality on pots (i suppose) |
| 09:17.42 | Aze` | if i link analog Line + splitter +adsl => fax machine = All works |
| 09:17.54 | _GiGi_ | mog_home: hmm, i must redirect connection to another number, when i send hold to another pbx, it redirect this connection. |
| 09:18.22 | _GiGi_ | Aze`: i cant help u now, im very busy :) |
| 09:18.24 | Antonie | When I compile asterisk-1.0.9 I get an error saying 'cannot find - lssl' :) |
| 09:18.42 | _GiGi_ | i must redirect this connection in my IVR :> |
| 09:18.43 | Aze` | np:) |
| 09:18.44 | Antonie | strange.. it has something to do with openssl I guess ? but I don't know what |
| 09:19.17 | Antonie | if there is anybody here who knows how to solve the above compilation error.. I would really appreciate that |
| 09:21.09 | *** join/#asterisk tobiasWolf (n=konversa@195.162.255.10) |
| 09:27.50 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 09:28.24 | *** join/#asterisk fishboy1669 (i=proxyuse@62.69.81.129) |
| 09:28.46 | fishboy1669 | morning guys |
| 09:30.47 | *** join/#asterisk duckz (n=duckz@193.192.46.26) |
| 09:31.36 | fishboy1669 | well god must have heard me |
| 09:31.50 | fishboy1669 | asterisk have announced on the website the 1.2.0 version |
| 09:32.07 | fishboy1669 | not quite the massive announcment i would have expected though |
| 09:32.12 | fishboy1669 | just a couple of lines |
| 09:32.31 | fishboy1669 | and still no list of whats been fixed and what the new functionality is |
| 09:32.52 | fishboy1669 | probably have to wait till the next release for the docs and manuals though :( |
| 09:33.41 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
| 09:34.03 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 09:34.27 | zoa | antonie |
| 09:35.08 | zoa | You must install these packages before proceeding with the installation of Asterisk. |
| 09:35.08 | zoa | * Linux 2.4 kernel sources or kernel 2.6 header files. (for libpri) |
| 09:35.08 | zoa | * bison and bison-devel packages (This is used to build Asterisk) |
| 09:35.08 | zoa | * ncurses and ncurses-devel packages (Used to build astman, etc.) |
| 09:35.08 | zoa | * zlib and zlib-devel packages |
| 09:35.09 | zoa | * openssl and openssl-devel packages |
| 09:35.12 | zoa | oops |
| 09:35.17 | zoa | that should have been in private |
| 09:35.23 | endre | thats ok |
| 09:35.33 | zoa | http://www.asteriskguru.com/tutorials/basic_installation_information_asterisk_from_source.html |
| 09:35.43 | zoa | what distro are you on Antonie ? |
| 09:36.47 | *** join/#asterisk oej_ (n=Olle@apollo.webway.se) |
| 09:37.04 | zoa | hey ho olle |
| 09:37.08 | Chotaire | duh, anyone here who could think about a reason why a "register =>" in sip.conf wouldn't even try registering according to "sip show registry"? |
| 09:37.35 | zoa | yes |
| 09:38.09 | Chotaire | and? |
| 09:42.38 | endre | peer=? |
| 09:42.43 | endre | or what |
| 09:42.57 | fishboy1669 | hi zoa |
| 09:42.57 | *** join/#asterisk nfi|ermes (n=ermsewrk@217.220.121.62) |
| 09:43.01 | endre | type=peer |
| 09:43.03 | endre | i think |
| 09:43.23 | Chotaire | endre: ? |
| 09:44.19 | Aze` | Anyone can help me with fxo shared with dsl + splitter ? |
| 09:45.56 | Chotaire | omg... I found the problem. |
| 09:46.08 | Chotaire | filing bug report. l8ers. |
| 09:46.27 | testmachine | hm |
| 09:46.33 | testmachine | i am bussy with the language packs now |
| 09:47.14 | *** join/#asterisk fourcheeze (n=rich@westbury.doilywood.org.uk) |
| 09:51.55 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 09:59.59 | *** join/#asterisk zoa (n=kkk@pirus.securax.be) |
| 10:00.02 | zoa | damn inet |
| 10:04.39 | *** join/#asterisk rking (n=rking@ip68-1-234-152.dl.dl.cox.net) |
| 10:04.39 | *** join/#asterisk cjk (n=cjk@80.92.64.103) |
| 10:05.03 | fishboy1669 | result found the list of changes for version 1.2.0 |
| 10:05.04 | fishboy1669 | http://www.digium.com/index.php?menu=ast12-announcement#changes |
| 10:05.13 | *** join/#asterisk viperdude (n=jon@84.45.193.6) |
| 10:05.18 | fishboy1669 | not oviouse to find though |
| 10:05.20 | fishboy1669 | :( |
| 10:05.35 | cjk | hi, does anyone know why asterisk writes a cdr log (duration=0, disposition=failed) when i do dial(SIP/user&SIP/user2) and user2 is not online. any way to change this behaviour |
| 10:14.49 | viperdude | i have a problem with a AGI app. if I send STREAM FILE with a file that does not exist then * says file not found, however if I send a file that does exist it does nothing, not even a error message, any ideas? |
| 10:15.37 | *** join/#asterisk gambolputty (n=gambolpu@cblmdm72-240-242-4.buckeyecom.net) |
| 10:18.33 | Chotaire | omg.. I have another problem.. on incoming calls (SIP), I will have one call to be ok and the next call will be silent.. e.g. nobody can hear anything... then the next call is ok, and then the next call will be screwed again.. anyone ever heard about a behaviour like this? what kinda misconfiguration can that be? |
| 10:19.02 | Chotaire | no matter what SIP provider I will be using. |
| 10:20.16 | Chotaire | voice is gone completely but I can hear the caller hit dtmf tones. |
| 10:24.48 | [Airwolf] | Chotaire, is your * server behind NAT ? |
| 10:25.06 | Chotaire | nope |
| 10:25.32 | fishboy1669 | chotaire what codecs are u using |
| 10:25.51 | fishboy1669 | chotaire do u have a firewall anywhere |
| 10:27.47 | Chotaire | ulaw for most |
| 10:27.57 | Chotaire | I dont consider this a firewall problem. |
| 10:33.48 | *** join/#asterisk ful|work (n=fulgas@209.8.233.198) |
| 10:36.19 | jcath | hey, asterisk supports an ISDN adapter as a FXO, right? |
| 10:36.26 | jcath | or not? |
| 10:38.25 | *** join/#asterisk Samoied (n=Samoied@popeye.opens.com.br) |
| 10:39.57 | zoa | yes |
| 10:39.58 | zoa | it does |
| 10:40.06 | zoa | although its not called that way :) |
| 10:41.26 | fishboy1669 | anyone know how to upgrade from version 1.0.9 to 1.2.0 |
| 10:45.03 | jcath | but does isdn adapter support normal analog phone line? but not ISDN line? |
| 10:45.54 | nextime | uhmm why in cvs head i can't find app_setcallerpres or app_callingpres? |
| 10:46.44 | zoa | its changed now |
| 10:46.52 | zoa | hmm |
| 10:46.53 | zoa | although |
| 10:46.59 | zoa | maybe its with set() now |
| 10:47.25 | nextime | uhmm, ok i will look on voip-info, thanks |
| 10:49.42 | cjk | how can i set the system amaflags default |
| 11:08.28 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 11:08.58 | puzzled | hi |
| 11:12.38 | viperdude | hi anyone around?: |
| 11:13.27 | Ahrimanes | yeah |
| 11:14.08 | *** join/#asterisk chapeaurouge (n=chap@85.201.80.249) |
| 11:14.20 | viperdude | hi i am trying to write AGI app on FreeBSD, but WAIT FOR DIGITS doesn't seem to work |
| 11:14.23 | viperdude | any ideas |
| 11:15.36 | Ahrimanes | viperdude: have a digium card in the machine or ztdummy loaded? might be a timing issue |
| 11:15.36 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
| 11:16.02 | viperdude | i think we have ztdummy loaded |
| 11:16.45 | viperdude | shoudl it be? |
| 11:16.46 | *** join/#asterisk razu_ (n=razu@217-159-181-27-dsl.prn.estpak.ee) |
| 11:16.49 | viperdude | should |
| 11:19.16 | Ahrimanes | yeah |
| 11:19.17 | Ahrimanes | kldstat |
| 11:19.18 | viperdude | we do have ztdummy loaded |
| 11:19.38 | Ahrimanes | ok |
| 11:19.39 | Ahrimanes | hm |
| 11:19.53 | viperdude | Id Refs Address Size Name |
| 11:19.53 | viperdude | <PROTECTED> |
| 11:19.53 | viperdude | <PROTECTED> |
| 11:19.53 | viperdude | <PROTECTED> |
| 11:19.53 | viperdude | <PROTECTED> |
| 11:19.54 | viperdude | <PROTECTED> |
| 11:19.56 | viperdude | <PROTECTED> |
| 11:19.58 | viperdude | <PROTECTED> |
| 11:20.00 | viperdude | <PROTECTED> |
| 11:20.02 | viperdude | <PROTECTED> |
| 11:20.04 | viperdude | 10 1 0xc28f2000 2000 ztdummy.ko |
| 11:20.05 | Ahrimanes | what language are you writing the agi in |
| 11:20.05 | tzafrir_laptop | ~pb |
| 11:20.07 | jbot | hmm... pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca/ |
| 11:20.07 | viperdude | sorry for the flood |
| 11:20.09 | Ahrimanes | argh.. dont paste |
| 11:20.11 | viperdude | PHP |
| 11:20.16 | viperdude | sorry |
| 11:20.23 | Ahrimanes | oh ok |
| 11:20.32 | Ahrimanes | there's phpagi libraries, using those? |
| 11:20.41 | tzafrir_laptop | viperdude, to get a one-liner with all the zaptel modules that are loaded, run: lsmod | grep zaptel |
| 11:21.00 | tzafrir_laptop | viperdude, to get a one-liner with all the zaptel modules that are loaded, run: lsmod | grep ^zaptel |
| 11:21.20 | viperdude | I will try the phpagi lib |
| 11:21.21 | tzafrir_laptop | (the first one is typically two-liner, sorry) |
| 11:21.23 | *** join/#asterisk schula (n=uli@dino.spamt.net) |
| 11:26.15 | *** part/#asterisk shanky (n=shanky@82.159.214.9) |
| 11:26.51 | Ahrimanes | tzafrir_laptop: freebsd, not linux |
| 11:30.24 | *** join/#asterisk bugant (n=bugant@host58-101.pool8534.interbusiness.it) |
| 11:36.42 | *** join/#asterisk psukogo (n=ctrix@81-174-32-147.f5.ngi.it) |
| 11:36.47 | psukogo | Hi |
| 11:37.20 | psukogo | I have a problem but I cannot solve it. |
| 11:37.35 | psukogo | I'm using asterisk with only SIP channels |
| 11:37.48 | psukogo | moh is choppy and RXfax does not work. |
| 11:38.13 | psukogo | If I call, answer() and RXfax() are called but I cannot hear anything. |
| 11:38.14 | *** join/#asterisk sysdebug (n=sysdebug@200.163.193.247) |
| 11:38.19 | psukogo | any hints ? |
| 11:40.02 | psukogo | I'm using 1.0.9 |
| 11:40.05 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
| 11:42.28 | *** join/#asterisk jgonzalez (n=jgonzale@187.Red-80-26-93.staticIP.rima-tde.net) |
| 11:42.32 | jgonzalez | hi |
| 11:42.54 | kippi | If I phoned BT and asked them for Framing, coding, and timing they should know what i am talking about? |
| 11:43.05 | jgonzalez | is possible to do a 3G videoconference using a PC with asterisk and a PCMCIA UMTS and a 3G movil phone ? |
| 11:54.35 | *** join/#asterisk DannyF (n=dannyf@c-454fe353.24-0099-74657210.cust.bredbandsbolaget.se) |
| 11:59.11 | *** join/#asterisk limepickle (n=chris@193.128.28.34) |
| 11:59.19 | limepickle | hi all |
| 12:00.50 | limepickle | i'm new to asterisk & IRC also, i'm looking for help concerning a TE110p |
| 12:06.00 | mut | heh |
| 12:06.11 | mut | i love how all the sudden now that this person has voip there is a problem with it |
| 12:06.15 | mut | the TAC case reads.. |
| 12:06.18 | mut | "Customer called in asking how secure VOIP was because sometimes they could hear thier nieghbors dialing numbers on thier phone. Customer is using a cordless phone and I am assuming the nieghbor is also. Informed customer that it is most likely the fact that they both use cordless phones and not a VOIP issue" |
| 12:07.23 | Exion | got a customer calling that they was hearing the radio on their phone... Same issue as you ;) hehe |
| 12:08.49 | limepickle | excuse me mut, would you know of anyone around here who could help a beginer with a TE110p card? |
| 12:09.07 | mut | what ya goin with it? |
| 12:09.20 | mut | doing* |
| 12:09.33 | limepickle | i'm wanting to connect an Ericcson BP250 to it and accept incoming calls to the conference extensions |
| 12:10.32 | limepickle | as in incoming FROM the BP250, so the asterisk will give it conf functionality |
| 12:11.24 | mut | ya used www.voip-info.org? |
| 12:11.33 | mut | looks like ya just wanna hook it up as a pri i assume? |
| 12:11.43 | *** join/#asterisk iDunno (i=brettp@stef.sommitrealweird.co.uk) |
| 12:11.44 | mut | just look for zaptel config |
| 12:11.49 | limepickle | yes, and no i havn't really looked at viop info |
| 12:11.52 | mut | and zapata.conf |
| 12:12.24 | limepickle | i've got a link on the web to all my settings if you want to see? |
| 12:14.15 | mut | but voip-info will give ya a good start |
| 12:14.34 | mut | any more specific questions people can answer quickly in here |
| 12:14.41 | mut | but to get ya setup i can't help ya right now |
| 12:14.51 | limepickle | ok, no problem, thats fair enough |
| 12:15.39 | limepickle | thanks |
| 12:18.22 | *** join/#asterisk vexorg (n=vexorg@CPE000021ded913-CM001225419164.cpe.net.cable.rogers.com) |
| 12:18.29 | RoyK | does anyone here have a DOS boot floppy? |
| 12:19.26 | mut | www.bootdisk.com |
| 12:19.28 | mut | lots of em |
| 12:20.14 | *** join/#asterisk limepickle (n=chris@193.128.28.34) |
| 12:31.01 | viperdude | :q |
| 12:32.01 | *** join/#asterisk PupenoL (n=pupeno@200.123.183.89) |
| 12:32.14 | limepickle | would anyone be able to offer any help to me in setting up a TE110P card to link to an Ericsson BP250? |
| 12:32.37 | limepickle | I am a total beginner I must add |
| 12:34.52 | mut | heh |
| 12:35.54 | limepickle | i'm not specically hassling you mut :), I'm just putting out a request, if anyone is able to help then great. |
| 12:35.54 | mut | http://www.voip-info.org/wiki/index.php?page=Asterisk+config+zapata.conf |
| 12:36.00 | mut | http://www.voip-info.org/wiki/view/Asterisk+config+zaptel.conf |
| 12:36.04 | limepickle | ty |
| 12:36.26 | mut | first will let ya configure the card itself |
| 12:36.33 | mut | 2nd will let ya configure it for use with asterisk |
| 12:36.53 | mut | you manage to get the driver loaded and everything i assume? |
| 12:37.27 | *** join/#asterisk Caziques (i=FateX@196.46.143.177) |
| 12:37.31 | limepickle | yes, it sees the channels etc, and I have a blinking red led, i'm just unsure of the next step |
| 12:37.50 | mut | ya got it configured for pri mode? |
| 12:37.51 | Caziques | can anyone please help me ? |
| 12:37.58 | Caziques | oh hi |
| 12:38.03 | limepickle | like i said earlier, when you have time, i have a link to all my info. I'm in no rush |
| 12:38.27 | limepickle | i'm trying to tread carefully mut :) |
| 12:38.29 | mut | just say yes or no.. ;) |
| 12:38.46 | limepickle | you beat me to that one, YES |
| 12:39.07 | Caziques | lo ? |
| 12:39.12 | mut | and the BP250 is also pri? |
| 12:39.25 | mut | both with the same encodings |
| 12:40.00 | limepickle | i havn't got as far as the bp250 :), I'm still trying to see what I'll need to configure for the asterisk |
| 12:40.09 | mut | well first things first |
| 12:40.12 | mut | get the pri up and green |
| 12:40.18 | mut | then ya can move onto using it |
| 12:40.33 | limepickle | ok, so green is good, right! |
| 12:40.53 | limepickle | thanks, i'll check those links out then |
| 12:41.17 | mut | heh yea... |
| 12:41.24 | mut | usually thats the way it goes no? |
| 12:41.30 | Caziques | does anyone have sample configs for using asterisk with sofphones only to try and learn the sw ? |
| 12:41.41 | limepickle | its an unknown device in ls pci btw, is that normal before i start? |
| 12:42.11 | mut | no |
| 12:42.20 | mut | it should be a tiger network something or other |
| 12:42.37 | limepickle | ok, i saw that somehwere |
| 12:42.45 | mut | well |
| 12:42.58 | *** join/#asterisk bugant (n=bugant@host58-101.pool8534.interbusiness.it) |
| 12:42.58 | mut | lsmod | grep wcte11xp |
| 12:43.02 | mut | does it show anything? |
| 12:43.06 | Caziques | yo .. anyone ? |
| 12:43.14 | limepickle | hang on... |
| 12:43.15 | mut | Caziques: www.voip-info.org |
| 12:43.27 | Caziques | thanx |
| 12:44.13 | Caziques | i'll come back when i've mastered this thing |
| 12:44.21 | limepickle | lsmod shows a zaptel wcusb wcte11xp |
| 12:45.25 | tzafrir_laptop | wcusb? any special reason for loading it? should be harmless, though |
| 12:46.17 | *** join/#asterisk coppice (n=chatzill@40.199.17.210.dyn.pacific.net.hk) |
| 12:46.26 | Caziques | any one know what this is from ...Ouch ... error while writing audio data: : Broken pipe Segmentation fault |
| 12:46.30 | limepickle | i just followed the digium setup :) |
| 12:46.47 | tzafrir_laptop | Caziques, probably an error from mpg123 |
| 12:47.03 | Caziques | is there a way to debug mpg123 ? |
| 12:47.05 | tzafrir_laptop | that is: from the pipe to the music-on-hold program |
| 12:47.10 | limepickle | caziques, i had that, it was because my zaptel & zapata.conf differed |
| 12:47.39 | Caziques | where do i disable music on hold .. just delete the conf file ? |
| 12:47.53 | limepickle | tyw mut, i'll try gettting it GREEN first ok, I'll get back to you guys ok? |
| 12:48.02 | mut | k |
| 12:48.05 | limepickle | ty |
| 12:51.11 | limepickle | please don't slap me, but how do i save my irc conversation out to a text file? |
| 12:51.59 | mut | copy & paste it? |
| 12:52.01 | mut | *shrug* |
| 12:52.09 | tzafrir_laptop | limepickle, what irc client? |
| 12:52.11 | mut | i log everything |
| 12:52.20 | limepickle | mirc, it doesnt let me copy/paste |
| 12:52.34 | tzafrir_laptop | e.g: on xchat it's easy to log everything |
| 12:52.49 | mut | limepickle: in mirc ya just highlight the text |
| 12:52.56 | mut | and it automatically copies it to the clipboard |
| 12:53.03 | limepickle | Ah ah! ok |
| 12:53.13 | Caziques | mut .... do ya know why i would be getting segmentation faults all the time ? |
| 12:53.42 | Caziques | only left with asterix and sip conf's and still getting a segmentation fault |
| 12:53.48 | mut | cuz it doesn't like ya? |
| 12:53.55 | mut | audio driver sucks? |
| 12:54.01 | *** join/#asterisk lilo (i=levin@freenode/staff/pdpc.levin) |
| 12:54.10 | mut | what ver ya using? |
| 12:54.16 | Caziques | 1.2 |
| 12:54.45 | mut | try checkin out the core dump |
| 12:54.49 | limepickle | one more thing before i go, where can i get a description of the LEDS on a TE110p, as I'm not sue whether its red cos nothings connected to it or cos its busted |
| 12:54.53 | Caziques | cool thanx |
| 12:54.58 | *** join/#asterisk dualarrow (n=Andrew@203-121-194-76.e-wire.net.au) |
| 12:55.00 | mut | http://www.voip-info.org/tiki-index.php?page=Asterisk%20debugging |
| 12:55.14 | mut | limepickle: red means no link |
| 12:55.17 | mut | green means link |
| 12:55.55 | limepickle | no link would fit, i have no cable yet :), bye for now! |
| 12:57.00 | Caziques | kay so leme understand ... Asterisk doensnt run like a server (ie squid) but you need to install it on the operator console ? |
| 12:57.27 | Caziques | hey forgive me but im new at this asterisk |
| 12:57.44 | *** join/#asterisk dualarrow (n=Andrew@203-121-194-76.e-wire.net.au) |
| 13:00.29 | tzafrir_laptop | it does run like a server (daemon is a more "technical" term) |
| 13:01.06 | Caziques | yea i know that |
| 13:01.32 | tzafrir_laptop | do you ask about the installation or about running asterisk at startup? |
| 13:01.54 | Caziques | installation is done now when i start it for the first tiem it does a seg afult |
| 13:02.04 | Caziques | seg fault |
| 13:02.05 | Flauto | goog morning |
| 13:02.13 | Caziques | afternoon |
| 13:02.28 | Flauto | okay |
| 13:02.33 | Flauto | just got up |
| 13:02.36 | Flauto | early morning |
| 13:02.39 | tzafrir_laptop | where do you see the segfault? how do you run asterisk? |
| 13:03.15 | tzafrir_laptop | early afternoon here |
| 13:03.19 | Caziques | started by asterisk -vvvvcd |
| 13:03.47 | *** join/#asterisk _Madar (n=tophe@219-84-129-163-adsl-tpe.static.so-net.net.tw) |
| 13:03.50 | Caziques | [chan_phone.so] => (Linux Telephony API Support) |
| 13:03.56 | Caziques | Segmentation fault |
| 13:04.07 | Caziques | those were the last two lines |
| 13:04.19 | tzafrir_laptop | what version of asterisk? what linux distro? was it installed over an earlier installation? |
| 13:04.34 | *** join/#asterisk lilo (i=levin@freenode/staff/pdpc.levin) |
| 13:04.37 | tzafrir_laptop | (the asterisk install, not the linux install) |
| 13:04.50 | Chotaire | can anyone assist me with this babe? Nov 18 14:04:09 ERROR[16211]: chan_capi.c:2098 start_pbx_on_match: ISDN2: did not find exten for '12345', ignoring call. |
| 13:04.52 | Caziques | 1.2.0 suse 9.1 no previous asterisk |
| 13:05.05 | Chotaire | prior version: 1.0.3 |
| 13:05.11 | Chotaire | now running: 1.2.0 |
| 13:05.19 | Caziques | 1.2.0 = asterisk ........suse 9.1 ......no previous asterisk |
| 13:05.44 | Chotaire | adding an extension [12345] didn |
| 13:05.46 | Chotaire | 't help. |
| 13:06.40 | tzafrir_laptop | ugly workaround: do away with chan_modem.so ('noload => chan_modem.so' in /etc/asterisk/modules.conf). |
| 13:06.50 | *** join/#asterisk razu_ (n=razu@217-159-242-106-dsl.est.estpak.ee) |
| 13:06.56 | Caziques | thanx leme try that |
| 13:07.32 | tzafrir_laptop | it is mostly used by the largely non-fuctional old isdn4linux support |
| 13:08.00 | Chotaire | tzafrir: any idea with my little isdn problem? |
| 13:08.13 | Chotaire | tzafrir: google knows nothing about start_pbx_on_match |
| 13:08.30 | Caziques | still segmentation fault |
| 13:08.54 | Flauto | from time to time, i got error message on that chan_modem thing |
| 13:09.03 | Chotaire | and for incoming calls, there's another bugger: |
| 13:09.03 | Chotaire | <PROTECTED> |
| 13:09.04 | Chotaire | Nov 18 14:06:40 NOTICE[16379]: app_dial.c:1011 dial_exec_full: Unable to create channel of type 'Capi' (cause 44 - Requested channel not available) |
| 13:09.15 | Chotaire | oops, that's for outgoing calls ;) |
| 13:09.49 | Chotaire | I seem to have run into severe configuration incompatibilities between 1.0.3 and 1.2.0... anyone awake who can point me to a direction? |
| 13:09.58 | tzafrir_laptop | Chotaire, sorry, but chan_capi is not my field |
| 13:09.58 | Flauto | what is that chan_modem thing? |
| 13:10.28 | *** join/#asterisk fugitivo (n=ajf@209.13.244.248) |
| 13:10.40 | Flauto | tzafrir, i have anothing message, saying, Nov 18 06:23:38 WARNING[14292]: chan_sip.c:9575 handle_response_register: Got 200 OK on REGISTER that isn't a register |
| 13:11.01 | tzafrir_laptop | not sure exactly. It was supposed to support all sorts of modems. It ended up supportingg almost only isdn4linux ISDN modems |
| 13:11.35 | Flauto | i dont' even have any modem on my linux machine |
| 13:11.42 | Flauto | so, can i noload it? |
| 13:11.52 | *** join/#asterisk r0d3nt (i=nobody@wsip-24-234-241-145.lv.lv.cox.net) |
| 13:11.58 | tzafrir_laptop | Flauto, do you register to another server (e.g: voip provider) or is it a sip phone registering to your server? |
| 13:12.17 | Chotaire | hrmpf |
| 13:12.32 | Flauto | yes, i use all sip on my asterisk |
| 13:12.34 | *** join/#asterisk astoria (n=chat@user-7e5a43.user.msu.edu) |
| 13:12.41 | astoria | Good Morning all. |
| 13:12.47 | Flauto | i use only fwd by iax2 |
| 13:12.48 | tzafrir_laptop | Flauto, you can unload it. But there are some other modules that depend on it (chan_modem_*.so), noload them as well |
| 13:12.50 | Chotaire | and now I just found out that updating to 1.2.0 still leaves me with the same damn problems... on incoming sip calls, sometimes the caller hears nothing and sometimes they do. |
| 13:13.06 | astoria | Can anyone here recommend a good company to do data recovery on a hard drive with files that were deleted, yet undelete cannot get them? |
| 13:15.57 | Flauto | is that register is okay but it is not a register thing needs to be fixed? |
| 13:16.22 | Flauto | 200 is an extension on my asterisk |
| 13:16.45 | Flauto | but the sip register id is a different one so i should not even show sip register as 200 |
| 13:16.50 | Flauto | that is the part i dont' understand |
| 13:17.23 | *** join/#asterisk oej_ (n=Olle@apollo.webway.se) |
| 13:19.41 | newl | astcryz: OnTrack. Be prepared to spend some money though. ;) |
| 13:20.00 | newl | erm.. astoria even. |
| 13:22.55 | astoria | newl: OnTrack? Have they been reliable? |
| 13:23.02 | astoria | newl: money is no object here. |
| 13:23.39 | newl | We used them heavily when I was at Ford Motor Co. |
| 13:23.55 | *** join/#asterisk SilentBob420 (n=silentbo@CPE-24-26-168-34.mn.res.rr.com) |
| 13:24.02 | psukogo | question: is therea a workaround to make asterisk transmit silence on Sip channels ? |
| 13:24.08 | astoria | For situations like mine? |
| 13:24.22 | psukogo | or better: to transmit something even if the client is not trasmitting silence ? |
| 13:25.54 | *** join/#asterisk darkskiez (n=darkskie@194.247.78.146) |
| 13:26.04 | *** join/#asterisk zeedo (n=zeedo@80.68.92.188) |
| 13:27.07 | newl | astoria: Yes. Also for the normal expected situations such as disk failure and the like. |
| 13:27.14 | docelmo | ITS FIRDAY! |
| 13:28.10 | newl | Only for another two and a half hours. =) |
| 13:28.34 | astoria | newl: GOod. Thanks. I appreciate your recommendation. I am going to call them at 9. |
| 13:28.42 | newl | np |
| 13:29.02 | astoria | If I say newl sent me, do I get a discount? |
| 13:29.21 | newl | haha doubt it. :) |
| 13:39.00 | Flauto | anyone knows why i get this message? Nov 18 06:23:38 WARNING[14292]: chan_sip.c:9575 handle_response_register: Got 200 OK on REGISTER that isn't a register |
| 13:40.36 | *** join/#asterisk epablo (n=epablo@200.75.139.188) |
| 13:40.52 | epablo | Hi guys |
| 13:41.00 | Flauto | hi epablo |
| 13:41.35 | epablo | I'm having some problems with a zaptel card. When I restart the service it hangs the server |
| 13:41.57 | epablo | I'm using 1.2.0, freshlly built last night |
| 13:42.39 | docelmo | Any asterisk C dev's in here? I have a quick question |
| 13:43.32 | twisted | docelmo, make it quick... i'm about to head to work |
| 13:43.46 | Flauto | epablo, i have the same problem sometimes but it does not really effect the usage of zaptel card |
| 13:43.48 | docelmo | is this valid for pulling channel VAR's? myvar = pbx_builtin_getvar_helper(chan,"MYVARIABLE"); |
| 13:44.03 | twisted | docelmo, should be, |
| 13:44.10 | docelmo | sweet.. thanks josh! |
| 13:44.35 | twisted | np |
| 13:45.03 | epablo | Flauto:?? |
| 13:46.51 | *** join/#asterisk heison (n=heison@216.126.67.39) |
| 13:47.59 | *** join/#asterisk mrtwister (n=mrtwiste@cable-9-42.cgates.lt) |
| 13:48.10 | Chotaire | omg, why did I update to 1.2.0... |
| 13:48.11 | Chotaire | <PROTECTED> |
| 13:48.22 | mrtwister | hello - is anyome know what is name of xorcom rapid irc channel? |
| 13:48.27 | *** join/#asterisk mychi (n=murphyx@pool-151-197-185-19.phil.east.verizon.net) |
| 13:48.33 | *** join/#asterisk jimmy_deanPB (n=jhodapp@72.244.232.226) |
| 13:49.44 | *** join/#asterisk tdonahue (n=tdonahue@208.51.101.201) |
| 13:50.49 | LostFrog | Chotaire: just out of curiosity what did you run before 1.2.0? |
| 13:51.12 | darkskiez | docelm0: what u making ? |
| 13:51.46 | Chotaire | 1.0.3 |
| 13:52.41 | tzafrir_laptop | mrtwister, #xorcom-rapid, naturally. But feel free to ask here, if the question is generic enough |
| 13:52.52 | Chotaire | chan_capi-0.4.0-pre1 won't even compile.. I'm fed up right now, downgrading... |
| 13:53.14 | tzafrir_laptop | Chotaire, try chan_capi-cm |
| 13:53.29 | Chotaire | I did |
| 13:53.31 | Chotaire | <PROTECTED> |
| 13:53.41 | fugitivo | what the hell is curl? |
| 13:53.44 | Chotaire | and more gayness like this.. I can't even get an early b3 dialtone to work. |
| 13:53.44 | mrtwister | tzafrir_laptop, sure, thanks! question is simple and to you :) did you making asterisk-2 repository? |
| 13:54.00 | tzafrir_laptop | mrtwister, 1.2, right? |
| 13:54.10 | *** join/#asterisk genmud (n=genmud@ip68-98-82-206.ph.ph.cox.net) |
| 13:54.21 | viperdude | curl.haxx.se |
| 13:54.27 | tzafrir_laptop | I've already uploadd zaptel 1.2 packages to the "experimental" repository |
| 13:54.35 | fugitivo | do i need that? |
| 13:54.41 | fugitivo | no i don't |
| 13:54.45 | tzafrir_laptop | there are some older libpri and asterisk there |
| 13:54.47 | fugitivo | why it comes as default? |
| 13:55.02 | tzafrir_laptop | fugitivo, doesn't your distro come with one? |
| 13:55.04 | Chotaire | <PROTECTED> |
| 13:55.05 | Chotaire | <PROTECTED> |
| 13:55.07 | Chotaire | great... not. |
| 13:55.15 | fugitivo | tzafrir_laptop: no, it's lfs (linux from scratch |
| 13:55.16 | fugitivo | ) |
| 13:55.19 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:55.58 | mrtwister | tzafrir_laptop, yes 1.2 |
| 13:56.13 | mrtwister | tzafrir_laptop, i remember, you had own 'beta' repository |
| 13:57.20 | tzafrir_laptop | http://xorcom-rapid.berlios.de/ has most of the links, it lacks some details. The 1.2 packages are still in a separate repository: |
| 13:57.42 | tzafrir_laptop | deb http://rapid.dotsrc.org/ experimental/ |
| 13:57.48 | *** part/#asterisk epablo (n=epablo@200.75.139.188) |
| 13:57.57 | mrtwister | tzafrir_laptop, thank you a lot! |
| 13:58.34 | tzafrir_laptop | Speaking of IRC channels: I'm also on #asterisk-il , if it's relevant to anybody |
| 13:58.56 | X-Files | i have one question in asterisk: Calling to me (use exten => 1,1,Dial(SIP/1,60,tTrhwg) ) but this Line busy, i need in talking process beep mode (about new call) |
| 14:02.04 | LostFrog | I think that is a phone feature, X-Files. |
| 14:02.13 | {zombie} | X-Files: you need to enable call-waiting on your sip device |
| 14:02.15 | {zombie} | if it supports it |
| 14:02.25 | X-Files | hmm |
| 14:02.54 | fugitivo | my * 1.2 didn't compile chan_modem |
| 14:02.56 | {zombie} | and do you really need all those options on your Dial string? |
| 14:03.34 | X-Files | {zombie}: maybe call_limit ? |
| 14:03.51 | {zombie} | I mean the tTrhwg |
| 14:04.06 | X-Files | {zombie}: this for test |
| 14:04.07 | X-Files | ;) |
| 14:04.14 | {zombie} | do you even know what they do? |
| 14:04.18 | X-Files | yep |
| 14:04.23 | X-Files | i read wiki |
| 14:04.24 | X-Files | ;) |
| 14:04.31 | {zombie} | ok |
| 14:05.39 | Chotaire | I found this stuff out.. another one for your notes: |
| 14:05.42 | Chotaire | exten => 9,1,SetCallerPres(prohib_not_screened) |
| 14:05.42 | Chotaire | exten => 9,2,Dial(${TRUNK}//b,45,ThHg) |
| 14:05.49 | Chotaire | the "//" did the trick |
| 14:06.02 | Chotaire | (chan_capi early b3 dialtone with clir) |
| 14:07.11 | Chotaire | TRUNK=Capi/g1 |
| 14:08.06 | Chotaire | (using asterisk 1.2.0 with chan_capi-cm-0.6.1) |
| 14:08.55 | fugitivo | does asterisk 1.2 use chan_modem.so? |
| 14:09.03 | Chotaire | hm? |
| 14:09.06 | Chotaire | yes |
| 14:09.16 | fugitivo | why it didn't compile it? |
| 14:09.23 | fugitivo | i don't have the module |
| 14:09.48 | Chotaire | are you sure? or is it just disabled in modules.conf? |
| 14:09.49 | {zombie} | only if you have one of those crappy voice modems |
| 14:09.54 | fugitivo | i think it doesn't use it anymore |
| 14:10.00 | {zombie} | if you don't, don't bother with that module |
| 14:10.26 | Chotaire | hunt for a line that says noload => chan_modem.so |
| 14:10.28 | fugitivo | i disable it from modules.conf and it works |
| 14:10.41 | fugitivo | and i use a clone x100p |
| 14:10.45 | {zombie} | speaking of which, I was visiting a prospective new customer today |
| 14:10.50 | {zombie} | they're a bartercard type company |
| 14:10.56 | Chotaire | you're right btw.. it's not compiled by default. |
| 14:10.58 | {zombie} | and have a phone line they use for "authorisations" |
| 14:11.13 | {zombie} | merchant rings up and supplies their card number, customer number, and sale amount |
| 14:11.16 | *** join/#asterisk langals (n=icechat5@196.7.14.183) |
| 14:11.17 | {zombie} | and it responds with an auth number |
| 14:11.30 | *** part/#asterisk darkskiez (n=darkskie@194.247.78.146) |
| 14:11.33 | {zombie} | .. it's running on a USB voice modem, and interfaced with VB script into MS Access |
| 14:11.36 | {zombie} | *shudder* |
| 14:11.43 | {zombie} | made me want to run far, far away |
| 14:11.45 | *** join/#asterisk kpettit (n=keith@69.15.174.114) |
| 14:11.52 | Chotaire | zombie: try hawaii |
| 14:12.14 | {zombie} | is it nice this time of year? |
| 14:14.11 | Chotaire | it's always nice. |
| 14:14.17 | Chotaire | hawaii has 25 degrees all year. |
| 14:14.27 | Chotaire | (centigrade) |
| 14:15.01 | {zombie} | sounds like a good option then |
| 14:15.09 | {zombie} | although I wouldn't be so much running as swimming |
| 14:15.15 | {zombie} | and it might take a while.. :) |
| 14:15.16 | lunk | just stay away from the, liquid hot mag-ma |
| 14:16.36 | blop | http://bugs.debian.org/cgi-bin/bugreport.cgi?bug=339417 -> any idea on how to resolv this ? |
| 14:17.47 | Chotaire | omg |
| 14:17.48 | Chotaire | <PROTECTED> |
| 14:17.51 | Chotaire | anyone seen that before? |
| 14:18.15 | yxa | WARNING[1325]: chan_iax2.c:7532 socket_read: Received mini frame before first full voice frame |
| 14:18.16 | langals | I am having a problem installing Asterisk1.2.0 on Ubuntu |
| 14:18.25 | langals | I think that I have installed all the dependencies |
| 14:18.45 | stbain | langals: Ubuntu dependencies can be a PITA |
| 14:18.56 | stbain | lot of *-dev packages to install |
| 14:18.59 | langals | throwing an error with chan_zap.c "dereferencing pointer to incomplete type" |
| 14:19.15 | langals | ja |
| 14:19.28 | langals | that is why it is better to install from the repository |
| 14:19.41 | langals | stbain - but they don't have 1.2.0 in the repository |
| 14:19.56 | langals | stbain - do you have any idea which dependency I could be missing |
| 14:20.17 | stbain | are you compiling by source or using the 1.0.9 package from the Ubuntu repository |
| 14:20.20 | stbain | ? |
| 14:20.33 | langals | stbain: no - compiling from source |
| 14:20.49 | langals | stbain - doing it from the repos works fine, but I want the latest version |
| 14:26.55 | Chotaire | ARGHS! |
| 14:26.55 | Chotaire | <PROTECTED> |
| 14:26.55 | Chotaire | <PROTECTED> |
| 14:27.20 | Chotaire | wtf is wrong here? it works for outgoing calls, so why should it be illegal on incoming calls? |
| 14:27.39 | *** join/#asterisk Abbas (n=Abbas@203.81.194.242) |
| 14:28.18 | *** join/#asterisk Eight (n=blake@12-227-165-54.client.mchsi.com) |
| 14:28.43 | stbain | langals: If I remember correctly, I had to install several library packages. I believe there may be a metapackage that will install most of the low level compiler tools and C libraries you need |
| 14:29.34 | langals | stbain - thanks - I will have another look |
| 14:30.54 | yxa | stbain build-essential |
| 14:33.59 | *** join/#asterisk A-Tuin|work (n=A-Tuin@nat.office.legend.net.uk) |
| 14:34.31 | jdv79 | does asterisk have any stastics or profiling abilites? |
| 14:34.35 | jdv79 | that'd be neat |
| 14:34.43 | stbain | jdv79: you mean call detail records? |
| 14:35.14 | jdv79 | i mean more like performance and call flow statistics |
| 14:35.34 | *** join/#asterisk stkn (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 14:35.43 | stbain | jdv79: http://www.voip-info.org/wiki/view/Asterisk+billing |
| 14:36.07 | Katty | morning |
| 14:36.17 | jdv79 | not billing related though |
| 14:36.19 | yxa | hi |
| 14:36.20 | jdv79 | eh |
| 14:36.22 | jdv79 | guess not |
| 14:36.35 | stbain | well... even if it isn't billing related |
| 14:36.50 | stbain | most of the information you're probably looking for is used in the billing-related processes |
| 14:37.59 | jdv79 | i want to know about call quality vs load and stuff like that |
| 14:38.08 | jdv79 | i'm not sure how that's billing related |
| 14:38.09 | *** join/#asterisk marsas (n=asfdsdaf@85.206.143.101) |
| 14:38.42 | *** part/#asterisk marsas (n=asfdsdaf@85.206.143.101) |
| 14:39.04 | Chotaire | ok I am entirely fed up.. downgrading for good. |
| 14:40.35 | _DAW | hello |
| 14:41.06 | *** join/#asterisk CRCC (n=crc@avc.proxy.astra-net.com) |
| 14:43.19 | *** join/#asterisk insurin1 (n=root@82-42-19-166.cable.ubr01.knor.blueyonder.co.uk) |
| 14:44.32 | CRCC | ping |
| 14:44.36 | CRCC | !ping |
| 14:45.12 | jdv79 | is pbx_realtime.c related to the -p flag? |
| 14:45.20 | tzafrir_laptop | no |
| 14:45.27 | *** join/#asterisk Gennaro (n=Gennaro@ppp-62-11-87-247.dialup.tiscali.it) |
| 14:45.29 | Gennaro | hi |
| 14:45.34 | jdv79 | just a guess... |
| 14:45.36 | fugitivo | any idea why mixmonitor and monitor is not leaving any file in /var/lib/asterisk/monitor? |
| 14:45.57 | Gennaro | someone can help me with the 1° installation of a phon in sip.conf |
| 14:46.07 | Gennaro | someone can help me with the 1° installation of a phon in extension.conf |
| 14:46.07 | fugitivo | i don't get any errors |
| 14:46.08 | Gennaro | ? |
| 14:46.18 | tzafrir_laptop | the -p flag is to run asterisk with real-time priority: that is: if asterisk needs to do something than it has top priority over everything but the kernel |
| 14:46.23 | Chotaire | downgraded.. and everythign seems working. |
| 14:46.55 | fugitivo | tzafrir_laptop: should i use that? |
| 14:47.47 | tzafrir_laptop | fugitivo, generally: yes. it gives a batter sound quality. however be aware that this means that if asterisk is in a 100% cpu loop, the system is hung |
| 14:48.23 | tzafrir_laptop | in my testing box I have an extension that runs Segfault (there is such an app, from bristuff) |
| 14:49.15 | tzafrir_laptop | It has proved useful for those cases. But I figure that you might as well control asterisk through the manager interface, because it uses nothing but asterisk and the kernel |
| 14:49.50 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 14:49.50 | *** mode/#asterisk [+o anthm] by ChanServ |
| 14:49.52 | Gennaro | when i do |
| 14:49.54 | Gennaro | [4035] |
| 14:49.54 | Gennaro | type=friend |
| 14:49.54 | Gennaro | contex=from-sip |
| 14:49.54 | Gennaro | fromuser=200 |
| 14:49.54 | Gennaro | callerid=200 <200> |
| 14:49.55 | Gennaro | host=192.168.50.42 |
| 14:49.57 | Gennaro | nat=no |
| 14:49.59 | Gennaro | canreinvite=yes |
| 14:50.01 | Gennaro | dtmfmode=info |
| 14:50.03 | Gennaro | incominglimit=1 |
| 14:50.05 | Gennaro | in sip |
| 14:50.09 | tzafrir_laptop | Gennaro, when you do that, you flood the channel. |
| 14:50.11 | tzafrir_laptop | ~pb |
| 14:50.13 | jbot | hmm... pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca/ |
| 14:50.21 | *** part/#asterisk Vhata (i=[U2FsdGV@platform.adept.co.za) |
| 14:50.21 | Gennaro | ops.. |
| 14:50.26 | Katty | Gennaro: (= |
| 14:50.40 | *** join/#asterisk areski (n=areski@polar.es6.egwn.net) |
| 14:50.50 | tzafrir_laptop | and he did that exactly after someone got an op |
| 14:51.05 | Katty | so? |
| 14:51.08 | Katty | it's not like he knew |
| 14:51.16 | tzafrir_laptop | NM |
| 14:51.36 | fourcheeze | Gennaro: what's the question? |
| 14:51.58 | fugitivo | Gennaro: please describe your problem if you want to get any help |
| 14:52.03 | Gennaro | i installed now asterisk and .. |
| 14:52.11 | Gennaro | i have an ip phone |
| 14:52.14 | tzafrir_laptop | Gennaro, you have a typo there: s/contex=/context=/ |
| 14:52.25 | Katty | context=from-sip |
| 14:53.06 | Gennaro | so i read.. thati i need to know how can i do to register a telephone to try an incoming and outcoming call |
| 14:53.10 | Gennaro | thats all |
| 14:53.36 | Gennaro | i see that i need to do something like this in sip config |
| 14:53.43 | Gennaro | and so i do |
| 14:53.53 | Gennaro | now?? |
| 14:55.42 | Gennaro | so .. i read a definition for any of these sip clients in sip.config enalbes logins and call to the server asterisk from clients. |
| 14:56.00 | *** join/#asterisk juanjoc (n=juanjoc@40-115-235-201.fibertel.com.ar) |
| 14:56.11 | *** join/#asterisk b0ngfrog (n=winston@c-67-190-81-76.hsd1.co.comcast.net) |
| 14:57.50 | Gennaro | fugitivo, ?? |
| 15:05.46 | b0ngfrog | morn |
| 15:06.18 | fugitivo | yes? |
| 15:07.01 | fugitivo | Gennaro: you setup the registration of your sip devices in sip.conf, then you need to edit your dial plan (extensions.conf) |
| 15:08.29 | Chotaire | please let me ask my most important question again, maybe someone's awake who knows the answer.. on incoming sip calls (no matter which SIP provider I use) i have a 50% chance that the person who calls me (from pstn -> voip number) will not hear anything.. on the next call he will, on the next call he won't, on the next call he will... |
| 15:08.35 | Chotaire | does anyone know what might be going wrong here? |
| 15:08.58 | Chotaire | I have had the same problems with asterisk 1.03 and 1.2.0 so I assume I have a hidden misconfiguration somewhere. |
| 15:09.13 | jdv79 | how can i tell * to not retry a failed call |
| 15:09.31 | jdv79 | cause its rapidly retrying on a single fail and flooding my gw |
| 15:09.42 | b0ngfrog | can you hear them??? |
| 15:09.46 | Chotaire | yes |
| 15:09.49 | tzanger | jdv79: with callfiles? |
| 15:09.57 | jdv79 | how'd you guess |
| 15:10.02 | tzafrir_laptop | Chotaire, is there a NAT involved? |
| 15:10.05 | jdv79 | i did set retry something to 0 though |
| 15:10.09 | tzanger | jdv79: there is a RetryTime |
| 15:10.12 | Chotaire | no, I use a static ip on the * box |
| 15:10.13 | tzanger | and a MaxRetries |
| 15:10.16 | Chotaire | and I have been trying with and without nat=yes |
| 15:10.30 | jdv79 | actually - no - this is another box, sorry |
| 15:10.38 | jdv79 | its just a Dial |
| 15:10.49 | b0ngfrog | Chotaire all ports between 10000 -20000 are open to the * box?? |
| 15:10.49 | tzanger | well something is causing the dial to retry |
| 15:10.53 | tzanger | dial does not redial on its own |
| 15:10.58 | jdv79 | ok |
| 15:11.10 | fugitivo | Chotaire: it looks like a nat or rtp problem |
| 15:11.18 | jdv79 | i just have an extension mapped directly to a dial |
| 15:11.22 | *** join/#asterisk Sedorox (i=brandon@smartserv/cna/Sedorox) |
| 15:11.24 | jdv79 | call it and it calls out |
| 15:11.27 | jdv79 | that sorta thing |
| 15:11.44 | Chotaire | on that ip: yes |
| 15:12.17 | Chotaire | fugitivo: this is really strange, I dont have this problem with outgoing calls. |
| 15:12.33 | *** join/#asterisk emakris2 (n=emakris@c-24-128-56-2.hsd1.ma.comcast.net) |
| 15:12.36 | jdv79 | i'm probably doing something stupid |
| 15:12.46 | fugitivo | Chotaire: is your * behind nat? |
| 15:12.48 | Chotaire | nope |
| 15:12.50 | Chotaire | static ip |
| 15:13.11 | tzanger | jdv79: again, * will receive a call on that exten and the dialplan will Dial() the other... unless you have a loop it will not retry |
| 15:13.19 | *** join/#asterisk moo[work] (n=moo@82.102.1.42) |
| 15:13.27 | moo[work] | hi, can i use includes in sip.conf? |
| 15:13.41 | Chotaire | when using the fritzbox 7050, it will work (behind a nat).. when using * without a nat, it won't work. |
| 15:13.46 | jdv79 | poking around... |
| 15:14.48 | fugitivo | Chotaire: are you using canreinvite? |
| 15:15.10 | Chotaire | no |
| 15:15.23 | b0ngfrog | so canreinvite=no ?????? |
| 15:15.29 | Chotaire | that's correct |
| 15:15.43 | Chotaire | for all sip providers. |
| 15:15.52 | *** join/#asterisk sahafeez (n=sahafeez@67.109.14.227.ptr.us.xo.net) |
| 15:15.57 | Chotaire | this is happening out of nowhere without a reason. it always worked before. |
| 15:16.01 | Chotaire | anyway I can debug this? |
| 15:16.15 | b0ngfrog | how about for your sip phones??? |
| 15:16.18 | *** join/#asterisk mhnoyes (n=mhnoyes@user-2ivfiur.dialup.mindspring.com) |
| 15:16.23 | Chotaire | they have nothing to do with it. |
| 15:16.33 | Chotaire | asterisk will take the voice call, it will "Answer" |
| 15:16.55 | Inkubot | :) |
| 15:16.58 | juanjoc | Chotaire: Are you sure that you have enough bandwidth? |
| 15:17.04 | Chotaire | oh yes ;) |
| 15:17.10 | *** part/#asterisk insurin1 (n=root@82-42-19-166.cable.ubr01.knor.blueyonder.co.uk) |
| 15:17.30 | juanjoc | Chotaire: Something similar happens to me when the network is overloaded. |
| 15:17.42 | b0ngfrog | again Does this happen with sip sets or just trunks |
| 15:17.45 | Chotaire | that's impossible, I have 5% load on my link |
| 15:17.57 | Chotaire | only with incoming calls through sip |
| 15:17.58 | Chotaire | not with pstn |
| 15:18.22 | Chotaire | it happens with all configured sip providers. |
| 15:18.33 | Chotaire | sometimes it works, sometimes it doesnt.. chances are pretty much 50% |
| 15:18.35 | b0ngfrog | again Does this happen with sip sets or just sip trunks |
| 15:18.46 | Chotaire | can you explain the difference? |
| 15:19.17 | Chotaire | give me a hint and I'll tell you everything you need to know |
| 15:19.33 | b0ngfrog | Chotaire - do you have any 'local' sets that connect to this * box? This way you can tell if it is a network issue or config.. |
| 15:20.13 | *** join/#asterisk Samoied (n=Samoied@popeye.opens.com.br) |
| 15:20.51 | Chotaire | it's actually not a network issue because the * box is directly connected to the internet. it can be an issue with the inet link though. I get a few maximum retries warnings here and there. |
| 15:21.29 | b0ngfrog | Any firewall (software or hard) between the * and inet?? |
| 15:21.39 | Chotaire | on the * box itself |
| 15:21.45 | Chotaire | iptables |
| 15:21.55 | b0ngfrog | are you running webmin on that box?? |
| 15:21.59 | Chotaire | hehe, nope. |
| 15:22.03 | Chotaire | I am not a mouse-clicker. |
| 15:22.12 | Chotaire | you referring to 10000/tcp I assume |
| 15:23.04 | Chotaire | tcp 0 0 127.0.0.1:10024 0.0.0.0:* LISTEN 0 1140415398 1279/amavisd (maste |
| 15:23.04 | Chotaire | tcp 0 0 127.0.0.1:10025 0.0.0.0:* LISTEN 0 1792423172 13758/master |
| 15:23.10 | Chotaire | that's all within that range, and on localhost only ;) |
| 15:23.45 | moo[work] | ppl, how does #include works in sip.conf? |
| 15:24.01 | moo[work] | can i do something like #include foo/bar.conf ? |
| 15:24.22 | Chotaire | if there was anything blocked, I would get syslog messages. |
| 15:25.53 | Chotaire | also, there is no error message in asterisk console (running -vvvvvvvvvvvvvc) when this error happens... maximum retries warning only happens sometimes, it doesnt seem related. |
| 15:26.26 | Chotaire | is it possible this can be a configuration issue in extensions.conf? |
| 15:26.32 | b0ngfrog | And you said this just started to happen?? |
| 15:26.56 | Chotaire | well, today I found out while testing my new sip provider. |
| 15:27.35 | Chotaire | I am sure it wasn't happening before, I would have found out. |
| 15:27.48 | Chotaire | or my customers would've told me. |
| 15:28.20 | b0ngfrog | i would sip debug and find a call that works and doesnot work ... compare rtp port numbers etc |
| 15:28.59 | b0ngfrog | what codec are you running to the providers?? |
| 15:31.16 | Chotaire | I allow alaw, ulaw, ilbc, gsm, g726 |
| 15:32.12 | *** join/#asterisk jcath (n=skycat@61.51.71.84) |
| 15:32.37 | Chotaire | let me restrict to ulaw for testing... |
| 15:33.06 | *** join/#asterisk sthbad (i=vladi@ggbit-peers.cores.bg) |
| 15:33.13 | sthbad | hello, i need some help |
| 15:33.39 | iDunno | really? |
| 15:33.41 | sthbad | asterisk prints this error "Unable to find a path from ulaw to g723" |
| 15:34.01 | iDunno | is 723 the patent encumbered one? |
| 15:34.01 | sthbad | when i'm trying to call voicemail, trying to hold a call, or trying to listen a music |
| 15:34.33 | Abbas | sthbad: asterisk does not support g723 other than passon |
| 15:34.36 | Abbas | pass on |
| 15:34.41 | *** join/#asterisk MuppetMaster (n=MuppetMa@62.37.169.64) |
| 15:34.53 | MuppetMaster | Hello |
| 15:35.07 | MuppetMaster | Anyone have any experience in a mulit-lingual vmail application on 1.2? |
| 15:35.23 | sthbad | Abbas, what can i use? |
| 15:35.41 | Chotaire | b0ngfrog: I have recorded sip debug logs of both working and non-working attempts with restriction set to ulaw. let me wander through it. |
| 15:35.44 | sthbad | can i see an example configuration of workin MusicOnHold ? |
| 15:35.53 | Abbas | sthbad: ULAW , GSM, ILBC |
| 15:36.35 | sthbad | i have to set this in voicemail.conf ? |
| 15:36.43 | sthbad | format=ulaw,gsm,ilbc |
| 15:37.21 | Chotaire | b0ngfrog: I see no difference. |
| 15:38.07 | fugitivo | anyone registered asterisk to a softswitch huawei softx3000? |
| 15:38.23 | jcath | friends, what is the effort to put VPN between 2 voip pbxs? |
| 15:38.33 | _Sam-- | on this line, the requested format=4 is an audio codec? AUTHENTICATED call from 192.168.0.10, requested format = 4 |
| 15:38.55 | *** part/#asterisk MuppetMaster (n=MuppetMa@62.37.169.64) |
| 15:39.07 | jcath | the latency or anything else? |
| 15:41.06 | sthbad | Abbas, in iax.conf there is allow option for formats |
| 15:41.19 | sthbad | did i have to disallow g723 format, now allow=all |
| 15:42.12 | Chotaire | b0ngfrog: absolutely no difference besides that the non-working one is at port 10426 and the working one is at port 17980 (peer audio rtp) |
| 15:42.34 | Chotaire | I can try to find out if the ip range has anything to do with it by retrying... |
| 15:43.08 | b0ngfrog | Chotaire - ip or port range??? |
| 15:43.18 | Chotaire | port range ofcourse ;) |
| 15:43.21 | Chotaire | Peer audio RTP is at port 217.10.67.4:10426 |
| 15:43.25 | Chotaire | that's udp, right? |
| 15:43.29 | b0ngfrog | yes |
| 15:43.39 | jdv79 | Auto fallthrough, channel 'SIP/10.8.6.3-0078dcb0' status is 'CONGESTION' |
| 15:43.47 | *** join/#asterisk funxion (n=nunya@host-64-110-51-254.hlm.ses-americom.net) |
| 15:43.51 | jdv79 | would that cause a retry normally? |
| 15:43.58 | funxion | hello |
| 15:44.07 | jdv79 | i'm not sure exactly why its happening still |
| 15:44.34 | funxion | does anyone happen to have any experience interconnecting a mitel sx2000 with universal NSU to a * box via E1 using euroisdn? |
| 15:44.56 | *** join/#asterisk Sobakai (n=jmwoodga@45.e6.d12c.cidr.airmail.net) |
| 15:45.33 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 15:45.51 | *** join/#asterisk Ariel_ (n=Ariel@adsl-068-157-125-248.sip.mia.bellsouth.net) |
| 15:45.57 | Ariel_ | morning folks |
| 15:46.03 | funxion | morning |
| 15:46.16 | jcath | evening :P |
| 15:46.20 | msw | quick question - how do I force a hangup on a zap channel that's stuck in a IVR menu? |
| 15:46.42 | Ariel_ | soft hangup |
| 15:46.49 | Flauto | good moring ariel |
| 15:46.53 | Ariel_ | then the channel number |
| 15:47.15 | msw | thanks |
| 15:48.33 | *** join/#asterisk limepickle (n=chris@193.128.28.34) |
| 15:49.33 | limepickle | i now have a link betwwen my asterisk and bp250, could use a little help to get a call through |
| 15:49.58 | Katty | Ariel_: :> |
| 15:50.16 | Katty | Ariel_: how's the family? (= |
| 15:50.22 | *** join/#asterisk Nico_Bdav (n=nico@shm67-2-82-227-191-194.fbx.proxad.net) |
| 15:50.28 | Ariel_ | Katty, hope everything is going well. Family is doing great thanks |
| 15:50.38 | Katty | Ariel_: all is good on this in (= |
| 15:50.42 | Katty | s/in/end/ |
| 15:50.46 | funxion | Ariel_ do you have any experience interconnecting a mitel sx2000 with universal NSU to a * box via E1 using euroisdn? |
| 15:51.02 | file | cookie? |
| 15:51.05 | Ariel_ | funxion, no |
| 15:51.10 | funxion | damn |
| 15:51.12 | Katty | file: i have no cookies. |
| 15:51.12 | funxion | thnx |
| 15:51.24 | Katty | file: i do have 24 albums of christmas though |
| 15:51.25 | file | that makes me sad |
| 15:51.28 | limepickle | funxion, you have any experience with te110p cards? |
| 15:51.33 | Katty | and 44oz of caffeine |
| 15:51.34 | file | Katty: ooh may I have one?!? |
| 15:51.38 | Katty | file: possibly. |
| 15:52.13 | file | but, that's not an answer! |
| 15:52.13 | funxion | limepickle yes |
| 15:52.13 | limepickle | ooh |
| 15:52.13 | Katty | file: mew?! |
| 15:52.13 | limepickle | could i possibly pick your brains? |
| 15:52.21 | file | limepickle: that might hurt |
| 15:52.21 | funxion | you can try but there is not much there |
| 15:52.22 | limepickle | hehe |
| 15:52.46 | Katty | file: what? you want me to post it all somewhere for you? |
| 15:52.52 | Katty | file: so you can be lazy and wget it all |
| 15:52.59 | limepickle | ok, whats a steady red led? I'm connecting to a BP250 and the BP250 seems to think the link is ok, but I'm getting steady red on the Asterisk TE110p |
| 15:53.03 | Katty | file: after i spent /hours/ ripping it from my collection? |
| 15:53.10 | *** join/#asterisk santoshr (i=1063@203.199.110.93) |
| 15:53.12 | funxion | hmm |
| 15:53.17 | limepickle | using E! |
| 15:53.21 | limepickle | *E1 |
| 15:53.22 | funxion | well red means no sync |
| 15:53.23 | limepickle | soory |
| 15:53.31 | limepickle | ok |
| 15:53.32 | santoshr | is it true tht asteriskv1.2 is requried for call transfer |
| 15:53.35 | file | Katty: yes. |
| 15:53.37 | fugitivo | limepickle: bad cable |
| 15:53.45 | santoshr | supervised call transfer tht is |
| 15:54.07 | limepickle | really? Its just that the Ericsson side says status 20 which is a good status I believe |
| 15:54.09 | funxion | if you line coding and framing all match and your cable is good check your clock source |
| 15:54.11 | Katty | file: k, moment |
| 15:54.25 | testmachine | hm |
| 15:54.27 | testmachine | woei |
| 15:54.31 | testmachine | everything works now |
| 15:54.34 | testmachine | what has to work here |
| 15:54.36 | testmachine | hah |
| 15:54.37 | *** join/#asterisk Corydon-w (i=grey@pdpc/supporter/sustaining/Corydon76-home) |
| 15:54.41 | *** join/#asterisk DarthClue (i=user76@wsip-68-99-73-32.tu.ok.cox.net) |
| 15:54.49 | testmachine | next week adding it to the phone centre |
| 15:54.53 | Katty | DarthClue: hi asl. |
| 15:54.54 | santoshr | hey file: please correct me if i am wrong. |
| 15:54.55 | limepickle | ok, where are those defined, I'm a newbie |
| 15:55.05 | santoshr | call transfer only works with astierk 1.2 |
| 15:55.14 | DarthClue | any chance somebody would be willing to fix ChanIsAvail? |
| 15:55.21 | Katty | DarthClue: no |
| 15:55.24 | Katty | DarthClue: go home |
| 15:55.29 | *** join/#asterisk r0d3nt (i=nobody@wsip-24-234-241-145.lv.lv.cox.net) |
| 15:55.33 | *** join/#asterisk Gennaro (n=Gennaro@ppp-62-11-85-238.dialup.tiscali.it) |
| 15:55.36 | Gennaro | hi |
| 15:55.49 | Katty | DarthClue: don't mind me, i'm grumpy in the morning /as usual/ |
| 15:55.49 | Gennaro | some one can help me with 1 call? |
| 15:55.54 | file | but I do know oej did some good call transfer changes in chan_sip |
| 15:56.06 | Gennaro | i tried to registred in sip.config and in extension.conf |
| 15:56.06 | *** join/#asterisk tracinet (n=tracinet@24-50-29-205.atlsfl.adelphia.net) |
| 15:56.20 | Gennaro | but i have erroe in asterisk |
| 15:56.50 | santoshr | lme: u aroung |
| 15:56.51 | kippi | I have incoming call setup, the extension is logged on but i get this error -- Extension '' in context 'default' from '' does not exist. Rejecting call on channel 0/1, span 1 |
| 15:56.54 | limepickle | excuse me funxion, do you know of any good tutorials or docs on this matter? |
| 15:57.14 | funxion | not really |
| 15:57.20 | kippi | any ideas |
| 15:57.27 | funxion | call digium for support its free for installation |
| 15:57.48 | Gennaro | ? |
| 15:58.17 | limepickle | ok, thanks. I'm actually awaiting an email from a guy called Kenny, I'm just impatient ;) |
| 15:58.53 | *** join/#asterisk wunderkin (i=kev@12-201-105-27.client.mchsi.com) |
| 15:59.22 | Damin | Oh my god! They killed Kenny! |
| 15:59.28 | Damin | You bastards! |
| 15:59.33 | fugitivo | lol |
| 15:59.41 | tracinet | hello all - been using official 1.2.0 for a day now and have found an issue when accepting incoming calls from Broadvox (incomng call works fine on preview 1.0.X version but does not seem to successfully send an "answer" on the 1.2.0 release) - i have captured the calls both on the old version which works and the new version that fails and have the snippet that is different between the 2 if anyone is willing to look at it |
| 16:00.04 | file | tracinet: pastebin. |
| 16:00.11 | Damin | tracinet: Can you post it to pastebin.ca please? |
| 16:00.17 | chapeaurouge | would 5090-5100 and 10000-20000 the ports range to be open when behind a NAT? |
| 16:00.22 | tracinet | sorry - bear with me - like my 3rd time on irc |
| 16:00.35 | tracinet | not sure i know how to do that |
| 16:00.41 | limepickle | so does anyone have any good docs on a TE110P at all? |
| 16:00.42 | Damin | tracinet: You might want to take this to #asterisk-dev |
| 16:00.43 | *** join/#asterisk santiago (n=santiago@208.195.215.124) |
| 16:00.48 | file | chapeaurouge: 5060 UDP and 10000-20000 UDP for SIP and RTP, 4569 UDP for IAX2 |
| 16:00.59 | chapeaurouge | file, thx |
| 16:01.04 | limepickle | please? |
| 16:01.11 | tracinet | Damin - thanks will do |
| 16:01.14 | santoshr | so file u r tellin me tht it was not supported with asterisk v1.0.x |
| 16:01.24 | Damin | tracinet: pastebin.ca is a website that lets you post a text snippet, and then we can all review it and see what the difference is.. |
| 16:01.39 | tracinet | ah - ok - will check that our |
| 16:01.48 | Damin | tracinet: This avoids blasting it into the channel.. ;) |
| 16:01.54 | file | which can be rude |
| 16:02.05 | tracinet | good thing i asked first :) |
| 16:02.11 | *** join/#asterisk naif (n=User@host250-27.pool62110.interbusiness.it) |
| 16:02.12 | naif | hello |
| 16:02.21 | *** join/#asterisk Leob (n=chatzill@pool-70-20-22-11.bstnma.fios.verizon.net) |
| 16:03.22 | limepickle | hmm, ok |
| 16:03.30 | blop | i'm looking for a up2date debian repository with asterisk 1.2.0 on it :) if someone knows an url :) |
| 16:03.44 | Leob | Asterisk 1.2 help needed: getting error "Ouch ... error while writing audio data: : Broken pipe" when running asterisk -v |
| 16:03.57 | limepickle | i get that too |
| 16:04.01 | file | that's perfectly normal |
| 16:04.34 | limepickle | TE110P docs anyone? |
| 16:04.39 | *** join/#asterisk Utah_Dave (n=boucha@0-1pool139-17.nas28.salt-lake-city1.ut.us.da.qwest.net) |
| 16:04.55 | jcath | hey, will asterisk support srtp? |
| 16:05.04 | file | jcath: 'tis on the bug tracker, so maybe |
| 16:05.31 | santoshr | has anybody ever user call transfer with asterisk v1.0.x |
| 16:05.36 | Corydon-w | limepickle: what type of docs? |
| 16:05.39 | mishehu | bah. anybody else having a symbol problem with codec_gsm? I saw somebody posted something to the list, but it had no replies to it. |
| 16:05.45 | mishehu | in 1.2.0 that is |
| 16:05.47 | fugitivo | i'm using mixmonitor for outgoing zap calls, i see "executing mixmonitor...." on the CLI, but i don't see any file under /var/lib/asterisk/monitor (this is the path set in asterisk.conf) |
| 16:06.06 | Leob | I'm also getting "undefined symbol: ast_parseoptions" ... any ideas on what is going on? |
| 16:06.08 | limepickle | corydon, anything relating to links, or anythnig |
| 16:06.14 | mishehu | good kitty kat, don't tear up the furniture |
| 16:06.22 | Katty | mishehu: oh, puh-lease. |
| 16:06.34 | Katty | mishehu: your bird's the one that likes to chew on the couch |
| 16:06.36 | Corydon-w | limepickle: Just the zaptel kernel driver |
| 16:06.57 | mishehu | Katty: it didn't successfully chew on it. I got it back to the cage before it did any. |
| 16:07.14 | Katty | mishehu: we obviously know who the dangerous one is. |
| 16:07.23 | Katty | mishehu: i just claw on humans. |
| 16:07.37 | jcath | so, file, any open source voip client support srtp now? |
| 16:07.40 | mishehu | Katty: next you'll tell me you're a vegan kat, and won't eat my bird. |
| 16:07.48 | Katty | mishehu: obviously. |
| 16:07.56 | Katty | mishehu: but the birdseed might look good. |
| 16:08.06 | file | jcath: Goooooooogle might know |
| 16:08.07 | iDunno | vegan cat - poor thing. |
| 16:08.18 | jcath | ok':P |
| 16:08.26 | mishehu | Katty: and I'll then start accepting applications from wolves to shepherd sheep... |
| 16:08.31 | Katty | iDunno: cats can't be vegan |
| 16:08.38 | Katty | iDunno: they require an animal source for health purposes. |
| 16:08.42 | *** join/#asterisk shmooz (n=shmooz@H142.C72.B0.tor.eicat.ca) |
| 16:08.54 | mishehu | because they'll say "but we're vegans too, just like that Kat" |
| 16:08.55 | Katty | mishehu: not falling for that one |
| 16:09.02 | iDunno | heh. |
| 16:09.24 | Katty | animals are friends - not food! |
| 16:09.30 | Katty | </bruce> |
| 16:09.35 | fugitivo | Katty: vegetarian? |
| 16:09.40 | limepickle | no probs coryden, i just want to link my BP250 to asterisk via E1 |
| 16:09.42 | Katty | fugitivo: vegan. |
| 16:09.55 | fugitivo | Katty: nice |
| 16:10.00 | Katty | fugitivo: i'm always nice. |
| 16:10.10 | Katty | fugitivo: except in the morning, when i attempt to steal file's orange juice |
| 16:10.18 | fugitivo | lol |
| 16:10.19 | syle | ohh god i hate vegans , they are like deformed creatures with no meat on their bones from not eating meat, most of them not mentally all there either missing minerals |
| 16:10.20 | file | minor detail |
| 16:10.27 | mishehu | Katty: I thought it was "Fish are friends, not food..." |
| 16:10.33 | Katty | mishehu: oh yeah. |
| 16:10.37 | mishehu | and that sharks said that |
| 16:10.38 | Katty | mishehu: same thing, only different. |
| 16:11.05 | mishehu | Katty: yep yep yep yep yep uh-huh. |
| 16:11.11 | Katty | ! |
| 16:11.13 | Katty | ALIENS |
| 16:11.37 | file | Katty: Carol of the Bells! |
| 16:11.46 | Katty | file: no |
| 16:11.49 | Katty | that's the muppets, silly boy |
| 16:11.51 | file | Katty: yes. |
| 16:11.52 | file | nope |
| 16:12.00 | mishehu | Katty: what a memory |
| 16:12.01 | Katty | john denver and the muppets |
| 16:12.20 | Katty | mishehu: that's the name of a scrapbook store out here |
| 16:12.26 | Katty | mishehu: oh dear, what a memory |
| 16:12.52 | mishehu | Katty: although I still remember some of the episodes of sesame st. that I saw when I was a kid.... |
| 16:12.53 | *** join/#asterisk lehel (n=asd@82.79.20.17) |
| 16:12.55 | lehel | hello |
| 16:13.08 | Katty | mishehu: and it's sad i still recognize it ;) |
| 16:13.09 | mishehu | yet I can't remember where I just placed the screwdriver... |
| 16:13.13 | fishboy1669 | muppets way out do seseme st |
| 16:13.18 | Katty | mishehu: back pocket |
| 16:13.27 | fishboy1669 | did u guys have bagpuss over there |
| 16:13.33 | fishboy1669 | and trumpton |
| 16:13.34 | mishehu | Katty: I'd know if it was in the back pocket, as it'd poke me in the butt. |
| 16:13.36 | Katty | mishehu: that's where i leave mine, and then end up sitting on it >.< |
| 16:13.38 | fishboy1669 | camblewick green |
| 16:14.07 | fourcheeze | camberwick |
| 16:14.13 | mishehu | Katty: one could argue that leaving the screwdriver in your back pocket is rather freudian... |
| 16:14.20 | file | I'm dreaming of a white Christmas! |
| 16:14.30 | fishboy1669 | get some white paint then |
| 16:14.32 | fishboy1669 | ;-) |
| 16:14.35 | *** join/#asterisk shmaltz (n=chatzill@69.28.255.210) |
| 16:14.36 | Katty | file: like the ones you used to know? |
| 16:14.45 | tzanger | ohh man don't do that to me |
| 16:14.49 | Katty | mishehu: hey! |
| 16:14.52 | file | OKAY, where are you... the guy who was talking about the broken pipe error |
| 16:14.54 | tzanger | now I miss so very much my grandmother's potato pancakes |
| 16:14.54 | fishboy1669 | katty better than leaving itin your front pocket |
| 16:14.56 | Katty | mishehu: don't throw perfectly good food at file! |
| 16:14.59 | file | Leob: you |
| 16:15.02 | tzanger | fuck now I'm gonna have to have that for breakfast tomorrow |
| 16:15.09 | Katty | fishboy1669: nothing fits in my front pocket. |
| 16:15.14 | Katty | fishboy1669: they are /female/ jeans |
| 16:15.15 | mishehu | Katty: didn't know what else to throw at him... a Hanukiyah would hurt too much... |
| 16:15.28 | tzanger | my grandma's potato pancakes were always the best because she grated them by hand, my mom used a food processor and the texture was totally different |
| 16:15.29 | Katty | fishboy1669: besides, it'd be stabbing me in the hipbone. oww. |
| 16:15.32 | file | Leob: when you do a make install of asterisk, it specifically tells you that some modules are like, not supposed to be there - and guess what, app_muxmon.so doesn't exist anymore so you should have deleted it :) |
| 16:15.39 | file | it's been replaced |
| 16:15.42 | mishehu | tzanger: did grandma put applesauce on them? |
| 16:15.53 | tzanger | fuck no |
| 16:15.54 | Katty | mishehu: how a big burnt piece of falafel? |
| 16:15.59 | tzanger | sugar and maple syrup |
| 16:16.05 | tzanger | well sugar OR maple syrup |
| 16:16.09 | fishboy1669 | katty whats so different about femail jeans? ive never bought a pair myself so wouldnt know |
| 16:16.11 | tzanger | applesauce is a desert |
| 16:16.13 | mishehu | tzanger: eww |
| 16:16.17 | shmaltz | anybody here has a 406 or 411 digium card? |
| 16:16.20 | tzanger | I have no idea how people put it on food like pancakes and ham |
| 16:16.24 | tzanger | shmaltz: yes |
| 16:16.34 | shmaltz | tzanger, did it help for echo? |
| 16:16.49 | fishboy1669 | god im hungry now |
| 16:16.52 | fishboy1669 | you bad people |
| 16:16.58 | shmooz | argh me 2 |
| 16:17.23 | Katty | fishboy1669: nothing fits in the front pockets. |
| 16:17.28 | shmooz | almost as contangeous as yawning in a train |
| 16:17.31 | Katty | fishboy1669: usually, not even hands, because female jeans usually tend to hug the hips. |
| 16:17.45 | Katty | file: i dumped another album on the server |
| 16:18.00 | tzanger | shmaltz: it worked about as well as KB1 |
| 16:18.06 | fishboy1669 | thought hips were on the sides not front? |
| 16:18.08 | tzanger | but MG2 is so much better so I stick with it |
| 16:18.26 | shmaltz | tzanger, does echocancel=32 in zapata.conf have any effect? |
| 16:18.37 | Katty | fishboy1669: go find a pair of female jeans |
| 16:18.40 | tzanger | shmaltz: no |
| 16:18.53 | tzanger | echocancel of anything other than 'no' uses the VPM if available, and the VPM is not adjustable |
| 16:18.54 | Katty | fishboy1669: hipbones are very much in front |
| 16:19.04 | *** part/#asterisk oej_ (n=Olle@apollo.webway.se) |
| 16:19.15 | fishboy1669 | mmm next girlfriend i get ill have to check that theory |
| 16:19.18 | shmaltz | tzanger, it's not helping me in any way, some old teriible echo :( |
| 16:19.21 | Katty | fishboy1669: k |
| 16:19.24 | *** join/#asterisk Gennarox (n=Gennaro@ppp-62-11-86-57.dialup.tiscali.it) |
| 16:19.26 | Katty | fishboy1669: and it's not a theory |
| 16:19.28 | *** join/#asterisk Chotaire (i=chotaire@chotaire.net) |
| 16:19.29 | Katty | fishboy1669: i sorta am a female. |
| 16:19.34 | Gennarox | hi |
| 16:19.38 | tzanger | shmaltz: I'm guessing your echo tail is longer than the card can handle |
| 16:19.40 | Katty | fishboy1669: i'm very well aware where my hipbones are, kthx |
| 16:19.44 | tzanger | Katty: haha |
| 16:19.47 | fishboy1669 | sorta? u dont sound 100% sure lol |
| 16:19.59 | shmaltz | tzanger, meaning the delay? |
| 16:20.06 | fishboy1669 | katty i wanst disputing that u didnt |
| 16:20.16 | Gennarox | some one can helpme to understand the meaning of sip.config, extension.config ?? |
| 16:20.18 | tzanger | shmaltz: yes. if the tail is too long the echo can doesn't try to chop it because it thinks it's the other side talking. (simple explanation) |
| 16:20.23 | Katty | that's funny, someone thinking i'm not a female. |
| 16:20.38 | fishboy1669 | hey i think your femail ist just how u wrote |
| 16:20.43 | Katty | k |
| 16:20.47 | shmaltz | tzanger, is there *any* solution for such a problem? |
| 16:20.57 | fishboy1669 | i sorta am femail sounds like u are not wuite sure |
| 16:21.00 | fishboy1669 | ;-) |
| 16:21.03 | *** join/#asterisk dalabera (n=dalabera@146.82.190.164) |
| 16:21.13 | tzanger | shmaltz: hardware echo can, or modify zaptel source to allow for longer tail lengths at the cost of more CPU |
| 16:21.30 | fishboy1669 | im not disputing u have all the right bits in the right places |
| 16:21.41 | shmaltz | tzanger, which line of the source? |
| 16:21.43 | fishboy1669 | and im not disputing u dont know where you jean pockets are |
| 16:21.58 | fishboy1669 | im a man and never noticed befor thats all |
| 16:22.04 | shmaltz | tzanger, its expressed in ms? or taps? |
| 16:22.32 | tzanger | shmaltz: taps |
| 16:22.36 | tzanger | 1 tap = 1/8ms |
| 16:22.49 | fishboy1669 | just seem to remember stood behind one of my exes with my hands in her front pockets and they were in kinda right place |
| 16:22.50 | shmaltz | and where in the source is it? |
| 16:22.55 | shmaltz | wct4xxp.c? |
| 16:22.55 | tzanger | so 256 is only 32ms |
| 16:23.02 | tzanger | shmaltz: no zaptel proper I think |
| 16:23.30 | fishboy1669 | katty are american jeans different to uk ones? |
| 16:23.45 | Katty | how would i know |
| 16:24.23 | fishboy1669 | sorry stupid question i suppose |
| 16:24.37 | fishboy1669 | never know u may have been shopping for jeans in the uk |
| 16:24.47 | fishboy1669 | just on the off chance |
| 16:26.25 | shmaltz | tzagner, I got this in wct4xxp.c: |
| 16:26.27 | shmaltz | <PROTECTED> |
| 16:26.28 | shmaltz | <PROTECTED> |
| 16:26.30 | shmaltz | <PROTECTED> |
| 16:26.33 | shmaltz | what shouol I modify? |
| 16:27.56 | dalabera | Hello everyone, I've installed 1.2, So far everything was working perfect , except that this morning a receive a call that users weren't able to verify the voicemail |
| 16:28.05 | funxion | does anyone happen to have any experience interconnecting a mitel sx2000 with universal NSU to a * box via E1 using euroisdn? |
| 16:28.15 | dalabera | Using the cisco 7940 |
| 16:28.56 | dalabera | my config on sip.conf is dtmfmode=rfc2833 , disallow all alow=ulaw allow=mlaw |
| 16:29.25 | malverian[work] | Anyone here using Snom 320 phones in production? |
| 16:29.26 | Qwell | mlaw? |
| 16:29.28 | dalabera | I've checked the bug 5780, but this is for g729 |
| 16:29.39 | dalabera | sorry ulaw |
| 16:29.52 | Qwell | you allow ulaw twice? |
| 16:30.33 | dalabera | nope |
| 16:32.29 | fishboy1669 | katty sorry if i offended u i didnt mean to |
| 16:32.46 | *** join/#asterisk shido6 (n=shido6@d221-68-216.commercial.cgocable.net) |
| 16:32.47 | kippi | can anyone help me with this error? |
| 16:32.48 | kippi | -- Extension '' in context 'default' from '' does not exist. Rejecting call on channel 0/1, span 1 |
| 16:32.58 | Gennarox | sorry... how can i register an account / a telephone / to call.... i'll study more,, |
| 16:33.03 | Gennarox | tanx |
| 16:33.10 | Gennarox | bye |
| 16:33.17 | *** join/#asterisk [TK]D-Fender (n=joe@toronto-HSE-ppp4122655.sympatico.ca) |
| 16:34.46 | Katty | fishboy1669: no big (= |
| 16:36.09 | shmaltz | tzagner, I got this in wct4xxp.c: |
| 16:36.10 | shmaltz | /* Setup Echo length (128 taps) */ |
| 16:36.12 | shmaltz | t4_vpm_out(wc, x, 0x022, (echotail >> 8)); |
| 16:36.13 | shmaltz | t4_vpm_out(wc, x, 0x023, (echotail & 0xff)); |
| 16:36.15 | shmaltz | what shouol I modify? |
| 16:37.11 | fishboy1669 | katty cool |
| 16:37.34 | mishehu | bah... did the cadences somehow change in zaptel 1.2.0? |
| 16:39.09 | *** join/#asterisk Kokey (n=Kokey@201.153.63.79) |
| 16:39.09 | mishehu | Katty: I don't know if you remember Tony, the guy who works for me, but he was just stuck out in the lobby for 5 minutes because the distinctive ring from th eintercom didn't pick up, the cadence is somehow different... bah! |
| 16:39.20 | mishehu | all I did was update from 1.2.0b1 to 1.2.0 |
| 16:39.25 | *** join/#asterisk ReD-MaN (i=redman@dhcp-0-2-b3-9a-4a-5b.cpe.quickclic.net) |
| 16:39.41 | Katty | mishehu: you should have had him run to starbucks. |
| 16:39.55 | mishehu | Katty: I'm not a coffee drinker |
| 16:40.05 | iDunno | that's alright, then... |
| 16:40.12 | iDunno | starbucks don't have coffee. |
| 16:40.26 | iDunno | they have something that looks like it *might* possibly be coffee. |
| 16:40.30 | fishboy1669 | lol |
| 16:43.09 | fishboy1669 | god u lot are just plane mean |
| 16:43.17 | mishehu | iDunno: "cup o' sludge" ? |
| 16:43.31 | fishboy1669 | first u make me hungry now im craving a cup of columbian coffee |
| 16:43.35 | *** join/#asterisk A-Tuin|work (n=A-Tuin@nat.office.legend.net.uk) |
| 16:43.37 | iDunno | :) |
| 16:43.43 | iDunno | mishehu: something like that :) |
| 16:43.52 | fishboy1669 | next it will be beer and a nice glas of port or wisky to finnish |
| 16:43.58 | docelmo | Any developers in here that can help me for a few minutes? Im having issues |
| 16:44.03 | *** join/#asterisk shanky (n=shanky@82.159.214.9) |
| 16:44.07 | fishboy1669 | and some czeck icecream |
| 16:44.13 | nfi|ermes | which version of mpg123 do you suggest me ofr CVS version ? |
| 16:44.13 | shanky | hi, good afternoon from spain |
| 16:44.19 | bweschke | docelmo: what's up? |
| 16:44.25 | *** join/#asterisk _Madar (n=tophe@219-84-129-163-adsl-tpe.static.so-net.net.tw) |
| 16:46.26 | iDunno | hmm. port and creamy cheese and crackers would be fantastic :) |
| 16:46.38 | Katty | port? |
| 16:46.44 | Katty | i hope that's a crisp wine |
| 16:46.46 | iDunno | yes, port. |
| 16:46.52 | iDunno | it's a fortified red wine :) |
| 16:46.55 | Katty | crisp wine with creamy cheese |
| 16:48.29 | mishehu | dammit! I don't know why I'm getting these errors, and no responses were posted to the other person who posted to the mailing list... |
| 16:48.31 | mishehu | codec_gsm.so: Undefined |
| 16:48.33 | mishehu | symbol "Short_term_analysis_filteringx" |
| 16:48.43 | mishehu | this is in 1.2.0 |
| 16:48.52 | file | bad build? |
| 16:49.53 | mishehu | file: everything else seems fine except the cadence change that happened with zaptel... |
| 16:50.30 | mishehu | not sure what else would cause a bad build. I built twice and got the same sha1sum for codec_gsm.so on both builds. |
| 16:51.05 | fishboy1669 | katty have u ever tried port its georgouse |
| 16:51.14 | puzzled | anyone know if the Asterisk Business Edition price of $995 is an annually recurring license fee? |
| 16:52.15 | shmaltz | anybody here using a hardware echo can not from digium? |
| 16:52.27 | iDunno | fishboy1669: depends on the port, there is some *really* crap port ;) |
| 16:52.34 | fishboy1669 | true |
| 16:52.37 | iDunno | a decent aged tawny is good, though :) |
| 16:52.55 | fishboy1669 | i went wine tasting past couple of weeks so get to try a few and buy one i like |
| 16:52.57 | puzzled | shmaltz: someone posted on -users today about an EC from Orion |
| 16:52.59 | Katty | fishboy1669: i don't drink. |
| 16:53.03 | fishboy1669 | oh |
| 16:53.07 | shmaltz | puzzled, thanks |
| 16:53.09 | fishboy1669 | katty how old are u |
| 16:53.23 | Katty | fishboy1669: old enough to know better, too young to care. |
| 16:53.29 | fishboy1669 | sorry that was a rude question |
| 16:53.43 | Himeko | not in all cultures |
| 16:53.44 | fishboy1669 | r u old enough to drink is better way of putting it |
| 16:54.05 | Katty | of course. |
| 16:54.11 | fishboy1669 | state side its 21 right |
| 16:54.19 | Katty | depends what country you're in |
| 16:54.20 | fishboy1669 | how come u dont drink? |
| 16:54.31 | Katty | how come you drink? |
| 16:54.40 | fishboy1669 | which contry u in? |
| 16:54.46 | Katty | ... |
| 16:54.49 | fishboy1669 | though it was america your from |
| 16:55.00 | fishboy1669 | cos some drinks are nice tasting |
| 16:55.01 | Katty | this isn't dal.net |
| 16:55.04 | Katty | kthx |
| 16:55.22 | fishboy1669 | what god u got me all wrong |
| 16:55.38 | *** join/#asterisk DrDeke (i=dekemar@deculator.engin.umich.edu) |
| 16:55.41 | fishboy1669 | and sometimes its nice to get drunk |
| 16:55.48 | Katty | if you say so. |
| 16:55.49 | mishehu | file: I guess it was the K6OPT thingie was enabled in the makefile. I disabled it, seems to work fine now. |
| 16:56.30 | fishboy1669 | escapeism the psycologistc call it |
| 16:56.35 | mishehu | heh damn, g729 conversion time < 10 ms |
| 16:56.44 | mishehu | (antyhign to g729) |
| 16:56.46 | Katty | mishehu: is that a good thing or a bad thing? |
| 16:56.54 | mishehu | Katty: the conversion time? |
| 16:56.58 | Katty | mishehu: yes |
| 16:57.03 | iDunno | Katty: you know, you're really missing out, none of the hangovers, none of the money wastage... it's terrible, you're letting down the economy ;) |
| 16:57.07 | mishehu | Katty: definitely a good thing |
| 16:57.10 | Katty | mishehu: kk |
| 16:57.19 | Katty | iDunno: pfft. |
| 16:57.23 | fishboy1669 | idunno lol |
| 16:57.28 | Himeko | i like playing games for escapism |
| 16:57.34 | mishehu | Katty: less time converting between codecs, faster transmission of audio. |
| 16:57.39 | Katty | mishehu: yay |
| 16:57.56 | fishboy1669 | but smaller packet size |
| 16:57.56 | mishehu | Himeko: been doing that too lately... urban terror 3 and kwak4 |
| 16:58.00 | mishehu | err quake4 |
| 16:58.02 | mishehu | heh |
| 16:58.11 | fishboy1669 | lol |
| 16:58.15 | fishboy1669 | poor dunno |
| 16:58.15 | Himeko | q4 sucked |
| 16:58.23 | fishboy1669 | drinking on a school night not good |
| 16:58.24 | Himeko | f.e.a.r. was cool |
| 16:58.25 | mishehu | iDunno: tequila? |
| 16:58.27 | *** join/#asterisk darkskiez (n=darkskie@194.247.78.146) |
| 16:58.28 | *** part/#asterisk juanjoc (n=juanjoc@40-115-235-201.fibertel.com.ar) |
| 16:58.29 | DrDeke | pff school |
| 16:58.35 | fishboy1669 | thats what ports good for as well |
| 16:58.42 | fishboy1669 | port and brandy settle the stomach |
| 16:58.48 | *** part/#asterisk psukogo (n=ctrix@81-174-32-147.f5.ngi.it) |
| 16:59.01 | mishehu | Himeko: well, i'm a bit disappointed in kwak4 too, all I can do is hope that they'll make an urban terror 4... |
| 16:59.02 | fishboy1669 | katty have u ever drunk |
| 16:59.20 | Katty | how can i put this any more clearly |
| 16:59.21 | iDunno | mishehu: nah - it was 3 pints of bitter, 2 double glen livits and a quadruple kahlua (there may have been other stuff, I can't remember :) |
| 16:59.23 | Katty | I Do Not Drink. |
| 16:59.37 | mishehu | fishboy1669: brandy has this sickly sweet taste to it that makes me want to gag |
| 17:00.06 | mishehu | DRINK! |
| 17:00.08 | fishboy1669 | katty have u ever though? |
| 17:00.17 | Himeko | i don't drink alcohol either |
| 17:00.26 | *** join/#asterisk slePP (n=slepp@S0106000f663692da.ed.shawcable.net) |
| 17:00.32 | fishboy1669 | hey im not saying thats a bad thing |
| 17:00.36 | Himeko | or coffee |
| 17:00.43 | fishboy1669 | i spent way to much on it when i was younger |
| 17:00.47 | iDunno | there's lots of people that don't, got no problem with that - they're useful people - specially the ones with licences that will drive you to and from the pub :) |
| 17:00.53 | mishehu | heh, darthclue was around? haven't seen him pop on by lately... or else I've been too busy |
| 17:00.56 | fishboy1669 | but still its nice at times |
| 17:01.22 | iDunno | Himeko: you don't drink coffee?! Woah! scary! do you drink any caffienated substance? |
| 17:01.31 | Katty | mishehu: you silly rabbit. |
| 17:01.35 | Katty | fishboy1669: no. |
| 17:01.37 | Himeko | sure |
| 17:01.38 | Katty | fishboy1669: i do NOT drink |
| 17:01.42 | Himeko | but coffee sucks |
| 17:01.46 | wunderkin | ~seen darthclue |
| 17:01.53 | jbot | darthclue <i=user76@wsip-68-99-73-32.tu.ok.cox.net> was last seen on IRC in channel #asterisk, 1h 6m 39s ago, saying: 'any chance somebody would be willing to fix ChanIsAvail?'. |
| 17:02.00 | fishboy1669 | i dont drink coffee much either |
| 17:02.03 | Himeko | i'd drink alcohol before coffee |
| 17:02.12 | wunderkin | oh yeah he just left heh |
| 17:02.15 | fishboy1669 | just when im really knackered |
| 17:02.23 | fishboy1669 | tea at odd tiems as thats good for u |
| 17:02.32 | fishboy1669 | and water rest of time |
| 17:02.42 | Himeko | i drink alot of tea |
| 17:02.47 | Himeko | and too much dr pepper |
| 17:02.49 | fishboy1669 | plus the odd beverage social |
| 17:03.02 | fishboy1669 | ive cut down on coke and dr pepper |
| 17:03.21 | fishboy1669 | the taste nice and good for enery but mess up your teeth |
| 17:03.26 | *** join/#asterisk RickTick (n=rpulido_@71.196.2.172) |
| 17:03.27 | Himeko | i take in way too much sugar sometimes |
| 17:03.31 | fishboy1669 | chocolate chip |
| 17:03.34 | Himeko | then get all lethargic |
| 17:03.41 | fishboy1669 | mmm |
| 17:03.43 | file | Katty: I don't think anybody can feel the way I feel about you now! |
| 17:03.48 | fishboy1669 | too much suger highs are not good |
| 17:04.09 | fishboy1669 | lol at file |
| 17:04.11 | Katty | file: that's because you've clearly insaned. |
| 17:04.32 | fishboy1669 | katty why u may be really nice |
| 17:04.42 | Katty | oh bother |
| 17:05.27 | fishboy1669 | right youll be glad to hear im going |
| 17:05.29 | fishboy1669 | time for home |
| 17:05.33 | fishboy1669 | have a good weekend |
| 17:07.13 | *** join/#asterisk Juggie (i=agony@CPE00c049d9f271-CM00137186c8d8.cpe.net.cable.rogers.com) |
| 17:09.24 | *** join/#asterisk fugitivo (n=ajf@209.13.244.248) |
| 17:10.38 | fishboy1669 | katty night |
| 17:10.40 | fishboy1669 | night guys |
| 17:10.41 | fishboy1669 | bye |
| 17:11.56 | RickTick | Hi all: anyone here got asterisk loaded on a WRT54GS router? |
| 17:13.30 | *** part/#asterisk darkskiez (n=darkskie@194.247.78.146) |
| 17:15.30 | Beirdo | not me, but I might try it sometime |
| 17:16.28 | Beirdo | I'm curious as to if the WRT54G or GS has the CPU power to successfully transcode ulaw to say G726 or speex or something |
| 17:16.57 | funxion | my thoughts too |
| 17:16.59 | *** join/#asterisk MnDBnDr (n=MnDBnDr@198.234.224.6) |
| 17:17.11 | Beirdo | I'd love to put my iaxy behind one if so |
| 17:17.20 | funxion | I'm wondering is I can do g729 end to end |
| 17:17.21 | RickTick | I visit a company yesterday ... that told me they did got it done. |
| 17:17.22 | file | speex? no |
| 17:17.24 | file | g726? maybe |
| 17:17.27 | Beirdo | sending ulaw from Puerto Rico to Canada... eek. |
| 17:17.39 | Beirdo | how about gsm? |
| 17:17.58 | Beirdo | gsm is reasonable quality for a LOT less bandwidth than ulaw :) |
| 17:18.00 | file | maybe is the answer for most of the middle codecs |
| 17:18.02 | funxion | what about no trancode just g729 on both ends of call |
| 17:18.06 | file | cause I haven't done it |
| 17:18.15 | Beirdo | I understand, file :)_ |
| 17:18.16 | file | if it's passthrough, anything will fly |
| 17:18.30 | funxion | really |
| 17:18.31 | file | as it's just shuffling the same frame through the core, no modifications |
| 17:18.31 | funxion | nice |
| 17:18.48 | Beirdo | yeah, I'd be looking at transcoding. the MIPS should have enough power to maybe do it. Fair enough |
| 17:18.55 | funxion | has there been further testing with the new t38 pass through? |
| 17:19.05 | Beirdo | without trying it, it's impossible to really say for sure, I understand. |
| 17:19.15 | Beirdo | but that's better than "hell no that will never work" :) |
| 17:19.23 | funxion | lol |
| 17:19.45 | Beirdo | like with the speex ) |
| 17:19.46 | Beirdo | heh |
| 17:20.43 | Beirdo | gsm is fine by me anyways. I'm just looking for ways to decrease the bandwidth requirements of the call while still using the iaxy remotely, and I'm NOT setting up a computer with asterisk at my fiancee's parent's house |
| 17:20.48 | *** join/#asterisk tamp4x (n=kkdkdkkk@204.124.238.248) |
| 17:20.54 | Beirdo | swapping out a wrt54G, that's doable :) |
| 17:21.09 | tamp4x | is there a command to hand up calls |
| 17:21.12 | viperdude | does anyone have a suggestion as to how to get a mobile phone as a agent in a queue? |
| 17:21.20 | file | tamp4x: soft hangup <channel name> |
| 17:21.26 | tamp4x | because when i use zap destory channel i cant bring thaat back up unless i restart the channel |
| 17:21.32 | tamp4x | file is that a cli command? |
| 17:21.35 | file | yes, it is |
| 17:21.39 | tamp4x | thanks |
| 17:21.40 | funxion | mobile phone as in cell phone? or wifi phone? |
| 17:21.41 | Beirdo | thanks for the input, file :) |
| 17:21.45 | viperdude | cell phone |
| 17:22.37 | Beirdo | the iaxy is driving us nuts a lot as her cable provider often has seriously limited upstream |
| 17:22.38 | funxion | viperdude you can try agentcallbacklogin |
| 17:23.15 | viperdude | so i set up agentcallbacklogin on a incoming pstn channel? |
| 17:23.22 | viperdude | and dial that from the cell phone? |
| 17:23.45 | mishehu | Beirdo: yeah, iaxy is terribly lacking in codecs |
| 17:23.52 | Beirdo | yup |
| 17:24.00 | funxion | yeah you would login remotely from the cell phone then enter the entire number for callback after login |
| 17:24.14 | Beirdo | I considered bringing down an SPA-3000 to replace it, but as I plan on moving there myself... |
| 17:24.20 | *** join/#asterisk MikeJ[Laptop] (n=ircatjer@d14-69-8-30.try.wideopenwest.com) |
| 17:24.29 | mishehu | Beirdo: I'd rather even get one of those 168 chipset phones, even if cheapie, instead. at least they support g729, gsm, etc. |
| 17:24.35 | Beirdo | I might just put a device at my mom's place. |
| 17:24.44 | Beirdo | well, she uses a cordless phone |
| 17:25.00 | mishehu | Beirdo: sipura? |
| 17:25.15 | Beirdo | the -3000? yes |
| 17:26.46 | Beirdo | stop discontinuing models I can run openwrt on, buttheads! |
| 17:27.01 | Nivex | you are not their business model. |
| 17:27.09 | af_ | where could I buy an x100p? |
| 17:27.16 | Beirdo | Nivex: no kidding. |
| 17:27.40 | Beirdo | although the use of WRT54g for openwrt has significantly affected their market share, I'd bet |
| 17:27.54 | mishehu | af_: probably a lot less places than you could before. |
| 17:28.17 | af_ | it looks like digium discontinued it |
| 17:28.31 | mishehu | Beirdo: you are an evil hacker. you should be brought to US soil and tried as a felon under provisions of the Darth Maul Corporate Agenda |
| 17:28.37 | af_ | what alternatives are for 1 fxo? |
| 17:28.38 | Beirdo | bah |
| 17:28.42 | bbz | anyone need to purcahse some polycom 501's @ cost? :) we got leftovers from a project =/ |
| 17:28.43 | coppice | with the incredibly high price of 802.11 boxes, linksys have obviously decided they need to burn some of the cash on royalties :-) |
| 17:29.15 | mishehu | af_: actually, intel discontinued the chipset it used |
| 17:29.19 | *** join/#asterisk Gennaro (n=Gennaro@ppp-62-11-85-245.dialup.tiscali.it) |
| 17:29.20 | tzanger | bbz: where are you located |
| 17:29.21 | af_ | oh I see |
| 17:29.31 | bbz | tzanger: STL, mO |
| 17:29.40 | Gennaro | sorry... a question.. |
| 17:29.44 | af_ | so what is good to have 1 fxo? |
| 17:29.45 | Beirdo | mishehu: shouldn't it be possible to recreate the code for another chipset? |
| 17:29.46 | tzanger | that's not too bad, I'm in ontario |
| 17:30.06 | Gennaro | i add telephone [name] in sip.config |
| 17:30.41 | LostFrog | What do manufacturers have against GSM? |
| 17:31.09 | Gennaro | and i say type=friend , username=100, secret=100 host dynamic |
| 17:31.17 | Gennaro | when i lunch asterisk.. |
| 17:31.23 | Gennaro | all ok |
| 17:31.25 | mishehu | LostFrog: it sounds too much like a cellular protocol |
| 17:31.53 | Gennaro | if i try to log with the phone on ip of my server... nothing.. |
| 17:31.54 | LostFrog | Yeah.. I was looking at a VOIP-GSM gateway.. And it doesn't have the GSM codec. |
| 17:31.55 | LostFrog | lol |
| 17:31.56 | Gennaro | why? |
| 17:32.24 | file | Gennaro: 1. Is the name in [] what you're using as the username on the phone? cause, like, username= is for when you're calling somewhere and need to do username/password authentication |
| 17:32.34 | file | Gennaro: 2. Have you done a sip debug to see if it's trying to register? |
| 17:32.48 | file | Gennaro: 3. Is there firewall/nat stuff? |
| 17:33.56 | Utah_Dave | bbz: how much are you asking for the polycoms? |
| 17:34.33 | Gennaro | [320] type=friend username=100 secret=100 host=dynamic |
| 17:34.48 | Gennaro | and i use on phone account 100 password 100 |
| 17:35.04 | bbz | Utah_Dave: $165 + shipping |
| 17:35.05 | RoyK | zoa: ping |
| 17:35.08 | LostFrog | Gennaro: then the heading has to be [100] |
| 17:35.16 | Gennaro | ah.. |
| 17:35.27 | LostFrog | As file was trying to tell you. |
| 17:35.29 | RoyK | ~seen zoa |
| 17:35.33 | jbot | zoa is currently on #asterisk (7h 35m 34s). Has said a total of 8 messages. Is idling for 6h 48m 34s |
| 17:35.40 | Utah_Dave | bbz: how many do you have? |
| 17:36.29 | bbz | Utah_Dave: 30 |
| 17:36.57 | *** join/#asterisk klicek (n=tk@nat.bratniak.ds.pw.edu.pl) |
| 17:36.57 | Gennaro | lostfrog, i do so.. |
| 17:37.02 | Gennaro | but.. nothing.. |
| 17:37.13 | Utah_Dave | bbz: my boss may want some of those. can I email you? |
| 17:37.15 | Gennaro | i stop now asterisk.. |
| 17:37.17 | file | then do a sip debug and see if the packets are even getting to the asterisk |
| 17:37.22 | Gennaro | and restart |
| 17:38.31 | Gennaro | lostfrog lunch in asterisk set debug |
| 17:38.48 | Gennaro | how can i do to see debug? |
| 17:39.00 | file | on the CLI type "sip debug" |
| 17:39.08 | Gennaro | ok |
| 17:39.10 | file | if packets are coming in from the phone, they'll fly across the screen |
| 17:39.10 | Gennaro | to read?! |
| 17:39.18 | file | if there's nothing - then you have other problems |
| 17:39.48 | Gennaro | i dont see file on screen... |
| 17:40.00 | Gennaro | i dont see packet |
| 17:40.18 | file | then your problems are network related most likely |
| 17:41.03 | Gennaro | i'm tring to see on firewall |
| 17:42.31 | bbz | is anyone else interested in buying some 501's @ cost? $165+ship (im in st.louis) -- client backed out of part of a deal, and we got 30 on us! i can email pics/info/etc. |
| 17:44.06 | *** join/#asterisk earthsound (n=webmonke@138.26.35.115) |
| 17:44.22 | *** join/#asterisk justinu (n=j2@72.18.13.48) |
| 17:44.26 | *** join/#asterisk BhaalWK (i=bhaal@freenode/staff/bhaal) |
| 17:45.10 | earthsound | if there are any digiums employees in here, could you private msg me, please? |
| 17:45.32 | mishehu | earthsound: why not just call digium up? |
| 17:45.49 | mishehu | thru iaxtel or something. |
| 17:46.11 | earthsound | mishehu: b/c their voicemail system isn't a person :| |
| 17:46.45 | earthsound | (I've tried that) |
| 17:46.55 | Corydon-w | It's not voicemail, it's an IVR |
| 17:46.57 | Corydon-w | and a queue |
| 17:47.01 | earthsound | same difference |
| 17:47.19 | Corydon-w | You wait in a queue until you can get to a live person |
| 17:47.22 | *** join/#asterisk juanjoc (n=jcomella@40-115-235-201.fibertel.com.ar) |
| 17:48.13 | earthsound | yes, you're correct. something I would do were I next to a phone and could wait x minutes :) |
| 17:48.25 | Corydon-w | Oh, goodie, a businessman... can't tell the difference between an IVR and voicemail... |
| 17:48.46 | earthsound | actually, I'm not a businessman. I see you're not clairvoyant, either |
| 17:48.58 | Corydon-w | Okay, a businesswoman |
| 17:49.03 | file[laptop] | just call and wait like every other normal person |
| 17:49.18 | earthsound | nor woman |
| 17:49.29 | Corydon-w | A businesscreature |
| 17:49.30 | *** join/#asterisk BhaalWK (i=bhaal@freenode/staff/bhaal) |
| 17:49.31 | earthsound | file: I'm not near a phone |
| 17:49.32 | Beirdo | businessdog? |
| 17:49.51 | file[laptop] | then find a phone? |
| 17:49.53 | Corydon-w | I knew some of them weren't human; now we have proof! |
| 17:50.10 | earthsound | wow, really intuitive aren't you, corydon? |
| 17:50.27 | *** join/#asterisk Gennaro (n=Gennaro@ppp-62-11-88-197.dialup.tiscali.it) |
| 17:50.35 | Corydon-w | Really impatient, aren't you, earthsound? |
| 17:50.46 | Beirdo | some people really like alienating those who might help them :) |
| 17:50.58 | earthsound | not really, or I wouldn't be "wasting" my time typing to a select group of asterisk geeks :P |
| 17:51.17 | file[laptop] | it's your time to waste. |
| 17:51.23 | Corydon-w | Are you actually having hardware issues? |
| 17:51.28 | earthsound | Beirdo: unless one of the people in here are @ digium, then none can help me |
| 17:51.48 | Beirdo | I'm sure that more than one person in here's working at digium |
| 17:52.08 | earthsound | as I said, I am not here b/c of asterisk, I am here to run someone down who works at digium...I am not near a phone & figured a digium employee may be scanning this channel |
| 17:52.11 | Beirdo | and I'm sure that if you WERE to wait on the phone, you may even be connected to one of them |
| 17:52.39 | earthsound | Beirdo: as I mentioned, I'm neither near a phone or able to access one. I wouldn't be in this channel if I were. |
| 17:52.58 | *** join/#asterisk oldbrat (n=daiviet@203.210.217.15) |
| 17:53.03 | Corydon-w | You can access the Internet, but not access a phone? |
| 17:53.04 | Beirdo | ah well, not much we can do for ya then |
| 17:53.32 | *** join/#asterisk Gennaro (n=Gennaro@ppp-62-11-88-197.dialup.tiscali.it) |
| 17:53.49 | earthsound | Corydon: yes, from where I am, you are correct |
| 17:54.06 | Gennaro | i still have the same prob. what can i do?!? |
| 17:54.07 | Corydon-w | Where is this mysterious place? |
| 17:54.26 | earthsound | a simple request for a digium employee turned into a long distraction from those who cannot help, it seems |
| 17:54.27 | Beirdo | funny, softphones would likely work |
| 17:54.40 | Gennaro | lost frog, don u know? |
| 17:54.54 | Corydon-w | Which Digium employee are you trying to track? |
| 17:54.57 | earthsound | mark |
| 17:55.05 | Corydon-w | Why? |
| 17:55.09 | *** join/#asterisk razu_ (n=razu@213-35-174-231-dsl.prn.estpak.ee) |
| 17:55.18 | *** join/#asterisk syzygyBSD (n=chatzill@66.226.228.204.cpe.speedyquick.net) |
| 17:55.31 | Corydon-w | Mark is a busy guy; you're unlikely to talk with him anyway |
| 17:56.19 | Beirdo | earthsound: I guess you don't want help. |
| 17:56.19 | Corydon-w | Heh, fine. Suit yourself. |
| 17:56.48 | earthsound | Beirdo: you're right, I don't want/need help. I just wanted to run an old friend down |
| 17:57.01 | Beirdo | ummm, email? |
| 17:57.04 | earthsound | I don't have my cell on me, or I would call his phone |
| 17:57.13 | Gennaro | if i digit sip degug i don't see nothing.. |
| 17:57.15 | earthsound | yeah, but that isn't immediate :| |
| 17:57.31 | Corydon-w | You need an immediate answer from Mark? |
| 17:57.34 | Katty | rehi. |
| 17:57.40 | Corydon-w | What's so important? |
| 17:57.40 | Beirdo | heya, Katty :) |
| 17:57.44 | Katty | Beirdo: (= |
| 17:57.47 | file[laptop] | yay KittyKatty(tm) |
| 17:58.00 | *** join/#asterisk TheCops (n=henri@206-248-136-187.dsl.teksavvy.com) |
| 17:58.04 | earthsound | Corydon: why do you care? I don't need asterisk "help" or anything else that is any of your business |
| 17:58.10 | justinu | lol |
| 18:00.27 | Beirdo | well, I think I might score another WRT54G and try it out tonight |
| 18:00.32 | Gennaro | port number for registration 5060, port number for RTP ? |
| 18:01.05 | TheCops | Someone have a PRI plugged into Asterisk and live to Quebec ?! |
| 18:01.54 | *** join/#asterisk SplasPood (i=nobody@paravolve.net) |
| 18:02.00 | justinu | gennaro: look at rtp.conf |
| 18:02.09 | Gennaro | ok i see |
| 18:02.45 | *** join/#asterisk morale (n=russell@S010600111155e117.cg.shawcable.net) |
| 18:04.02 | earthsound | I wish I had an asterisk related problem for you guys. :) you're willingness to help a stranger is exemplary |
| 18:04.37 | Katty | file[laptop]: i ate it. |
| 18:04.40 | Katty | file[laptop]: with cinnamon. |
| 18:05.00 | Gennaro | justinu nothing.. |
| 18:05.21 | mog_home | earthsound thats what crack'll do to you |
| 18:05.25 | TheCops | Someone is using RDNIS in quebec?! |
| 18:05.31 | file[laptop] | bah |
| 18:05.31 | Gennaro | i put [100] in sip.config as in guides and i cant register telephone.. |
| 18:05.36 | file[laptop] | it's an e-mail about a fax complaint |
| 18:05.41 | earthsound | mog_home: lol |
| 18:05.49 | Katty | file[laptop]: hence the cinnamon addition |
| 18:06.35 | X-Files | i have one question in asterisk: Calling to me (use exten => 1,1,Dial(SIP/1,60,tTrhwg) ) but this Line busy, i need in talking process beep mode (about new call). My gateway not support call_wait, this can configure to asterisk ?? |
| 18:06.47 | bbz | earthsound: once you cut away the egos and bullshit, you'll find its very very helpful :) |
| 18:07.48 | Flauto | is iaxtel acturally working? |
| 18:08.02 | file[laptop] | Flauto: probably not |
| 18:08.13 | Flauto | that is what i thought |
| 18:08.26 | Flauto | i tried a couple of times, it did not even register |
| 18:08.29 | earthsound | bbz: too true; same goes for many other forums where technically minded/gifted ppl congregate |
| 18:08.57 | Beirdo | heh |
| 18:09.05 | [TK]D-Fender | X-Files : you need to set "callerid=yes" for [1] in sip.conf |
| 18:09.05 | file[laptop] | iaxtel works once on a blue moon |
| 18:09.13 | [TK]D-Fender | X-Files : then reload |
| 18:09.26 | *** join/#asterisk SplasPood (i=nobody@paravolve.net) |
| 18:09.34 | file[laptop] | [TK]D-Fender: bad bad, giving useless information! |
| 18:09.53 | [TK]D-Fender | X-Files : you need to set "callwaiting=yes" for [1] in sip.conf <- my bad... |
| 18:10.10 | file[laptop] | yet again, bad bad, giving useless information! |
| 18:10.15 | [TK]D-Fender | shup yuo! |
| 18:10.31 | [TK]D-Fender | I've helped all sort of people here... |
| 18:10.36 | file[laptop] | chan_sip has no idea about call waiting, all it does is throw another call at your SIP device - if it doesn't accept it, too bad |
| 18:10.51 | Gennaro | some one can illumiate me with.. sip.config and the other passage to registrer an ip phone? |
| 18:11.07 | [TK]D-Fender | really? hmmm, Then I'm wasting valuable bytes here! |
| 18:11.30 | earthsound | in the unlikely case that mark does pop in before the end of the day, tell him that david stopped by looking for him. I'll call him later, when I get to my phone. :) |
| 18:11.50 | Beirdo | oh we'll be sure to do that... *snicker* |
| 18:12.09 | Katty | you silly sods. |
| 18:12.17 | Katty | you've clearly all insaned. |
| 18:12.21 | Katty | the lot of you! |
| 18:12.27 | Beirdo | many many months ago, Katty :) |
| 18:12.33 | Katty | ;> |
| 18:12.37 | *** join/#asterisk DeeJayTwo (i=deejay2@215-238.sh.cgocable.ca) |
| 18:13.09 | DeeJayTwo | what is iaxprov.conf meant to? |
| 18:14.49 | *** join/#asterisk Hmmhesays (i=negative@66.173.103.108) |
| 18:15.04 | DrDeke | yes, we _clearly_ have all insaned :) |
| 18:15.40 | DrDeke | hehe |
| 18:15.59 | syzygyBSD | is there a way to make it so when a caller calls in, he gets put into a queue but it still rings for 20 seconds? |
| 18:16.05 | *** join/#asterisk tainted_ (n=somewher@mail.k2usa.com) |
| 18:16.09 | Katty | [TK]D-Fender: careful. people who chide me often get clawed. |
| 18:16.15 | *** join/#asterisk Sedorox (i=brandon@smartserv/cna/Sedorox) |
| 18:16.15 | Katty | [TK]D-Fender: and not in a nice way. |
| 18:16.20 | [TK]D-Fender | ... |
| 18:16.26 | [TK]D-Fender | had my hopes up there ;) |
| 18:16.39 | Katty | aww, poor baby. |
| 18:16.40 | Katty | </sarcasm> |
| 18:16.50 | tainted_ | are u guys flirting? |
| 18:16.53 | [TK]D-Fender | You forgot the opening tag... |
| 18:16.57 | Flauto | is enumlook up still working? |
| 18:16.59 | tainted_ | that's kind of disturbing |
| 18:17.00 | DrDeke | lol |
| 18:17.16 | *** join/#asterisk kanobuho (n=kanobuho@201.124.179.200) |
| 18:17.21 | tainted_ | [TK]D-Fender you're sick |
| 18:17.21 | [TK]D-Fender | :O |
| 18:17.21 | Katty | ... |
| 18:17.25 | Beirdo | tainted_: if you have to ask, you need to get out more :) |
| 18:17.31 | DrDeke | out? |
| 18:17.32 | DrDeke | where is that? |
| 18:17.32 | [TK]D-Fender | A man of many hats, I am... |
| 18:17.44 | tainted_ | he's trying to pick up chix on #asteriks |
| 18:17.46 | DrDeke | Do they have a lot of jitter there? |
| 18:17.46 | DrDeke | ;) |
| 18:17.56 | tainted_ | i'm just jealous b/c i didn't think of it first |
| 18:18.15 | Katty | tainted_: he's a male. |
| 18:18.19 | Katty | tainted_: a /male/ |
| 18:18.31 | tainted_ | he's a Kip Dynamite |
| 18:18.32 | Katty | tainted_: why would i flirt with a male? |
| 18:18.42 | Katty | tainted_: that's just insane. |
| 18:18.45 | DeeJayTwo | at what condition is the iaxy.bin firmware sent to a iaxy unit? |
| 18:18.52 | Beirdo | Katty: for fun? :) because you're bored? |
| 18:18.58 | DeeJayTwo | what do I need to do to ensure it would be sent if it had to? |
| 18:19.01 | tainted_ | katty: and you're LaFawnduh |
| 18:19.03 | Katty | Beirdo: is that what normal people do? |
| 18:19.11 | Beirdo | don't ask me, I'm not normal |
| 18:19.14 | Katty | k |
| 18:19.46 | [TK]D-Fender | "normal is just another word for 'boring as dirt'" |
| 18:19.58 | Katty | anyone who knows me in teh real world knows i do not flirt with males ;) |
| 18:20.11 | coppice | neither do I |
| 18:20.12 | Beirdo | :) |
| 18:20.16 | Flauto | hehe |
| 18:20.23 | Beirdo | no worries, Katty. |
| 18:20.30 | Katty | now a female.......that's a different story |
| 18:20.45 | *** join/#asterisk sellout_ (n=asdf@S01060002e302d95b.vs.shawcable.net) |
| 18:20.49 | coppice | Yep, we match again |
| 18:20.51 | mishehu | Katty: so can we at least watch? |
| 18:20.52 | mishehu | hehehe |
| 18:21.00 | Katty | mishehu: you can watch |
| 18:21.01 | Flauto | hehe |
| 18:21.08 | Katty | screw the rest of the channel |
| 18:21.19 | coppice | what? all of them? |
| 18:21.20 | tainted_ | Katty u can handle all of us? |
| 18:21.22 | sellout_ | wow this is a way more active channel than efnet#asterisk :> |
| 18:21.23 | Katty | ... |
| 18:21.26 | tainted_ | that's a lotta cox |
| 18:21.29 | mishehu | half of you are gay |
| 18:21.30 | [TK]D-Fender | Damn, neither Linksys or Sipura have any docs or even real pages for the SPA-941 up anywhere it seems.... |
| 18:21.33 | Katty | ok, that's sick |
| 18:21.34 | justinu | the real question is... does anyone give a shit? |
| 18:21.37 | justinu | no, didn't think so |
| 18:21.43 | Katty | in the words of bkw |
| 18:21.44 | Katty | NEXT |
| 18:21.45 | mishehu | and there's not many females around here. |
| 18:21.51 | *** join/#asterisk ecto (n=ectospas@69.85.202.2) |
| 18:22.04 | tainted_ | Katty here's a quarter and a dime, go call someone who cares :D |
| 18:22.09 | Beirdo | Katty: I agree, NEXT :) |
| 18:22.11 | Katty | tainted_: kthx |
| 18:22.12 | mishehu | Katty: and in the words of eric cartman "screw you guys, I'm going home!" right? ;-) |
| 18:22.18 | Katty | mishehu: sure |
| 18:22.26 | Beirdo | "screw you! home!" |
| 18:22.33 | Beirdo | my usual way to leave work |
| 18:22.34 | Beirdo | :) |
| 18:22.42 | *** join/#asterisk zoa (n=kkk@pirus.securax.be) |
| 18:22.45 | zoa | aha! |
| 18:22.51 | Corydon-w | zoa! |
| 18:23.09 | sellout_ | I'm having trouble getting the MusicOnHold to play randomly, anyone able to give me a hand? |
| 18:24.02 | sellout_ | I've got "randon=yes" set in the .conf file, but when I put a 'm' at the end of any Dial() command I always start with the same song |
| 18:24.25 | DeeJayTwo | with my Iaxy it looks like as soon as I dial * with 2 digits, it's not sent to the asterisk server... |
| 18:24.29 | *** join/#asterisk zeedo (n=zeedo@80.68.92.188) |
| 18:24.29 | DeeJayTwo | like for *67 |
| 18:24.43 | DeeJayTwo | :/ |
| 18:24.50 | tamp4x | why do i not see the proper caller id when i pump calls thru a t1? i only see the number? do i need to have it go thru a pri that has an atached e911 db? |
| 18:24.57 | wunderkin | sellout_; random=yes? |
| 18:25.01 | sellout_ | wunderkin: ayup |
| 18:25.06 | Corydon-w | sellout_: moh is suspended on the same song when it's inactive |
| 18:25.26 | Corydon-w | sellout_: the "random" portion of moh just means it chooses the NEXT song randomly |
| 18:25.38 | DeeJayTwo | also, when I'm using *67 |
| 18:25.38 | sellout_ | Corydon-w: even if multiple people are dialed in? |
| 18:25.43 | DeeJayTwo | the remote peer can't hear me... |
| 18:25.46 | DeeJayTwo | but I can hear him.. |
| 18:25.48 | Corydon-w | sellout_: but everybody on moh in the same class hears exactly the same thing at the same time |
| 18:25.50 | DeeJayTwo | Only with *67 |
| 18:25.50 | fugitivo | native moh is volume is too low |
| 18:26.28 | Corydon-w | sellout_: plus or minus a few frames |
| 18:26.40 | iCEBrkr | hrrm, what's this crap about udev? |
| 18:27.07 | Samoied | hello all! |
| 18:27.09 | syzygyBSD | where is a good place to get moh music? |
| 18:27.10 | sellout_ | Corydon-w: ok |
| 18:27.20 | iCEBrkr | syzygyBSD: the freemusic place. :P |
| 18:27.24 | Samoied | symmetrix RTp is implemented in asterisk? |
| 18:27.31 | sellout_ | I've just noticed that it always seems to start on song 1 |
| 18:27.34 | Samoied | I have problems with clients behind nat |
| 18:27.34 | fugitivo | www.freeplaymusic.com |
| 18:27.38 | sellout_ | I'll dial in from a few phones and try it out |
| 18:27.44 | iCEBrkr | fugitivo: yea, that place |
| 18:27.49 | justinu | iCEBrkr: what's your problem with udev? |
| 18:27.51 | justinu | :P |
| 18:28.04 | Samoied | because them send a port, but NAT changes them |
| 18:28.09 | iCEBrkr | justinu: line 0: Unable to open master device '/dev/zap/ctl' |
| 18:28.20 | Corydon-w | iCEBrkr: drivers not loaded? |
| 18:28.26 | justinu | iCEBrkr: did you follow the instructions in README.udev? |
| 18:28.26 | iCEBrkr | I'm assuming I'm getting that error due to FC3 and udev??? |
| 18:28.32 | fugitivo | iCEBrkr: did you add the udev rules? |
| 18:28.34 | justinu | then run "udevstart" |
| 18:28.38 | iCEBrkr | I'm just getting there now |
| 18:28.41 | justinu | then modprobe ztdummy |
| 18:28.50 | iCEBrkr | justinu: Do I have to use ztdummy? |
| 18:28.54 | justinu | if you want meetme |
| 18:28.56 | iCEBrkr | I have a digium card. |
| 18:28.58 | justinu | oh |
| 18:29.00 | justinu | then no |
| 18:29.04 | justinu | modprobe zaptel? |
| 18:29.07 | iCEBrkr | ok, didn't think so |
| 18:29.15 | iCEBrkr | yeah the modprobe is happy |
| 18:29.17 | justinu | the udev stuff still applies |
| 18:29.27 | iCEBrkr | dmesg shows ok. |
| 18:29.30 | iCEBrkr | ok |
| 18:30.42 | iCEBrkr | I'm only assuming that this is a udev issue. I'll follow the readme and see if that clears it up |
| 18:30.57 | justinu | yeah... remember to run udevstart, and wait like 30 seconds |
| 18:31.04 | justinu | it seems to take a while to create the device nodes |
| 18:31.08 | justinu | then ls -l /dev/zap |
| 18:31.10 | iCEBrkr | right on |
| 18:31.14 | justinu | if the files are there, you're good |
| 18:32.13 | *** join/#asterisk aaronz (n=yoyo@pdpc/supporter/student/aaronz) |
| 18:33.12 | iCEBrkr | piece of cake.. |
| 18:33.13 | iCEBrkr | It works. |
| 18:33.34 | Katty | chocolate? |
| 18:33.40 | iCEBrkr | Katty: If you prefer :P |
| 18:33.46 | Katty | i'll take two |
| 18:33.55 | iCEBrkr | Katty: ok lardo! |
| 18:34.04 | Beirdo | hehe |
| 18:34.04 | Katty | i'm hardly lardo ;) |
| 18:34.05 | iCEBrkr | :P |
| 18:34.07 | Juggie | if its cvs-head or 1.2 i think you can run 'make udev' cant you? |
| 18:34.10 | aaronz | in creating my app, what is the most appropriate way to do asynchronous i/o with the voice channels? can i start my own threads, or should i be using existing asterisk routines? |
| 18:34.18 | Katty | iCEBrkr: i'm rather small. |
| 18:34.30 | iCEBrkr | Katty: Lemme guess, you can eat and eat and eat and never gain weight? *jealous look* |
| 18:34.36 | Katty | iCEBrkr: no, i'm vegan |
| 18:34.42 | Katty | iCEBrkr: i chew on trees all day |
| 18:34.44 | iCEBrkr | That explains a lot :D |
| 18:35.21 | iCEBrkr | No offense, but the vegans I've met. They're-- well? COO-COO! COO-COO! |
| 18:35.34 | Beirdo | be nice to vegans :) just because they don't eat meat doesn't mean they won't beat you to a pulp if you piss em off, just like anyone else :) |
| 18:35.49 | fugitivo | vegans are stronger and smarter |
| 18:35.50 | mishehu | they just won't eat the pulp |
| 18:35.59 | iCEBrkr | Beirdo: I went camping with about 8 vegans. Oh, it was QUITE interesting. |
| 18:36.03 | mishehu | Katty: you must have strong teeth |
| 18:36.07 | Beirdo | my soon-to-be mother-in-law is vegan |
| 18:36.38 | Katty | mishehu: how does one tell? |
| 18:36.45 | tamp4x | hamburgers and veggie burgers tatse the same |
| 18:36.46 | Katty | mishehu: they're sharp! |
| 18:36.50 | mishehu | Katty: if you can compete with a beaver... |
| 18:36.55 | Katty | blah |
| 18:36.58 | mishehu | TIMBER! |
| 18:37.00 | Katty | i'd hug the beaver |
| 18:37.01 | iCEBrkr | haha |
| 18:37.01 | tamp4x | swap her veggie burger with a slabe of beef |
| 18:37.23 | [TK]D-Fender | vegan? Careful : you are what you eat ;) |
| 18:37.26 | aaronz | anyone a proper asterisk developer? |
| 18:37.38 | Katty | haha |
| 18:37.39 | Katty | proper |
| 18:37.44 | tamp4x | dis asterisk program be massive |
| 18:37.46 | Corydon-w | Proper? |
| 18:37.58 | Beirdo | nah, I think they are all trying to be improper |
| 18:38.02 | aaronz | rather than users that can answer technical questions |
| 18:38.07 | aaronz | for hacking asterisk |
| 18:38.15 | [TK]D-Fender | aaronz : just ask what it is you really want to ask and see what you get... |
| 18:38.20 | Corydon-w | hacking? |
| 18:38.28 | aaronz | d-fender: i did with no reply |
| 18:38.33 | sellout_ | hrmmm my new Aastra 480i CT is URL escaping the # sign in it's registration message |
| 18:38.42 | Katty | it's linux |
| 18:38.45 | Katty | you don't need to hack ;) |
| 18:38.49 | sellout_ | so instead of registering as #xxx it's registering as %23xxx |
| 18:38.54 | aaronz | corydon-w: hacking doesnt have to imply security you know |
| 18:39.01 | sellout_ | and asterisk doesn't like that |
| 18:39.04 | sellout_ | anyone have any ideas? |
| 18:39.05 | tamp4x | hacking asterisk code wise go learn some c, hacign asterisk script kiddie wise go play with ser |
| 18:39.11 | tamp4x | heh heh heh |
| 18:39.18 | [TK]D-Fender | aaronz : ask again, not everyone here at one point in time is as inclined or as skilled as any other. |
| 18:39.30 | file[laptop] | mmm? |
| 18:40.26 | aaronz | what is the proper way to do asychronous i/o against a voice channel in an app. can i start my own threads? does asterisk provide functions for this? im trying to create an app that essentially streams full-duplex the voice channel over tcp to another local server |
| 18:40.38 | aaronz | which i will release back to teh community once its done |
| 18:41.01 | sellout_ | shouldn't asterisk handle a url escaped registration properly? |
| 18:41.08 | [TK]D-Fender | there we go! a real question, and one I'm completely out of my league on! File? |
| 18:41.32 | justinu | sellout_: asterisk doesn't handle a lot of sip stuff properly |
| 18:41.37 | Katty | file is busy eating cookies |
| 18:41.42 | Katty | please leave a message after the beep |
| 18:41.43 | Katty | BEEP |
| 18:41.45 | sellout_ | justinu: really? |
| 18:41.53 | fugitivo | give me a cookie |
| 18:41.54 | sellout_ | bummer |
| 18:41.59 | sellout_ | even in 1.2? |
| 18:41.59 | justinu | yeah, like unicode SIP uris crash it |
| 18:42.03 | justinu | yeah, even in 1.2 |
| 18:42.07 | sellout_ | eeek |
| 18:42.20 | sellout_ | man this 480i is gonna suck then |
| 18:42.28 | justinu | hmm.... |
| 18:42.30 | fugitivo | is asterisk going to use yate code for nat problems? |
| 18:42.30 | justinu | i have one on the way |
| 18:42.32 | justinu | supposedly |
| 18:42.34 | *** join/#asterisk L|NUX (n=linux@202.5.145.58) |
| 18:42.37 | sellout_ | it can't even dial a number that has a # in it |
| 18:42.49 | sellout_ | like the call doesn't even make it to the asterisk box |
| 18:42.55 | [TK]D-Fender | sellout_ : you should be able to use # in the dialstring... |
| 18:43.02 | Corydon-w | aaronz: see the source for app_chanspy |
| 18:43.17 | sellout_ | [TK]D-Fender: it might be a moot point since I can't register as #113 |
| 18:43.20 | zoa | hmm |
| 18:43.28 | zoa | somebody know how the iax2 mwi is called in the sources ? |
| 18:43.32 | zoa | im trying to find out when it was added |
| 18:44.14 | sellout_ | justinu: it's a neat phone, but I had to go to the beta firmware (1.3) to get it working "decent" |
| 18:44.20 | [TK]D-Fender | sellout_ You shouldn't have "#" in your registration info, but no reason you shouldn't be able to DIAL itr |
| 18:44.28 | sellout_ | and the docs for the config file are weak |
| 18:44.38 | sellout_ | [TK]D-Fender: why no # in the registration? |
| 18:44.45 | sellout_ | it's always worked with the pap2's |
| 18:45.05 | [TK]D-Fender | I suspect it is an illegal char in context names, but I may be mistaken.... |
| 18:45.11 | sellout_ | [TK]D-Fender: and this phone (I'm sure it's a prob with the phone itself) will not dial any string that starts with # |
| 18:45.14 | justinu | sellout: hmm |
| 18:45.33 | aaronz | corydon-w: would it be better to try and create a channel driver & bridge it? |
| 18:45.35 | [TK]D-Fender | sellout_ : cheaked your MGCP dial string on it? |
| 18:46.02 | [TK]D-Fender | sellout_ : the 480i is considered one of the better phones for # and Sayson/Aastra seems pretty involved in keeping it that way |
| 18:46.12 | [TK]D-Fender | s/#/* |
| 18:46.15 | sellout_ | [TK]D-Fender: sorry you kinda lost me? do you mean this string from the config? |
| 18:46.18 | file[laptop] | # is used on SIP phones to indicate "hey - I'm done dialing..." |
| 18:46.19 | sellout_ | sip dial plan: "[2-9]11|9911|1[01]xx|[2-9]xxxxxx|1[2-9]xxxxxxxxx|011x+#|xx*|#xxx|xxxxxxxxxxxxxxxxx#" |
| 18:46.51 | sellout_ | bummer so I was using a hack with the pap2 I guess |
| 18:46.52 | [TK]D-Fender | Hmmm, so you wan "#101" to dial that out? |
| 18:47.00 | [TK]D-Fender | sellout_ : looks legit I think. |
| 18:47.08 | sellout_ | I want #101 to dial the ext #101 |
| 18:47.27 | *** join/#asterisk JohnJacob (n=JohnJaco@pcp0011543623pcs.mainf01.in.comcast.net) |
| 18:47.28 | sellout_ | those are "internal" extensions, this way people don't have to prefix a outbound call with 9 or whatever |
| 18:47.38 | sellout_ | like I said, it works fine on the pap2's |
| 18:47.41 | [TK]D-Fender | and you have exten => #101,1,something here" in your dialplan and it doesn't work? |
| 18:47.52 | sellout_ | [TK]D-Fender: no that works just fine |
| 18:48.02 | [TK]D-Fender | just not on the 480i? |
| 18:48.05 | sellout_ | right |
| 18:48.08 | [TK]D-Fender | hmmm |
| 18:48.15 | sellout_ | "call failed" and no message whatsoever on the asterisk console |
| 18:48.21 | sellout_ | like, instant call failed |
| 18:49.46 | TheCops | Someone have a PRI plugged into Asterisk and live to Quebec ?! |
| 18:50.17 | sellout_ | [TK]D-Fender: did that 'sip dial plan' line I pasted look like a proper MGCP dial string? |
| 18:50.33 | sellout_ | sorry I'm not 100% on my terminalology but with the right hints, I'm a mean googler |
| 18:51.02 | justinu | you're running MGCP? |
| 18:51.08 | sellout_ | nope |
| 18:51.14 | [TK]D-Fender | sellout_ : yeah it looks perfectly legit and I'm reading the MGCP RFC on it now .... |
| 18:51.21 | [TK]D-Fender | Might be a bug in tht firmware rev. |
| 18:51.23 | sellout_ | just asterisk sip, pap2's and now this Aastra 480i ct |
| 18:51.38 | sellout_ | well I went to the beta firmware (1.3) and it still does it |
| 18:52.26 | syle | list |
| 18:56.43 | *** join/#asterisk Agiofws (n=AC90CESS@athe730f-0956.otenet.gr) |
| 18:57.58 | *** join/#asterisk simprix (n=simprix@gw001.cdsoc.org) |
| 18:58.04 | simprix | What does a error 486 mean |
| 18:58.13 | justinu | busy here? |
| 18:58.16 | Agiofws | i have just made a wifi network http://www.paranoid.gr/~agiofws/1st-LINK.jpg <---- can i use VOIP from box-A to box-C ? what software do i use and do i bye any hardware ? |
| 18:58.18 | simprix | yea |
| 18:58.27 | justinu | it means the phone is busy |
| 19:01.38 | jlewis | is there some reason the t option to Dial isn't compatible with Dial(IAX2...)? or have we done something wrong? |
| 19:02.29 | malverian[work] | Hmm.. |
| 19:02.34 | *** join/#asterisk santiago (n=santiago@208.195.215.124) |
| 19:02.38 | justinu | hey jlewis: passed the level3 interop :) |
| 19:02.38 | jlewis | I'm using IAX to get to PRI (for dialout) on another server...and wanting the called party to be able to do a #transfer back on the server that initiated the call |
| 19:03.03 | jlewis | justinu: I think I have too...I asked them to just N/A the last few "silly tests"...the sip conf bridge crap |
| 19:03.06 | malverian[work] | Anyone know of an EAGI or similar that will allow you to save incoming fax to a PDF file but also pass it through to a normal zap channel? |
| 19:03.36 | malverian[work] | And I want it to be passthrough, I don't want to receive the fax, save it, then resend it to a zap line. |
| 19:04.00 | justinu | jlewis: feels good to have it out of the way |
| 19:04.17 | jlewis | did you do those last few 5.x.x cases? |
| 19:04.31 | justinu | no... for whatever reason they weren't in our test plan |
| 19:04.41 | justinu | i think because we told them we don't do conferencing |
| 19:05.01 | jlewis | the test plan depends on your answers to the initial setup questionaire...we must have told them we wanted access to more features |
| 19:05.09 | justinu | yeah - i didn't fill that out |
| 19:05.17 | justinu | some forienger who barely speaks english did |
| 19:05.34 | malverian[work] | No one tried anything like that before? |
| 19:06.57 | wunderkin | so whats the minimum requirement for minutes now on level3? |
| 19:07.28 | jlewis | no idea |
| 19:08.24 | TheCops | Someone have a PRI plugged into Asterisk and live to Quebec ?! |
| 19:08.28 | jlewis | bit of a PITA to get them working though |
| 19:08.40 | mishehu | TheCops: appearantly not. |
| 19:09.01 | jlewis | and they're anal about things like e164 format callerid...on our gblx voip term, I can setcidnum to whatever I want and they don't care |
| 19:09.03 | justinu | they want 4million a month |
| 19:09.17 | TheCops | mishehu, I'll probably cry if I dont find someone before thuesday |
| 19:09.22 | wunderkin | wow |
| 19:09.58 | mishehu | TheCops: why do you need somebody with a PRI in Quebec? |
| 19:10.05 | Leob | file, thanks for your suggestion... things seem to be working fine now! |
| 19:10.06 | TheCops | mishehu, RDNIS testing |
| 19:10.18 | file[laptop] | Leob: Kevin yelled at you too on the mailing list |
| 19:11.08 | mishehu | TheCops: bah, PCMCIA! People Can't Memorize Computer Industry Acronyms (or telecommunications ones for that matter) I can't remember now what RDNIS is. but guess I can't be of help cuz I'm not around there. |
| 19:11.25 | justinu | redirected DNIS |
| 19:12.07 | mishehu | justinu: that takes care of the R.... |
| 19:12.08 | mishehu | heh |
| 19:12.13 | mishehu | I'd have to look it up, no biggie. |
| 19:12.18 | justinu | ~dnis |
| 19:12.19 | jbot | [dnis] A telephone service that identifies the number that the caller dialed for the receiver of the call. DNIS is a common feature of 800 and 900 services, and can identify the number originally dialed when multiple 800 or 900 numbers terminate on the same destination trunks. DNIS works by passing the dialed number to the destination device, which can act upon ... |
| 19:12.34 | Leob | file[laptop], hmmm, Ihaven't got that message, yet... I'm subscribed on digest mode... |
| 19:12.35 | ikarus | Oh, DID |
| 19:13.19 | ikarus | btw, in telecom land, why does everything have to have 3 acronyms, usually a American, a European and a German one |
| 19:13.31 | ikarus | and you have to be lucky to remember what acronyms mean the same |
| 19:13.36 | sellout_ | cause there's no such thing as a standard? |
| 19:13.40 | mishehu | so I guess when my provider "passes digits" on to my pri that's dnis. |
| 19:13.42 | sellout_ | I blame nortel |
| 19:13.56 | mishehu | I blame canada |
| 19:14.03 | sellout_ | sure thing eH? |
| 19:14.07 | mishehu | and the spanish inquisition |
| 19:14.12 | justinu | q.931 calls it "Called Party Identification" |
| 19:14.13 | sellout_ | we're pretty mellow and won't hold it against you |
| 19:14.14 | mishehu | because nobody expects... |
| 19:14.30 | ikarus | in all Asterisk docs it is called DID |
| 19:14.40 | ikarus | iirc |
| 19:14.44 | sellout_ | did is direct inward dial |
| 19:14.45 | Beirdo | mishehu: don't MAKE me hunt you down :) |
| 19:14.49 | ikarus | or I am sleeping |
| 19:14.56 | justinu | DNIS is probably some AT&T term |
| 19:14.57 | sellout_ | which is kinda different |
| 19:15.24 | justinu | i remember having analog DID lines from ILEC |
| 19:15.30 | justinu | trying to remember how they worked |
| 19:15.31 | Beirdo | muhahah |
| 19:15.36 | sellout_ | you would use dnis to make did work if you had calls comming in on multiple channels |
| 19:15.53 | ikarus | sellout_: DID sends you which number you got dialed on aswell iirc |
| 19:15.55 | justinu | i think they were groundstart lines |
| 19:16.13 | ikarus | justinu: that is the common case |
| 19:16.21 | sellout_ | meh I worked for ipnetworking @sprint not the POTS side of things |
| 19:16.25 | mishehu | Beirdo: hmm let me see... which would I respect more for the "hunting down" bit... you, the CIA, or the Mossad? |
| 19:16.29 | ikarus | I luckily only have to deal with ISDN and a CLUELESS TELCO |
| 19:16.30 | justinu | i also remember having switched 56k digital lines |
| 19:16.37 | justinu | they had real phone numbers |
| 19:16.43 | Beirdo | mishehu: me. I won't screw it up like the CIA |
| 19:16.59 | mishehu | Beirdo: there were 3 choices there though... |
| 19:17.03 | Beirdo | and you're not Arab, so the Mossad won't care much about ya |
| 19:17.07 | ikarus | ISDN here has 1 real number and several MSN's |
| 19:17.22 | mishehu | Beirdo: unless my name was Mordechai Vannunu... |
| 19:17.44 | justinu | switched 56k was weird stuff |
| 19:17.48 | mishehu | but I can't remember if it was the shaba"k or Mossad that got Vannunu. |
| 19:18.07 | Beirdo | I'll just have to make a quick detour next time I fly through O'Hare :) |
| 19:18.07 | Beirdo | hehe |
| 19:18.16 | *** join/#asterisk funxion (n=nunya@host-64-110-51-254.hlm.ses-americom.net) |
| 19:18.19 | *** join/#asterisk razu_ (n=razu@213-35-174-231-dsl.prn.estpak.ee) |
| 19:18.30 | sellout_ | justinu: like a DS0 from the telco? |
| 19:18.35 | mishehu | Beirdo: I'll just have to make a call to Homeland Security next time to fly to O'Hare. ;-) |
| 19:18.37 | sellout_ | like centrix? |
| 19:18.47 | Beirdo | now now |
| 19:18.50 | Beirdo | that's just not nice |
| 19:19.03 | Beirdo | no need to call the Nazis |
| 19:19.09 | justinu | sellout_: something like that... but it was digital, and only used to transfer data. |
| 19:19.14 | ikarus | justinu: here they only offer a few insane services, ISDN BRI is prolly the worst if you need special tricks, I need 2 BRI lines cascaded and sharing outbound permitted numbers, they said they can technicly do it, but that administrativly it might get complex |
| 19:19.18 | justinu | we used adtran switched 56k "modems" for it |
| 19:19.23 | mishehu | Beirdo: they're more like storm troopers from the original star wars series... |
| 19:19.25 | mishehu | dumb. |
| 19:19.30 | Beirdo | heh |
| 19:19.35 | Beirdo | Hitler youth |
| 19:19.37 | sellout_ | justinu: sounds like frame relay to me |
| 19:19.43 | justinu | it wasn't FR |
| 19:19.46 | mishehu | and customs is no better. I got harrassed because I came back from vietnam alone... |
| 19:19.48 | sellout_ | unless it was something serial? |
| 19:19.50 | *** join/#asterisk Rav1974 (n=r@static-70-19-119-112.ny325.east.verizon.net) |
| 19:19.51 | justinu | it may have been FR inside the telco cloud |
| 19:19.55 | Beirdo | alone? |
| 19:19.58 | sellout_ | justinu: point to point on your end? |
| 19:20.01 | mishehu | so I must have been going to pick up and smuggle home all sorts of vietnamese girlies |
| 19:20.04 | Beirdo | they expected you to bring back a whore? |
| 19:20.16 | justinu | yeah, you'd just "call" the other number, and it would establish a 56k serial connection to it |
| 19:20.23 | sellout_ | x25? |
| 19:20.28 | justinu | it even had an AT command set |
| 19:20.32 | mishehu | Beirdo: donno, but a guy just can't travel the world can he |
| 19:20.37 | Beirdo | I guess not |
| 19:20.47 | justinu | it wasn't x25 either |
| 19:20.59 | sellout_ | huh |
| 19:21.03 | justinu | because you were dialing PSTN numbers |
| 19:21.14 | justinu | you could call them from voice lines too, but it would just give dead air |
| 19:21.14 | sellout_ | weird |
| 19:21.19 | justinu | even tho the modem would see the call |
| 19:21.29 | ikarus | justinu: and you are sure it was packet switched ? |
| 19:21.30 | justinu | that was back in the day before v92 56k modems existed |
| 19:21.38 | justinu | no, it was circuit switched |
| 19:21.54 | sellout_ | what provider |
| 19:21.57 | ikarus | justinu: ah, then it was just routed via special telco circuits, that could handle 56k |
| 19:22.02 | justinu | Pacific Bell |
| 19:22.10 | justinu | now SBC |
| 19:22.12 | *** join/#asterisk MRH2 (n=Mr_happy@fcirc-adsl.demon.co.uk) |
| 19:22.14 | ikarus | justinu: I have seen such systems before (but only on higher speeds, like 1Mbit) |
| 19:22.16 | sellout_ | ahh the days when all phone lines were actual ds0's |
| 19:22.20 | justinu | yeah |
| 19:22.25 | sellout_ | and they didn't compress the living hell out of your calls |
| 19:22.35 | sellout_ | anyone else remember the "pin drop" commercials? |
| 19:22.38 | *** join/#asterisk ozant (n=on@81.215.219.201) |
| 19:22.38 | justinu | we eventually upgraded to a T1 network |
| 19:22.41 | justinu | yeah of course... sprint |
| 19:22.44 | sellout_ | heh |
| 19:22.46 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-170.dslextreme.com) |
| 19:22.55 | sellout_ | should come up with a new slogal for them |
| 19:23.04 | sellout_ | "so compressed, even grandma sounds like darth vader" |
| 19:23.13 | *** join/#asterisk Seggy (i=rbutler@tsss.org) |
| 19:23.14 | ikarus | sellout_: remember when calling long distance you had to wait for the next switch office to send you the dial tone after you called the area code |
| 19:23.15 | tamp4x | why do i not see the proper caller id when i pump calls thru a t1? i only see the number? do i need to have it go thru a pri that has an atached e911 db? |
| 19:23.18 | sellout_ | not quite the same punch as the original ads, but it's more correct |
| 19:23.18 | justinu | back when sprint wasn't just a shitty wireless company |
| 19:23.38 | Beirdo | yeah, they did other things in shitty ways too :) |
| 19:23.40 | sellout_ | ikarus: not really |
| 19:23.46 | justinu | ikarus: before my time |
| 19:23.56 | ikarus | Here that system was in use until 1993 or so |
| 19:24.07 | MRH2 | ? there is no difference in the cabiing between to the exchange between a normal phone like and isdn2/bri? - it is just terminated differently at either end is that right? |
| 19:24.16 | file[laptop] | tamp4x: what do you mean proper caller id? |
| 19:24.18 | justinu | ikarus: i bet you whisteled 2600hz into the phone to make it think you had dropped the trunk, then you MF outpulsed a new number |
| 19:24.21 | ikarus | the last switch that actually needed it was taken offline that year |
| 19:24.31 | ikarus | justinu: nope, didn't work here |
| 19:24.34 | justinu | bummer |
| 19:24.48 | tamp4x | well the caller id that u set in sip/zapata.conf , the name does not show up when u call a cell phone |
| 19:24.49 | ikarus | justinu: to begin with the technology they actually used on most of the network was already digital |
| 19:25.03 | justinu | i like the way the old telco tone plants used to sound |
| 19:25.08 | file[laptop] | tamp4x: name isn't delivered over the PSTN usually |
| 19:25.08 | justinu | the big analog ones with rotating drums in them |
| 19:25.15 | justinu | busy tone sounded like you were killing a goose |
| 19:25.17 | tamp4x | orly |
| 19:25.18 | file[laptop] | tamp4x: it's done via lookup at the end |
| 19:25.21 | ikarus | tamp4x: caller id (depending on the technology) is normally only a number even for e911 |
| 19:25.22 | file[laptop] | against a database |
| 19:25.35 | ikarus | Only in Belgium do they have a caller ID that can send a name |
| 19:25.56 | MRH2 | dr evil? |
| 19:26.13 | ikarus | justinu: relays man, relays, that was the last system here, it sounded like hell (they did a bit on the news on it when they where to take it offline) |
| 19:26.13 | file[laptop] | in some instances you can send callerid name within the same network, ie if you're calling over a Bell circuit and you call a Bell landline, it may traverse internally across their switches |
| 19:26.19 | tamp4x | i thought on normal telephone lien su can see caller id or id that if your both usign the same carrier |
| 19:26.30 | tamp4x | yea |
| 19:26.39 | file[laptop] | but even that's not guaranteed to work |
| 19:27.53 | ikarus | GSM signalling doesn't support it at all last I checked |
| 19:28.09 | ikarus | Although there is a work around for that with certain cell messages |
| 19:28.24 | ikarus | (Works on about 80% of all phones) |
| 19:28.31 | Rav1974 | Do you know a prodcution level iax or sip carrier for phone service |
| 19:28.45 | tamp4x | so caller id is not a standard thign absically |
| 19:28.49 | ikarus | Rav1974: Cable & Wireless |
| 19:28.49 | Rav1974 | voxee & voipjet keep going down from time to time |
| 19:28.50 | tamp4x | basically |
| 19:28.52 | *** join/#asterisk DeeJayTwo (i=deejay2@215-238.sh.cgocable.ca) |
| 19:28.57 | ikarus | tamp4x: it is standard, numerical caller id that is |
| 19:29.13 | MRH2 | where Rav1974? |
| 19:29.20 | Rav1974 | MRH2 in NY |
| 19:29.27 | file[laptop] | callerid number you can usually send across... provided your provider allows it |
| 19:29.34 | file[laptop] | and number will traverse across most everything... |
| 19:29.49 | MRH2 | <--in scotland |
| 19:30.03 | ikarus | file[laptop]: even internationally |
| 19:30.12 | mishehu | ikarus: depends |
| 19:30.14 | file[laptop] | ikarus: sometimes... |
| 19:30.21 | ikarus | mishehu: it works within Europe and Russia |
| 19:30.31 | ikarus | atleast to mobile carriers |
| 19:30.44 | file[laptop] | internationally is one of those, "if it works - good for you, if not - meh too bad" |
| 19:30.47 | ikarus | No idea about fixed line, never had to call people in russia fixed line |
| 19:30.50 | justinu | ikarus: i liked that old funky sound |
| 19:30.57 | justinu | warbling dial tone |
| 19:31.07 | mishehu | in .il I only received CIDNum from somebody who called me from florida, but ppl who called me from the midwest and california did not send the cidnum in the manner it expected I guess |
| 19:31.09 | justinu | quaking busy tones |
| 19:31.12 | justinu | quacking |
| 19:31.13 | ikarus | justinu: dial tone here had been stable for years |
| 19:31.18 | justinu | gave the PSTN "character" |
| 19:31.25 | justinu | every switch had it's own unique sound |
| 19:31.40 | ikarus | justinu: too much was already digital to allow such deviation |
| 19:31.54 | justinu | i've heard some recordings of phone calls from the early 60s |
| 19:31.58 | justinu | way different sound then now |
| 19:32.11 | ikarus | Rav1974: ah, NY, that might not be handy on C&W :) (C&W is big in the UK and Europe) |
| 19:32.31 | ikarus | justinu: hmmmmm, I heard some aswell and the quality here improved mainly |
| 19:32.44 | ikarus | justinu: only local calls lost some quality |
| 19:32.48 | ikarus | But not much |
| 19:33.18 | justinu | not saying it was better then |
| 19:33.22 | justinu | LD calls sounded like shit |
| 19:33.29 | justinu | FDM sucked in those days |
| 19:33.51 | ikarus | justinu: LD here used to have alot of whistle and clipping |
| 19:33.56 | justinu | hehe |
| 19:34.04 | ikarus | with digital tech it is just straight quality |
| 19:34.16 | justinu | where do you live? |
| 19:34.24 | ikarus | justinu: The Netherlands |
| 19:34.30 | justinu | i was in romania in 2000, and they still had manual switchboard operators |
| 19:34.39 | justinu | you just lift the phone and tell them what number you want to dial |
| 19:35.16 | justinu | usually the operators knows all the numbers of people in the village, so you just tell them the name of the person you want to call if it's local |
| 19:35.38 | justinu | she has to log the start time, and end time of each call with paper and pencil |
| 19:36.02 | ikarus | justinu: I have no experience with that |
| 19:36.13 | *** join/#asterisk KranZ (n=user@sme.bestline.net) |
| 19:36.34 | ikarus | although here phone phreaking is kind of fun, there was a switch that had the UUCP comm/upgrade functionality open on a public phone number |
| 19:36.44 | ikarus | That one got abused the hell out of |
| 19:36.51 | Katty | ikarus: your remind me of kidikarus |
| 19:36.55 | Katty | or was it kidicarus |
| 19:36.59 | ikarus | Katty: wrong memory |
| 19:37.03 | ikarus | think greek mythology |
| 19:37.04 | Katty | some old nes game |
| 19:37.09 | Katty | oh ah |
| 19:37.12 | Katty | kthen |
| 19:38.10 | *** join/#asterisk kanobuho (n=kanobuho@201.124.179.200) |
| 19:38.20 | ikarus | Most insane thing was done here by the telco, the renumbered almost the whole country to go to a 10 digit plan for all fixed and mobile numbers (with a 3 or 4 digit area code, which always starts with 0) |
| 19:39.22 | ikarus | It does make writing dial plans relativly easy |
| 19:39.28 | ikarus | although I have some bugs to kill |
| 19:40.35 | sellout_ | [TK]D-Fender: I got the phone to register as #113 |
| 19:40.41 | KranZ | has a feature been added yet which allows for voicemail checking when you hit * or something while the greeting is playing? |
| 19:40.42 | sellout_ | pedantic=yes |
| 19:40.53 | file | KranZ: that has been in existence for a longggg time |
| 19:40.58 | KranZ | good |
| 19:41.03 | KranZ | ....now what is it! |
| 19:41.07 | sellout_ | but this is real slow now |
| 19:41.31 | file | KranZ: If the caller presses '*' during the prompt, the call jumps to |
| 19:41.32 | file | extension 'a' in the current context. |
| 19:41.41 | KranZ | hmm |
| 19:41.56 | KranZ | then you run voicemailmain? |
| 19:42.06 | file | yes |
| 19:42.13 | KranZ | coo, thx |
| 19:42.13 | mishehu | grrr where is fedex |
| 19:42.25 | KranZ | i will now test it |
| 19:42.27 | syzygyBSD | Is there a common reason why music on hold stops immediatly for a sip phone? I haven't tested it with any other type |
| 19:42.52 | ikarus | Bleah, annoying, I can't work on the dial plan tuesday (next day I officially work), need to go to training |
| 19:43.06 | [TK]D-Fender | sellout_ : So all's good with your 480i? |
| 19:43.27 | file | [TK]D-Fender: your dialplan is showing |
| 19:43.40 | [TK]D-Fender | file : ? |
| 19:44.01 | file | did my bad humor overload your brain? |
| 19:44.20 | [TK]D-Fender | What humour? |
| 19:44.22 | [TK]D-Fender | ;P |
| 19:44.35 | file | good point |
| 19:44.41 | *** join/#asterisk toddf (n=toddf@ns0.fries.net) |
| 19:45.04 | sellout_ | [TK]D-Fender: well I still can't dial a number that starts with # |
| 19:45.09 | sellout_ | call fails instantly |
| 19:45.17 | sellout_ | but at least I'm registering as the right extension now |
| 19:45.21 | [TK]D-Fender | :/ |
| 19:46.23 | sellout_ | is there anything you could suggest I look into on this phone to allow that? |
| 19:46.44 | sellout_ | that sip dial plan: config directive looks like it might be it |
| 19:46.51 | [TK]D-Fender | Not particularly as I don't own one myself.... |
| 19:46.51 | ikarus | Or with # |
| 19:46.58 | ikarus | so it won't collide with local calls |
| 19:47.02 | sellout_ | I just can't find docs on how it should be formatted |
| 19:47.08 | sellout_ | ikarus: it's what we do, works nicely |
| 19:47.10 | sellout_ | on pap2's |
| 19:47.19 | [TK]D-Fender | Google "MGCP dialplan RFC" |
| 19:48.24 | *** join/#asterisk L|NUX (n=linux@202.5.145.58) |
| 19:48.47 | sellout_ | I'm gonna owe you a bunch of beer when this is done |
| 19:48.49 | ikarus | sellout_: well, I will have to figure out a way to make goto's, 484 (early-dial) and my BudgeTones work together (possibly severly hacking asterisk) |
| 19:49.02 | sellout_ | hey, toronto! |
| 19:49.11 | Beirdo | Toronto sucks :) |
| 19:49.15 | fugitivo | my asterisk is crashing |
| 19:49.27 | fugitivo | (1.2) |
| 19:49.37 | sellout_ | Beirdo: yea but we don't tell them that |
| 19:49.46 | sellout_ | they *are* the centre of the universe after all |
| 19:49.57 | ikarus | I need to upgrade to 1.2 to see if it is still b0rked |
| 19:50.11 | Beirdo | yeah, I live at the centre of the universe. hehe |
| 19:50.28 | fugitivo | i think mixmonitor is crashing my asterisk |
| 19:50.36 | [TK]D-Fender | Beirdo : Oh yeah? Try a Montreal winter on for size if you think you have something to complain about ;) |
| 19:50.41 | Beirdo | hehe |
| 19:50.46 | Beirdo | I lived in Ottawa for years |
| 19:50.51 | Beirdo | and grew up in Parry Sound |
| 19:50.54 | sellout_ | I spent 6 years in exile in Edmonton |
| 19:50.55 | sellout_ | beat that |
| 19:50.55 | Juggie | ottawa is cold as hell |
| 19:50.59 | Beirdo | Montreal winters are pale |
| 19:51.02 | Juggie | freeze the balls of a brass monkey |
| 19:51.15 | Beirdo | Edmonton's got Ottawa beat though |
| 19:51.20 | [TK]D-Fender | Edmonton... *shudder* Where the land is so flat you can see your dog running away from you for DAYS :D |
| 19:51.26 | sellout_ | zing! |
| 19:51.28 | Katty | Edmond Dantez |
| 19:52.11 | Katty | name the novel ;) |
| 19:52.29 | *** join/#asterisk GordonF (n=SumDude@myw-stp-196-35-171-40.sentechsa.net) |
| 19:52.29 | *** join/#asterisk Montag___ (n=Montag@194-185-151-70.f5.ngi.it) |
| 19:52.29 | Montag___ | anyone use grandstream ata 286 ???? |
| 19:52.35 | Katty | Montag___: hi |
| 19:52.44 | Katty | Montag___: how're you today? |
| 19:52.56 | Montag___ | Katty: fine thanks... |
| 19:53.06 | GordonF | Hi all. Anyone have any experience with A@H1.5 or is there somewhere I can go for help please? |
| 19:53.15 | Katty | Montag___: just a little impatient eh? |
| 19:53.35 | fugitivo | damn, it crashed again |
| 19:54.42 | funxion | hmmm |
| 19:55.29 | [TK]D-Fender | We need an A@H dedicated IRC channel.... |
| 19:55.46 | mog_work | there is one |
| 19:55.50 | mog_work | as well as an amp one |
| 19:55.51 | [TK]D-Fender | GordonF : just ask and see who'll answer. |
| 19:56.07 | [TK]D-Fender | mog_work : needs better advertising then! |
| 19:56.15 | file | mog_work: Mattttttttt |
| 19:56.21 | GordonF | mog_work : Where are they please? |
| 19:57.03 | bbz | GordonF: whats your q? |
| 19:57.42 | [TK]D-Fender | I helped a guy un-A@H himself :) Straight CLI baby! |
| 19:58.20 | chapeaurouge | im trying to call one computer with another, using linphone, on my private LAN. But it says it cannot find the users... they're there. wtf. can anyone take a look? |
| 19:58.29 | chapeaurouge | http://rafb.net/paste/results/m4eYRt88.html |
| 19:58.47 | GordonF | I installed A@H and got the Sangoma working. Created SIP users for X-Lite to break out thru the PSTN but they can't. It doesn't seem to see the context which is definatly there. They can dial one another etc tho just not out. If I create a context then I can dial out. |
| 19:59.42 | bbz | GordonF: are you exclusively using a@h or are you modifying the configs by hand as well? |
| 20:00.24 | GordonF | bbz: exclusively till I created my own context to test with |
| 20:00.38 | bbz | ok - 2nd question, is your zapata setup correctly? |
| 20:01.07 | GordonF | bbz : Yes otherwise my experience tells me I would'nt be able to use the PRI at all |
| 20:01.49 | bbz | when you place the call, and watch the CLI what is it telling you that is happening |
| 20:02.08 | GordonF | thats the scary part it does nothing with verbose at 50 |
| 20:02.41 | bbz | well, then it sounds like your dialing contexts arent matching anything |
| 20:02.46 | bbz | NXXXNXXXXX |
| 20:02.54 | juanjoc | fugitivo: Are you using straight 1.2.0 with no patches? |
| 20:03.17 | juanjoc | fugitivo: Can you provide a backtrace? |
| 20:03.45 | juanjoc | fugitivo: Check http://bugs.digium.com/view.php?id=5778 and see this is what you're experiencing. |
| 20:03.54 | *** join/#asterisk acertain (n=acertain@200.105.75.2) |
| 20:04.03 | bbz | Gordon: edit your extensions_additional.conf , scroll to bottom, you should see outrt-0xx-xxxxx , what do you have listed in that context? |
| 20:04.56 | *** join/#asterisk CRCC (n=crc@sharpstrong.proxy.astra-net.com) |
| 20:06.06 | acertain | Hi all |
| 20:06.08 | GordonF | bbz : I tried the 7/10 from the AMP default and verified that the context in the extadd.conf was correct |
| 20:06.36 | bbz | GordonF: /msg me and paste me your outrt context plz |
| 20:08.54 | tzafrir_laptop | could anybody point me at where exactly zaptel uses libpri? why do I need to rebuild zaptel after libpri update? |
| 20:12.40 | jdv79 | anyone know how to generate ulaw files with sox? |
| 20:13.48 | }cytrak{ | hi I just got a TE110p card but I already have asterisk compiled ... do I need to recompile it again after compiling the zapata modules ? |
| 20:14.20 | }cytrak{ | it that a requirement everytime? |
| 20:14.31 | bbz | no |
| 20:14.37 | bbz | are you using a@h ? |
| 20:14.48 | }cytrak{ | a@h ? |
| 20:14.54 | bbz | asterisk @ home |
| 20:14.59 | *** join/#asterisk marc324 (n=marc3234@206-248-153-235.dsl.teksavvy.com) |
| 20:15.03 | tzafrir_laptop | }cytrak{, if you have already built zaptel and chanzap, there is no reason to rebuild |
| 20:15.08 | }cytrak{ | no |
| 20:15.11 | Beirdo | I use asterisk in my home, but not A@H |
| 20:15.21 | bbz | lol |
| 20:15.34 | mog_work | i run aterisk@biz at home |
| 20:15.38 | }cytrak{ | tzafrir_laptop: I have not I have just build asterisk |
| 20:16.00 | bbz | cytrak: try genzaptelconf -s -d |
| 20:16.01 | }cytrak{ | anyway I need to update my 1.2 beta to the release version |
| 20:16.33 | *** join/#asterisk CRCC (n=crc@dslgw1.astra-net.com) |
| 20:17.40 | tzafrir_laptop | BTW: genzaptelconf is now part of latest debian packages, not to mention all xorcom packages. but IIRC genzaptelconf of a@h does not support PRI cards |
| 20:17.55 | *** join/#asterisk harryvv (n=none@S010600a0c93f6f7e.vs.shawcable.net) |
| 20:19.33 | docelmo | Anyone in here know how I work with remainders in C? |
| 20:19.41 | docelmo | I cant find shit on the net for it. |
| 20:20.01 | tzafrir_laptop | docelmo, what do you want to do? |
| 20:20.17 | docelmo | round down a number so I can kill the remainder |
| 20:20.46 | docelmo | or round up for that matter |
| 20:20.49 | gordonjcp | what, convert a real to an int? |
| 20:20.51 | tzafrir_laptop | docelm0, if it is an int, then simply / |
| 20:21.02 | Beirdo | to round down... (int)blah |
| 20:21.04 | docelmo | either way.. I need to round.. its not.. double |
| 20:21.16 | gordonjcp | docelmo: ? |
| 20:21.20 | Beirdo | to round (normally): (int)(blah + 0.5) |
| 20:21.23 | gordonjcp | yes |
| 20:21.31 | gordonjcp | that rounds to the nearest integer |
| 20:21.43 | *** join/#asterisk sgtpepper (n=apple@200-47-10-93.comsat.net.ar) |
| 20:21.43 | *** join/#asterisk abatista (n=Ariel@adsl-068-157-125-248.sip.mia.bellsouth.net) |
| 20:22.10 | docelmo | so I would do something like: (int)(10.5 + 1) and I would get 12? |
| 20:22.32 | sgtpepper | IronHelix: u there? |
| 20:22.35 | Beirdo | no |
| 20:22.41 | Beirdo | you would get 11 |
| 20:23.03 | Beirdo | what's the integer part of (10.5 + 1)? |
| 20:23.09 | docelmo | so the exact syntax is (int)(10.5 +1) |
| 20:23.16 | docelmo | 10 |
| 20:23.23 | Beirdo | no |
| 20:23.33 | Beirdo | 10.5 + 1 = 11.5 |
| 20:23.40 | KranZ | wont MOD or DIV help |
| 20:23.44 | KranZ | i forget which returns what |
| 20:23.45 | *** part/#asterisk kanobuho (n=kanobuho@201.124.179.200) |
| 20:24.03 | Beirdo | and yes, the syntax is (int)(blah + 0.5) for a normal rounding to the nearest integer of a float |
| 20:24.28 | KranZ | you dont have to use blah, bleh works just as well |
| 20:24.51 | Beirdo | the + 0.5 is to make anything 0.5 and over up to a 1.x, and anything below that will still be 0.x |
| 20:25.02 | sgtpepper | excuse me... i need help... |
| 20:25.03 | Beirdo | basic rounding :) |
| 20:25.38 | *** part/#asterisk Samoied (n=Samoied@popeye.opens.com.br) |
| 20:27.34 | docelmo | is floor a command in C |
| 20:27.38 | docelmo | err a function? |
| 20:27.57 | sgtpepper | excuse me... for a trunk... i'm using exten => 9.,1,Dial(oh323/anylink.dyndns.org/93${EXTEN:1}) |
| 20:28.03 | sgtpepper | i dial 9113 |
| 20:28.05 | *** join/#asterisk RoyK (n=roy@host-81-191-147-248.bluecom.no) |
| 20:28.09 | sgtpepper | and the answer i get is not found |
| 20:28.17 | sgtpepper | the context is added |
| 20:28.23 | Juggie | sigh you think (int) will do rounding? :P |
| 20:28.24 | bbz | try _9xxx, |
| 20:28.32 | Juggie | int willl just chop off any decimals |
| 20:28.33 | bbz | will it always be 3 digits? |
| 20:28.39 | KranZ | or .9 |
| 20:28.41 | bbz | 9xxx ? |
| 20:28.57 | iDunno | or _9. |
| 20:29.06 | bbz | yea for unlimtied digits _9. |
| 20:29.11 | iDunno | but that's bad, you should use _9x. |
| 20:29.16 | iDunno | (really) |
| 20:29.17 | docelmo | sigh I give up.. screw rounding.. |
| 20:29.19 | bbz | otherwise to insure just 3 |
| 20:29.50 | sgtpepper | with _9XXX |
| 20:29.53 | sgtpepper | still not found |
| 20:29.54 | Juggie | hahah |
| 20:30.06 | Juggie | if you cant round, then i dont forsea a big future for you in c programming |
| 20:30.24 | docelmo | I will figure it out later Jug.. |
| 20:30.30 | KranZ | google it |
| 20:30.33 | docelmo | I need to get this done now.. dont have time.. |
| 20:30.39 | *** join/#asterisk Uther_P (n=uther_p@66.180.120.82) |
| 20:30.44 | KranZ | http://code.dreamincode.net/snippet285.htm |
| 20:30.49 | sgtpepper | any idea bbz? |
| 20:31.01 | bbz | _9. |
| 20:31.19 | KranZ | sgtpepper: you're missing the _ |
| 20:31.22 | bbz | and if thta doesnt work, its probably not in the right context |
| 20:31.44 | bbz | <sgtpepper> with _9XXX <- he had it |
| 20:31.57 | KranZ | oh |
| 20:32.13 | KranZ | prolly needs to include the context |
| 20:32.21 | sgtpepper | exten => _9XXX,1,Dial(oh323/anylink.dyndns.org/93${EXTEN:1}) <= not working |
| 20:32.32 | KranZ | did u reload? |
| 20:32.32 | KranZ | heh |
| 20:32.36 | Juggie | just look at math.h |
| 20:32.37 | bbz | lol |
| 20:32.39 | sgtpepper | yep |
| 20:32.40 | sgtpepper | reload |
| 20:32.42 | sgtpepper | in cli |
| 20:32.51 | bbz | sgt: are you sure the call is being made in that context? |
| 20:32.55 | KranZ | does it show it registered that exten? |
| 20:32.59 | KranZ | set verbose 3 |
| 20:33.39 | sgtpepper | no debug output |
| 20:34.01 | KranZ | it should list the extensions when u reload of the verbosity is 3 or higher |
| 20:34.23 | KranZ | ...how hard was it for anyone to move from extensions.conf to a mysql db? |
| 20:34.33 | sgtpepper | <PROTECTED> |
| 20:34.44 | KranZ | what are you calling from |
| 20:34.47 | sgtpepper | <PROTECTED> |
| 20:34.52 | KranZ | k |
| 20:34.55 | sgtpepper | from a local client |
| 20:35.02 | KranZ | on the computer? |
| 20:35.07 | sgtpepper | i'm just starting with asterisk |
| 20:35.08 | sgtpepper | yep |
| 20:35.14 | docelmo | quit |
| 20:35.20 | iDunno | close... |
| 20:35.22 | bbz | put a / first |
| 20:35.24 | *** join/#asterisk TK9 (n=Administ@p54B2C6B2.dip0.t-ipconnect.de) |
| 20:35.28 | docelmo | wrong window |
| 20:35.46 | bbz | sgtpepper: what context is the phone set to? |
| 20:35.57 | KranZ | bbz, he's calling from the * box |
| 20:35.58 | bbz | [101] |
| 20:35.59 | bbz | context= ? |
| 20:36.16 | bbz | sgtpepper: define local client |
| 20:36.21 | *** part/#asterisk TK9 (n=Administ@p54B2C6B2.dip0.t-ipconnect.de) |
| 20:36.43 | sgtpepper | 192.168.1.31 is my vmware test enviroment |
| 20:36.54 | sgtpepper | 192.168.1.30 is a sip softphone |
| 20:36.57 | bbz | so do you have a softphone for your mac? |
| 20:37.02 | sgtpepper | yep |
| 20:37.04 | sgtpepper | sjphone |
| 20:37.11 | KranZ | what's the context in sip.conf |
| 20:37.11 | bbz | k so i assume he's using that kranz |
| 20:37.25 | sgtpepper | demo KranZ |
| 20:37.35 | KranZ | context should be local |
| 20:37.36 | bbz | sgt: where you register the phone in sip.conf, it should be [xxx] <-- extension and listed will be a context= |
| 20:37.38 | iDunno | which isn't the local context ;) |
| 20:37.50 | bbz | so you need context=local |
| 20:37.57 | KranZ | ^ |
| 20:38.05 | sgtpepper | i-ll try it |
| 20:38.08 | sgtpepper | <PROTECTED> |
| 20:38.11 | sgtpepper | is the softphone |
| 20:38.15 | *** part/#asterisk acertain (n=acertain@200.105.75.2) |
| 20:38.24 | KranZ | or include local in demo |
| 20:39.07 | Damin | Anyone running a non production 1.2.0 system they can test something for me on? |
| 20:40.25 | sgtpepper | <PROTECTED> |
| 20:40.26 | sgtpepper | <PROTECTED> |
| 20:40.42 | sgtpepper | that means that the h323 fxo gateway is down right? |
| 20:41.01 | bbz | it showed ip$localhost in the cli? |
| 20:41.09 | sgtpepper | yes |
| 20:41.16 | bbz | thats no good |
| 20:41.18 | KranZ | or the hostname doesnt resolve, or wrong port |
| 20:41.35 | jdv79 | mp3/wav to ulaw - anyone know a way? |
| 20:41.36 | sgtpepper | the router is listening in the default h323 port |
| 20:42.05 | sgtpepper | i'm using sip with gsm for the local peer |
| 20:42.06 | Damin | Anyone? |
| 20:42.13 | sgtpepper | and hh323 g723 for the remote |
| 20:42.22 | KranZ | did u dial 3185 when you called? |
| 20:42.30 | sgtpepper | nope |
| 20:42.33 | sgtpepper | 9113 |
| 20:42.41 | sgtpepper | and it shoul dial on the remote 93113 |
| 20:43.06 | MRH2 | has asterisk-sounds gone from 'v1-2-0' |
| 20:46.29 | sgtpepper | KranZ: pm ok? |
| 20:48.39 | MRH2 | and what happened to chan_modem.so? |
| 20:49.16 | xbmodder_lappy | ? |
| 20:49.16 | sgtpepper | KranZ: now i have ping to the gateway and everything.... but why can't asterisk connect to the gw? |
| 20:49.36 | bbz | sgt: if you see -- H.323 call 'ip$localhost/3185' cleared, reason 9 (Connection failure) |
| 20:49.38 | bbz | in the cli |
| 20:49.42 | sgtpepper | it should connect to port 1720 on than ip |
| 20:49.45 | bbz | are you removing ip$localhost ? |
| 20:49.46 | sgtpepper | yes bbz |
| 20:49.55 | bbz | because that shouldnt be a variable |
| 20:50.03 | sgtpepper | from where? |
| 20:51.04 | *** part/#asterisk santiago (n=santiago@208.195.215.124) |
| 20:53.36 | MRH2 | oh nevermnind about the modem thing |
| 20:53.42 | MRH2 | ;) |
| 20:56.11 | sgtpepper | where i change ip$localhost |
| 20:56.56 | sgtpepper | where do i remove ip |
| 20:57.07 | sgtpepper | $localhost bbz |
| 20:57.08 | sgtpepper | ? |
| 20:57.57 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 20:58.29 | sgtpepper | bbz: ? |
| 21:00.29 | *** join/#asterisk xtr (n=01928375@S01060012174cc0e1.vf.shawcable.net) |
| 21:00.37 | sgtpepper | :S |
| 21:00.41 | sgtpepper | :-S |
| 21:00.45 | lesouvage | IRQ-latency seems to be important. I checked it on my system with lspci -vvvv and I found out that the latency for the interrupts the system is using is 32. Is this a good or bad value for the irq-lantency? |
| 21:02.25 | jdv79 | this is incredible that converting from mp3 to ulaw would be such a problem |
| 21:03.07 | *** join/#asterisk konfuzed (n=KonfuzeD@H129.C72.B0.tor.eicat.ca) |
| 21:03.57 | sgtpepper | anyone? |
| 21:04.47 | sgtpepper | jdv79: u there? |
| 21:05.04 | jdv79 | i am |
| 21:05.12 | sgtpepper | a question... |
| 21:05.22 | *** join/#asterisk _Saback_ (n=_Saback_@20150081156.user.veloxzone.com.br) |
| 21:05.42 | sgtpepper | i'm trying to establish a call between a sip softphone using gsm |
| 21:05.59 | sgtpepper | and pstn behind a h323 g723 gateway |
| 21:06.02 | _Saback_ | Hi All |
| 21:06.14 | _Saback_ | What does it means? !! Got reject for frame 117, retransmitting frame 117 now, updating n_r! |
| 21:06.19 | sgtpepper | <PROTECTED> |
| 21:06.23 | sgtpepper | <PROTECTED> |
| 21:06.25 | sgtpepper | <PROTECTED> |
| 21:06.29 | sgtpepper | any thoughts jdv79 ? |
| 21:07.38 | sgtpepper | ? |
| 21:07.45 | *** part/#asterisk Utah_Dave (n=boucha@0-1pool139-17.nas28.salt-lake-city1.ut.us.da.qwest.net) |
| 21:10.27 | *** join/#asterisk rob314 (n=root@207.58.194.55) |
| 21:10.30 | MRH2 | ok asterisk sounds is in head but not in version tag 1-2-0 is that they way it should be? |
| 21:11.05 | MRH2 | not ruling out wiredness at this end... |
| 21:11.45 | jdv79 | sgtpepper, looks like you're not interpolating something you should be |
| 21:12.03 | sgtpepper | i'm new in asterisk |
| 21:12.11 | sgtpepper | so.. define interpolating? |
| 21:12.15 | sgtpepper | yo mean transcoding? |
| 21:12.26 | jdv79 | maybe its your fault - maybe its the drivers fault |
| 21:12.45 | *** part/#asterisk rob314 (n=root@207.58.194.55) |
| 21:12.55 | sgtpepper | where do i check? |
| 21:12.56 | jdv79 | interpolating is not asterisk specific - it means substituting in variable values |
| 21:13.07 | jdv79 | ask someone who knows a lot more than you and I:) |
| 21:13.08 | *** join/#asterisk rob314 (n=root@207.58.194.55) |
| 21:13.16 | sgtpepper | ohh ok |
| 21:13.36 | jdv79 | ip$localhost looks like its supposed to be something else |
| 21:13.54 | *** join/#asterisk file[laptop] (n=jcolp@mctnnbsa24w-142167049176.nb.aliant.net) |
| 21:14.01 | jdv79 | but for some reason it was never "interpolated" with the correct values |
| 21:16.24 | lesouvage | sgtpepper: if you copy and paste the relevant lines to pastebin.com I can take a look. I'm not an asterisk guru but maybe it helps. |
| 21:17.00 | sgtpepper | with the relevant lines u mean the config lines? |
| 21:17.17 | *** join/#asterisk the_jeka (n=the_jeka@20132219254.user.veloxzone.com.br) |
| 21:17.58 | iCEBrkr | http://www.voip-info.org/wiki/view/NVLineDetect |
| 21:18.01 | iCEBrkr | Is this for real? |
| 21:18.08 | iCEBrkr | or is it someone fishing for email addresses? |
| 21:18.43 | lesouvage | sgtpepper: the extensions of the phones (don't paste passwords) and the line you used to dial from one phone to the other. and the cli output. |
| 21:19.10 | sgtpepper | ok |
| 21:20.10 | lesouvage | set the verbose level to 10 in the cli or start the cli with asterisk -vvvvvvvvvr |
| 21:20.44 | lesouvage | THis way details of what is happiening will show up. |
| 21:22.18 | sellout_ | noooooooooo |
| 21:22.24 | sellout_ | my aastra is a brick now |
| 21:22.28 | sellout_ | failed firmware update |
| 21:22.33 | sgtpepper | lesouvage: http://pastebin.com/434699 |
| 21:22.45 | sellout_ | and running tcpdump on the interface it's on doesn't show any network activity |
| 21:22.52 | DrDeke | that's bad |
| 21:23.09 | sellout_ | :< |
| 21:23.59 | sellout_ | anyone ever worked with bricked aastra devices |
| 21:24.11 | harryvv | aastra a brick now? |
| 21:24.14 | sellout_ | any special key sequences I could hit to get it to try and pull a default fw from somewhere? |
| 21:24.27 | sellout_ | harryvv: I thinkso |
| 21:24.27 | DrDeke | i love devices that do that, but I don't know anything about aastras |
| 21:24.29 | sellout_ | boo hoo |
| 21:24.48 | sellout_ | like there should be a rom on it somewhere that holds a default firmware |
| 21:24.58 | harryvv | never heard of aastra |
| 21:26.18 | firelord | Does anyone know if Asterisk supports DVI4? |
| 21:27.09 | lunk | how can i get user input from either the dial-plan directly, or from an AGI script? |
| 21:27.54 | jdv79 | what is a .ulaw faile? |
| 21:27.59 | jdv79 | file rather |
| 21:31.45 | jdv79 | is it s wav file? |
| 21:32.44 | diamon | Similar, it's an audio format. |
| 21:33.06 | chapeaurouge | how do i detach from the asterisk console without shutting down asterisk? |
| 21:33.20 | lunk | exit ? |
| 21:33.23 | mog_work | exit |
| 21:33.27 | mog_work | but if you did asterisk -c |
| 21:33.29 | chapeaurouge | hmmm |
| 21:33.29 | mog_work | you cant |
| 21:33.32 | chapeaurouge | ok |
| 21:33.38 | mog_work | never start asterisk -c |
| 21:33.38 | chapeaurouge | i was wondering... |
| 21:33.41 | mog_work | unless your debugging |
| 21:33.43 | chapeaurouge | mog_work, ok |
| 21:33.49 | chapeaurouge | thx |
| 21:33.50 | mog_work | asterisk -g |
| 21:33.54 | mog_work | is the way to go |
| 21:34.17 | twisted[asteria] | or you could do it the safe route |
| 21:34.23 | twisted[asteria] | <PROTECTED> |
| 21:34.25 | twisted[asteria] | ;) |
| 21:34.46 | twisted[asteria] | that way it'll auto-restart if it crashes/gets inadvertently killed |
| 21:36.33 | jdv79 | is there anywhere i can find info on exactly what the .ulaw files are that * needs |
| 21:38.37 | *** part/#asterisk DrDeke (i=dekemar@deculator.engin.umich.edu) |
| 21:39.30 | *** join/#asterisk jmjones (n=jmjones@adsl-223-72-14.aep.bellsouth.net) |
| 21:42.06 | *** join/#asterisk fiber0pti (n=johndoe@207.114.199.98) |
| 21:42.40 | jmjones | i'm trying to use the "Monitor()" function to record both ends of a conversation. when i'm talking to the person on the other end, it sounds really clear, really smooth. when i listen to the recorded audio, it's really choppy. |
| 21:42.46 | jmjones | any tips? |
| 21:45.34 | diamon | I've seen oddness with Monitor on the mailing lists. One suggestion was to record to a ramdisk as I recall. |
| 21:46.03 | jdv79 | there are links about that too |
| 21:46.16 | jdv79 | the disk activity for recording is supposedly intense |
| 21:46.20 | }cytrak{ | when installing the zaptal modules do I first need to do a make modules_prepare on a 2.6 kernel ? |
| 21:46.51 | diamon | Darn if I know why * needs a 20ms write-time chunk, but hey. |
| 21:46.57 | }cytrak{ | I just did a make clean , make and I got a make26 error regarding no rule to make target 'modules' |
| 21:47.24 | jdv79 | why not buffer a bit? |
| 21:47.32 | fiber0pti | There any tools that have GUIs to utilize Asterisk RealTime or does everything have to be done with queries for right now? |
| 21:47.34 | *** join/#asterisk morale (n=russell@2001:5c0:8fff:fffe:0:0:0:3e5f) |
| 21:47.42 | h3x | shit, ive set up boxes that record 96 lines |
| 21:47.45 | h3x | so two quad t1 cards |
| 21:47.50 | h3x | and didnt have any problems like that |
| 21:48.05 | Nivex | ebbeh? chan_zap.c:10992: error: dereferencing pointer to incomplete type |
| 21:48.14 | h3x | maybe your computer sucks |
| 21:48.39 | ikarus | h3x: are you working for the CIA that you needed to do that :) |
| 21:49.00 | jmjones | h3x - it *does* suck, but i didn't think it sucked that bad..... |
| 21:49.08 | h3x | uh no, the cia dosent have to set up their own stuff coz CALEA is already built into the big iron switches |
| 21:49.23 | jmjones | maybe i'll try putting it on another machine....or maybe recording to ramdisk |
| 21:49.25 | jmjones | thx all |
| 21:49.31 | h3x | jmjones: what kind of machine is it |
| 21:49.47 | ikarus | h3x: I am kidding, but I seriously wonder why on earth you need that much monitoring |
| 21:49.55 | h3x | ikarus: call centers |
| 21:50.16 | h3x | hence the reason Monitor() and MonMux() Was written |
| 21:50.48 | ikarus | h3x: ah well, I don't see the use in recording every fucking thing unless the people you call are sue happy (erm, ah....) |
| 21:50.55 | diamon | Oh, about the stuttering, you may have some luck raising your PCI latency to 64. I just thought of that. |
| 21:51.02 | h3x | well one of them i set up is for people to activate their credit card |
| 21:51.13 | diamon | ikarus: Or a call center... |
| 21:51.22 | h3x | manual activations anyway |
| 21:51.26 | znoG | guys, how do I make a sip client NOT native bridge ?? |
| 21:51.32 | anthm | the file being recorded with choppyness in it is not related to the performance |
| 21:51.34 | *** join/#asterisk stkn (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 21:51.35 | *** join/#asterisk danalien (n=danalien@unaffiliated/danalien) |
| 21:51.48 | anthm | its bad math or something somewhere |
| 21:51.54 | h3x | anthm: I was about to get to that but ... we were talking some BS :) |
| 21:51.54 | znoG | when I call from SIP client A, through Asterisk A to another Asterisk B, SIP tries to talk to Asterisk B directly |
| 21:52.17 | h3x | I usually hear "that" when i play back ulaw recorded sounds as alaw |
| 21:52.21 | znoG | maybe nat=yes |
| 21:52.29 | h3x | i guess the same goes for slin to either |
| 21:52.39 | ikarus | diamon: the only call center I actually got to see the internals didn't record everything unless they hit a button |
| 21:53.04 | h3x | ikarus: mostly that is because dialogic cards suck balls |
| 21:53.14 | h3x | they dont have resources to record every call |
| 21:53.25 | ikarus | h3x: they had VoIP connectivity |
| 21:53.28 | h3x | unless you put in extra MSI cards you dont need |
| 21:53.34 | h3x | asterisk based? |
| 21:53.41 | ikarus | h3x: partially |
| 21:53.52 | h3x | well it depends on the application |
| 21:54.19 | ikarus | h3x: but they didn't have to record anything by law, so they didn't bother with it unless for training or harrasment |
| 21:54.34 | infinity1 | is voipjet down or is there another bug in my extensions? |
| 21:54.47 | h3x | the problem with call recordings is its always the one you didnt record that you needed. |
| 21:54.49 | infinity1 | it was working yesterday. hrmz. |
| 21:56.36 | juanjoc | znoG: Set canreinvite=no in sip.conf |
| 21:56.49 | jmjones | h3x it's a pIII 550 with about half a gig of ram, 2 160GB drives, nonraided |
| 21:56.57 | funxion | anyone here have any experience connecting * to a mitel sx2000 via e1 euroisdn? |
| 21:57.05 | ikarus | h3x: ah well, I sadly personally do not have to deal with such things (I would love to play more with VoIP, but I only have to deploy a small setup) |
| 21:57.52 | anthm | what format are you recording to and what are the formats of the 2 channels |
| 21:58.31 | *** join/#asterisk jebba (n=jebba@200-127-51-15.cab.prima.net.ar) |
| 21:58.54 | anthm | I bet you have someone with a 30 ms codec talking to a 20ms codec and some of the data is being ovelooked due to buffering samplecount bs |
| 21:59.11 | }cytrak{ | could someone give me a hand ... what do I have to do before compiling the zaptal modules ? the info at voip-info is not being very helpful neither the book i got |
| 21:59.41 | jmjones | anthm i've tried gsm and wav |
| 21:59.58 | jmjones | we're both using sjphone and using the same codecs as far as i know.... |
| 22:00.07 | anthm | there are 3 formats to consider 2 caller and the file |
| 22:00.08 | h3x | which codec |
| 22:02.56 | jmjones | anthm - what 3 formats are there? the asterisk book only showed 2 - gsm and wav. |
| 22:03.25 | jmjones | h3x - we are both using the asterisk default, which is i think a gsm.... |
| 22:03.25 | *** join/#asterisk kn0x (n=nunya@tor/session/x-56d02932150d725a) |
| 22:03.41 | kn0x | is anyone elso using Stanaphone? |
| 22:03.56 | *** join/#asterisk Ariel_ (n=Ariel@dsl-20-177.cofs.net) |
| 22:03.59 | h3x | no dude he means theres 3 endpoints that have a codec |
| 22:04.05 | kn0x | my registrations are failing after ztdummy compile... want to see if its just a cawinkydink |
| 22:04.05 | anthm | i am talking about the codecs for the 2 callers and the codec that you are saving the file to |
| 22:04.22 | jmjones | h3x oh - ok. sorry. |
| 22:04.27 | h3x | wav isnt a codec by the way |
| 22:04.40 | h3x | it can contain ulaw, alaw, gsm, and some other things |
| 22:04.51 | h3x | but that should be somewhat automatic |
| 22:05.01 | *** join/#asterisk Slmick (n=doobie@188.80-202-17.nextgentel.com) |
| 22:05.07 | firelord | Can asterisk recieve raw wav? |
| 22:05.12 | Slmick | Hi all |
| 22:05.12 | anthm | in this case when you say wav you mean slin |
| 22:05.16 | anthm | with a header |
| 22:05.19 | h3x | well, yeah |
| 22:05.48 | h3x | i encode gsm wavs though |
| 22:06.02 | Slmick | can i spit out my problem and ask for a opinion on it? or is that rude :) |
| 22:06.20 | anthm | yes i like those better asterisk calls them wav40 |
| 22:06.23 | anthm | 49 |
| 22:06.23 | jmjones | i feel like an amateur twit when it comes to codes and encoding and what not... |
| 22:06.38 | anthm | to debug why it's choppy you need to determine 3 values |
| 22:06.48 | jmjones | actually, i *am* an amateur twit.... |
| 22:06.56 | file[laptop] | Airespring, I swear I will hunt down all your employees and and teach them a lesson... |
| 22:06.57 | anthm | the format of each caller and the format of the file |
| 22:06.58 | h3x | dude, make the call and type "show channels" in your asterisk cli and it will tell you |
| 22:07.17 | h3x | what codec is being used on each channel |
| 22:07.32 | anthm | and gobbles it up |
| 22:07.42 | anthm | mmm |
| 22:07.54 | *** join/#asterisk Navman (n=p_e@62.108.206.82) |
| 22:08.06 | Slmick | have trouble getting asterisk to register against a voip sip account, get registration xxxx timed out constantly |
| 22:08.25 | anthm | file, i see, to have found rtp that doesnt suck! |
| 22:08.27 | jmjones | my other party is gone.....i'm recording locally and getting a pretty good recording. i'll see what i've got with all the pieces being on my lan... |
| 22:08.30 | anthm | seem |
| 22:08.36 | *** part/#asterisk Navman (n=p_e@62.108.206.82) |
| 22:08.41 | file[laptop] | anthm: did you kill someone to get it? |
| 22:08.59 | Slmick | might have something to do with nat.. have a setup like this: asterisk -> fw-nat -> adslmodem-nat -> internet -> voip provider |
| 22:08.59 | anthm | no just had to do a lot of evaluation and trial and err |
| 22:09.44 | Slmick | is it all the nat that makes the trouble? tried connecting with x-ten to the sip account and that worked great.. shouldnt asterisk work just like a sip client? |
| 22:09.54 | h3x | Slmick: Uhm. double nat dosent work with sip. |
| 22:10.03 | jdv79 | i'll ask again for sheer fun - does anyone know how to convert other audio files into ulaw that * can use? |
| 22:10.03 | anthm | run away while you still can!!! |
| 22:10.22 | stevek | jdv79: use sox |
| 22:10.35 | anthm | they call it sox in unix land and wavepad in windows world |
| 22:10.37 | Slmick | h3x when asterisk is a client too? thought that was more with external sip clients going in over double nat |
| 22:10.46 | h3x | dosent matter |
| 22:10.56 | h3x | sip just does call control |
| 22:11.09 | Slmick | hmm oki..so have to degrade it to single nat..then it _might_ work |
| 22:11.11 | h3x | it sets up another udp connection for rtp to transport voice (and video)... |
| 22:11.20 | jmjones | ok - i'm doing show channels and don't see what looks like an encoding type |
| 22:11.28 | jdv79 | oh - i might suck - thanks stevek for pointing me back there |
| 22:11.39 | jmjones | can i paste about 3 lines of what gets returned from "show channels"? |
| 22:11.39 | anthm | proto specific |
| 22:11.49 | anthm | sip show channels or iax2 show channels |
| 22:11.53 | anthm | to see the fmt |
| 22:11.53 | Slmick | h3x what if i specify the port range in rtp.conf and forward all those ports..? |
| 22:11.55 | h3x | haha |
| 22:12.00 | anthm | or show channel xyz with the exact name |
| 22:12.02 | jmjones | valentine*CLI> show channels |
| 22:12.02 | jmjones | <PROTECTED> |
| 22:12.02 | jmjones | SIP/jmjones-l1-495c (default 1 ) Up Bridged Call SIP/192.168.1.100-0815c740 |
| 22:12.02 | jmjones | SIP/192.168.1.100-0815c740 (default 100 2 ) Up Dial SIP/jmjones-l1|20|Ttr |
| 22:12.02 | jmjones | 2 active channel(s) |
| 22:12.13 | h3x | um, how about sip show channels |
| 22:12.14 | jmjones | sorry - that was about 5 :-( |
| 22:12.25 | h3x | you dont need to paste it :P |
| 22:12.41 | *** join/#asterisk Timoti (n=asqsa@85.99.35.211) |
| 22:12.54 | jmjones | the "Format" for both is gsm |
| 22:13.09 | anthm | and what are you writing to ? |
| 22:13.20 | jmjones | wav here. |
| 22:13.25 | jmjones | and this is pretty smooth. |
| 22:13.36 | h3x | try writing to .gsm |
| 22:13.40 | h3x | coz it makes more sense anyway |
| 22:13.47 | anthm | even better, WAV |
| 22:13.50 | h3x | yeah |
| 22:13.50 | jmjones | i'll probably have to wait until my other party is back and try sip show channels. |
| 22:13.59 | Rav1974 | I am going to be running a sip proxy on the firewall in front of ASTARO. I guess the benifit would be that I don't need to do any port forwarding right? |
| 22:14.00 | jmjones | but i did record gsm with him and it was really choppy. |
| 22:14.01 | anthm | then you get cake+eat it |
| 22:14.02 | h3x | capital WAV is wav header on a gsm formatted file |
| 22:14.28 | Timoti | Hi I have a technical question .... from my provider there is always Tarif announment " 5 EURO sent per min or etc " unfortunately I can not disable it .. for that reason are there any way to make 2 min pause at the beginning of call not to hear that traif announcment ? |
| 22:14.46 | Slmick | h3x, why does x-ten work over double nat? it uses sip too..? just confused |
| 22:14.48 | Timoti | sorry I meant 2 second |
| 22:14.50 | h3x | jmjones: your poor pIII is decompressing both sides of your call just to save it into files. |
| 22:14.59 | h3x | even though that should work, |
| 22:15.09 | h3x | it would make a hell of a lot more sense to just save it as .gsm or .WAV |
| 22:15.17 | anthm | i have a module on pbxfreeware called b64 that reads and writes the file as ascii mime text encoded WAV |
| 22:15.41 | anthm | so it's the same as WAV but it looks like email |
| 22:15.41 | h3x | what for emails? |
| 22:15.44 | anthm | you can cat the file |
| 22:15.58 | Timoti | no recommendation or hel for my question ? |
| 22:16.01 | anthm | and playback can play it |
| 22:16.16 | anthm | and if you cat it | sendmail the recip can play it too |
| 22:17.13 | *** join/#asterisk ComputerWarm (i=Computer@66.244.235.210) |
| 22:17.29 | docelmo | anyone know why printf("%f",float) would return a number like 30 characters long when a .005678 number went in? |
| 22:17.34 | ComputerWarm | Hello all question, is there anyway to turn the volume up with asterisk? |
| 22:18.03 | jmjones | you guys are fantastic. i'm going to put asterisk on a faster machine (maybe my laptop), monitor in gsm and check the "sip show channels" when my friend is available again.... |
| 22:18.05 | docelmo | Do it on the ATA or Zap cards |
| 22:18.22 | infinity1 | when trying to use voipjet ... i'm getting Nov 18 14:17:42 NOTICE[26138]: chan_iax2.c:2818 auto_congest: Auto-congesting call due to slow response |
| 22:18.28 | *** join/#asterisk santiago (n=santiago@208.195.215.124) |
| 22:18.33 | ComputerWarm | anyone? |
| 22:18.33 | funxion | I'm having an issue with an isdn e1 connecting to a mitel sx2000 the * box can call to the mitel but when the mitel sends a call to * it gets an immediate reorder tone and I see no action in * |
| 22:18.45 | funxion | please help me |
| 22:19.12 | infinity1 | ComputerWarm: zap channel? |
| 22:19.19 | ComputerWarm | infinity1 no its all sip |
| 22:19.26 | ComputerWarm | from the carrier |
| 22:19.26 | infinity1 | ComputerWarm: don't think so |
| 22:19.45 | ComputerWarm | infinity1 ya me neither but i can always hope; thanks |
| 22:20.51 | kn0x | NOTE: ALL THOSE WITH STANAPHONE FOR DID SERVICE- REGESTRATION USERNAME HAS CHANGED (www.stanaphone.com) |
| 22:21.06 | Slmick | which ports do i need to forward to * when i have * behind nat and want it to connet to a voip provider with sip? |
| 22:21.50 | ComputerWarm | interesting they come in here to spam now lol |
| 22:22.03 | docelmo | Slmick, typically with asterisk 10-20000 UDP |
| 22:22.27 | docelmo | as far as ATA use STUN it varies |
| 22:22.29 | Slmick | docelmo, or what i define in rtp.conf ? |
| 22:22.38 | h3x | and they cant spell, either. |
| 22:22.41 | *** part/#asterisk jdv79 (n=jdv79@u1057064.ul.warwick.net) |
| 22:22.57 | docelmo | you can define anything in rtp.conf but you have to open the same ports in your firewall also |
| 22:22.58 | Slmick | or have i missunderstood rtp.conf too.. :) |
| 22:23.11 | funxion | I'm having an issue with an isdn e1 connecting to a mitel sx2000 the * box can call to the mitel but when the mitel sends a call to * it gets an immediate reorder tone and I see no action in * can someone point me in the right direction |
| 22:23.13 | Slmick | oki |
| 22:23.19 | Slmick | thanks |
| 22:23.31 | infinity1 | anyone use voip jet and having a problem today? |
| 22:23.41 | *** part/#asterisk Slmick (n=doobie@188.80-202-17.nextgentel.com) |
| 22:23.48 | docelmo | funxion, check your inbound context |
| 22:24.03 | docelmo | make your you have exten => s,1,... or the DID being calls |
| 22:24.10 | funxion | docelmo inbound context has _X.,1, |
| 22:24.21 | funxion | but I dont see anything happen in cli |
| 22:24.30 | funxion | its like the call isnt hitting me |
| 22:24.49 | funxion | shouldn't I see something if a call is attempted inbound |
| 22:24.49 | docelmo | then make sure your context is correct for the mitel |
| 22:24.54 | docelmo | no.. |
| 22:25.02 | docelmo | you would have to do sip debug |
| 22:25.06 | docelmo | watch the trace |
| 22:25.10 | funxion | its not sip |
| 22:25.13 | funxion | its zap |
| 22:25.22 | docelmo | then check your zap config |
| 22:25.23 | funxion | e1 |
| 22:25.26 | docelmo | but you can zap debug also |
| 22:25.45 | funxion | whats the command for zap debug |
| 22:27.32 | fugitivo | pbx.c:785 in pbx_find_extension: Maximum PBX stack exceeded |
| 22:27.36 | fugitivo | what's that? |
| 22:27.46 | docelmo | ahh hell Im going home |
| 22:27.49 | docelmo | zap debug |
| 22:27.57 | docelmo | sigh.. |
| 22:27.58 | docelmo | ~docs |
| 22:28.00 | jbot | i guess docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com |
| 22:28.11 | *** part/#asterisk Timoti (n=asqsa@85.99.35.211) |
| 22:28.14 | fugitivo | ? |
| 22:29.36 | fugitivo | anyone? |
| 22:30.21 | fugitivo | i know that it's a stack overflow, but why? |
| 22:36.27 | *** join/#asterisk Utah_Dave (n=boucha@0-1pool139-17.nas28.salt-lake-city1.ut.us.da.qwest.net) |
| 22:40.21 | syzygyBSD | anyone know why asterisk wouldn't be recieving all of the digits dialed on the phone via sip? 2 different phones |
| 22:40.28 | justinu | fugitivo: infinite loop? |
| 22:40.44 | syzygyBSD | it catches at least 1 with a sipura, and none on the polycom |
| 22:42.33 | *** join/#asterisk klicek (n=tk@nat.bratniak.ds.pw.edu.pl) |
| 22:42.50 | infinity1 | anyone having a problem with voipjet? |
| 22:43.48 | fugitivo | justinu: you're saying that * has an infinite loop in zapata? |
| 22:44.41 | *** join/#asterisk bjohnson (n=bjohnson@i216-58-60-57.cybersurf.com) |
| 22:46.20 | *** join/#asterisk jdv79 (n=jdv79@u1057064.ul.warwick.net) |
| 22:47.06 | justinu | fugitivo: i'm saying that maybe your dial plan has an infinite loop |
| 22:48.04 | fugitivo | i don't think it's a problem with the dial plan |
| 22:48.31 | *** join/#asterisk lorinc (n=ang@caracas-1230.adsl.interware.hu) |
| 22:50.05 | *** join/#asterisk nobell (n=jdegraff@70.103.228.158) |
| 22:51.11 | nobell | can somebody explain to me what the cdrs are/is and why the h extension might mess it up? also, I can't get the h extension to run in a macro, any ideas? |
| 22:52.30 | bweschke | nobell: the h extension will not run in a macro |
| 22:52.58 | syzygyBSD | did asterisk add a new minimum length for dtmf tones? it isn't catching all of mine |
| 22:53.35 | nobell | I should be able to run the h extension from the context that called the macro though, right? |
| 22:54.06 | bweschke | h will make a mess of a CDR if you don't fork the CDR and do a NoCDR() on the CDR in the 'h' extension, instead of the dialed number as your destination you'll now have 'h' |
| 22:54.22 | justinu | snom is weird |
| 22:54.32 | justinu | default behavior is to play dial tone when you put someone on hold |
| 22:54.32 | *** join/#asterisk doodle (n=doodle@26.166.171.66.subscriber.vzavenue.net) |
| 22:54.39 | doodle | hi list |
| 22:54.42 | justinu | bizzare |
| 22:54.44 | bweschke | hello |
| 22:55.26 | bweschke | nobell: a macro isn't a call. so termination of a macro doesn't mean that 'h' will be the next iteration in the dial plan after the macro finishes |
| 22:55.57 | *** join/#asterisk sozo (n=sozo@pppoecl90208.minlos.no) |
| 22:57.49 | doodle | would anyone be interested in a non-phone person but IT based overview of Asterisk? a client of mine needs a new PBX solution and I hear this is a great tool to use, if one can figure it out. |
| 22:57.54 | nobell | if I goto a macro from context A, and make a call in that macro, where will asterisk look for extension h? (I've tried all kinds of stuff already) |
| 22:58.02 | *** join/#asterisk fulgas (n=fulgas@a81-84-117-79.cpe.netcabo.pt) |
| 22:58.09 | doodle | basically, i need to know the basics! |
| 23:00.13 | nobell | okay, so I should probably be using the CDRs to finish up the call and send info to a web server about the call, is that right? |
| 23:00.42 | jdv79 | i'm getting choppy audio on a call but i get no packet loss, and low latency and jitter - what else could it be? |
| 23:01.01 | doodle | (Guess that's a no...haha, thanks anyway gang - I'll lurk for a while if anyone would be willing to explain the basics to me) |
| 23:02.16 | justinu | doodle, going rate is $150/hr |
| 23:02.49 | doodle | for talk or setup? |
| 23:03.01 | justinu | talk is part of setup, no? |
| 23:03.14 | jdv79 | 150/hr - where are you that you can charge that much? |
| 23:03.16 | many | easy talkin is 150, setup is 250. |
| 23:03.32 | justinu | los angeles |
| 23:03.32 | harryvv | digium charges 175 per hour |
| 23:03.43 | jdv79 | and they wrote it |
| 23:03.44 | doodle | guess it could be, $150/hr doesn't scare me, i hear this is a good system but i need basic info on it to decide if it's for my client |
| 23:03.48 | justinu | for $150/hr, i'm available now, btw. |
| 23:03.50 | jdv79 | that's rather brazen a rate |
| 23:04.20 | justinu | depends on who you are, i suppose. |
| 23:04.27 | many | and where. |
| 23:04.35 | harryvv | considering that propriatory pbx vendors charge |
| 23:04.41 | justinu | yeah |
| 23:04.58 | many | and actually, being a storyteller reading some webpages to doodle should cost 300. |
| 23:05.02 | *** join/#asterisk zeedo (n=zeedo@80.68.92.188) |
| 23:05.03 | harryvv | asterisk is not a simple system and the more features you add the higher the complexity. |
| 23:05.07 | justinu | doodle: what locale? |
| 23:06.07 | doodle | Texas - I can read docs all day long and get no where or lots of places, but if this is the place to go for setup help and not discussion, I'll be back, if I can find out more someplace. |
| 23:06.20 | justinu | mainly tech info here. |
| 23:06.29 | doodle | I figured $150/hr for a setup guy, which is reasonable! |
| 23:06.44 | doodle | Roger that justinu...thanks anyways |
| 23:06.55 | justinu | you can find me here if you want to talk |
| 23:07.00 | doodle | :-) |
| 23:08.07 | *** join/#asterisk BoRiS (i=boris@S010600112f38a61e.wp.shawcable.net) |
| 23:12.41 | syzygyBSD | stupid asterisk 1.2, I am not even going to bother trying to figure out why dtmf tones arn't working, it works great on 1.0.9 |
| 23:12.52 | justinu | syzygyBSD: i know why they don't work |
| 23:13.01 | syzygyBSD | oh, please tell me? |
| 23:13.03 | justinu | and it's an easy fix. |
| 23:13.14 | *** part/#asterisk doodle (n=doodle@26.166.171.66.subscriber.vzavenue.net) |
| 23:13.18 | syzygyBSD | just spent a while tring to figure out why it only catches some of them |
| 23:13.25 | justinu | http://bugs.digium.com/view.php?id=5780 |
| 23:16.06 | harryvv | possible to run two seperate ivrs in asterisk? |
| 23:16.20 | harryvv | two or more for incomming sip |
| 23:16.26 | syzygyBSD | you can run as many as you want |
| 23:16.33 | harryvv | Thats what I thought |
| 23:22.27 | *** join/#asterisk Gennaro (n=Gennaro@ppp-62-10-136-193.dialup.tiscali.it) |
| 23:22.43 | Gennaro | hi some one can help me with 1 installation? |
| 23:23.02 | Gennaro | i installed * on 2 machines.. |
| 23:23.11 | Gennaro | and i have 2 different prob.. |
| 23:23.22 | Gennaro | can i explain? |
| 23:23.29 | *** join/#asterisk qpkt (n=alex@S0106006097940f68.vw.shawcable.net) |
| 23:24.55 | qpkt | Does anyone know whether asterisk will convert sip inband dtmf into rtp control packets when sending over iax channel ? I got sip on lan and iax trunk to my voip provider. |
| 23:24.55 | Gennaro | is there someone?!? |
| 23:26.13 | justinu | snom 360 is fucked! |
| 23:26.24 | justinu | i had one line on hold, and I call another person... |
| 23:26.31 | justinu | when I hang up call 2, it drops both calls! |
| 23:26.32 | justinu | wtf? |
| 23:26.46 | qpkt | From what I understand iax should forward dtmf out of band but I could be wrong. |
| 23:26.55 | h3x | justinu: maybe the 1st caller hung up on you :P |
| 23:27.00 | justinu | not possible. |
| 23:27.01 | bweschke | nobell: sorry. walked away... it should be looking for 'h' in the context where the call is ending |
| 23:27.02 | justinu | i was testing. |
| 23:27.07 | bweschke | at least - that's been my experience with it |
| 23:27.11 | h3x | did you upgrade your firmware |
| 23:27.20 | justinu | qpkt: it will do that on SIP -> SIP calls... not sure about IAX |
| 23:27.28 | Gennaro | some one can tell me what should i do to use xlite / sjphone on my asterisk? |
| 23:27.54 | Gennaro | i remove ; from Sip.conf but... |
| 23:28.07 | qpkt | yeah I also tried changing codec on iax channel to ulaw/alaw but this did not help much |
| 23:28.22 | qpkt | since I thought gsm was not passing inband |
| 23:28.37 | *** join/#asterisk RoyK (n=roy@55.80-202-161.nextgentel.com) |
| 23:29.29 | qpkt | on SIP to SIP it works fine, but my setup is analog -> SIP -> SIP -> IAX -> PSTN |
| 23:29.48 | qpkt | obviously iax is terminating inband somewhere |
| 23:30.29 | justinu | can't really help you much with IAX, sorry |
| 23:32.01 | qpkt | thanks |
| 23:32.19 | qpkt | maybe I could try sip trunk instead of iax |
| 23:32.37 | qpkt | do I need anything special to have sip trunk work over nat ? |
| 23:32.41 | *** join/#asterisk SuperB (n=chatzill@206.80.108.124) |
| 23:33.13 | Gennaro | some one can say me why i cant use my server as local provier? |
| 23:34.29 | Gennaro | i read that i need to change sip and i do so.. i delete ; from xterm |
| 23:34.31 | SuperB | Quick Question - Is anyone still having problems with chan_modem.so not being created even after downloading the updated 1.2 release? This is on a red hat 8.0 system |
| 23:34.42 | fugitivo | me |
| 23:34.58 | fugitivo | just remove the line load => chan_modem.so from your modules.conf |
| 23:35.10 | SuperB | thanks |
| 23:36.23 | justinu | oic why the snom hung up call one! |
| 23:36.28 | justinu | it tried to do a transfer |
| 23:36.32 | justinu | wtf? hold is not transfer! |
| 23:36.36 | Gennaro | in my sip .. [xlite1] Turn off silence suppression in X-Lite ("Transmit Silence"=YES)! Note that Xlite sends NAT keep-alive packets, so qualify=yes is not needed type=friend regexten=1234 ; When they register, create extension 1234 username=xlite1 callerid="Jane Smith" <5678> host=dynamic nat=yes ; X-Lite is behind a NAT router canreinvite=no ; Typically set to NO if behind NAT |
| 23:36.36 | Gennaro | disallow=all allow=gsm ; GSM consumes far less bandwidth than ulaw allow=ulaw allow=alaw so |
| 23:36.44 | Gennaro | can i use a phone? |
| 23:36.52 | fugitivo | Gennaro: pastebin |
| 23:37.42 | *** join/#asterisk grimse (n=grimse@p5481FAF5.dip.t-dialin.net) |
| 23:39.20 | Gennaro | fugitivo, i'm very affraid can u say me who it works? |
| 23:39.25 | Gennaro | fugitivo, i'm very affraid can u say me how it works? |
| 23:39.40 | justinu | lol |
| 23:39.44 | justinu | afraid of what? pastebin? |
| 23:40.04 | Gennaro | no.. |
| 23:40.20 | fugitivo | ~pastebin |
| 23:40.21 | jbot | i guess pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca/ |
| 23:40.30 | X-Files | i have one question in asterisk: Calling to me (use exten => 1,1,Dial(SIP/1,60,tTrhwg) ) but this Line busy, i need in talking process beep mode (about new call). My gateway not support call_wait, this can configure to asterisk ?? |
| 23:40.44 | fugitivo | Gennaro: paste your lines there, so it'll be easier to read |
| 23:40.48 | Gennaro | how .. |
| 23:40.51 | Gennaro | ok.. |
| 23:40.56 | Gennaro | i'm going... |
| 23:41.01 | Gennaro | sorry.. |
| 23:41.29 | kippi | anyone know much aboutABE login |
| 23:41.47 | *** join/#asterisk simprix (n=simprix@miwi1dsl-a283.wi.tds.net) |
| 23:41.50 | *** part/#asterisk simprix (n=simprix@miwi1dsl-a283.wi.tds.net) |
| 23:41.55 | *** join/#asterisk simprix (n=simprix@miwi1dsl-a283.wi.tds.net) |
| 23:42.43 | simprix | I have a asterisk 1.0.9 box with voicepulse connect. I have a x-ten client attached. When i am using it for a outgoing call there is a slight hiss on the line when i am silent ? |
| 23:43.19 | *** join/#asterisk klictel (n=klictel@modemcable185.108-200-24.mc.videotron.ca) |
| 23:43.53 | Gennaro | http://pastebin.ca/29161 |
| 23:43.56 | Gennaro | fugitivo |
| 23:44.47 | justinu | simprix: codec? |
| 23:45.31 | simprix | I have tried all the codecs |
| 23:45.44 | simprix | it doesn't matter what client i use |
| 23:45.50 | simprix | either iax or sip |
| 23:45.54 | *** join/#asterisk santiago (n=santiago@208.195.215.124) |
| 23:46.18 | fugitivo | simprix: did you turn off silent supression? |
| 23:46.36 | justinu | simprix: does it still happen if you mute xten? |
| 23:46.36 | simprix | Transmit Silence = Yes |
| 23:46.38 | hardwire | my queues aren't timing out |
| 23:46.42 | hardwire | anybody have experience with that? |
| 23:46.50 | fugitivo | Gennaro: what's the problem? |
| 23:46.58 | simprix | it doesn't happen on any client if i mute the mic |
| 23:47.05 | justinu | sounds like echo then |
| 23:47.20 | justinu | your own background noise reflected back |
| 23:47.32 | Gennaro | i do so becouse my guide say that i can register a telephone if i do so.. |
| 23:47.32 | simprix | is there anyway to fix that |
| 23:47.37 | justinu | better headset? |
| 23:47.41 | simprix | ok |
| 23:47.46 | Gennaro | but i don't know rtp port |
| 23:47.53 | simprix | what is a good headset |
| 23:47.54 | justinu | maybe playing with the audio tuning wizard too |
| 23:48.05 | Gennaro | i belive and other param to pass to get registred |
| 23:48.08 | justinu | simprix: sorry, i only use hardphones |
| 23:48.12 | justinu | don't have any advice for you |
| 23:48.21 | justinu | maybe your mic gain is too high |
| 23:48.39 | simprix | where can i adjust that |
| 23:48.45 | Gennaro | at work i have an pbx based on askerisk that work with INT and i don't know what should i do,, |
| 23:48.45 | justinu | right on the phone panel |
| 23:48.59 | Gennaro | dont registed. |
| 23:49.57 | Gennaro | and the 2° prob is that on my machine in this moment if i call asterisk -vvvg dont' open console to sip debug |
| 23:50.03 | justinu | ok, finally got the snom to behave like a rational phone |
| 23:51.15 | Gennaro | .. asterisk ready end |
| 23:51.42 | Gennaro | fugitivo can u say me how to configure a sip phone on asterisk? |
| 23:51.53 | *** join/#asterisk pandect (n=ender@69-162-25-193.stcgpa.adelphia.net) |
| 23:52.59 | pandect | anybody have any ideas why "100*", agent sign-in, wont' work from an ATA extension, but it will work from other extensions? |
| 23:53.15 | pandect | When I dial 100* from the ATA extension, it just rings unavailable, fast busy signal |
| 23:53.33 | fugitivo | pandect: maybe your ata configuration is doing something with the * |
| 23:53.42 | Qwell | So...the words "cue" and "queue". |
| 23:53.47 | pandect | fugitivo: I was thinking about that, but I can't track anything down |
| 23:53.52 | Qwell | Does one "cue" music, or do they "queue" music? |
| 23:53.53 | pandect | fugitivo: It's an SPA-1001 |
| 23:53.53 | fugitivo | pandect: what ata? |
| 23:54.05 | justinu | cue is like ff/rewind |
| 23:54.12 | Qwell | justinu: I know the answer. :p |
| 23:54.22 | justinu | then why are you asking? |
| 23:54.26 | Qwell | but, which does one do? |
| 23:54.28 | fugitivo | pandect: i don't have any spa, but i bet it's the ata that does something with the *, did you check the config? |
| 23:54.32 | Qwell | I have a point... |
| 23:54.40 | fugitivo | .cue, .bin |
| 23:54.48 | pandect | fugitivo: Yeah, I can't locate any options that have any mention of a "*" |
| 23:54.58 | Qwell | one could do either. essentially, they could both mean the same thing |
| 23:55.14 | Qwell | If you cue the music, it plays instantly. If you queue it, it could also play instantly, if the queue is empty. |
| 23:55.14 | simprix | is jitterbuffer for SIP |
| 23:55.26 | Qwell | therefore, cue = queue |
| 23:55.31 | Qwell | I love the English language |
| 23:57.00 | *** join/#asterisk Flauto (n=zhao@c-24-14-197-214.hsd1.il.comcast.net) |
| 23:57.08 | Gennaro | fugitivo works.. |
| 23:57.17 | `Sauron | cue != queue |
| 23:57.30 | `Sauron | You don't play with a pool queue |
| 23:57.30 | Gennaro | so can i ask u another thing ? |
| 23:57.33 | Qwell | `Sauron: in certain circumstances, they do the same thing |
| 23:57.36 | `Sauron | you do, however, play with a pool cue |
| 23:57.40 | Qwell | s/do/are/ |
| 23:58.20 | Qwell | in my opinion, there should never be ambiguity like that in languages... |
| 23:58.35 | `Sauron | Qwell: Welcome to english. |
| 23:58.42 | Qwell | That was my point. :p |
| 23:58.52 | justinu | qwell: you should learn lojban then... |
| 23:58.59 | `Sauron | german/norwegian doesn't have that kind of ambiguity |
| 23:59.09 | `Sauron | Feel free to speak german from now on. :) |
| 23:59.18 | Qwell | `Sauron: no words sound the same at all? |
