| 00:00.12 | jaytee | voip-info is gone for good? or just offline at the moment? |
| 00:02.10 | digitalirony | looks like its down |
| 00:04.03 | rue_mohr | [TK]D-Fender, you still in a descent mood or should I ask tommorow |
| 00:05.08 | Daejeo | jaytee: JT? |
| 00:05.14 | Daejeo | same guy? |
| 00:05.19 | jaytee | no |
| 00:06.20 | Daejeo | Defraz: what are you trying to achieve ? |
| 00:06.20 | *** join/#asterisk path_ (n=path@pc-15-190-86-200.cm.vtr.net) |
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| 00:08.49 | rue_mohr | ok if I give each line on the analog card an extension, and make the line keys speed dials to those number, and asterisk dials the phones on the accounts of the buttons for the lines, then you will be able to answer them cause their ringing, and when their ringing your not dialing out on them |
| 00:08.57 | rue_mohr | am I making sense? |
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| 00:16.46 | jaytee | rue_mohr, no sense at all, are we talking FXO ports or FXS? what exactly are you trying to accomplish? |
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| 00:22.24 | rue_mohr | fxo |
| 00:22.45 | rue_mohr | I'm trying to fully assiciate line keys with channels on the card |
| 00:23.20 | *** join/#asterisk m57 (n=m57@dhcp16201.myzipnet.com) |
| 00:24.29 | rue_mohr | ok each line button logs in with its own account |
| 00:24.50 | rue_mohr | each line rings the same buttons account on all phones |
| 00:24.58 | rue_mohr | that does incomming calls |
| 00:25.22 | rue_mohr | now for outgoing calls, I need it so that when you hit the line key, it gives you the line |
| 00:25.55 | rue_mohr | so I need to make each line button a speed dial to the extension for the card channel |
| 00:26.00 | rue_mohr | that make sense? |
| 00:26.59 | jaytee | rue_mohr, in a very odd sort of way but probably not the way you're hoping it would make sense. |
| 00:27.18 | rue_mohr | each one is a seperate number for a seperate business |
| 00:27.35 | rue_mohr | the office staff have to be able to choose what number their call goes out on |
| 00:27.42 | rue_mohr | aka what analog line |
| 00:28.24 | jaytee | rue_mohr, so basically you want to segment your dialplan to support multiple customers? and guarantee that when they make an outbound call they always use the same FXO line? |
| 00:29.08 | rue_mohr | right! |
| 00:29.23 | rue_mohr | accept each of the 4 business goes to all 4 phones |
| 00:29.39 | rue_mohr | cause no staff member belongs to a specific business |
| 00:29.46 | rue_mohr | :) fun eh? |
| 00:30.22 | jaytee | rue, strange but doable |
| 00:31.18 | jaytee | you need to figure out how contexts work in asterisk. |
| 00:32.10 | rue_mohr | its not asterisk I'm having trouble with its the polycom601 |
| 00:32.31 | rue_mohr | it wont allow me to do a line button as a speed dial |
| 00:32.56 | jaytee | rue_mohr, line buttons are not speed dial buttons |
| 00:33.02 | rue_mohr | :) |
| 00:33.11 | rue_mohr | there is a setting I'm trying to get working |
| 00:33.20 | rue_mohr | call.autoOffHook.1.enabled="1" |
| 00:33.30 | rue_mohr | and its not working |
| 00:34.34 | *** join/#asterisk hypa7ia (i=hypatia@antenora.aculei.net) |
| 00:34.44 | rue_mohr | jaytee, its as close to a key'd system as their gonna get |
| 00:36.36 | jaytee | rue_mohr, do you have Polycom FTP provisioning working? |
| 00:36.54 | jaytee | or are you trying to configure the phone through the web interface? |
| 00:40.11 | rue_mohr | hell no |
| 00:40.25 | rue_mohr | ok, how the heck do i make these line buttons do this |
| 00:41.49 | jaytee | you can't alter that setting on the Polycom through the web browser. you need to put it in the config file for the particular phone you're programming using the FTP provisioning but using that function only creates more problems. |
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| 00:43.14 | rue_mohr | like I said, I'm NOT using the web interface |
| 00:43.32 | rue_mohr | your distracted, cause I posted an xml value just a min ago |
| 00:43.43 | rue_mohr | call.autoOffHook.1.enabled="1" |
| 00:43.53 | rue_mohr | is not working for me |
| 00:44.14 | jaytee | right, and call.autoOffHook.1.Contact="somenumber" |
| 00:44.37 | rue_mohr | wehre contact is something in a list or its an extension number? |
| 00:44.41 | rue_mohr | they aren't clear |
| 00:44.59 | jaytee | it's an extension or dialable number |
| 00:45.04 | rue_mohr | oh sip url |
| 00:45.10 | rue_mohr | hmm |
| 00:45.27 | rue_mohr | but I put '20' in it (valid extension) and it didn't work |
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| 00:47.20 | rue_mohr | I'll try again |
| 00:47.23 | jaytee | so you've got 4 analog lines and 4 polycom 601's? |
| 00:47.54 | rue_mohr | one of the line icons is a keypad, Its from the default config, It wont go away |
| 00:48.15 | rue_mohr | the phones are not each related to a line |
| 00:48.19 | rue_mohr | all lines go to all phones |
| 00:48.35 | rue_mohr | each person takes and makes calls for all 4 companies |
| 00:48.46 | rue_mohr | they choose the line depending what their doing |
| 00:49.00 | rue_mohr | right now they basically have 4 analog sets on each desk |
| 00:49.12 | rue_mohr | with 4 call display boxes |
| 00:51.25 | rue_mohr | would be nice if the web interface had a reboot button |
| 00:51.26 | *** part/#asterisk Defraz (n=T0tal@fw-poky.fuzecore.com) |
| 00:52.05 | rue_mohr | seeing as its not good for much else |
| 00:54.06 | *** part/#asterisk hypa7ia (i=hypatia@antenora.aculei.net) |
| 00:54.47 | jaytee | so setup 4 incoming contexts for each channel and 4 separate contexts for outgoing calls ( one for each company and pointing to it's dedicated analog line. Have each analog channel's ). Program internal SIP extensions for each phone so the line numbers are unique to each phone but related to a particular company, then when a call comes in one of the analog lines it dials the SIP lines on all the phones that are associated with that company. When |
| 00:54.48 | jaytee | one of the phones wants to make an outbound call if your outbound context is included in another context for that phone it will only have that analog trunk available to use. |
| 00:55.47 | rue_mohr | haha its been working, I moofed one of the dialplan entires (for the only button I been testing) |
| 00:56.00 | jaytee | well, carry on then |
| 00:56.32 | rue_mohr | jaytee, just reading your comment again, sounds like what I'v done |
| 00:57.13 | rue_mohr | yes, incomming calls are easy, the outgoing calls I have to do with that auto pickup thing |
| 00:59.46 | jaytee | rue_mohr, think of each line on your Polycom as a separate SIP device, it has it's own entry in sip.conf so it can also have it's own context. even if the number of phones is more than the number of analog lines you have you can still do it all in the dialplan without a "speed dial" button. |
| 01:00.16 | rue_mohr | yes, if I start thinking of it as a direct replacement for the 4 analog phones I should be ok |
| 01:00.43 | jaytee | so you can have two contexts included in the context for each phone, one for all the internal extensions and one for the outbound trunk with the outbound trunk context being unique. |
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| 01:13.44 | rue_mohr | I'm setting up the fxo channels with extension numbers |
| 01:14.02 | rue_mohr | so the line keys dial them automatically when you pick them up |
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| 01:40.30 | jondecker76 | can someone help me with configuring an SPA 3000? |
| 01:40.50 | jondecker76 | I'm trying to use the Voxilla configuration wizard, but it doesn't make sense |
| 01:41.08 | jondecker76 | I've also read the entire "Asterisk: The future of Telephony" book |
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| 02:11.06 | jaytee | jondecker76, try this link: http://wiki.2l2o.com/index.php/Linksys_SPA-3102 |
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| 02:13.56 | SlicerDicer | yo [TK]D-Fender whats that polycom you love so much? |
| 02:14.16 | SlicerDicer | the Joygasam bedside phone ;-) |
| 02:15.03 | SlicerDicer | and how do I adjust music on hold volume? the book does not say volume control really? |
| 02:15.13 | SlicerDicer | do I need to recode it? |
| 02:16.14 | outtolunc | a pillow works nicely <G> |
| 02:17.09 | SlicerDicer | nvm ip 301 joygasam is what he had |
| 02:17.15 | SlicerDicer | history ftw |
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| 02:33.00 | [TK]D-Fender | SlicerDicer: No, I just said I'd rather use my IP 301 than the 57i CT that I had |
| 02:40.01 | *** join/#asterisk wtsexton00 (n=tim@75.151.195.117) |
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| 02:43.02 | wtsexton00 | anyone ever have a polycom with two lines registered using the second line instead of the first if you select it |
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| 03:02.45 | wtsexton00 | polycoms are funky |
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| 03:09.47 | joako | wtsexton00: So you are saying if you press the line2 key you get a dialtone, but it goes out line 1? |
| 03:10.00 | wtsexton00 | say you pick up line one |
| 03:10.15 | wtsexton00 | dial the voicemail extension, in my case 5080 |
| 03:10.34 | wtsexton00 | asterisk sees it coming from line2 |
| 03:11.32 | drmessano | Sounds like a phone programming issue |
| 03:11.34 | joako | wtsexton00: Are you pressing the line 1 button, or just picking up the handset / pressing "new call"? |
| 03:11.44 | wtsexton00 | pressing the line |
| 03:11.50 | wtsexton00 | drmessano, I hope so |
| 03:11.59 | wtsexton00 | hence me seeking help :) |
| 03:12.24 | joako | wtsexton00: Have you gone through the config file and made sure everything is correct? What phone and what firmware version? |
| 03:12.37 | wtsexton00 | 550, 3.1 |
| 03:12.51 | drmessano | Go over the config.. Its almost completely obvious that line1 button seems to pick up line2 |
| 03:13.29 | wtsexton00 | which is odd, since they ring correctly |
| 03:13.44 | NovceGuru | asdf |
| 03:13.52 | NovceGuru | ok, ssh session didn't die. |
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| 03:13.57 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
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| 03:15.08 | [TK]D-Fender | wtsexton00: Nothing is obvious since we aren't looking at your configs and the SIP degub of the "confused" call. |
| 03:15.33 | wtsexton00 | I'm working on posting that up |
| 03:17.18 | *** join/#asterisk dandate2 (n=dandate2@c-71-202-125-220.hsd1.ca.comcast.net) |
| 03:17.21 | dandate2 | hey guys now that i have my asterisk box configured can I also have it perform tasks like web hosting, smtp relays? |
| 03:18.06 | joako | wtsexton00: any chance the two SIP accounts have the same password? I've noticed asterisk doesn't seem to like diffrent SIP users/peers with the same password |
| 03:18.14 | [TK]D-Fender | dandate2: Would you like fries with that, sir? |
| 03:18.27 | wtsexton00 | no, they don't use passwords |
| 03:18.29 | wtsexton00 | just peers |
| 03:18.57 | *** join/#asterisk Hanif08 (n=bucoo77@netop.jaring.my) |
| 03:19.09 | joako | so you auth using what? IP? username only? |
| 03:19.21 | [TK]D-Fender | CRAZY |
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| 03:20.22 | dandate2 | just wondering if it was possible, godaddy.com is running a bill up on me and was wondering if i could use the same linux box for various tasks or only one thing |
| 03:20.26 | drmessano | [TK]D-Fender: I already has this out with someone else earlier wants to install FreePBX/Asterisk on an 8 core box with almost zero load since "We already have the box and its being underutilized" |
| 03:20.54 | drmessano | This is when i hope for awesome XSS attacks to validate what I already know to be true |
| 03:21.18 | *** join/#asterisk CrazyTux (n=brandon@ip68-111-67-4.oc.oc.cox.net) |
| 03:24.34 | wtsexton00 | http://c3pfro.org/~tim/ |
| 03:24.36 | mmlj4 | analog gateways (FXO banks).... what's a good brand? reliable and not too expensive? |
| 03:26.03 | wtsexton00 | There are the phone configs if someone would like to look at them |
| 03:26.32 | wtsexton00 | running a fairly basic configuration |
| 03:26.38 | [TK]D-Fender | wtsexton00: Where is the SIP debug of the failed call? Where are your * configs? |
| 03:26.54 | *** join/#asterisk LemensTS (n=customgt@adsl-70-238-154-243.dsl.stlsmo.sbcglobal.net) |
| 03:27.05 | LemensTS | anyone here use sourceforge cvs |
| 03:30.45 | wtsexton00 | I don't think its in the * config |
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| 03:32.27 | wtsexton00 | * receives the line2 address when line1 is used |
| 03:32.52 | [TK]D-Fender | wtsexton00: You're 1/2 right... |
| 03:33.13 | wtsexton00 | I'm not on site so I can't produce a debug atm |
| 03:33.29 | [TK]D-Fender | wtsexton00: Well if you want to fix something try to be in a position to. |
| 03:33.38 | [TK]D-Fender | wtsexton00: Ask again when you have access to it |
| 03:34.59 | wtsexton00 | ah, ok |
| 03:37.23 | joako | wtsexton00: Try setting a password for each account |
| 03:39.16 | Hanif08 | hi asterisk friends.... |
| 03:39.32 | [TK]D-Fender | joako: He doesn''t have access... |
| 03:39.48 | wtsexton00 | yea I'm at home, and thats at work |
| 03:39.59 | Hanif08 | anyone can lead me to enable my asterisk rx incoming calls from my cisco gw |
| 03:40.22 | Hanif08 | i always get [Jan 13 22:58:06] NOTICE[22507]: chan_sip.c:13952 handle_request_invite: Call from '' to extension '86575080' rejected because extension not found. |
| 03:40.41 | wtsexton00 | searching about it has been fruitless for me |
| 03:41.54 | Hanif08 | i created alredy exten 86575080.. but my phone no ring |
| 03:42.05 | [TK]D-Fender | wtsexton00: Searching is worthless when you aren't looking at SIP debug. Now THAT is ffruitless. |
| 03:42.27 | [TK]D-Fender | Hanif08: Go look at the SIP DEBUG of YOUR call and see whats really going on. |
| 03:42.36 | [TK]D-Fender | Hanif08: PASTEBIN is your friend. |
| 03:42.36 | wtsexton00 | looked at debug all day, polycom was sending the address for line 2 instead of line 1 |
| 03:42.37 | [TK]D-Fender | ~pb |
| 03:42.38 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste, or , http://bin.cakephp.org/ |
| 03:42.40 | [TK]D-Fender | ^^^^^^^666 |
| 03:42.49 | [TK]D-Fender | lol, get shift-slip |
| 03:42.53 | [TK]D-Fender | great* |
| 03:42.57 | Hanif08 | ops.. sorry jbot. |
| 03:43.01 | Hanif08 | noted on that |
| 03:47.10 | *** join/#asterisk dsp2877 (i=dsp28777@28.227.48.60.cbj02-home.tm.net.my) |
| 03:47.16 | dsp2877 | hi all |
| 03:47.48 | dsp2877 | am trying to install asterisk 1.4.22.1 with a digium card, can i still use the zaptel drivers or i need to use this DAHDI thing |
| 03:49.02 | wtsexton00 | you can still use them |
| 03:50.01 | dsp2877 | okay thanks |
| 03:50.19 | wtsexton00 | to refer to them as zap, you'll need to add 'dahdichanname = no' under [options] in asterisk.conf |
| 03:50.21 | Deeewayne | dsp2877, if you are starting from scratch you might as well use dahdi |
| 03:51.03 | NovceGuru | woooh....color polycom! |
| 03:51.19 | *** join/#asterisk MindTheGap_ (n=MindTheG@189.59.202.234) |
| 03:51.43 | wtsexton00 | 670? |
| 03:51.44 | dsp2877 | deeewayne, i need them not only for the digium card, but also for meetme, so im not sure how meetme would work as it requires timing or something from the zttimer/ztdummy |
| 03:51.48 | NovceGuru | wtsexton00: yeah |
| 03:51.52 | wtsexton00 | gigabit too |
| 03:52.10 | NovceGuru | yeah, my PoE switch is 100mb :( |
| 03:52.24 | NovceGuru | I ran seperate copper for the pc though |
| 03:52.35 | wtsexton00 | I've got a network running the 560s |
| 03:52.45 | wtsexton00 | the non-color version |
| 03:52.54 | Deeewayne | dsp2877, you aren't going to have problems with meetme by going to dahdi |
| 03:53.01 | NovceGuru | yeah, the default background is fugly |
| 03:53.32 | wtsexton00 | I hate the backgrounds |
| 03:54.05 | NovceGuru | I would like to put their company logo but I'm using switchvox to provision the phones and I don't feel like hacking closed stuff :| |
| 03:54.35 | dsp2877 | deeewyne : ok let me give it a go and see |
| 03:54.43 | NovceGuru | only using it because anytime the user adds a contact it recreates the config and reboots the phone automagically, I suppose I could script something to pull configs every 5 minutes...meh |
| 03:54.53 | wtsexton00 | hmm on my dev asterisk box I also have the same issue with line 2 showing when line 1 is used |
| 03:54.58 | dsp2877 | thanks |
| 03:55.17 | Deeewayne | dsp2877, going with dahdi will make it easier in the future when you want to move to a new version |
| 03:55.28 | wtsexton00 | really, I just flap power normally |
| 03:56.13 | wtsexton00 | I'm struggling to get multiple lines working correctly. |
| 03:56.15 | dsp2877 | ok thanks man, noted that |
| 03:56.24 | *** join/#asterisk Winkie (n=urmom@ur.fa.gs) |
| 04:00.07 | wtsexton00 | hmm |
| 04:01.07 | [TK]D-Fender | NovceGuru: PWNED |
| 04:01.23 | wtsexton00 | Executing [286@incoming-from-phones:1] GotoIf("SIP/211a-b02dd970", "3?10:20") in new stack |
| 04:01.28 | NovceGuru | [TK]D-Fender: whats that? heh |
| 04:01.45 | wtsexton00 | 211a, is the second line |
| 04:02.44 | [TK]D-Fender | wtsexton00: Seriously. Ther is no point until you've got the SIP DEBUG and configs to back it. |
| 04:02.57 | [TK]D-Fender | NovceGuru: SwitchVox FUBAR-ig your phones. |
| 04:03.05 | [TK]D-Fender | NovceGuru: Provision them yourself. |
| 04:03.16 | wtsexton00 | well, that blows |
| 04:03.59 | NovceGuru | yeah I need to, their provisioning system seems lacking. But I was saying I did like how it lets the end user update their contact list and reboot their phone view the web ui, without bugging me :) |
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| 04:05.11 | [TK]D-Fender | NovceGuru: Want to update your contact list... do it ont he phone |
| 04:05.43 | NovceGuru | well it can sync their contact lists with outlook etc |
| 04:05.58 | NovceGuru | much easier then pecking a name into a 9 digit pad :| |
| 04:06.03 | [TK]D-Fender | NovceGuru: How desperate is that! WOOHOO! |
| 04:06.06 | *** join/#asterisk jks (n=jks@193.189.93.254) |
| 04:06.16 | [TK]D-Fender | NovceGuru: 1 shot deal. |
| 04:06.23 | [TK]D-Fender | NovceGuru: What a bunch of whiners |
| 04:06.24 | *** join/#asterisk beek (n=klinebl@65.211.106.242) |
| 04:06.24 | [TK]D-Fender | :p |
| 04:06.27 | NovceGuru | I know :P |
| 04:06.36 | NovceGuru | they are so in love with their outlook :\ |
| 04:06.41 | *** join/#asterisk mog (n=mog@c-68-62-217-121.hsd1.al.comcast.net) |
| 04:06.41 | *** mode/#asterisk [+o mog] by ChanServ |
| 04:06.45 | NovceGuru | it is slick, i'll give switchvox that |
| 04:07.29 | [TK]D-Fender | NovceGuru: Yeah, you can hardly tell you're getting the whole shaft... until its too late ;) |
| 04:08.02 | *** join/#asterisk dec3pti0n (n=dec3pti0@pdpc/supporter/student/dec3pti0n) |
| 04:08.13 | NovceGuru | they just tease you with the tip |
| 04:08.14 | NovceGuru | sICK! |
| 04:09.10 | NovceGuru | well it seems ok, the switchboard was a selling point for it |
| 04:09.29 | NovceGuru | trixbox has their HUD, which i'd like to see in other places besides fuxbox :| |
| 04:11.36 | dec3pti0n | ok it's been a while since I messed with asterisk and I'm wondering if the latest version still requires me to have that zdummy module in case we don't have a digium card that does the timing |
| 04:12.01 | *** join/#asterisk clyrrad (n=darryl@99.250.101.208) |
| 04:12.42 | *** join/#asterisk SkramX (i=mark@phalse.2600.COM) |
| 04:12.58 | [TK]D-Fender | dec3pti0n: * doesw not require it. Certain apps require it however, and IAX2 trunking |
| 04:13.00 | jaytee | dec3pti0n, nope, no more ztdummy......you need dahdi_dummy now |
| 04:13.38 | dec3pti0n | ok cool, I'm planning on trying out IAX2 and SIP trunking , always prefered IAX2 though |
| 04:14.36 | wtsexton00 | not getting anywhere tonight |
| 04:14.39 | dec3pti0n | do you guys know/recommend any VOIP providers out there that I could test an * voip deployment ? hopefully one that provides IAX2 trunking ? |
| 04:14.54 | [TK]D-Fender | dec3pti0n: And why does IAX2 matter? |
| 04:14.59 | dec3pti0n | I have heard of bandwidth.com but .. |
| 04:15.17 | [TK]D-Fender | dec3pti0n: and whats to "test"? |
| 04:16.27 | dec3pti0n | [TK]D-Fender, the whole NATing issue, going through FWs unless that has also changed of course + plus on IAX2 it used to be true that it will use less bandwidth |
| 04:16.51 | dec3pti0n | [TK]D-Fender, connection and testing quality that the VOIP provider will provide |
| 04:17.11 | [TK]D-Fender | dec3pti0n: it will use less if you are trunking multiple calls at a time. As for NAT it is very rare that its a problem for anyone |
| 04:17.39 | dec3pti0n | cool thanks |
| 04:17.49 | [TK]D-Fender | dec3pti0n: IAX tends to come out with more jitter, etc and overall lower audio quality for some reason. |
| 04:19.25 | dec3pti0n | hmm, really ? when one uses IAX trunking ? good to know .... the old system that I had setup in 2003 was all IAX2 but I did no IAX2 trunking though |
| 04:19.51 | dec3pti0n | I just got my slicehost and I will try to install * on it now :) |
| 04:20.58 | hardwire | [TK]D-Fender: doesn't it tend to have more jitter when multiple non-trunked iax2 calls go to between the same sources? I thought some routers delayed packets to see if they could buddy them up at all. |
| 04:20.58 | wtsexton00 | hmm looks like someone else had this issue changing from a peer to a friend with a username resolved it |
| 04:22.41 | dec3pti0n | hehe just noticed the asterisk.org uses bandwidth.com :) so I guess that would be a good VOIP provider ! |
| 04:22.43 | [TK]D-Fender | hardwire: Not sure on the finer points, just hearing plenty of first hand feed-back that switching even with the same provider showed a significant difference |
| 04:23.34 | hardwire | that's kind of a bummer |
| 04:23.53 | hardwire | I wonder how different the modules themselves are in efficiency between sip and iax. |
| 04:24.14 | *** part/#asterisk AndyML (n=quassel@pool-72-78-117-135.phlapa.fios.verizon.net) |
| 04:25.46 | *** join/#asterisk CunningPike (n=arodgers@S01060014bf81366b.vc.shawcable.net) |
| 04:29.21 | *** part/#asterisk wtsexton00 (n=tim@75.151.195.117) |
| 04:31.13 | *** join/#asterisk jameswf (n=james@unaffiliated/jameswf-home) |
| 04:31.52 | *** join/#asterisk momo1431 (n=mattm@h-66-134-89-2.snvacaid.covad.net) |
| 04:32.12 | jameswf | so i got an email from microsoft to download windows 7 beta... do you think http://xkcd.com/528/ is true?? |
| 04:33.06 | dsp2877 | lol |
| 04:33.19 | dsp2877 | probably fake |
| 04:35.03 | dec3pti0n | I always go with the rule-of-thumb "don't believe anything microsoft says" :) |
| 04:36.17 | *** join/#asterisk Carlos_PHX (n=Carlos@ip68-3-162-244.ph.ph.cox.net) |
| 04:36.56 | dandate2 | i'm using godaddy to host my webserver and domain www.newlineequity.com , could I just use apache on my asterisk box to do this as well? |
| 04:38.50 | [TK]D-Fender | dandate2: Try asking in#doesthislooklikeawebhostingchanneltoyou |
| 04:40.40 | *** join/#asterisk dmz (n=dmz@64.203.203.232.dyn-cm-pool-64.hargray.net) [NETSPLIT VICTIM] |
| 04:44.47 | NovceGuru | now how the hell is switchvox provisioning the polycoms without any settings in the provision server settings...grr |
| 04:44.48 | *** join/#asterisk `Sauron (n=sauron@dsl001-130-033.aus1.dsl.speakeasy.net) |
| 04:45.20 | NovceGuru | probably via some known method in the manual. but still! |
| 04:45.22 | SlicerDicer | [TK]D-Fender: I just had to give you hell man :) |
| 04:45.26 | SlicerDicer | I could not remember the model |
| 04:45.39 | *** join/#asterisk xuser (n=xuser@unaffiliated/xuser) |
| 04:45.45 | Qwell | jameswf: I hear it's quite pleasant and hardly Hitler-y at all. |
| 04:46.07 | NovceGuru | jameswf: I hear it's vista plus a little different gui and less nagging |
| 04:46.54 | jameswf | 2.44gig download kinda a turn off, maybe a method of anti-piracy |
| 04:46.55 | [TK]D-Fender | NovceGuru: Could be direct HTTP <- |
| 04:47.05 | [TK]D-Fender | NovceGuru: no files, just live-generated on wget |
| 04:47.14 | [TK]D-Fender | NovceGuru: I'd believe it.. |
| 04:47.33 | NovceGuru | [TK]D-Fender: I don't think there's a way to add a contact? I don't think there's an API on the phones httpd ? |
| 04:47.53 | jameswf | here goes we will bloat our downloads to the point no one will use the bandwith |
| 04:48.31 | NovceGuru | [TK]D-Fender: seems to be quite a bit more configuring being done then what you can do in the web config |
| 04:48.57 | [TK]D-Fender | NovceGuru: did I say "web config"? NO. |
| 04:49.01 | [TK]D-Fender | NovceGuru: provisioning <- |
| 04:49.19 | [TK]D-Fender | NovceGuru: FTp is jsut a stupid protocol to pick up a file. same with HTTP |
| 04:49.43 | jameswf | craig newmark is a little too obama fanboy for my taste..... |
| 04:49.44 | NovceGuru | oh I see, when the phones access http://blah.com/mac.cfg it generates it ont he fly |
| 04:50.04 | NovceGuru | is that what you are saying? |
| 04:50.47 | [TK]D-Fender | thinks he might see a dim light flickering... |
| 04:50.51 | Qwell | jameswf: 2.44 gb as an anti-piracy measure? |
| 04:50.59 | NovceGuru | I understand that, but I don't understand how the provision server on the polycom is set to ftp with address 0.0.0.0 and it still picks up it's config |
| 04:51.02 | Qwell | jameswf: I don't know about you, but the last time I saw a game download... |
| 04:51.37 | NovceGuru | and there isn't an ftpd running on the switchvox |
| 04:51.47 | momo1431 | Question - Has anyone got imap storage to work on Asterisk? Where can i go to look at diagnostics messages? There is nothing showing up in verbose mode, and i can't find any docs. I don't even know where to start! |
| 04:52.00 | momo1431 | Any suggestions on where to start? |
| 04:52.02 | NovceGuru | momo1431: I was there a few days ago and gave up! |
| 04:52.36 | ricko73 | google.com |
| 04:52.38 | NovceGuru | I couldn't get anything in the console or in /var/log/asterisk |
| 04:52.50 | [TK]D-Fender | NovceGuru: And you clearly haven't read the admin guide anything close to thoroughly |
| 04:53.17 | NovceGuru | this is true, just going off what I see |
| 04:53.20 | NovceGuru | reads |
| 04:53.42 | ricko73 | is that a statement affirming your ability or a statement of action? |
| 04:53.50 | NovceGuru | just didn't seem logical to me that its getting configs from http when it's set to ftp |
| 04:54.34 | ricko73 | NovceGuru: it shouldn't be |
| 04:55.28 | NovceGuru | unless theres a firewall on it doing some mac based filtering..there isn't an ftp server running on it |
| 04:55.48 | momo1431 | Re: polycoms - how are they getting the IP addresses? |
| 04:55.51 | ricko73 | it could be running with xinet.d |
| 04:56.09 | NovceGuru | via a dhcp server without any option 66 enabled |
| 04:56.35 | momo1431 | ohh. |
| 04:56.41 | NovceGuru | I'm pretty sure ARP is being used to look for polycom mac addresses |
| 04:58.10 | momo1431 | I don't know anything about Switchvox, but my polycoms either need a provisioning server or are setup via the phone or through the web. |
| 04:58.33 | momo1431 | is this in a working network? |
| 04:59.01 | NovceGuru | it's on a test network but yes it's working ;) |
| 04:59.07 | NovceGuru | - wink + :) |
| 04:59.48 | NovceGuru | SV said they used http provisioning |
| 05:00.12 | NovceGuru | but the "sever settings" in the phone say they're set to ftp at 0.0.0.0 BUT have the login credentials that SV created |
| 05:00.28 | momo1431 | and did you do any configuration on the polycoms under Settings - Advanced - Admin Settings - ...? |
| 05:00.48 | NovceGuru | nope just plugged it in from factory defaults |
| 05:01.31 | NovceGuru | out of the box and onto the network where it recieved an IP address from an unrelated (nothing configured for option-66 aka vanilla) dhcp server |
| 05:02.06 | *** join/#asterisk `Sauron (n=sauron@dsl001-130-033.aus1.dsl.speakeasy.net) [NETSPLIT VICTIM] |
| 05:02.13 | NovceGuru | SV picked it up (assume through arp) and pushed settings to it (somehow) and rebooted it all nicely configured |
| 05:02.22 | momo1431 | Looking at my phone - the server address is also 0.0.0.0, but i know it's set to something because i have option 66 enabled. |
| 05:03.06 | momo1431 | that must only be set if you have manually set a server. |
| 05:03.07 | NovceGuru | is the server type correct? |
| 05:03.18 | NovceGuru | yeah, must be the case |
| 05:03.29 | momo1431 | is the user set to PlcmSpip? That's what my user is set to and that is incorrect. |
| 05:04.11 | *** join/#asterisk animositisomina (n=chatzill@201-75-60-134-ma.cpe.vivax.com.br) |
| 05:04.36 | NovceGuru | the user and pass seem to be set to what the SV set for it |
| 05:05.11 | NovceGuru | no go here, menu -> status -> platform -> config |
| 05:05.30 | NovceGuru | I have boot server: PC |
| 05:05.44 | NovceGuru | boot server type: ftp |
| 05:06.03 | NovceGuru | config: phone<mac>.cfg, sip<mac>.cfg |
| 05:06.24 | NovceGuru | but port 21 is not open on the SV! |
| 05:06.38 | NovceGuru | and the dhcp server is not giving it option-66...grr! |
| 05:07.01 | momo1431 | Well - if i could get everything working without configuring it perhaps i should switch! It might help me get started with IMAP voicemail storage! |
| 05:07.29 | NovceGuru | :) |
| 05:07.50 | NovceGuru | I tried setting it up on a whim and couldn't get any diags to dump and I haven't been back to it yet |
| 05:09.15 | NovceGuru | well it's obvious it has to be doing http provisioning...thats what they said and also the only port open on the box |
| 05:10.30 | NovceGuru | im gonna have to figure it out because im pretty sure the pretty picture of a flower on the background isn't going to fly! |
| 05:12.50 | momo1431 | Sorry i can't help you more. I'm sure it's documented somewhere, but i'm not using that. |
| 05:14.55 | NovceGuru | it's np i'm just ranting |
| 05:15.01 | NovceGuru | hate when I can't figure something out |
| 05:17.38 | *** join/#asterisk SkywaIker (n=pirch@125.25.236.255.adsl.dynamic.totbb.net) |
| 05:21.40 | *** join/#asterisk sah-work (n=Bawbatos@adsl-76-203-174-215.dsl.pltn13.sbcglobal.net) |
| 05:22.31 | joako | Is there opposite of Dial() option M? |
| 05:23.13 | *** join/#asterisk Winkie (n=urmom@ur.fa.gs) |
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| 06:08.13 | [TK]D-Fender | checkout time. Later all |
| 06:09.10 | dec3pti0n | I need to compile the DAHDI and libpri before * right ? |
| 06:16.23 | dsp2877 | yeap |
| 06:21.56 | NovceGuru | app_meetme.c:1620 conf_run: Unable to set flags: Inappropriate ioctl for device |
| 06:21.59 | NovceGuru | wooh! |
| 06:24.25 | *** join/#asterisk frogonwheels (n=michaelg@203.59.141.93) |
| 06:26.19 | frogonwheels | I'm having partial success with sending SMS from asterisk (with smsq) to a local SMS enabled uniden phone that's attached to a PAP2T... |
| 06:27.06 | frogonwheels | it all seems to work - sms is received - except that the Handset doesn't stop ringing (thouh the PAP2T reports that the line has been hung up) |
| 06:28.38 | frogonwheels | anybody got any tips on where to look - I'm drawing blanks on my google searches. I've shot an email off to uniden support - but not sure at all what that will give me |
| 06:30.12 | frogonwheels | I'm (now) pretty sure that it's to do with PAP2T interacting wtih the phone. and nothing to do with asterisk. |
| 06:37.01 | *** part/#asterisk animositisomina (n=chatzill@201-75-60-134-ma.cpe.vivax.com.br) |
| 06:38.04 | dsp2877 | am trying to load chan_alsa for asterisk 1.4.22, but keep getting this : |
| 06:38.12 | dsp2877 | reror: /usr/lib/asterisk/modules/chan_alsa.so: undefined symbol: snd_pcm_open |
| 06:38.29 | dsp2877 | any reason for that |
| 06:38.37 | dsp2877 | make/make install all compiled ok |
| 06:40.38 | dandate2 | does anyone know if 1 box will be enough to handle asterisk, apache, and postfix ? |
| 06:40.54 | dec3pti0n | has anyone here tried to install * on a XEN guest before ? |
| 06:41.00 | carrar | http://tinyurl.com/9656om |
| 06:41.20 | frogonwheels | dandate2: I have a router running asterisk, a web server and Xmail. it all depends on how much you are \doing! |
| 06:41.23 | carrar | dandate2, depends on your call volumn and box size obviousy |
| 06:41.47 | carrar | dec3pti0n, asterisk works great in xen |
| 06:41.59 | carrar | again it depends on the load of the host |
| 06:42.51 | dec3pti0n | I would assume so I'm just having issues with dahdi-linux not compiling since it can't find the proper linux source |
| 06:42.57 | dec3pti0n | kernel source |
| 06:43.07 | carrar | dec3pti0n, http://tinyurl.com/79vfll |
| 06:43.31 | dec3pti0n | hehe thank you sir |
| 06:49.21 | dan__t | Is there any way to like "kick" a peer, in order to get my iax2 trunk to authenticate and register? |
| 07:02.26 | *** join/#asterisk misc-- (n=misc@202-154-80-42.people.net.au) |
| 07:02.37 | misc-- | hello... is dns needed for asterisk to register sip devices? |
| 07:06.23 | ricko73 | ~weather KSBM |
| 07:06.35 | ricko73 | brrr |
| 07:06.50 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 07:06.51 | ricko73 | misc--: are you using IP addresses for the sip devices? |
| 07:07.31 | misc-- | ricko7: no just the IP address |
| 07:08.04 | misc-- | however when running a tcpdump on the asterisk box, I can see the phone sending a request out on port 5060 but the server simply doesn't reply. The server can ping the phone, etc... |
| 07:08.35 | ricko73 | is there a firewall between the two devices? |
| 07:12.10 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
| 07:13.34 | *** join/#asterisk obnauticus (n=obnautic@about/windows/regular/obnauticus) |
| 07:15.24 | misc-- | ricko73: no - strange... the dns server for the asterisk box was set to a file server which had no internet access. When the file server was turned off, phones started to register again |
| 07:19.03 | *** join/#asterisk sah-work (n=Bawbatos@adsl-76-203-174-215.dsl.pltn13.sbcglobal.net) |
| 07:19.04 | *** join/#asterisk phix (n=threat@123-243-44-131.tpgi.com.au) |
| 07:20.46 | dan__t | Hrm. |
| 07:20.52 | dan__t | Why would I not have any 'sip' menus in the console? |
| 07:21.41 | *** join/#asterisk Thorn (n=thorn@unaffiliated/thorn) |
| 07:22.49 | dan__t | What did I do this time. |
| 07:22.50 | dan__t | *sigh* |
| 07:25.18 | dan__t | heh. |
| 07:28.07 | dan__t | Yeah, I don't have any sip or iax2 module names in here.... guess that means modules were not loaded eh? |
| 07:29.18 | *** join/#asterisk botox93 (n=botox93@213.221.82.242) |
| 07:30.08 | *** join/#asterisk SkywaIker (n=pirch@125.25.248.225.adsl.dynamic.totbb.net) |
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| 07:49.49 | dan__t | Well. Crap. |
| 07:50.03 | dec3pti0n | does DAHDI Linux 2.1.0.3 require glibc 2.4 ? |
| 07:54.52 | dandate2 | i'm configuring apache to host my webserver, won't port 80 tie with my freepbx gui and cause problems? |
| 07:57.30 | denon | ~freepbx |
| 07:57.31 | jbot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 08:01.10 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
| 08:05.01 | carrar | dandate2, why do you keep asking freepbx questions in here? |
| 08:05.06 | carrar | You never stop |
| 08:05.21 | *** join/#asterisk gamercod4 (n=daffyduc@ABayonne-257-1-29-68.w90-55.abo.wanadoo.fr) |
| 08:05.31 | gamercod4 | Hi all |
| 08:06.02 | gamercod4 | i don't know if i'm in the good channel but i've a question about Vtiger and Asterisk integration |
| 08:06.30 | carrar | try |
| 08:07.10 | gamercod4 | the click and dial is OK in vtiger, but no have popup in vtiger 5.1 for know the calling information |
| 08:07.35 | gamercod4 | and i don't know if the configuration is in asterisk (send an url or other) |
| 08:09.53 | drmessano | How do you have Asterisk + Vtiger working? |
| 08:10.34 | gamercod4 | i've install the vtiger 5.1 version and load the PBXManager module |
| 08:10.52 | gamercod4 | and run the php file for communicate with asterisk |
| 08:11.01 | drmessano | AHHH I see now |
| 08:11.50 | *** join/#asterisk grEvenX (n=even@apb9hb.ip.ssc.net) |
| 08:12.10 | gamercod4 | but normally, when i received a call, i've a popup within vtiger session |
| 08:12.40 | gamercod4 | and i haven't this function :/ |
| 08:17.13 | *** join/#asterisk Gopher_77 (n=Jim@cpe-71-72-19-206.neo.res.rr.com) |
| 08:17.39 | Gopher_77 | can someone recommend a good metered sip provider for low volume? |
| 08:19.57 | carrar | Gopher: http://tinyurl.com/97d6s9 |
| 08:20.41 | Gopher_77 | :P |
| 08:20.41 | Gopher_77 | that doesn't give recommendations though |
| 08:20.47 | carrar | this isn't a sales channel |
| 08:21.03 | Gopher_77 | I know |
| 08:21.25 | drmessano | ~itsp |
| 08:21.25 | jbot | [~itsp] An ITSP is an Internet telephony Service Provider or "VoIP Telephone company". The allow you to either PLACE calls to the PSTN (called Termination), RECEIVE calls from the PSTN (called Origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs |
| 08:22.20 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
| 08:23.25 | *** join/#asterisk `Sauron (n=sauron@dsl001-130-033.aus1.dsl.speakeasy.net) [NETSPLIT VICTIM] |
| 08:23.25 | Gopher_77 | ~itsplist-us |
| 08:23.26 | jbot | [~itsplist-us] Here are some popular ITSPs (USA) starting with the more respected ones : http://www.teliax.com , http://connect.voicepulse.com , http://www.nufone.net , http://www.broadvoice.com, http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net |
| 08:23.48 | dan__t | Alright, I'm not sure what's up here. When I fire up Asterisk, then head to a console... I'll do a 'reload', then a 'help', then 'help' will just go to a newline. |
| 08:24.23 | dan__t | I think thre's an issue with my modules being loaded. Further proof is: [Jan 14 00:22:55] NOTICE[23290]: loader.c:869 load_modules: 1 modules will be loaded, [Jan 14 00:22:55] NOTICE[23290]: loader.c:869 load_modules: 157 modules will be loaded. |
| 08:24.37 | dan__t | Even further, I don't see a 'sip' or 'iax2' context menu in the console. |
| 08:25.17 | drmessano | Sounds like modules.conf is hosed |
| 08:26.41 | dan__t | That's what I figured, but I'm trying to figure out how it got that way. |
| 08:26.57 | dan__t | Naturally, I expect Fedora 9 RPMs to blow up when compiled under EL5. |
| 08:30.07 | dan__t | 'cause I haven't found a solid set made natively for EL4/5 |
| 08:31.26 | *** join/#asterisk HeMan (n=jimmy@ssh.southpole.se) |
| 08:31.46 | *** join/#asterisk andy-x_l (i=425c01d1@gateway/web/ajax/mibbit.com/x-b484766cf7ca516b) |
| 08:33.36 | dan__t | Well. This modules.conf is identical to another, which is working fine. |
| 08:34.03 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 08:34.13 | dan__t | http://pastebin.com/m5f937412 , if you wouldn't mind taking a peek. |
| 08:44.26 | dan__t | What's wrong with me? I'm rocking out to Public Enemy. |
| 08:46.17 | ScriptFanix | listening using MoH ? :p |
| 08:47.45 | dan__t | heh. not yet. |
| 08:47.54 | dan__t | I just might, though, to celebrate this victory. |
| 08:48.11 | dan__t | I suspect my menu options were fubar'd to some degree |
| 08:48.16 | dan__t | thus, no modules are enabled by default. |
| 08:48.41 | frogonwheels | I tried this qq earlier in the day: I'm having partial success with sending SMS from asterisk (with smsq) to a local SMS enabled uniden phone that's attached to a PAP2T... |
| 08:48.53 | frogonwheels | it all seems to work - sms is received - except that the Handset doesn't stop ringing (thouh the PAP2T reports that the line has been hung up) |
| 08:49.05 | frogonwheels | anybody got any tips on where to look - I'm drawing blanks on my google searches. I've shot an email off to uniden support - but not sure at all what that will give me |
| 08:49.36 | dan__t | Timeouts? |
| 08:50.10 | frogonwheels | dan__t: on where? as far as asterisk is concerned - and the PAP2t - it's all hung up.. |
| 08:50.33 | dan__t | Got it. |
| 08:50.46 | dan__t | Not famiiar, sorry, just throwing stuff out there. |
| 08:51.13 | frogonwheels | dan__t: Nah that's ok. I'm kinda dried up for ideas. |
| 08:51.41 | frogonwheels | I'm thinking something on the PAP2T to do with it doing things in a different order.. |
| 08:52.22 | frogonwheels | dan__t: I tried timeouts on a bunch of places b4 realising it wasn't the problem :( |
| 08:55.26 | dan__t | I'm not familiar with PAP2T, what is it? |
| 08:55.30 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 08:55.47 | frogonwheels | dan__t: it's a linksys 2 port ATA |
| 08:55.47 | *** part/#asterisk Hanif08 (n=bucoo77@netop.jaring.my) |
| 08:56.05 | frogonwheels | dan__t: it's a rebadged SIPURA <somenumber> |
| 08:58.09 | dan__t | Got it. |
| 08:59.18 | dan__t | So, in 'make menuselect', what does a XXX mean when an item is selected? |
| 08:59.48 | frogonwheels | shrugs. |
| 08:59.59 | frogonwheels | I build asterisk in the OpenWRT build system. :) |
| 09:02.46 | dan__t | heh |
| 09:02.50 | dan__t | which version? |
| 09:02.56 | *** join/#asterisk grndslm (n=grndslm@24-119-80-142.cpe.cableone.net) |
| 09:02.56 | tzafrir_laptop | frogonwheels, what version? any patches you needed to apply? |
| 09:03.15 | frogonwheels | tzafrir_laptop: for the OpenWRT you mean? |
| 09:03.54 | frogonwheels | tzafrir_laptop: OpenWRT has a great system where it allows a bunch of ordered patches that get applied after the tarball is extracted... |
| 09:04.22 | frogonwheels | tzafrir_laptop: it's in the latest Kamikaze. |
| 09:04.41 | tzafrir_laptop | frogonwheels, every packaging system is basically like that |
| 09:04.55 | frogonwheels | tzafrir_laptop: Yeah possibly true ;P |
| 09:04.56 | tzafrir_laptop | but I was asking about the version of Asterisk |
| 09:05.07 | frogonwheels | tzafrir_laptop: lol. . 1.4. ermm... |
| 09:06.11 | frogonwheels | 1.4.22 |
| 09:06.42 | *** join/#asterisk kippi (n=chriso@83-244-164-130.cust-83.exponential-e.net) |
| 09:06.46 | kippi | Morning |
| 09:07.28 | kippi | In asterisks I have two queues, is there a command in asterisk that will pick at random which queue the call will go to |
| 09:07.47 | *** part/#asterisk gamercod4 (n=daffyduc@ABayonne-257-1-29-68.w90-55.abo.wanadoo.fr) |
| 09:11.37 | dan__t | There is a RAND() dialplan function |
| 09:11.40 | dan__t | I think you can be creative with that. |
| 09:11.51 | dan__t | RAND([min], [max]) |
| 09:12.13 | dan__t | RAND([min][|max]) |
| 09:12.54 | *** join/#asterisk styelz (n=yoohoo@m0o0.mooo.com) |
| 09:17.38 | kippi | dan__t: Just looking at that now, just wondered if there was anything built into Queue that could do it |
| 09:21.04 | dan__t | Not that I've read, no. |
| 09:23.05 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 09:25.00 | dan__t | checking for speex_preprocess_ctl in -lspeex... no |
| 09:25.02 | dan__t | why not?! |
| 09:33.10 | *** join/#asterisk h-idrisi (n=h-idrisi@212.100.196.195) |
| 09:33.19 | dsp2877 | will dahdi work on asterisk 1.4.22 version and below? |
| 09:33.37 | *** join/#asterisk phix (n=threat@123-243-44-131.tpgi.com.au) |
| 09:33.52 | dsp2877 | or is it only for the latest versions of asterisk or something |
| 09:37.40 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 09:45.21 | *** join/#asterisk Avelino (n=Avelino@mail.paterno.com.br) |
| 09:45.43 | kippi | can anyone see why small dial plan isn't working? http://pastebin.com/mc4760a it always seems to go to 1 |
| 09:50.18 | dan__t | Alright. Well. Someone mind pastebin'ing a stock modules.conf please? I messed up somewhere. |
| 09:51.46 | dan__t | kippi, shouldn't GotoIf($["${RAND}" != "1"]?LOL:LOL1) become GotoIf($["${RND}" != "1"]?LOL:LOL1) ? |
| 09:51.54 | dan__t | RND opposed to RAND |
| 09:52.00 | *** join/#asterisk `Sauron (n=sauron@dsl001-130-033.aus1.dsl.speakeasy.net) |
| 09:53.55 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
| 09:54.20 | kippi | ah |
| 09:55.45 | dsp2877 | hi guys |
| 09:55.49 | dsp2877 | just wondering if anyone can help |
| 09:55.57 | dsp2877 | i installed dahdi with asterisk 1.4.22 |
| 09:56.07 | dsp2877 | and i need to roll back to 1.4.21 |
| 09:56.12 | kippi | dan__t: thanks, working like a dream |
| 09:56.15 | dan__t | Good. |
| 09:56.19 | dsp2877 | do i need to uninstall dahdi and install zaptel instead? |
| 09:56.42 | dan__t | Suppose I can get a view at your modules.conf, kippi, if you haven't hacked it up so much yet? |
| 10:00.42 | kippi | its been changed a load |
| 10:03.52 | dan__t | Oh, ok. |
| 10:05.47 | dan__t | I don't get it. |
| 10:05.57 | dan__t | I'm comparing one install to another - identical. One loads modules, the other does not. |
| 10:09.11 | *** join/#asterisk axisys (n=axisys@ip68-98-177-71.dc.dc.cox.net) |
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| 10:15.09 | *** part/#asterisk zydoon (n=zydoon@41.225.177.131) |
| 10:18.23 | dan__t | I'm mad. |
| 10:18.29 | dan__t | How come * isn't seeing any of my modules. |
| 10:20.10 | frogonwheels | dan__t: can you get at a console? |
| 10:20.26 | dan__t | I can. |
| 10:20.34 | dan__t | It loads... what does it say, basic module or something |
| 10:20.59 | *** join/#asterisk nexu (n=nexu@a82-95-43-129.adsl.xs4all.nl) |
| 10:21.44 | frogonwheels | and if you show modules - how many loaded? |
| 10:21.54 | dan__t | http://pastebin.com/m46f44eb5 |
| 10:22.00 | dan__t | Quite a few but it looks like they're all core modules. |
| 10:22.24 | dan__t | What's interesting is that after a 'reload' from console, it appears that the core module itself goes away. |
| 10:22.57 | frogonwheels | dan__t: I've had problems with reload - but I think that's a uclibc bug. |
| 10:23.15 | frogonwheels | dan__t: so once it's restarted, what does list modules give you? |
| 10:23.41 | frogonwheels | ermm show modules |
| 10:24.22 | frogonwheels | .. then try module load one that isn't and see what happens maybe? |
| 10:24.28 | frogonwheels | gg. kids and all. |
| 10:24.39 | dan__t | hrm, how do i load one from console? sorry |
| 10:25.32 | dan__t | No such command 'show modules' (type 'help show modules' for other possible commands) |
| 10:25.33 | dan__t | very nice. |
| 10:25.57 | dan__t | hehe, because its 'module show' |
| 10:26.49 | dan__t | Ok... loading a module by hand appears to blow it up, too. |
| 10:29.33 | dan__t | I need to just do this the right way, and not use this hack of an RPM. |
| 10:29.47 | dan__t | I'll consider rolling my own for RHEL here soon. When I do, I'll go ahead and release it. |
| 10:32.18 | *** join/#asterisk CrazyTux (n=brandon@ip68-111-67-4.oc.oc.cox.net) |
| 10:45.42 | *** join/#asterisk raasdnil (n=mikel@60-241-138-146.static.tpgi.com.au) |
| 10:46.30 | raasdnil | hey all. Got an asterisk box with several IAX trunks to different providers. Currently, when calling our number on one of our providers, we are getting congestion. .. |
| 10:46.45 | raasdnil | I have about 40 active calls going... so the console is just insane trying to watch it right now |
| 10:47.07 | raasdnil | How can I whittle down the console log messages to say "only incomming from iax provider blah" ? |
| 10:47.11 | raasdnil | or is that not possible? |
| 10:48.35 | raasdnil | Asterisk 1.6.0 |
| 10:51.49 | *** join/#asterisk qdk_ (n=qdk@94.191.233.118.bredband.3.dk) |
| 10:56.28 | dan__t | Haha... even installing from source, even forcing a module to load with load =>, the module still doesn't show up. |
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| 11:06.38 | frogonwheels | :| |
| 11:06.45 | *** join/#asterisk mikkel (n=mikkel@130.226.36.170) |
| 11:07.34 | frogonwheels | raasdnil: I think you'll find that (I'm talking 1.4 here.. ) you can sip set debug peer <peername> |
| 11:09.24 | *** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim) |
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| 11:15.21 | raasdnil | frogonwheels: hmm.. doesn't look like that option exists in IAX... at least for 1.6 |
| 11:16.14 | frogonwheels | ah. iax. sorry |
| 11:16.42 | yang | Another Poortech went byebye |
| 11:19.09 | *** join/#asterisk strk (n=strk@ip-123-78.static.adsl.cheapnet.it) |
| 11:19.18 | strk | woo, full of people here, good sign :) |
| 11:19.31 | *** join/#asterisk magronez (n=eusei@unaffiliated/magrao/x-2903) |
| 11:19.45 | strk | can asterisc help me routing multiple sources onto a single SIP device ? |
| 11:20.24 | strk | ie: I've a linksys (hardware ATA) with only 2 lines input, but I'd like to have both lines (registrars) end up ringing the same single phone |
| 11:20.45 | strk | I don't think the linksys will allow this, and in any case I've more then 2 SIP addresses to deal with |
| 11:20.56 | *** join/#asterisk xnixan (n=xnixan@unaffiliated/xnixan) |
| 11:21.13 | xnixan | verywiseman, 2fashtak |
| 11:21.30 | strk | so I was thinking, can I configure asterisk to accept them all, and transfer to a single local SIP use against which I'd have the linksys register ? |
| 11:21.44 | verywiseman | xnixan, i also |
| 11:22.15 | xnixan | fain ya 7elo |
| 11:22.27 | verywiseman | xnixan, spa run , Thank of ALLAH |
| 11:22.55 | xnixan | mashy ya 3am el mazanger |
| 11:23.24 | verywiseman | xnixan, 7lash baqa , Rahet 3aleeek |
| 11:23.58 | xnixan | ya 7ag ro7 el channel bta3tak |
| 11:38.37 | *** join/#asterisk chendy (n=chendy@121.34.152.100) |
| 11:38.47 | *** join/#asterisk jgoo (n=jgoo@ppp204-192.adsl.forthnet.gr) |
| 11:39.04 | jgoo | what is the current consensus - cheapest *reliable* sip desktop handset |
| 11:39.30 | jgoo | grandstream / polycom / ??? approx. price in USD / EUR / GBP |
| 11:39.54 | jgoo | Also, for 30 extensions - what set offers the best look / feel ? |
| 11:40.03 | jgoo | that is, to monitor 30 extensions |
| 11:46.23 | verywiseman | there is linksys spa 1001 , it have 1 fxs 1 eth |
| 11:46.44 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 11:46.44 | verywiseman | or audiocodes mp-124 , have 24 fxs |
| 11:47.00 | *** part/#asterisk strk (n=strk@ip-123-78.static.adsl.cheapnet.it) |
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| 11:53.43 | jgoo | verywiseman, so, get 24 fxs and run analogue phones? |
| 11:53.43 | kippi | How can I check for congestion and if there is do something else |
| 11:54.10 | verywiseman | jgoo, yes |
| 11:54.23 | jgoo | Also, I just spoke, they'd like 8 of the phones to be base + handset phones (I prefer that for charging reasons rather than chocolate bar wifi phones) |
| 11:54.44 | jgoo | verywiseman, problem is, this would interface with their existing underground phone wiring - which sucks (hurrah for wifi) |
| 11:54.50 | jgoo | so I'll look linksys spa 1001 |
| 11:55.12 | jgoo | But if you know a good handset which has local wireless (like normal wireless phones) let me know |
| 11:56.02 | jgoo | Linksys sucks (that is a given) but how much is that SPA1001 |
| 11:56.15 | jgoo | I just realised I used 3 in one install, but my inebriated brain is struggling |
| 11:56.38 | jgoo | I also used a similar linksys device, with a diff model number... damn if I can recall why |
| 12:00.07 | kerx | Anyone know the equipment inside the Fonality PBX boxes? |
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| 12:30.20 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 12:30.48 | ccesario | hello, somebody know say me if includes command have major priority than extension s (in ael) ? |
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| 12:51.59 | defswork | I'm looking at using asterisk as a SIP gateway and need some idea of number of concurrent calls that can be handled - are there any charts that show this per CPU speed etc.. ? |
| 12:55.10 | tzafrir_laptop | defswork, highly depends on the codec you use |
| 12:55.46 | tzafrir_laptop | And naturally: depends on whether the media (RTP) actually passes through Asterisk. |
| 13:05.45 | defswork | if the codec from handset to afar peer use the same codec - and there isnt a direct route so asterisk has to route the rtp stream - does asterisk simply pass the stream through - un tampered - so less overhead too ? |
| 13:08.53 | *** join/#asterisk Floppe (i=muppet@crew.cluster-lan.org) |
| 13:10.07 | Floppe | is it possible to have an AGI script work in the background doing callerid lookups? |
| 13:14.43 | *** join/#asterisk sasargen (n=chatzill@173-100-8-38.pools.spcsdns.net) |
| 13:18.16 | tuxx- | an agi script is mostly run from the dialplan i think, so in your dialplan you can call the agi script and let it do the callerid lookup. |
| 13:18.53 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 13:24.00 | Floppe | have done that. sometimes the web is rather slow, so thought if one could do the lookup in the background so the dialplan can continue. |
| 13:24.51 | *** part/#asterisk LemensTS (n=customgt@adsl-70-238-154-243.dsl.stlsmo.sbcglobal.net) |
| 13:25.50 | *** join/#asterisk edwin_quijada (n=macaruch@200.26.172.98) |
| 13:25.55 | edwin_quijada | Hi! |
| 13:26.14 | edwin_quijada | There is a something way to do a call from CLI just for testing? |
| 13:27.07 | Gido-E | dial application? |
| 13:29.21 | tzafrir_laptop | edwin_quijada, originate |
| 13:31.06 | [TK]D-Fender | edwin_quijada: "help console dial" |
| 13:32.35 | tzafrir_laptop | dial is only good for one concurrent call (per sound card) |
| 13:32.47 | tzafrir_laptop | and requires having a working local sound card |
| 13:33.04 | tzafrir_laptop | usually is the case. but not always |
| 13:33.07 | [TK]D-Fender | tzafrir_laptop: Originate has 2 ends as opposed to just a basic OSS/ALSA dial. Its a trade-off |
| 13:33.13 | [TK]D-Fender | tzafrir_laptop: :) |
| 13:33.20 | [TK]D-Fender | tzafrir_laptop: He did jsut say a basic test... |
| 13:33.26 | tzafrir_laptop | yeah. but you normally have a phone |
| 13:33.37 | [TK]D-Fender | tzafrir_laptop: 6 of 1, half a dozen of the other... |
| 13:33.45 | tzafrir_laptop | originate Zap/g0/number Application Echo |
| 13:33.50 | tzafrir_laptop | to test a trunk |
| 13:33.54 | [TK]D-Fender | tzafrir_laptop: If he had a phone... he wouldn't need CLI for this :) |
| 13:34.16 | tzafrir_laptop | Or better: |
| 13:34.20 | *** join/#asterisk animositisomina (n=chatzill@201-75-60-134-ma.cpe.vivax.com.br) |
| 13:34.22 | [TK]D-Fender | tzafrir_laptop: points can be made for either case... |
| 13:34.24 | tzafrir_laptop | originate Zap/g0/number Application Playback tt-monkeys |
| 13:34.35 | tzafrir_laptop | (to make sure the call comes from Asterisk :-) |
| 13:35.26 | *** join/#asterisk stevetotaro (n=Steve@pool-71-254-231-87.hrbgpa.east.verizon.net) |
| 13:35.33 | [TK]D-Fender | tzafrir_laptop: I dunno.. tt-monkeys sounds a lot like the normal ambient background noise in my office ;) |
| 13:36.20 | *** join/#asterisk jmacz (n=jmacz@190.144.75.22) |
| 13:36.42 | eppigy | lol |
| 13:40.05 | *** part/#asterisk animositisomina (n=chatzill@201-75-60-134-ma.cpe.vivax.com.br) |
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| 13:46.31 | edwin_quijada | [TK]D-Fender: can i do this: asterisk -rx 'dial 8093701212@context' |
| 13:46.49 | [TK]D-Fender | edwin_quijada: read the INSTRUCTIONS |
| 13:48.15 | edwin_quijada | this commando just talk about extension I wanna know if I can dial a phone number to PTSN |
| 13:49.22 | [TK]D-Fender | edwin_quijada: read the INSTRUCTIONS <---- |
| 13:51.33 | *** join/#asterisk unasi7 (n=unasi7@84-75-23-200.dclient.hispeed.ch) |
| 13:55.23 | tuxx- | Floppe: you dont run the agi script on the same server/ |
| 13:58.17 | *** join/#asterisk doolph (n=doolph@cpe-001ee57d592e.cpe.cableonda.net) |
| 13:58.28 | Floppe | why? |
| 13:58.33 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
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| 14:07.28 | Katty | pamples things |
| 14:08.17 | eppigy | define pample please |
| 14:10.12 | verywiseman | [TK]D-Fender, i configured spa 3102 with my asterisk server,analog phone and softphone. and i could to make call between soft phone and analog phone ,but i can't make any external call ,and these are configurations http://fpaste.org/paste/1369 |
| 14:11.11 | [TK]D-Fender | verywiseman: you did not define [pstn] in sip.conf <--- |
| 14:11.20 | [TK]D-Fender | verywiseman: Your dial's have nowhere to go |
| 14:11.33 | [TK]D-Fender | verywiseman: You only set up the FXS port, not the FXO |
| 14:11.37 | eppigy | your dials are flawed |
| 14:11.52 | Katty | eppigy: http://angorakittyquilts.com/kitty1.gif |
| 14:12.39 | verywiseman | [TK]D-Fender, i am sorry , i forget to paste it http://fpaste.org/paste/1370 |
| 14:12.54 | verywiseman | but i defined pstn already |
| 14:13.12 | [TK]D-Fender | verywiseman: then show me the failed call attempt with SIP debug enabled |
| 14:13.16 | *** join/#asterisk kannan (n=kannan@121.246.242.95) |
| 14:13.45 | verywiseman | [TK]D-Fender, how? |
| 14:13.51 | [TK]D-Fender | verywiseman: and your port is wrong for [pstn] |
| 14:13.53 | kannan | hello all. In voicemail to email alert , whatever be the subjectline i set, * provides a default line of its own. How do i do this? |
| 14:14.09 | Katty | how do you do what? |
| 14:14.13 | Katty | that is a statement, not a question |
| 14:14.13 | [TK]D-Fender | kannan: its defined in sip.conf |
| 14:14.24 | [TK]D-Fender | kannan: err... [general] in voicemail.conf |
| 14:15.03 | *** join/#asterisk coppice (n=chatzill@201.193.17.210.dyn.pacific.net.hk) |
| 14:15.16 | Katty | hugs coppice |
| 14:16.07 | kannan | [TK]D-Fender ,hmm. but i want to define a different subject per context in voicemail.conf |
| 14:16.39 | [TK]D-Fender | kannan: You can't personalize per box. |
| 14:16.49 | eppigy | oic |
| 14:17.55 | *** join/#asterisk fskrotzki (n=fskrotzk@host198.textwise.com) |
| 14:17.56 | Katty | oh do you now |
| 14:18.00 | Katty | hugs fskrotzki |
| 14:18.03 | Katty | DO YOU REALLY |
| 14:18.17 | eppigy | yes |
| 14:18.20 | Katty | k |
| 14:18.22 | kannan | [TK]D-Fender , ok thank you |
| 14:18.32 | eppigy | i am very perceptive |
| 14:18.58 | Katty | but can you dance? |
| 14:19.03 | Katty | dance like rachel brice?! |
| 14:19.16 | eppigy | negative |
| 14:19.20 | Katty | bummer. |
| 14:19.27 | eppigy | :[ |
| 14:19.32 | Katty | she's a lovely dancer |
| 14:20.10 | Katty | REF: http://www.youtube.com/watch?v=XqJsaqykhLo |
| 14:20.56 | eppigy | id hit that |
| 14:21.13 | eppigy | im not gonna lie |
| 14:21.18 | dlewis | lol |
| 14:21.33 | Katty | i've never seen a dancer like that. |
| 14:21.43 | Katty | it's like a cross between robot and bellydancing |
| 14:21.54 | Katty | and that girl from queen of the damned |
| 14:22.18 | *** join/#asterisk dlewis_ (i=c7340d68@about/security/staff/dlewis) |
| 14:23.07 | verywiseman | [TK]D-Fender, which port i must bind to pstn? |
| 14:23.18 | [TK]D-Fender | verywiseman: the one you chose in your device's config <- |
| 14:23.31 | *** join/#asterisk Failrar (n=Failrar@fsm.xs4all.nl) |
| 14:26.37 | verywiseman | [TK]D-Fender, is it must to be 5060? |
| 14:26.52 | [TK]D-Fender | verywiseman: the one you chose in your device's config <- |
| 14:27.07 | verywiseman | [TK]D-Fender, yes it is same |
| 14:27.59 | [TK]D-Fender | verywiseman: not for the PSTN port it isn't |
| 14:28.14 | [TK]D-Fender | verywiseman: the LINE1 port and the PSTN port are DIFFERENT |
| 14:31.08 | *** join/#asterisk thansen (n=thansen@firewall.ssisecurity.com) |
| 14:31.47 | verywiseman | [TK]D-Fender, ok , line1 and pstn are binded to 5060, is it right? |
| 14:34.09 | *** join/#asterisk dmaayani (i=hidden-u@fw-int.transbeam.com) |
| 14:35.15 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 14:35.19 | verywiseman | [TK]D-Fender, ok , line1 and pstn are binded to 5060, is it right? |
| 14:35.22 | [TK]D-Fender | verywiseman: No. Go look |
| 14:35.53 | verywiseman | [TK]D-Fender, is it must to be different? |
| 14:36.07 | *** join/#asterisk mikkel (n=emilie@eye.exoro.dk) |
| 14:37.54 | mikkel | I'm in serious trouble, I have this one way sound problem again, people calling me can hear what I'm saying but I can't here them. Why is this problem so random, my phones can work for weeks and then suddenly I get this problem ? |
| 14:38.17 | mikkel | No change in anything |
| 14:38.39 | kannan | mikkel , maybe its going thry RTP ports not covered in asterisk rtp.conf, but the phones use those ports |
| 14:38.43 | verywiseman | [TK]D-Fender, is it must to be different? |
| 14:38.59 | kannan | or sipnat |
| 14:39.21 | [TK]D-Fender | verywiseman: Yes |
| 14:39.21 | mikkel | There is no NAT involved all is no LAN |
| 14:39.52 | [TK]D-Fender | mikkel: Describe your setup in detail |
| 14:40.52 | mikkel | I have a Sangoma ISDN2 card A500 and a Linksys SPA 2102 with a Analog phone connected. |
| 14:43.24 | verywiseman | [TK]D-Fender, i bind line1 to 3456 and pstn to 3457,but also i can't make external call |
| 14:43.43 | [TK]D-Fender | mikkel: pastebin a failed call with SIP debug enabled @ verbose 10 |
| 14:43.59 | [TK]D-Fender | verywiseman: And you aren't showing me anything. |
| 14:44.07 | verywiseman | how |
| 14:44.13 | mikkel | Peer audio RTP is at port 192.168.30.242:16384 |
| 14:44.20 | verywiseman | [TK]D-Fender, what is u mean? |
| 14:44.24 | mikkel | That is my Linksys box |
| 14:46.12 | mikkel | http://pastebin.com/m1b338187 |
| 14:47.01 | x86 | [TK]D-Fender: dont you love "what is u mean?" |
| 14:47.05 | x86 | tis classic ;) |
| 14:47.58 | x86 | MAIN SCREEN TURN ON |
| 14:48.06 | x86 | FUR GREAT JUSTICE! |
| 14:48.48 | ricko73 | #asterisk Now with Engrish |
| 14:49.43 | ricko73 | I mean I do feel for people who don't speak english as their native language, just makes for a good headache on some days |
| 14:50.44 | Gido-E | ricko73 your native language is english? |
| 14:51.16 | ricko73 | you sir, are correct |
| 14:51.43 | ricko73 | you win the prize! A drink from the firehose |
| 14:52.01 | *** join/#asterisk gamercod4 (n=gamercod@ABayonne-257-1-41-66.w81-49.abo.wanadoo.fr) |
| 14:52.05 | gamercod4 | hi all |
| 14:52.22 | gamercod4 | somebody have already install vtiger with asterisk module ? |
| 14:54.54 | *** join/#asterisk alibb (n=ali@41.226.243.205) |
| 14:56.08 | alibb | hi every one , is there , somewhere a user manual for Asterisk/Digium GUI 2.0 ? |
| 14:56.14 | kannan | gamercod4, i believe vicidial has an option of using vtigr as SRM |
| 14:56.21 | kannan | crm |
| 14:56.37 | *** join/#asterisk russellb (n=russellb@asterisk/digium-open-source-team-lead/russellb) |
| 14:56.37 | *** mode/#asterisk [+o russellb] by ChanServ |
| 14:57.05 | *** join/#asterisk BipBip (n=BipBip@194.65.5.235) |
| 14:58.07 | kannan | alibb , i think the book has a chapter on it |
| 14:58.39 | alibb | kannan, on gui 2.0 ? |
| 14:59.42 | *** join/#asterisk propellerhead (n=yogurt2u@host15.190-30-186.telecom.net.ar) |
| 15:02.33 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 15:03.55 | *** join/#asterisk wtsexton00 (n=tim@potatosalad.worldspice.net) |
| 15:06.25 | *** join/#asterisk dec3pti0n (n=dec3pti0@pdpc/supporter/student/dec3pti0n) |
| 15:08.05 | *** join/#asterisk rwaite (n=fieldyca@rrcs-74-218-125-86.central.biz.rr.com) |
| 15:12.03 | *** join/#asterisk saftsack (n=oliver@g230230093.adsl.alicedsl.de) |
| 15:12.25 | *** join/#asterisk rwaite (n=fieldyca@rrcs-74-218-125-86.central.biz.rr.com) |
| 15:16.19 | *** join/#asterisk putnopvut (n=putnopvu@nat/digium/x-a4fbd4ea419e7a89) |
| 15:16.19 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 15:17.38 | [TK]D-Fender | x86: Take off every Zig! |
| 15:18.03 | [TK]D-Fender | alibb: Virtually nothing |
| 15:18.57 | [TK]D-Fender | ricko73: Look up... look down... look at Mr. Frying Pan! *wham* |
| 15:19.19 | eppigy | WHAT YOU SAY |
| 15:20.05 | *** join/#asterisk lowtek (n=no@adsl-074-185-089-046.sip.bhm.bellsouth.net) |
| 15:21.01 | [TK]D-Fender | eppigy: YOU KNOW WHAT YOU ARE DOING |
| 15:21.14 | ricko73 | lol |
| 15:21.16 | alibb | [TK]D-Fender, u mean phisically it's somewhere ? |
| 15:21.34 | ricko73 | [TK]D-Fender: I forgot that one |
| 15:21.44 | [TK]D-Fender | alibb: No I mean it has almost no documentation |
| 15:22.02 | *** join/#asterisk jameswf (n=james@unaffiliated/jameswf-home) |
| 15:22.41 | alibb | thx [TK]D-Fender |
| 15:23.30 | *** join/#asterisk zeeesh (n=zeeesh@203.215.179.43) |
| 15:23.52 | zeeesh | how to check AGI.pm is installed or not ? |
| 15:24.17 | [TK]D-Fender | zeeesh: "man find" |
| 15:24.27 | *** join/#asterisk daniev (n=dnv@190.144.60.154) |
| 15:24.55 | wtsexton00 | time to own some polycoms |
| 15:25.13 | errr | try to use it and if perl pisses and moans then its at least not in your perl module path |
| 15:26.12 | citats | zeeesh: if you call 'perl -MAsterisk::AGI' from the command line it will complain if you dont have it. if you do it will just sit waiting for code input |
| 15:26.36 | zeeesh | No manual entry for AGI.pm |
| 15:26.53 | zeeesh | i have manually installed but its showing this error ? |
| 15:27.06 | zeeesh | should i install through cpan prompt? |
| 15:27.16 | citats | zeeesh: its showing that error when you type 'man AGI.pm' right? |
| 15:27.25 | zeeesh | yeah |
| 15:27.26 | [TK]D-Fender | zeeesh: Error? What error? You didn't show us anything |
| 15:27.33 | citats | zeeesh: there is no man page associated with it |
| 15:27.50 | zeeesh | ok |
| 15:28.02 | errr | you normally use perldoc to read "man pages" about perl modules |
| 15:28.22 | citats | zeeesh: type: perl -MAsterisk::AGI -e 'new Asterisk::AGI;' |
| 15:28.35 | citats | zeeesh: if you get an error you dont have it. if you just get returned your good to go |
| 15:28.51 | wtsexton00 | ah ha, sip headers show it coming from the correct line but asterisk matches it to the wrong peer |
| 15:29.41 | russellb | everyone praise citats, he wrote AGI.pm :) |
| 15:29.59 | russellb | jbot: citats++ |
| 15:29.59 | russellb | heh |
| 15:30.12 | *** join/#asterisk maddog01 (n=minotaur@d38-1-100.commercial1.cgocable.net) |
| 15:30.16 | *** join/#asterisk mog (n=mog@nat/digium/x-f42b9e5595c6c835) |
| 15:30.16 | *** mode/#asterisk [+o mog] by ChanServ |
| 15:30.41 | zeeesh | <citats>: it sends me to this promt > not doing anything |
| 15:31.45 | citats | zeeesh: sounds like you probably didn't close the ' ' |
| 15:31.56 | [TK]D-Fender | citats: But for all your suggestions I woulds till have liked to have seen an ERROR from him :) |
| 15:32.55 | zeeesh | <citats>: Missing argument to -M. |
| 15:33.05 | *** join/#asterisk maddog01 (n=minotaur@d221-65-55.commercial.cgocable.net) |
| 15:34.07 | errr | zeeesh, are you just learning perl? |
| 15:34.40 | zeeesh | no i just need to install one perl module which is giving errror .. |
| 15:34.43 | citats | zeeesh: here this is a little better since it will return a result. paste the following into the command line exactly |
| 15:34.47 | citats | zeeesh: perl -MAsterisk::AGI -e 'print $Asterisk::VERSION . "\n";' |
| 15:36.02 | *** join/#asterisk wonderworld (n=w@p57BDBE94.dip.t-dialin.net) |
| 15:37.51 | wonderworld | hi, we are running asterisk on a PRI. we want to hear the messages from telco side, when a call failes (like "number out of service" , "wrong number", etc. asterisk hangs up on those numbers and returns a dialstatus. i tried to set priindication to "inband" in zapata.conf, but it didn't change the behaviour. any ideas? |
| 15:38.49 | *** join/#asterisk Deeewayne (n=Deeewayn@nat/digium/x-dfb56a229ccba6d2) |
| 15:38.49 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 15:38.51 | Linuturk | I need some assistance with echo cancel and faxes. It seems my echo canceller isn't detecting when faxes come in, and it is screwing up our faxes. |
| 15:39.43 | Linuturk | I have a Ditech Quad 2 T1 echo canceller, and I have the ability to turn off echo cancel on specific channels on our pri |
| 15:40.10 | Linuturk | the problem is, our pri isn't channelized, meaning our DID's don't always come in on a specific channel |
| 15:41.41 | Linuturk | if it was like that, I could simply turn off echo cancel for a particular channel and be done with this |
| 15:41.53 | [TK]D-Fender | Linuturk: Go read your Ditech's manual... this has nothing to do with * |
| 15:43.08 | *** join/#asterisk SuPrSluG (n=SuPrSluG@firewall-a.buf.ny.i-evolve.net) |
| 15:43.10 | [TK]D-Fender | wonderworld: You could always check your failure codes and playback messages yourself... |
| 15:43.29 | wonderworld | the problem is, that many different events result into the same error message |
| 15:44.50 | wonderworld | and it's important for us to know what exactly was the cause for a filed call. |
| 15:45.10 | *** join/#asterisk pecanha (n=e@189.106.159.162) |
| 15:45.18 | wonderworld | for example: you get CHANUNAVAIL if a number is not in service, but you get it also if a cellphone is switched of |
| 15:45.25 | wonderworld | off |
| 15:46.05 | file | HANGUPCAUSE will give you the Q.931 cause code and there is an option (inbanddisconnect) which sets whether the audio will be sent after the call is released |
| 15:48.09 | wtsexton00 | [TK]D-Fender switching from peer to friend fixed my issue from last night |
| 15:48.30 | wonderworld | file: thanks. that seems to be what i was looking for. could you point me to the documentation for that feature? |
| 15:48.38 | file | wonderworld: what feature? |
| 15:48.38 | wtsexton00 | it matches it to the correct registration :) |
| 15:48.57 | wonderworld | where would i set "inbanddisconnect" |
| 15:50.27 | *** join/#asterisk DarkRift (n=dark@65.92.165.123) |
| 15:50.55 | file | I found it in chan_dahdi.conf.sample |
| 15:51.47 | *** join/#asterisk ScarEye (n=scareye@12.27.87.66) |
| 15:52.49 | wonderworld | thanks. we are running a quiet old asterisk here (actually the one that came with the latest debian stable) is there a way to find out when a feature was implemented into asterisk? inbanddsiconnect isn't in the documentation for zapata.conf |
| 15:53.08 | *** join/#asterisk IsUp (n=nocturne@unaffiliated/isup) |
| 15:53.08 | file | it was added a few months ago |
| 15:53.14 | IsUp | hello |
| 15:53.27 | *** join/#asterisk qdk_ (n=qdk@81.7.168.130) |
| 15:53.36 | wonderworld | ok, so my only option would be to upgrade to the latest version, right? |
| 15:53.41 | file | yes |
| 15:53.45 | file | or backport everything involved |
| 15:53.53 | IsUp | anyone using asterisk-video? like mp4play, mp4save tools? |
| 15:53.56 | *** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim) |
| 15:54.29 | *** join/#asterisk vAd0r (n=IceChat7@216-201-139-51.res.logixcom.net) |
| 15:54.37 | vAd0r | what is the command to restart the asterisk service |
| 15:54.55 | lowtek | vAd0r: Depends on your Linux distro. |
| 15:55.12 | vAd0r | how do i checkj |
| 15:55.13 | *** join/#asterisk pikachu2000 (n=pikachu2@196-209-199-207-rrdg-esr-2.dynamic.isadsl.co.za) |
| 15:56.04 | *** join/#asterisk shido6 (n=shido6@74-132-200-214.dhcp.insightbb.com) |
| 15:56.16 | *** join/#asterisk Defraz (n=T0tal@fw-poky.fuzecore.com) |
| 15:56.38 | wonderworld | vAd0r: lsb_release |
| 15:56.47 | lowtek | vAd0r: What distro? |
| 15:56.49 | vAd0r | anyone got a working config for an 877w to allow asterisk traffic through |
| 15:57.04 | vAd0r | not sure what distro |
| 15:57.04 | wonderworld | or try "cat /proc/version" |
| 15:57.12 | vAd0r | i switched from pfsense to 877w and now i cant get viatalk to connect |
| 15:57.32 | vAd0r | redhat |
| 15:57.37 | lowtek | pfsense is based on the freebsd kernel afaik and there has always been issues with NAT states |
| 15:57.43 | IsUp | "service asterisk restart" |
| 15:58.02 | vAd0r | that worked thx |
| 15:58.05 | Linuturk | [TK]D-Fender: I wish I had a manual for it :( |
| 15:58.06 | IsUp | np |
| 15:58.11 | wtsexton00 | I've not had issues with pfsense and states |
| 15:58.12 | vAd0r | i had pfsense working |
| 15:58.18 | vAd0r | just cant get the cisco 877w |
| 15:58.34 | doolph | is pfsense any good for qos? |
| 15:58.50 | *** join/#asterisk tobias (n=tobias@cpe-069-134-127-101.nc.res.rr.com) |
| 15:59.02 | lowtek | While I appreciate the pfsense project, I would buy a commercial firewall such as a Forigate. |
| 15:59.48 | wtsexton00 | pfsense uses openbsd packet filtering |
| 16:00.40 | jameswf | deffinately screw flexibility much much better to do it someone elses way.... hackers will never break a template |
| 16:00.53 | IsUp | anyone using asterisk-video? like mp4play, mp4save tools? |
| 16:00.59 | *** join/#asterisk Thorn_ (n=thorn@unaffiliated/thorn) |
| 16:01.35 | pecanha | hey guys, is it expected to use from-trunk? or should I create my own dialplan for this? |
| 16:02.30 | pecanha | I have a sip account, that's currently registered. I've set context=phones. And tried (my first line of phones context) |
| 16:02.30 | pecanha | "/etc/asterisk/extensions.conf" 56L, 1166C |
| 16:02.32 | Gido-E | we just wrote a lot off documentation about the traficshaper for the Linux kernel. |
| 16:02.33 | pecanha | ops |
| 16:02.41 | pecanha | exten => s,1,Answer() |
| 16:02.45 | pecanha | Am I wrong? |
| 16:03.08 | IsUp | yes |
| 16:03.10 | IsUp | its wrong |
| 16:03.11 | Gido-E | 50% was undocumenten. And some pictures where wrong, even on wikipedia. All are patched now. |
| 16:03.24 | IsUp | what you trying to do pecanha? |
| 16:04.02 | pecanha | IsUp: I would like to answer, do some background IVR (already did most part), and then when pressed 1 for example, it should redirect to my extension |
| 16:04.28 | IsUp | whats your incoming context? |
| 16:04.59 | pecanha | I set on sip.conf (my peer) as context=phones |
| 16:05.20 | pecanha | :/ |
| 16:05.48 | *** join/#asterisk mark_csi (n=csiadmin@host217-41-18-3.in-addr.btopenworld.com) |
| 16:06.43 | *** join/#asterisk Dr-Linux|home (n=Nothing@119.63.130.34) |
| 16:08.01 | Dr-Linux|home | question, let's say I'm in Mexico and dial US toll-free number, will i get charge for this call? |
| 16:08.37 | eppigy | TRABAJO |
| 16:09.14 | Dr-Linux|home | what |
| 16:11.27 | Qwell | Dr-Linux|home: probably, yes |
| 16:11.58 | Dr-Linux|home | Qwell: and if i call the same toll free from US? |
| 16:12.18 | Qwell | then yes, it should be |
| 16:12.28 | Dr-Linux|home | :S |
| 16:12.55 | *** join/#asterisk angryuser (n=gdobrovo@LPuteaux-151-42-35-99.w193-251.abo.wanadoo.fr) |
| 16:13.18 | pecanha | Qwell: can you give me some help please? :/ |
| 16:13.36 | Dr-Linux|home | Qwell: what's the difference if the toll-free is in US pointed to my server, now my customer calls me from US .. and other customer calls from the Mexico? |
| 16:13.47 | Dr-Linux|home | both will be charged same? :S or how does it work |
| 16:13.49 | Qwell | Dr-Linux|home: It's still too early in the morning to think. |
| 16:14.07 | Dr-Linux|home | Qwell: lol |
| 16:14.25 | Dr-Linux|home | here is 9:15pm |
| 16:14.34 | angryuser | hello i had "Forbidden - wrong password on authentication for INVITE to '"PEER NAME" <sip:0975XXX@IP.IP.IP.IP>;tag=as7cd935fd'" after 5 minutes it is gone without any changes for server, it's a trunk provider problem ? ideas ? |
| 16:14.37 | eppigy | Dr-Linux|home: 1 is the country code you are dialing |
| 16:14.42 | eppigy | dialing 1 from mexico |
| 16:14.46 | eppigy | i an international call |
| 16:14.53 | eppigy | *is |
| 16:14.58 | angryuser | *when calling out |
| 16:15.08 | *** join/#asterisk Dror_W (i=d4b38cc2@gateway/web/ajax/mibbit.com/x-a55c851a4a3a5994) |
| 16:15.08 | Dr-Linux|home | angryuser: you are using the same sip client on both of servers |
| 16:15.10 | Dr-Linux|home | use one |
| 16:16.28 | Dr-Linux|home | angryuser: i mean same peer on both servers, disable one and you are good to go :) |
| 16:16.57 | Dr-Linux|home | eppigy: yes |
| 16:17.04 | eppigy | ok |
| 16:17.06 | angryuser | Dr-Linux|home: hm i this it is aastra phones with a sip provider, i had this when they tryed to call out, i have no 2 server's but only one.... |
| 16:17.26 | angryuser | i think* |
| 16:17.49 | vAd0r | what is the debug verbose configs |
| 16:18.39 | Dr-Linux|home | angryuser: okay then do this in the dialplan before then you sending call to provider "exten => XXXXXXXX,1,Set(CALLERID(NUM)=any-number) |
| 16:18.42 | Dr-Linux|home | replace patern |
| 16:18.43 | bmoraca | Dr-Linux|home: whether or not a call is billed by a telco is up to that telco. |
| 16:18.54 | *** join/#asterisk manxpower (n=Administ@router.asteriasgi.com) |
| 16:19.22 | Dror_W | hi, I need some help with jabberreceive |
| 16:19.25 | bmoraca | if you purchased an 800 number and are paying for rates from mexico, then your customers in mexico will probably not be billed |
| 16:19.31 | Dr-Linux|home | bmoraca: i'm confused, becuase one of my customer asked this question, he is from mexico |
| 16:19.43 | Dr-Linux|home | i see |
| 16:19.55 | bmoraca | when you purchased your 800 number, did you purchase it to be used in mexico and canada? |
| 16:20.24 | bmoraca | the rates are different, and usually significantly higher |
| 16:20.58 | bmoraca | this is the reason that you typically will see "international numbers" advertised by companies as their local phone number and not as their toll-free number |
| 16:22.00 | Dr-Linux|home | bmoraca: no, the DID is only for US |
| 16:22.23 | angryuser | Dr-Linux|home: ok i will try, thank you, but what is happenning ? why ? |
| 16:22.42 | angryuser | some peers have caller id problems ? |
| 16:23.01 | angryuser | or their provider require outgoing caller id to be set ? |
| 16:23.14 | manxpower | angryuser: what specific issue are you having? |
| 16:23.56 | Dr-Linux|home | angryuser: actually i'm expereince this issue when same peer exists on both gateways, basically when you send INVITE message, it replies to peer on same server becuase it exists there as well |
| 16:24.12 | angryuser | manxpower: had "Forbidden - wrong password on authentication for INVITE to '"PEER NAME" <sip:0975XXX@IP.IP.IP.IP>;tag=as7cd935fd'" after 5 minutes it is gone |
| 16:24.35 | angryuser | it's sometimes its happening |
| 16:24.40 | tzafrir_laptop | jbot, tell Dror_W about ask |
| 16:25.18 | angryuser | manxpower: when calling out with one trunk |
| 16:25.19 | manxpower | I've never seen a "wrong password" message being caused by anything other than a wrong password. |
| 16:25.35 | manxpower | angryuser: I can't help you with trunks, just with SIP peers |
| 16:26.12 | bmoraca | Dr-Linux|home: that probably means that dialing your 800 number will get the customer in mexico one of two places--absolutely nowhere or to the wrong company |
| 16:27.03 | Dr-Linux|home | bmoraca: so how they will dial my toll-free? |
| 16:27.10 | Dr-Linux|home | using 1 before TF? |
| 16:27.28 | angryuser | manxpower: hm, yes but, sometimes they apper, sometimes they dont ;) strange coz they use the same outgoing peer |
| 16:27.32 | angryuser | appear |
| 16:27.39 | bmoraca | you'll have to contact your telco or a mexican telco to see whether or not you can get a toll-free number in mexico |
| 16:27.48 | bmoraca | chances are it will not be the same number as you have now |
| 16:28.23 | bmoraca | barring that, the customer will call your local telephone number and pay applicable long-distance charges |
| 16:28.24 | pecanha | Hey, can I group 5 extensions, and mention the group it on Dial() application? How can I do that? |
| 16:28.26 | Dr-Linux|home | i see |
| 16:28.57 | Dr-Linux|home | what if start mexico customer dials 1 before the number? |
| 16:29.02 | bmoraca | pecanha: declare a variable with the appropriate value you would have put into Dial() and then call that variable |
| 16:29.34 | Dror_W | I try to checkout jabberrecieve trunk and get an error: svn: PROPFIND of '/view/asterisk/team/phsultan/jabberreceive': 301 Moved (http://svn.digium.com) |
| 16:29.58 | bmoraca | Dr-Linux|home: my point is that your 800 number doesn't exist outside of the US |
| 16:30.07 | pecanha | bmoraca: is it better to create a context and then call it, or a variable? |
| 16:30.14 | Dr-Linux|home | ok thanks |
| 16:30.22 | tzafrir_laptop | Dror_W, you should use /svn , not /view |
| 16:30.37 | bmoraca | pecanha: that wasn't your question and isn't applicable to the solution to what you asked |
| 16:30.50 | Dror_W | svn co http://svn.digium.com/view/asterisk/team/phsultan/jabberreceive/ jabberreceive |
| 16:31.02 | pecanha | bmoraca: yes, your answer is enough thanks |
| 16:31.02 | *** join/#asterisk Maximilian_81 (n=Maximili@200-113-117-193.static.tie.cl) |
| 16:31.14 | Maximilian_81 | hi everyone, |
| 16:31.18 | bmoraca | pecanha: what you wanted to know was whether or not you could dial a group of extensions from the Dial() command without directly mentioning the extensions |
| 16:31.32 | bmoraca | pecanha: where you do that in your dialplan is entirely up to you |
| 16:32.07 | tzafrir_laptop | Dror_W, svn co http://svn.digium.com/svn/asterisk/team/phsultan/jabberreceive/ jabberreceive |
| 16:32.25 | Dror_W | thanks tzafrir :-) It works! :-)) |
| 16:32.27 | tzafrir_laptop | was that checkout command written in some document? |
| 16:32.51 | pecanha | bmoraca: I can setup something like: TEST=SIP/1000&SIP/1001 ? |
| 16:33.35 | Maximilian_81 | i have a question in extensions.conf, may i can configure an extencion generic , something like "exten => *,1,Macro(phone)" ???? |
| 16:33.48 | jameswf | nortel has failed.... does this mean a bailout for telecom? |
| 16:34.40 | Dr-Linux|home | mbranca: He tried and he can dial my TF using 1 prefix before the number |
| 16:34.59 | angryuser | Dr-Linux|home: it was globad default DID from another trunk overiding the DID needed to be send out, you had the reason :) |
| 16:35.09 | Qwell | jameswf: yes, get in line. I was here first. |
| 16:35.39 | vAd0r | anyone got a working config for an 877w to allow asterisk traffic through |
| 16:35.49 | jameswf | Tailgate party in the bailout line.... oh wait is that to flashy when your lookin for a handout |
| 16:35.50 | bmoraca | pecanha: yes, that's what you need to do |
| 16:35.52 | Dr-Linux|home | angryuser: is it working? |
| 16:35.58 | Qwell | jameswf: nope! |
| 16:36.01 | angryuser | Dr-Linux|home: yes |
| 16:36.04 | anonymouz666 | jameswf: nortel has failed 'cause lots so many customers to me. |
| 16:36.11 | anonymouz666 | lost |
| 16:36.21 | Qwell | jameswf: I mean, it's not like we're cooking the steaks in champagne or anything |
| 16:36.22 | anonymouz666 | he he |
| 16:36.25 | Qwell | or...are we? |
| 16:36.31 | bmoraca | Dr-Linux|home: be careful, though, as he probably has to pay international rates to call that number |
| 16:36.35 | Qwell | because that does sound pretty good |
| 16:36.46 | Qwell | anonymouz666: pretty much, heh |
| 16:37.16 | Dr-Linux|home | bmoraca: yes that what i think |
| 16:37.17 | bmoraca | Dr-Linux|home: you might benefit from reading this: http://en.wikipedia.org/wiki/Toll-free_telephone_number |
| 16:37.20 | *** join/#asterisk fskrotzki (n=fskrotzk@host198.textwise.com) |
| 16:37.35 | jameswf | its kangaroo meat |
| 16:37.37 | Dr-Linux|home | bmoraca: ohh i can also call this TF from pakistan using 1 prefix |
| 16:37.46 | jameswf | mmm i love jack in the box |
| 16:37.52 | Qwell | jameswf: I hate you. |
| 16:38.16 | Qwell | I have to drive to Nashville if I want jitb... |
| 16:38.20 | bmoraca | Dr-Linux|home: in addition to him paying rates, you'll have to pay usage on your toll-free number. no sense in paying twice. advertise your local phone number for international use. |
| 16:38.31 | jameswf | has always wondered what that meat textured paste they put in to jack in the box tacos is made of.... |
| 16:38.36 | Qwell | jameswf: soy |
| 16:38.45 | Qwell | (seriously) |
| 16:38.55 | jameswf | i love greasy soy mmmmm |
| 16:39.01 | Qwell | but man are they delicious |
| 16:39.03 | coppice | short for soylent green |
| 16:39.04 | bmoraca | jack in the box tacos are pure heaven |
| 16:40.21 | jameswf | Textured Vegetable Protein (Soy Flour, Caramel Color), Soy Grits, Seasoning [Chili Powder (Chili Peppers, Spices, Salt, Garlic Powder), Wheat Flour, |
| 16:40.56 | jameswf | Imitation Beef Extract |
| 16:41.03 | jameswf | oh i am sooooo hungry |
| 16:41.17 | Qwell | jameswf: better be bringing me a couple... |
| 16:41.43 | jameswf | I can send em ups ground they will arrive in 5-10 days |
| 16:41.53 | Qwell | not like they'll go bad or anything |
| 16:43.39 | Maximilian_81 | in extensions.conf, may i can configure an extension generic , something like "exten => *,1,Macro(phone)" for all my users? |
| 16:45.24 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 16:45.37 | Maximilian_81 | i try exten => ${EXTEN},1,Macro(phone), but dosent work... "extension '101' rejected because extension not found." |
| 16:45.43 | *** join/#asterisk Thorn_ (n=thorn@unaffiliated/thorn) |
| 16:46.17 | *** part/#asterisk gego (n=rick@b238085.customer.hansenet.de) |
| 16:51.08 | Maximilian_81 | anyone can help me |
| 16:51.55 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 16:52.06 | angryuser | ~book |
| 16:52.06 | jbot | hmm... book is probably Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://tfot.leifmadsen.com or see ~buybook |
| 16:52.07 | mark_csi | Maximilian_81: that should work fine |
| 16:52.08 | [TK]D-Fender | Maximilian_81: gO READ cHAPTER 5 AGAIN |
| 16:52.16 | [TK]D-Fender | Maximilian_81: And no, it should NOT work |
| 16:52.30 | [TK]D-Fender | Maximilian_81: exten => ${EXTEN},1,Macro(phone) <-- you cannot just reference a VARIABLE as an exten there |
| 16:52.53 | *** join/#asterisk jfergus (n=jfergus@smtp-3.cmcflex.com) |
| 16:52.56 | [TK]D-Fender | Maximilian_81: exten => NOTAVARHERE,priritiy,App |
| 16:53.49 | *** join/#asterisk saftsack (n=oliver@g228066121.adsl.alicedsl.de) |
| 16:53.59 | *** join/#asterisk CunningPike (n=arodgers@204.239.10.119) |
| 16:54.03 | Maximilian_81 | let me check :) thanks |
| 16:54.13 | angryuser | a tip exten => 101, |
| 16:55.09 | Maximilian_81 | yep if i put exten => 101,1,Macro(phone) , its work |
| 16:55.33 | jfergus | Anyone ever try to make custom states so that for example, I could put a *7815 for "Out to Lunch" then have a STATE=LUNCH, so that when someone calls that extension they get a playback of "Out to lunch"? |
| 16:56.40 | Maximilian_81 | but i want to generate muy user automatically so, if one new user comes i have to edit the extensions.conf and add the macro for this user, thats why i want to do a generic rule for this users |
| 16:56.54 | angryuser | jfergus: it's easy if you use astdb and gotoif check before, but there are a lot of ways doing the same thing |
| 16:57.48 | angryuser | jfergus: to set a var in astdeb use Set() and DB() |
| 16:57.53 | angryuser | astdb* |
| 16:57.54 | jfergus | I currently have the dnds set to use astdb. I'm just trying to figure out how to get that to correlate to when someone calls, instead of just going to their voice mail or temp message, to playback a specific dnd message. |
| 16:58.25 | *** join/#asterisk Ericounet (n=Ericoune@ACaen-151-1-9-29.w86-215.abo.wanadoo.fr) |
| 16:58.33 | angryuser | jfergus: gotoif , again |
| 16:58.57 | [TK]D-Fender | Maximilian_81: You seem to have failed to grasp the concept of PATTERNS. Go read the book |
| 16:59.32 | [TK]D-Fender | jfergus: Split usage between the Unavailable and Busy messages |
| 17:00.18 | mmlj4 | analog gateways (FXO banks).... what's a good brand? reliable and not too expensive? |
| 17:00.46 | jfergus | Wouldn't that just send the call to either the unavailable or busy message? |
| 17:01.04 | [TK]D-Fender | mmjDescribe your precise needs |
| 17:01.19 | [TK]D-Fender | jfergus: Do you just want a message, or also to hit VM? |
| 17:01.26 | *** join/#asterisk hfb (n=hfb@pool-96-229-38-185.lsanca.dsl-w.verizon.net) |
| 17:01.49 | *** part/#asterisk Ericounet (n=Ericoune@ACaen-151-1-9-29.w86-215.abo.wanadoo.fr) |
| 17:01.54 | jfergus | exten => *7803,n,UserEvent(ASTDB|Channel: ${CHANNEL}^Family: DND^Value: In Meeting) |
| 17:01.54 | jfergus | exten => *7803,n,Set(DB(DND/${AMPUSER})=YES) |
| 17:02.18 | jfergus | I already have this. I would like it to basically say "In a Meeting" (in that example) and then go to voice mail. |
| 17:02.54 | [TK]D-Fender | jfergus: 1 gotoif to check your DB state. If its not set, jump right to Voicemail, otherwise play a sound file first |
| 17:05.52 | jfergus | That's what I thought, I'm just trying to see WHERE. :D And WHICH. There are several ways I can think of doing it, just wasn't sure what the best solution to it would be. |
| 17:05.53 | mmlj4 | [TK]D-Fender: 4 POTS lines in, SIP out to * box |
| 17:06.03 | *** join/#asterisk boynas (n=garyflor@wsip-98-190-136-194.ph.ph.cox.net) |
| 17:06.14 | [TK]D-Fender | mmlj4: Any plans for expansion? |
| 17:06.21 | mmlj4 | not for this project |
| 17:06.35 | [TK]D-Fender | jfergus: Clearly right before you would normally call VoiceMail <- |
| 17:06.42 | *** join/#asterisk botox93 (n=botox93@213.221.82.242) |
| 17:06.46 | [TK]D-Fender | mmlj4: Then an external solution really isn't cost effective |
| 17:07.09 | [TK]D-Fender | mmlj4: AudioCodes & Mediatrix both make decent external gateways if thats what you really want. |
| 17:07.20 | boynas | Quick question... Every time that you connecta a pri port to a Channel bank timing should be 0 right? Because channel banks are always slaves of asterisk. |
| 17:07.21 | *** join/#asterisk ChkDigit (n=mike@static24-72-71-175.regina.accesscomm.ca) |
| 17:07.34 | mmlj4 | you'd recommend a digium or clone 4-port FXO card instead? |
| 17:07.36 | *** join/#asterisk fogo (n=Paul@69.169.132.35.provo.static.broadweave.net) |
| 17:07.41 | [TK]D-Fender | boynas: Typically yes |
| 17:08.05 | boynas | tk Fender: can you tell me as scenario where that is not the case? |
| 17:08.23 | [TK]D-Fender | mmlj4: http://www.telephonydepot.com/Catalog/Sangoma-B600/B600D-Analog-Voice-Card |
| 17:08.24 | bmoraca | mmlj4: i've had very good luck with Sangoma echo cancelling TDM cards |
| 17:08.52 | [TK]D-Fender | boynas: Almost never happens. Maybe if the CB sits between * and the telco. |
| 17:09.11 | boynas | yeah, that is not the case... |
| 17:09.17 | mmlj4 | rhino cards any good? same ballpark price as the 2 externals you mentioned, the sangoma's $150+ more |
| 17:09.25 | [TK]D-Fender | boynas: that'd be the other 99% where indeed you set timng for it |
| 17:09.37 | mmlj4 | but yeag, I've repeatedly heard sangomas are good |
| 17:09.43 | [TK]D-Fender | mmlj4: So far history says "flakey" |
| 17:09.51 | mmlj4 | fair enough |
| 17:09.54 | jfergus | [TK]D-Fender, yeah, I realize that. What I meant was where the dial bit for that is. Mostly talking out loud here. But my question was answered. Thank you. |
| 17:09.55 | boynas | well the thing is that i am pretty much reproducing an existing setup, and that is the setup and is working fine.. but i am between implementing that same thing or do my own setup |
| 17:10.04 | *** part/#asterisk jfergus (n=jfergus@smtp-3.cmcflex.com) |
| 17:10.11 | [TK]D-Fender | mmlj4: YMMV and maybe they've improved, but you hear almost nothing new from them in here. |
| 17:10.42 | mmlj4 | "flakey" scared me off :-) |
| 17:10.52 | pecanha | Guys, having problem with this simple dialplan: http://pastebin.com/m16c289d0 . When a wrong extension is pressed it goes nowhere, and when occupied I would like to remain dialing. What I'm doing wrong? |
| 17:10.55 | dan__t | 'morning. |
| 17:11.29 | [TK]D-Fender | pecanha: you are missing COMMAS |
| 17:11.48 | *** join/#asterisk erth64net (n=erth64ne@69-30-67-191.dq1sn.easystreet.com) |
| 17:11.49 | mmlj4 | I had my heart set on a linksys boxlet, but of course they discontinued that model |
| 17:12.09 | [TK]D-Fender | pecanha: You also tried making more priorities for your "3" exten and copy&pasted and left "2"'s in there |
| 17:12.14 | [TK]D-Fender | pecanha: VERY sloppy |
| 17:12.42 | [TK]D-Fender | mmlj4: AFAIK the SPA-9000 is still out there, but you don't get port-level control over it. |
| 17:12.58 | [TK]D-Fender | mmlj4: Which I find is jst a piss-off. |
| 17:13.38 | pecanha | [TK]D-Fender: true! where am I missing commas? |
| 17:13.51 | [TK]D-Fender | pecanha: Also your IVR has a fixed playback which forces them to wait till it finishes before the background that follows starts during which they can at least cut that one short. |
| 17:14.07 | [TK]D-Fender | pecanha: exten => i,1Playback(opcaoinvalida) <- you tell ME |
| 17:14.43 | slima | I have GotoIf($["${CALLERID(num)}" != "Unknown"]?s,5) what can i do something like that: if CALLERID(num) != Unknown AND dupa.8 go to s,5 |
| 17:15.02 | pecanha | [TK]D-Fender: yes, that was my objective, to force playback, before background starts to play |
| 17:15.12 | [TK]D-Fender | slima: && |
| 17:15.23 | slima | o, thx |
| 17:15.37 | [TK]D-Fender | slima: Go read up on Asterisk Expressions on the WIKI and in your docs folder |
| 17:16.08 | EI5GTB | anyone know wher i would come across an oslec patch for dahdi> im following the guide for installing it with zaptel.. and it tells me to patch zap |
| 17:16.42 | EI5GTB | so naturally, i assume that a zap patch wont work with dahdi |
| 17:16.52 | mmlj4 | oh, wait... SPA400 might do what I want, only $320 from voipsupply (it's not the vanilla device they dropped, though) |
| 17:17.05 | [TK]D-Fender | EI5GTB: http://www.freepbx.org/forum/freepbx/installation/asterisk-1-6-dahdi-oslec |
| 17:17.14 | *** join/#asterisk SlicerDicer (n=kvirc@24-119-155-26.cpe.cableone.net) |
| 17:17.26 | mmlj4 | audiocodes and the other one may work, but I've used sipura devices for 3+ years now |
| 17:17.27 | [TK]D-Fender | mmlj4: Pretty dumb device though... and less faith in the EX on it. |
| 17:17.29 | [TK]D-Fender | EC* |
| 17:17.38 | EI5GTB | [TK]D-Fender, hmm, im on 1.4.22 |
| 17:17.40 | mmlj4 | hrm, yeah |
| 17:17.52 | [TK]D-Fender | mmlj4: Sipura/Linksys is good for ATA's, just not so great for FXO |
| 17:17.59 | mmlj4 | ah. |
| 17:18.05 | EI5GTB | the dahdi/oslec isnt asterisk dependant? |
| 17:18.19 | slima | [TK]D-Fender: thanks, its working ;) |
| 17:18.33 | [TK]D-Fender | EI5GTB: No. |
| 17:18.44 | [TK]D-Fender | slima: You're welcome |
| 17:18.49 | *** join/#asterisk Thorn (n=thorn@unaffiliated/thorn) |
| 17:19.00 | EI5GTB | [TK]D-Fender, ah, ok, just to confirm: no = no, not dependant |
| 17:19.14 | pecanha | [TK]D-Fender: perfect, everything seems to be working fine now. Yet, after 20 seconds I would like to playback a ocuppied file and dial again. I thought using goto, but how can I call exten => 1,n,Dial for example? if the priority is n? |
| 17:19.36 | [TK]D-Fender | pecanha: Go read up on LABELS |
| 17:19.42 | [TK]D-Fender | pecanha: this is all dialplan 101 |
| 17:20.06 | [TK]D-Fender | EI5GTB: DAHDI is not inherently tied to *. |
| 17:20.15 | [TK]D-Fender | EI5GTB: You could use it for raw DATA for instance |
| 17:20.28 | [TK]D-Fender | EI5GTB: Clearly the productive value would vary |
| 17:20.34 | EI5GTB | great jub, but * will be able to see it an odlec |
| 17:20.42 | EI5GTB | oslec** |
| 17:20.42 | pecanha | [TK]D-Fender: tks, i will for it |
| 17:20.54 | [TK]D-Fender | EI5GTB: * will if you set it up |
| 17:21.02 | EI5GTB | good good |
| 17:21.14 | EI5GTB | i currently have * running 100% with *, just no oslec |
| 17:21.20 | [TK]D-Fender | EI5GTB: Remember that * doesn't see OSLEC. DADHI just "does its thing" |
| 17:21.24 | EI5GTB | 100% with dahdi |
| 17:21.33 | EI5GTB | yea |
| 17:22.12 | jameswf | md5sum on a dvd iso takes forever |
| 17:22.14 | EI5GTB | it might be ok for people in the know, but heh, this name change has been very confusing for people like me |
| 17:22.14 | Maximilian_81 | thank everyone |
| 17:22.48 | *** join/#asterisk vAd0r (n=IceChat7@wsip-70-182-92-115.ok.ok.cox.net) |
| 17:22.50 | vAd0r | what are the debug commands |
| 17:22.54 | [TK]D-Fender | EI5GTB: You should know that you NEVER told * what EC to use... |
| 17:23.24 | EI5GTB | yea, well, i never hadn an ec anyway |
| 17:24.52 | *** join/#asterisk grx0 (n=dave@64-150-178-3.kansascity.abac.net) |
| 17:24.55 | EI5GTB | great, iv to rebuild dahdi, again |
| 17:25.03 | EI5GTB | must be my 4th time now :) |
| 17:25.57 | pecanha | [TK]D-Fender: understood labels, thanks again. There is any option to play a audio file while using Dial too? |
| 17:26.22 | [TK]D-Fender | pecanha: "core show application dial" |
| 17:27.19 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 17:28.27 | EI5GTB | [TK]D-Fender, one last q, that guide abviously assumes oslec is compiled and installedf? yes? |
| 17:28.28 | *** join/#asterisk wonderworld (n=ww@ip-62-143-28-129.unitymediagroup.de) |
| 17:29.02 | [TK]D-Fender | EI5GTB: Guess |
| 17:29.10 | EI5GTB | im guessing it does |
| 17:29.18 | [TK]D-Fender | EI5GTB: SMRT |
| 17:29.25 | EI5GTB | but what seemed abvious to me in the past has been wrong.. |
| 17:29.29 | EI5GTB | ;) thanks |
| 17:29.59 | [TK]D-Fender | EI5GTB: Past failure is not an excuse to become completely co-dependent :p |
| 17:30.34 | EI5GTB | haha, indeed.. im very cautious now about ploughing into a compile of somthing i dont know to much about.. |
| 17:30.44 | EI5GTB | im learning tho! |
| 17:30.57 | EI5GTB | someday i will recover my past confidence, and begin to break things again! |
| 17:31.16 | *** join/#asterisk ReDNeQ (n=ReDNeQ@70.114.226.70) |
| 17:33.26 | *** join/#asterisk vgster (n=vgster@94-194-190-189.zone8.bethere.co.uk) |
| 17:33.34 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 17:34.24 | EI5GTB | in soviet russia bots sit quietly in the corner ;) |
| 17:35.04 | dan__t | and doesn't ask for a bailout |
| 17:39.58 | macli | Anyone here has tried dahdi-linux-2.1.0.3 with oslec echo cancel? |
| 17:40.17 | EI5GTB | im in the process |
| 17:40.33 | EI5GTB | EI5GTB: http://www.freepbx.org/forum/freepbx/installation/asterisk-1-6-dahdi-oslec |
| 17:40.43 | EI5GTB | thanks to : [TK]D-Fender |
| 17:42.08 | *** join/#asterisk ray_tru` (n=ray_tru`@96-28-43-212.dhcp.insightbb.com) |
| 17:42.53 | ray_tru` | <SIP/ray-081cf760> Playing 'digits/3' (language 'fr') <--- But it is playing the sounds from /usr/share/asterisk/sounds, and not the subdirectory "FR" ... is there something i'm missing here? |
| 17:43.12 | ray_tru` | "fr" is the directory, not "FR" |
| 17:43.34 | mmlj4 | [TK]D-Fender: your wisdom is recorded for posterity: http://joeykelly.net/scratchpad/index.cgi?FXOs |
| 17:43.45 | macli | EI5GTB: Thanks, I will try the steps |
| 17:44.08 | EI5GTB | macli, np, as i said, im just in the middle of doing it as we speek |
| 17:44.15 | EI5GTB | ill keep you updated if you do same |
| 17:44.41 | ray_tru` | http://www.voip-info.org/wiki/view/Asterisk+cmd+SetLanguage |
| 17:45.00 | ray_tru` | It says there that if the file exists under fr/digits/ then it will play that file if language is set to "fr" |
| 17:46.32 | *** join/#asterisk Thorn (n=thorn@unaffiliated/thorn) |
| 17:47.08 | kannan | hello , i have set a ringall of 10 SIP phone, with timeout of 30 secs. If no one picks, in exten =>t,1 i send it a voicemail box. It takes 5-6 seconds to go to the exten t. How to do it immediate, with no blanks in the middle for the Calling party? |
| 17:48.00 | ruben23 | http://pastebin.com/m4daba090 |
| 17:48.19 | ruben23 | anyone have idea on this error http://pastebin.com/m4daba090 |
| 17:48.28 | *** join/#asterisk Thorn (n=thorn@unaffiliated/thorn) |
| 17:48.48 | Juggie | kannan, dial has an option to continue dialplan execution |
| 17:49.02 | kannan | like +103 , in the ringall exten itself? |
| 17:49.19 | Juggie | no |
| 17:49.27 | Juggie | jumping is gone |
| 17:49.30 | kannan | or a flag in dial cmd ? |
| 17:50.10 | Juggie | its a flag |
| 17:50.11 | Juggie | g |
| 17:50.28 | Juggie | so you'd have Dial(..thephones..,30,g) |
| 17:50.41 | Juggie | so if your dial was exten=>s,1,Dial.... |
| 17:50.53 | Juggie | then if no one picked up, it would go to exten=>s,2 |
| 17:50.58 | Juggie | which is where you want to put your voicemail |
| 17:51.09 | Juggie | if someone picks up then the next step isnt executed |
| 17:51.40 | *** part/#asterisk russellb (n=russellb@asterisk/digium-open-source-team-lead/russellb) |
| 17:52.14 | *** join/#asterisk rafiks_ (n=rafiks@c-68-56-151-58.hsd1.fl.comcast.net) |
| 17:52.27 | ray_tru` | So has anyone had problems with the SET(language) command? |
| 17:52.46 | ray_tru` | It always looks in /usr/share/asterisk/sounds instead of the subdirectory for the language |
| 17:53.14 | *** join/#asterisk thetrooper7 (n=thetroop@adsl56-150.kln.forthnet.gr) |
| 17:53.50 | ruben23 | anyone have idea..? |
| 17:54.16 | rafiks_ | something weird is going on w/my asterisk box... i got this lot of garbage in the console as if |
| 17:54.41 | thetrooper7 | hello ppl |
| 17:54.46 | rafiks_ | somebody is ttrying to make a call |
| 17:55.59 | outtolunc | ruben23: maybe you should ask on the vicidial forums, also no one is gonna help you with a 'make clean' error <G> |
| 17:56.44 | rafiks_ | it keeps on dialing somwhere with random 1757XXXXXXX numbers |
| 17:57.11 | ruben23 | outtolunc:ok thanks.. |
| 17:57.32 | rafiks_ | i tried to disable one sip trunk but still does not work.. |
| 17:58.29 | rafiks_ | i tinkered with extensions.conf ,no luck,, |
| 17:59.27 | rafiks_ | the last thing I did before I noticed this is I tried out the jukebox.agi |
| 17:59.47 | rafiks_ | but now i disbled it ,along with festival server |
| 18:03.10 | rafiks_ | http://pastebin.com/m1bf01df1 |
| 18:03.16 | *** join/#asterisk ta^3 (n=tacvbo@189.146.193.18) |
| 18:04.18 | rafiks_ | thats the snippet of the console output |
| 18:05.12 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 18:06.01 | *** join/#asterisk dieguito84 (n=diego@host204-186-dynamic.23-79-r.retail.telecomitalia.it) |
| 18:06.08 | riddlebox | anyone around |
| 18:10.20 | rafiks_ | jsut me |
| 18:10.28 | rafiks_ | i guess,blabbering |
| 18:13.25 | [TK]D-Fender | kannan: If noone picks up it'll just hit the next priority. You do NOT just to "t" for things like this |
| 18:14.08 | macli | EI5GTB, It seems working following the steps, except I changed /usr/src/dahdi/linux/drivers/staging path to usr/src/dahdi/drivers/staging |
| 18:14.29 | EI5GTB | uea, after much figuring out i realised that too |
| 18:14.32 | EI5GTB | making atm |
| 18:14.47 | ruben23 | anyone can help a make clean zaptel compile error....http://pastebin.com/m55a6136 |
| 18:15.07 | EI5GTB | i just made the dir /usr/src/dahdi/linux and copied the contents of the dahdi svn trunk into it |
| 18:15.27 | EI5GTB | macli, so its up and running for you./ |
| 18:15.29 | EI5GTB | ? |
| 18:15.48 | macli | Yes, it is running, I just made a test call |
| 18:15.55 | EI5GTB | cool cool |
| 18:16.11 | *** join/#asterisk `paul (n=kutimoy@119.111.117.37) |
| 18:16.18 | EI5GTB | someone should really not thet "dahdi/linux" in can be a bit confusing |
| 18:16.20 | `paul | i added a new field on cdr table but it doesnt get populated when i do a set(CDR(myfield)='test') in my dial plan.... help pls |
| 18:16.22 | *** join/#asterisk hiddenman (n=hiddenma@altlinux/developer/hiddenman) |
| 18:16.25 | hiddenman | hi all |
| 18:16.41 | ray_tru` | For anyone who cares, it seems that say.conf handles the multi language of digits, in reference to my problem above. |
| 18:16.47 | *** join/#asterisk Greek-Boy (n=greek@41.222.89.77) |
| 18:16.49 | ray_tru` | Good work on the documentation on that one btw. |
| 18:17.01 | hiddenman | i'm trying to establish a fax session and checking it according to this doc: http://tools.ietf.org/html/draft-mule-sip-t38callflows-01#section-5.2 |
| 18:17.35 | macli | I have echocanceller=oslec,1,4 in /etc/dahdi/system.conf and dahdi started ok, the call went through too, so I assume oslec is working :) |
| 18:18.00 | [TK]D-Fender | macli: "dahdi-cfg -vvvv" should confim what it uses |
| 18:18.01 | hiddenman | after receiving F11 invite my asterisk immediately sends SIP/2.0 488 Not acceptable here to the termination side. what happenned? how does asterisk know it's acceptable or not? |
| 18:18.19 | [TK]D-Fender | hiddenman: Typically means "can't agree on codecs" |
| 18:18.25 | hiddenman | may be special "codec"? udptl_t38=yes is everywhere |
| 18:18.39 | [TK]D-Fender | hiddenman: go pastebin the entire SIP debug for the call attempt |
| 18:18.42 | [TK]D-Fender | ~pb |
| 18:18.43 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste, or , http://bin.cakephp.org/ |
| 18:18.44 | [TK]D-Fender | ^^^^^^^^^^^ |
| 18:18.56 | *** join/#asterisk stevetotaro (n=Steve@pool-71-254-231-87.hrbgpa.east.verizon.net) |
| 18:18.57 | [TK]D-Fender | hiddenman: just saying what you set doesn't say what they are offering. LOOK |
| 18:18.59 | hiddenman | [TK]D-Fender: but it's a t.38 session, there are no separated codecs, are there? |
| 18:19.04 | file | hiddenman: it is t38pt_udptl |
| 18:19.05 | `paul | whats the variable for the calling sip extension in asterisk??? |
| 18:19.07 | file | not udptl_t38 |
| 18:19.18 | EI5GTB | macli, cool cool, im just in the process of restarting, for some reason i caoulnt get my dahdi chan to come back up |
| 18:19.27 | hiddenman | file: yeah, i was wrong, just bad memory ;-) |
| 18:19.28 | macli | Channel 01: FXO Kewlstart (Default) (Echo Canceler: oslec) (Slaves: 01) |
| 18:19.38 | EI5GTB | :D |
| 18:19.46 | [TK]D-Fender | `paul: Isn't a ral fixed value for this |
| 18:19.47 | hiddenman | [TK]D-Fender: i know pastebin, thanks. so... |
| 18:20.11 | [TK]D-Fender | `paul: you can get part based on the CHANNEL name |
| 18:20.38 | `paul | o you mean chop the things before and after the channel name? |
| 18:20.46 | `paul | to get the extension |
| 18:20.48 | macli | EI5TGB, I rebooted my computer once after installing dahdi |
| 18:21.04 | [TK]D-Fender | `paul: thats one way. Another is if the callerid matches, or you set a var in your peer config. |
| 18:21.10 | macli | check /dev/dahdi exist or not |
| 18:21.11 | EI5GTB | macli, yea, ok, my servers running :D time to see if it works |
| 18:21.17 | *** join/#asterisk BBHoss (n=bbhoss@c-68-62-168-242.hsd1.al.comcast.net) |
| 18:21.36 | EI5GTB | hmmm |
| 18:21.39 | EI5GTB | still a horrible echo |
| 18:21.47 | BBHoss | anybody know the story behind "Message: Thanks for all the fish."? I get it as a response to Action: Logoff in AMI |
| 18:21.51 | EI5GTB | macli, you had to add in that bit about the echo canceler? |
| 18:21.51 | *** join/#asterisk rdgr (n=rich@82-46-0-91.cable.ubr01.aztw.blueyonder.co.uk) |
| 18:21.59 | EI5GTB | in system.conf |
| 18:22.04 | [TK]D-Fender | BBHoss: Go read some Douglas Adams |
| 18:22.08 | *** join/#asterisk grEvenX (n=even@cB78D5AC1.dhcp.bluecom.no) |
| 18:22.09 | `paul | ok thanks ill just set their callerid to their extension itbe easier that way |
| 18:22.13 | macli | Yes, explicitly |
| 18:22.15 | BBHoss | [TK]D-Fender: ? |
| 18:22.17 | EI5GTB | aha |
| 18:22.35 | BBHoss | ah i see |
| 18:22.39 | thetrooper7 | ~ask no voice |
| 18:22.58 | thetrooper7 | ~help |
| 18:23.09 | macli | I don't have much echo problem though even if I don't use oslec |
| 18:23.20 | BBHoss | wonder who put that in there |
| 18:23.26 | stevetotaro | ~freepbx |
| 18:23.27 | jbot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 18:23.41 | EI5GTB | macli, yea, im on a cheapo x100p |
| 18:23.43 | hiddenman | [TK]D-Fender: so here is the full sip session: http://pastebin.ca/1308201 look at the lines 680-722 |
| 18:24.32 | EmleyMoor | My N95 won't let me configure it the way I need to now - it complains that "Profiles in the same realm must have the same username." - but they do! |
| 18:25.09 | hiddenman | [TK]D-Fender: the scheme is: D-Link ATA with full and working T.38 support,NAT, port 5065->GW->Asterisk with public IP->Mera MVTS+SIPHIT with public IP->terminator side with working T.38 |
| 18:25.39 | hiddenman | [TK]D-Fender: call immediately ends after first INVITE (F11) according to the http://tools.ietf.org/html/draft-mule-sip-t38callflows-01#section-5.2 |
| 18:26.12 | EI5GTB | macli, :D no echo :D:D:D:DD: |
| 18:26.20 | EI5GTB | <- is happy |
| 18:26.51 | macli | Congratulations :-) |
| 18:26.57 | [TK]D-Fender | hiddenman: Well its in response to the attempted T38 whose value you were told you got wrong.. I'd look at this all over if I were you |
| 18:27.42 | hiddenman | [TK]D-Fender: sorry, didn't get, what was wrong with T.38 value? |
| 18:28.01 | [TK]D-Fender | hiddenman: Go look at your configs again |
| 18:28.34 | hiddenman | do you mean t38pt_udptl = yes ? |
| 18:28.44 | *** join/#asterisk waverly360 (n=waverly@ns2.dalcon.com) |
| 18:28.58 | hiddenman | hmmm. are these spaces errors? |
| 18:29.20 | hiddenman | no, there aren't, according to example |
| 18:31.38 | *** join/#asterisk MICKEY17 (n=knoppix@217.116.243.93) |
| 18:32.31 | *** join/#asterisk riddlebox (n=user@IP-16-235.wustl.edu) |
| 18:37.03 | ricko73 | [TK]D-Fender: this could be your standard reply in the future http://www.youtube.com/watch?v=qileP4bAzek&feature=related |
| 18:37.23 | *** join/#asterisk sah-work (n=Bawbatos@65-119-47-100.dia.static.qwest.net) |
| 18:37.23 | hiddenman | [TK]D-Fender: could you please point me to the error in my configs? t38pt_udptl=yes is everywhere, in the general section as well |
| 18:38.00 | hiddenman | and i immediately gets a BYE after fax preamble has been sent |
| 18:38.38 | hiddenman | and i get [Jan 14 18:36:02] WARNING[4820]: chan_sip.c:5162 process_sdp: Unsupported SDP media type in offer: image 19128 udptl t38 |
| 18:39.33 | *** join/#asterisk Joe_CoT (n=Joe_CoT@ubuntu/member/joeterranova) |
| 18:40.07 | Joe_CoT | I've confused by the splitup of asterisk-sounds into a bunch of different files. Do I need to download all the different codecs? Is there a single one I can download that covers all of them? |
| 18:41.12 | ricko73 | Joe_CoT: the idea is to use the ones you want. If you're not using gsm or g729, don't download those |
| 18:41.46 | Qwell | Joe_CoT: The defaults that get installed are fine for most people. There are a few reasons you would want different ones... |
| 18:41.57 | Qwell | 1) Higher quality base |
| 18:42.01 | Qwell | 2) Less transcoding |
| 18:44.18 | hiddenman | is it theoretically possible to send fax from the ATA located behind NAT to the fax machine outside? |
| 18:44.34 | jaytee | wow, anyone read the news about Nortel filing for bankruptcy in the U.S.? |
| 18:44.42 | Qwell | jaytee: Nortel who? |
| 18:44.52 | jaytee | Qwell, :-) |
| 18:45.05 | grx0 | lol |
| 18:45.31 | jaytee | I tell 3 people that I'm migrating from Nortel to Asterisk and look what happens! |
| 18:45.58 | *** join/#asterisk lanning (n=lanning@66.151.128.195) |
| 18:46.04 | grx0 | heh... stupid telcom giants and their financial irresponsibility |
| 18:47.38 | lowtek | jaytee: I think that was all that was left (3 people) |
| 18:48.25 | eppigy | TRABAJO |
| 18:49.08 | hiddenman | jaytee: url please |
| 18:49.31 | jaytee | hiddenman, url to what? |
| 18:49.42 | hiddenman | jaytee: to the news about Nortel |
| 18:50.43 | Qwell | hiddenman: news.google.com, q=nortel |
| 18:50.45 | Qwell | take your pick :) |
| 18:50.47 | hiddenman | :) |
| 18:50.59 | jaytee | http://www.msnbc.msn.com/id/28654346/ |
| 18:51.54 | grx0 | I never realized Canada put the Nor in Nortel. |
| 18:52.43 | hiddenman | jaytee: thx |
| 18:54.28 | hiddenman | btw, what are the reasonable settings in the udptl.conf? can't find any docs about them |
| 18:55.28 | hiddenman | i have t38UDPFEC instead of t38UDPRedundancy for error correction, but peer offered me the second one...hmm |
| 18:55.33 | riddlebox | has anyone used heartbeat before? |
| 18:57.10 | *** part/#asterisk ipso (n=ipso@S0106005004c32d38.ok.shawcable.net) |
| 18:57.15 | hiddenman | changing didn't help :( |
| 18:57.52 | *** join/#asterisk SlicerDicer (n=kvirc@24-119-155-26.cpe.cableone.net) |
| 18:59.38 | *** join/#asterisk dlewis (i=c7340d68@about/security/staff/dlewis) |
| 18:59.49 | *** join/#asterisk Alric (n=Alric@masq.hyperusa.com) |
| 18:59.51 | *** join/#asterisk DukeOfURL (n=chatzill@introspect.com) |
| 19:01.19 | [TK]D-Fender | DukeOfURL: Funneh |
| 19:01.26 | *** part/#asterisk dlewis (i=c7340d68@about/security/staff/dlewis) |
| 19:01.36 | DukeOfURL | Hello. I have a new McLeod (PAETEC) PRI, and I'm unable to receive incoming calls. Can anyone tell me if McLeod is NI-2 or DMS100? Thanks |
| 19:03.15 | jaytee | Mcleod would be the ones to answer that question |
| 19:03.34 | DukeOfURL | The tech doesn't know... |
| 19:03.34 | drmessano | Is that like a Delorean? |
| 19:04.05 | DukeOfURL | You have to be a child of the 60s to appreciate my handle. |
| 19:04.11 | [TK]D-Fender | DukeOfURL: Looks like NI2 |
| 19:04.24 | [TK]D-Fender | DukeOfURL: off a 5ESS switch. |
| 19:05.04 | DukeOfURL | NI-2 (national) isn't working. I see the call arrive in PRI debug, but it says channel not available and terminates |
| 19:05.22 | DukeOfURL | it also says "unable to handle ROSE operation" |
| 19:05.29 | [TK]D-Fender | DukeOfURL: PB your configs and debug |
| 19:06.05 | hiddenman | i'm a stupid one. i put t38pt_udptl = yes under the [authentication] section. |
| 19:06.28 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) |
| 19:07.41 | hiddenman | now it works. thank you all, i'm going to kill my self |
| 19:08.18 | pecanha | [TK]D-Fender: I found that the audio of the first playback is playing faster than the caller start to hear the message. Can you suggest something, like increase delay of answer? |
| 19:08.40 | DukeOfURL | http://pastebin.com/d54dc6fa0 |
| 19:08.48 | *** join/#asterisk xnixan (n=xnixan@unaffiliated/xnixan) |
| 19:08.53 | [TK]D-Fender | pecanha: "core show application wait" |
| 19:08.58 | *** join/#asterisk path_ (n=path@249-110-21-190.adsl.terra.cl) |
| 19:09.22 | *** join/#asterisk rue_mohr (n=rue@24.207.122.10) |
| 19:09.27 | rue_mohr | hi all |
| 19:09.41 | DukeOfURL | line 17 is concatenated in PB |
| 19:09.59 | [TK]D-Fender | pecanha: try "ni1" just for fun... |
| 19:10.46 | [TK]D-Fender | DukeOfURL: rather |
| 19:10.52 | [TK]D-Fender | pecanha: strike that |
| 19:10.57 | DukeOfURL | D-Fender: it's the second T1 (span 2) |
| 19:11.26 | rue_mohr | BLI or BLF? |
| 19:12.38 | rue_mohr | er and whats the name of that asterisk wiki? open telephony? |
| 19:12.54 | [TK]D-Fender | ~wikis |
| 19:12.55 | jbot | [~wikis] VoIP Wiki covering Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE. http://www.voip-info.org (c) Arte Marketing Inc / CommPartners |
| 19:13.01 | rue_mohr | thats it |
| 19:13.08 | *** join/#asterisk reneger (n=reneger@dslb-088-078-125-227.pools.arcor-ip.net) |
| 19:13.11 | rue_mohr | the busy lamps, I or F? |
| 19:13.17 | rue_mohr | or am I completely wrong? |
| 19:13.24 | [TK]D-Fender | rue_mohr: BLF. |
| 19:13.27 | *** join/#asterisk Howie69 (n=Owner@host-64-234-5-166.nctv.com) |
| 19:13.30 | rue_mohr | ta |
| 19:13.31 | [TK]D-Fender | ~blf |
| 19:13.32 | jbot | [blf] Busy Lamp Field, aka little lights next to speed dials that light up when the person is on the phone and blink when that line is ringing. hint extensions are static mapped to SIP or other channels. |
| 19:15.04 | rue_mohr | wow, nortels news has been more widespread than anything, i thought they just filed for protection, not bankrupcy? |
| 19:15.44 | [TK]D-Fender | rue_mohr: Chapter 11 : Bankrptcy Protection |
| 19:16.34 | drmessano | Theres no distinction |
| 19:16.51 | rue_mohr | drat, broken link, (http://www.voipproducts.org/articles/article002.html) can you help me find the page on setting up blf? |
| 19:16.53 | drmessano | "Bankruptcy protection" is "BANKRUPTCY" |
| 19:17.02 | rue_mohr | drmessano, oh, heh |
| 19:17.22 | rue_mohr | so that leaves cisco? (and asterisk) |
| 19:17.48 | rue_mohr | there was a great page on blf, I dont know how to find it |
| 19:18.05 | *** join/#asterisk propellerhead (n=yogurt2u@host29.201-253-153.telecom.net.ar) |
| 19:18.21 | drmessano | Cisco, 3Com, and others |
| 19:19.53 | rue_mohr | 3com is doing phone systems? |
| 19:20.19 | rue_mohr | ok, asterisk / blf, webpage, help? |
| 19:20.54 | rue_mohr | http://www.voip-info.org/wiki/view/Asterisk+presence |
| 19:21.01 | rue_mohr | I think thats what I'm looking for actually |
| 19:23.58 | [TK]D-Fender | drmessano: No, is rather distinct. Ch 11 allows them protection from creditor claims during a reoganization to recapitalize so as not to get shut down. |
| 19:24.18 | rue_mohr | I can see why people write scripts for these configs, I'm on to 4? files now with each extension listed in them |
| 19:24.20 | [TK]D-Fender | rue_mohr: 3com has been doing PBX for ages |
| 19:24.54 | rue_mohr | wonders what their systems look like and cost |
| 19:29.56 | *** join/#asterisk davidef1986 (n=davide@2002:4e86:4a47:7b:b83e:7fff:fe8f:6802) |
| 19:29.59 | *** part/#asterisk davidef1986 (n=davide@2002:4e86:4a47:7b:b83e:7fff:fe8f:6802) |
| 19:30.45 | *** join/#asterisk ACK-NAK (n=Miranda@home.chicagoventure.com) |
| 19:32.03 | ACK-NAK | OPINION: Is there a preferred hunting convention? Ascending vs Decending and LIDL vs MIDL? |
| 19:32.29 | ACK-NAK | Seems like MIDL would 'uncover' a bad channel sooner |
| 19:32.37 | ACK-NAK | ...maybe |
| 19:32.55 | rue_mohr | usually see out starting at last line going down, in from first line going up, fax on dedicated or last |
| 19:33.14 | carrar | I use incoming ascending, outgoing decending |
| 19:33.42 | drmessano | I start from the middle, move out, and back in.. Like the lights on the front of KITT |
| 19:33.50 | carrar | haha |
| 19:34.01 | rue_mohr | heh |
| 19:34.10 | carrar | you have to put it in red alarm 1st |
| 19:34.13 | ACK-NAK | drmessano: LOL |
| 19:34.36 | rue_mohr | you have to have an array of systems that you can sequqence alarms on |
| 19:34.36 | drmessano | Nothing wrong with Super Pursuit mode on a PBX |
| 19:34.46 | drmessano | Just hard getting all those protrusions back in |
| 19:34.56 | pecanha | [TK]D-Fender: I already tried with Wait(), but couldn't fix |
| 19:35.11 | rue_mohr | anyone used dial for hold music? |
| 19:35.21 | grx0 | mmmm pbx with weapons integrated mmmmmmm |
| 19:35.26 | rue_mohr | er dtmf... |
| 19:35.30 | ACK-NAK | rue_mohr: ...why fix the fax to a given channel. Is there a theory there? |
| 19:35.39 | [TK]D-Fender | pecanha: because you should ANSWER before you "wait", or start with "playback(silence/3) |
| 19:36.02 | rue_mohr | well, I'm used to the fax being used as the 'if all else fails' outgoing line |
| 19:36.07 | pecanha | [TK]D-Fender: exactly, I did that! |
| 19:36.16 | [TK]D-Fender | pecanha: pastebin... |
| 19:36.16 | *** join/#asterisk dethstar (n=dethstar@unaffiliated/dethstar) |
| 19:36.16 | pecanha | [TK]D-Fender: answer before wait |
| 19:36.29 | ACK-NAK | About the Least-Idle vs Most Idle thing. Is there a best practice? |
| 19:36.39 | carrar | ACK-NAK, for incoming faxes they just come in on any channel |
| 19:36.43 | pecanha | [TK]D-Fender http://pastebin.com/m66aa8376 |
| 19:36.44 | carrar | (for us) |
| 19:38.16 | [TK]D-Fender | pecanha: Do the Silence sample I gave you instead. Also your exten looping like that means your phones can have 5,000,000 mised calls for 1 guy who sites around.. |
| 19:38.37 | [TK]D-Fender | pecanha: also that it rings, then stops for the playbck and they can try to grab onlyt o miss by a second |
| 19:39.26 | pecanha | [TK]D-Fender: do you recommend a hangup instead? or better way to allow user to keep on the line? |
| 19:39.44 | pecanha | trying playback(silence now |
| 19:39.46 | [TK]D-Fender | pecanha: I suggest you go read for a bit and think about this |
| 19:40.20 | pecanha | [TK]D-Fender: :s |
| 19:40.29 | carrar | heh |
| 19:40.51 | Alric | Anyone know how many Digium quad T1 cards your average new Xeon server could support, assuming the T1s weren't very active? |
| 19:41.09 | rue_mohr | ok, now to the aastra phones |
| 19:41.27 | grx0 | I think that has more to do with IRQ issues than the T1 activity |
| 19:41.38 | carrar | Alric, Sagnoma makes 8 pri port cards |
| 19:41.54 | carrar | not sure how many of those you could fit in a single box |
| 19:42.03 | carrar | and work |
| 19:42.04 | pecanha | [TK]D-Fender: playback silence worked well |
| 19:42.23 | Alric | carrar: Interesting, I wasn't aware of those. |
| 19:42.34 | carrar | defintely at least 2 digium quad port cards |
| 19:42.39 | carrar | we have that here |
| 19:43.20 | carrar | need to be aware of whats using your interrupts also |
| 19:43.31 | Alric | It looks like I've got to terminate 13 to 16 T1s, 12-15 of those being station side for FXS ports. |
| 19:43.48 | carrar | Alric, might terminate those on a media gateway |
| 19:43.55 | carrar | then pass sip to asterisk |
| 19:44.16 | carrar | 16 T1's might just bring that in on a DS3 |
| 19:44.34 | Alric | I mean those T1s are going out to analog stations |
| 19:44.40 | carrar | (to a media gateway) |
| 19:44.55 | DukeOfURL | D-Fender: were you able to check http://pastebin.com/d54dc6fa0 ? Thanks |
| 19:45.28 | carrar | You could go from the media gateway to a channel bank also (digital to anaglog) |
| 19:45.33 | carrar | dual purpose |
| 19:45.46 | *** join/#asterisk Greek-B0y (n=greek@41.222.89.77) |
| 19:46.28 | grx0 | group=2channel => 25-47 |
| 19:46.34 | carrar | 15 T1's is a lot of analog lines |
| 19:46.36 | carrar | err 16 |
| 19:47.30 | Alric | It is. |
| 19:47.32 | grx0 | might wanna make that 2 seperate lines in the config Duke...may or may not matter...but doesn't look like it'd be helping anything just in case asterisk didnt like parsing that :) |
| 19:49.07 | *** part/#asterisk Joe_CoT (n=Joe_CoT@ubuntu/member/joeterranova) |
| 19:49.36 | bmoraca | carrar: what kind of endpoints are you using? |
| 19:50.30 | bmoraca | if you are deploying Asterisk in an existing environment that wants to keep their current phones/cabling, you might want to look at the Citel Portico TVA |
| 19:51.00 | *** join/#asterisk CanWood (n=Scott@24.108.64.80) |
| 19:51.16 | [TK]D-Fender | bmoraca: EW... only for digital sets, and even then... EW |
| 19:51.30 | bmoraca | don't like them? |
| 19:52.21 | bmoraca | i've only seen them in demos, never production, but i think it's a decent idea for someone who doesn't want to replace a hundred phones...though they are still kind of expensive |
| 19:52.55 | [TK]D-Fender | bmoraca: those are only for DIGITAL sets... |
| 19:53.02 | bmoraca | i'm aware of that |
| 19:53.03 | EmleyMoor | I will probably tinker a bit more with my N95 as situations crop up |
| 19:53.07 | Corydon76-dig | They're an option for in-house IT, but I would never deploy that as a consultant |
| 19:53.08 | [TK]D-Fender | bmoraca: For analog you're far better off with AudioCodes / Mediatrix |
| 19:53.09 | bmoraca | that's why i asked what kind of endpoints he had |
| 19:53.09 | grx0 | gee I wish I'd seen that before selling about 150 avaya phones last summer :D |
| 19:53.15 | EmleyMoor | (I lost my old settings) |
| 19:53.24 | EmleyMoor | It works at home now |
| 19:53.31 | [TK]D-Fender | bmoraca: You already know.. he said FXS and you know tis hitting channel banks. |
| 19:55.08 | EmleyMoor | It looks like I might have to get a cname set up for the asterisk box - if I can't get back to how it was... |
| 19:56.50 | *** join/#asterisk kannan (n=kannan@121.246.242.95) |
| 19:56.58 | CanWood | Any ActivaTSP users out there? I'm curious to know if anyone has details on setting up Windows to use something other than the default windows "dialer.exe" for hadling calls |
| 19:57.39 | CanWood | Random application -> dialer.exe -> ActivaTSP -> Asterisk -> my phone works just fine but seems a bit convoluted |
| 20:01.36 | *** join/#asterisk nexu (n=nexu@86.85.169.45) |
| 20:08.57 | *** join/#asterisk rue_mohr (n=rue@24.207.122.10) |
| 20:13.19 | CanWood | Random application -> softphone -> ActivaTSP -> Asterisk would be hallier |
| 20:14.56 | *** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
| 20:16.01 | MICKEY17 | Which VoIP hardware works with OpenBSD? |
| 20:16.42 | [TK]D-Fender | MICKEY17: What does "VoIP hardware" have to do with an OPERATING SYSTEM? |
| 20:17.27 | *** join/#asterisk theHub (n=theHub@69.177.93.21) |
| 20:17.44 | MICKEY17 | uhm.. |
| 20:18.21 | DukeOfURL | How can I eliminate the "Really destroying SIP dialog " messages from the console? |
| 20:18.37 | [TK]D-Fender | DukeOfURL: Mod the source. |
| 20:22.56 | *** join/#asterisk angryuser (n=mirggi@90-156-167-83.reverse.alphalink.fr) |
| 20:28.20 | *** join/#asterisk ZX81 (n=matt@202.20.97.211) |
| 20:31.15 | *** join/#asterisk rickross (n=rickross@supporter/active/rickross) |
| 20:31.42 | rickross | darn! I just had two segfaults in a row when ppl tried to use the MeetMe conference bridge |
| 20:33.37 | *** join/#asterisk harry_v (n=pcsuppor@S0106001d7e52cc78.vs.shawcable.net) |
| 20:34.08 | loather-work | haha. people tell me i look like rick ross |
| 20:34.11 | loather-work | but i'm white. |
| 20:34.30 | loather-work | rickross: anyhow. what version of asterisk are you using? |
| 20:34.50 | rickross | loather - 1.6.0.3 on Centos 5.2 |
| 20:35.23 | *** join/#asterisk jpcansa (n=jpbenavi@201.203.102.70) |
| 20:35.31 | loather-work | just sip trunking or are you using isdn pri/bri/etc? |
| 20:35.59 | loather-work | or standard telephony (fxo/fxs) |
| 20:36.18 | rickross | iax trunk to Vitelity |
| 20:36.57 | rickross | I just recompiled and redeployed it this morning with "Optimize" turned off in "make menuselect" because of another segfault that was voicemail-related |
| 20:36.57 | loather-work | ok. in order to have timing for the meetme conference rooms you need dahdi installed for 1.6. |
| 20:37.08 | harry_v | vitelity has been a good provider |
| 20:37.10 | rickross | I believe the conf bridge had been working fine before that |
| 20:37.19 | jpcansa | getting this error when trying to transfer an outside Zap call to an internal ext. any idea "chan_sip.c:13388 attempt_transfer: SIP Transfer attempted with no appropriate bridged calls to transfer" |
| 20:37.29 | loather-work | did you recompile dahdi too? |
| 20:38.10 | harry_v | jp, is this a new install? |
| 20:38.33 | jpcansa | harry_v: no new install |
| 20:38.41 | harry_v | just dahdi issues |
| 20:38.53 | jpcansa | harry_v: i can make the transfer from any other ext. but this |
| 20:39.01 | CrashSys | I seem to be having a problem with Asterisk 1.4.21.2 when using AgentCallbackLogin. After the agent logs in, when the queue tries to use that agent, the system locks up, and I have to do a soft-hangup on the agent channel... |
| 20:39.14 | harry_v | interesting |
| 20:39.29 | harry_v | are thay all the same phones? |
| 20:39.51 | jpcansa | harry_v: yes, all linksys 942 |
| 20:40.26 | harry_v | and you have compared each sip.conf line by line with another one known to work? |
| 20:40.38 | harry_v | or any extention.conf line for each phone. |
| 20:40.41 | jpcansa | harry_v: well actually no, this one is a 962 |
| 20:40.59 | jpcansa | all others are 942 |
| 20:42.04 | harry_v | btw, I am looking for phone alternatives to polycom. Got a quote and well, the 12 polys exceed what another company is giving. Aastra or other. Its for a doctors office. |
| 20:42.57 | harry_v | I dont know about those models. did you look in the phones documentation to see if there was something different compared to the rest of the models? |
| 20:43.57 | *** join/#asterisk ajohnson (n=ajohnson@63.147.46.186) |
| 20:44.01 | lowtek | harry_v: The Polycom's are the best. |
| 20:44.18 | *** join/#asterisk GazD (n=GazD@spc1-stkp5-0-0-cust801.bagu.broadband.ntl.com) |
| 20:44.20 | *** join/#asterisk andy-x_l (i=425c01d1@gateway/web/ajax/mibbit.com/x-bbb43f13a469ae2f) |
| 20:44.22 | CrashSys | Polycom or Snom... best choices around :) |
| 20:45.06 | jpcansa | harry_v: both linksys models are almost the same, 962 has more lines |
| 20:45.24 | *** join/#asterisk qdk (n=qdk@94.191.233.118.bredband.3.dk) |
| 20:45.38 | GazD | silly question... what is an Index telephone systems? |
| 20:45.38 | *** join/#asterisk jshriver (n=jshriver@72.240.39.37) |
| 20:45.42 | jshriver | Greetings |
| 20:45.46 | jshriver | can you use more than 1 digium card in 1 box? |
| 20:45.53 | harry_v | Low, I know I have a polycom but just in the total phone cost along not including the server or 4 port fxo card it exceeded what he was quoted never less programming labor. |
| 20:47.13 | jpcansa | jshriver: yes |
| 20:47.36 | jshriver | ty |
| 20:47.44 | jshriver | anyone here use asterisk under Ubuntu? |
| 20:48.44 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 20:49.52 | *** join/#asterisk geoffl (n=geoff@gjctech.plus.com) |
| 20:51.29 | jpcansa | harry_v: this is what i get in CLI: http://pastebin.com/m68bd4a54 |
| 20:52.58 | rickross | loather - sorry, had some local stuff |
| 20:53.26 | rickross | no, the dahdi did not appear to be dependent on the menuselect of asterisk, but maybe I overlooked that? |
| 20:54.14 | [TK]D-Fender | jshriver: Many have, move along to your actual question |
| 20:55.31 | jpcansa | Hi Fender, do you have any idea why i got this issue transfering external calls? |
| 20:56.06 | jpcansa | some times it works, sometimes it just hung up the Zap channel |
| 20:56.47 | daniev | hello. i saw your comments about polycom |
| 20:57.24 | daniev | actually i have two of them and i'm very happy. but i must to buy other phones for my company |
| 20:57.30 | [TK]D-Fender | jpcansa: Looks like you're trying an attended transfer when the other side hasn't picked up yet |
| 20:57.50 | daniev | i've been thinking about cisco, but ... what do you think ??? |
| 20:58.03 | daniev | at least two phones per comparison |
| 20:58.35 | [TK]D-Fender | daniev: where are you located? |
| 20:59.01 | daniev | [TK]D-Fender: in colombia |
| 20:59.51 | daniev | i heard about some licenses or something ... but i haven't a clear idea about that |
| 20:59.59 | [TK]D-Fender | daniev: Porbably should bother looking at anything else. Linksys is an economical option if there is a hard import premium. |
| 21:00.04 | kannan | ChanSpy on a SIP Exten, is giving internittent voice. The Baring In party is on ZAP and so is the Other leg of the SIP CHannel. The SIP phone is within the LAN and that conversation is crystal clear. How to start troubleshooting this one? |
| 21:00.11 | [TK]D-Fender | daniev: Cisco is costly and troublesome |
| 21:01.07 | daniev | [TK]D-Fender : troublesome ? that would be a reason for not buy. I know is costly, but quality costs |
| 21:01.26 | daniev | however i don't want troublesome with my asterisk |
| 21:02.01 | *** part/#asterisk geoffl (n=geoff@gjctech.plus.com) |
| 21:02.23 | [TK]D-Fender | daniev: Polycom > All |
| 21:02.51 | daniev | ok. that's what i need to know. thanks [TK]D-Fender |
| 21:07.18 | daniev | [TK]D-Fender : i know that grandstream are budget phones. But is it a decent option for employees that don't need more else? |
| 21:07.41 | *** join/#asterisk telecos (n=sergio@34.167.219.87.dynamic.jazztel.es) |
| 21:07.44 | CanWood | daniev: we run'em |
| 21:07.52 | CanWood | they do the job for us |
| 21:08.03 | *** join/#asterisk geoffl (n=geoff@gjctech.plus.com) |
| 21:09.38 | [TK]D-Fender | daniev: ... |
| 21:09.40 | [TK]D-Fender | ~gs |
| 21:09.40 | jbot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
| 21:09.41 | *** part/#asterisk Deeewayne (n=Deeewayn@nat/digium/x-dfb56a229ccba6d2) |
| 21:09.43 | [TK]D-Fender | ~grandstream |
| 21:09.44 | jbot | hmm... grandstream is the Yugo of VoIP hardware. Run. Run away now. |
| 21:09.53 | CanWood | heh |
| 21:09.55 | *** join/#asterisk etfonhomey (n=chatzill@74-143-192-75.static.insightbb.com) |
| 21:09.57 | *** join/#asterisk Deeewayne (n=Deeewayn@nat/digium/x-dfb56a229ccba6d2) |
| 21:09.57 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 21:11.36 | *** join/#asterisk zerko (n=zerko@208.90.225.221) |
| 21:12.18 | CrashSys | Yeah, but what if it's a yugo Convertible? |
| 21:12.26 | daniev | CanWood : ok. thanks for the info |
| 21:12.34 | daniev | and to [TK]D-Fender too |
| 21:12.53 | daniev | grandsuck hehehe |
| 21:13.06 | CanWood | NP. We've had issues, bu no show stoppers. Sounds like we've just been lucky |
| 21:15.37 | *** join/#asterisk chi6IT41 (n=chigital@tmo-100-101.customers.d1-online.com) |
| 21:24.28 | *** join/#asterisk dodver (n=dodver@mask1.vekoll.vein.hu) |
| 21:25.15 | dodver | hello for everyone |
| 21:26.23 | *** join/#asterisk pfn (n=pfnguyen@hanhuy.com) |
| 21:26.28 | *** part/#asterisk pfn (n=pfnguyen@hanhuy.com) |
| 21:26.33 | *** join/#asterisk pfn (n=pfnguyen@hanhuy.com) |
| 21:31.26 | *** join/#asterisk bbryant (n=brett@c-68-59-20-153.hsd1.sc.comcast.net) |
| 21:32.05 | kerx | hi if i use g.711 and 50 agents, how much voip traffic is required? |
| 21:34.39 | [TK]D-Fender | kerx: depends how many calls. Depends what protocols. |
| 21:35.07 | kerx | 50 calls simultaneous |
| 21:35.10 | rickross | hmm, I think I found something connected to my MeetMe crash |
| 21:35.30 | rickross | [Jan 14 16:06:19] WARNING[18644] app_meetme.c: No DAHDI channel available for conference, conference recording disabled (is chan_dahdi loaded?) |
| 21:35.41 | kerx | [TK]D-Fender, I'm getting like 3.2megs using G.726 |
| 21:35.49 | kerx | w/ out payload/overhead, etc. |
| 21:36.04 | jaytee | BULLETIN: Ricardo Montalban has died at 88. "Khaaaaaaannnnn!!!!!" |
| 21:37.36 | drmessano | DA PLANE |
| 21:37.50 | harry_v | ohh really |
| 21:37.56 | rickross | Dere's a PLANE? |
| 21:38.01 | harry_v | he was a great actos |
| 21:38.06 | harry_v | actor |
| 21:38.18 | rickross | ahhh, that little guy - Herve someone |
| 21:38.29 | harry_v | That was the age my great uncle, a well known stuntman passed on. |
| 21:38.40 | drmessano | Herve Villanachonuevediaz |
| 21:38.41 | drmessano | I think |
| 21:39.07 | drmessano | Not to be confused with Fernando Valenzuela |
| 21:40.27 | rue_mohr | check_auth: username mismatch, have <12b>, digest has <12d> |
| 21:40.27 | rue_mohr | <PROTECTED> |
| 21:41.00 | rue_mohr | can I clear its digester? |
| 21:41.44 | rue_mohr | wonder if I goofed the config files |
| 21:41.50 | *** join/#asterisk Ericounet (n=Ericoune@ACaen-151-1-9-29.w86-215.abo.wanadoo.fr) |
| 21:41.58 | rue_mohr | yup |
| 21:44.27 | *** join/#asterisk Ericounet (n=Ericoune@ACaen-151-1-9-29.w86-215.abo.wanadoo.fr) |
| 21:44.40 | rickross | if I am only using dahdi's dummy driver, then is chan_dahdi.conf.template all I need for an actual chan_dahdi.conf? |
| 21:45.11 | rue_mohr | rickross, |
| 21:45.21 | [TK]D-Fender | rickross: Basically a pretty much empty files with a [channels] header should do it |
| 21:45.29 | [TK]D-Fender | rickand no declared channel |
| 21:45.34 | rickross | thank you |
| 21:45.51 | rickross | would NOT having this file in place prevent chan_dahdi from loading? |
| 21:46.37 | *** join/#asterisk freakazoid0223 (n=matt@pool-71-242-212-37.phlapa.east.verizon.net) |
| 21:46.41 | *** join/#asterisk clive- (i=ident@dsl-242-146-16.telkomadsl.co.za) |
| 21:47.13 | [TK]D-Fender | rickross: Yes. |
| 21:47.16 | clive- | I have a question... how do I download an old version of asterisk 1.2.17 ? |
| 21:47.28 | [TK]D-Fender | clive-: www.asterisk.org |
| 21:47.46 | *** join/#asterisk CanWood (n=Scott@24.108.64.80) |
| 21:47.51 | clive- | D-Fender thanks, let me go look |
| 21:49.03 | *** join/#asterisk nix8n82 (n=nate@63.162.26.149) |
| 21:51.01 | jaytee | drmessano, his name is Herve Villechaize. He was voted the World's Sexiest Man by the readers of Midget Magazine in 1982. |
| 21:51.03 | *** join/#asterisk kisu (n=kisu@2001:5c0:1100:9900:8453:e522:acd1:99dc) |
| 21:51.45 | drmessano | Well, excuse the eff out of me |
| 21:51.57 | drmessano | You wikislut |
| 21:51.58 | *** join/#asterisk voxter (n=voxter@76.77.95.2) |
| 21:52.24 | rue_mohr | grr aastra is givin me gears |
| 21:54.29 | *** part/#asterisk CanWood (n=Scott@24.108.64.80) |
| 21:57.27 | harry_v | [TK]D-Fender how would you compare snom and linksys phones? |
| 21:57.59 | [TK]D-Fender | harry_v: Very little case where I'd ever suggest Snom. |
| 21:58.03 | harry_v | cost wise is a concern |
| 21:58.19 | [TK]D-Fender | harry_v: Linksys if no need for PoE and esp if not in NA |
| 21:58.34 | harry_v | NA? |
| 21:58.39 | [TK]D-Fender | North America |
| 21:58.48 | *** join/#asterisk uwe (n=uwe@a18-90.adsl.paltel.net) |
| 21:58.59 | harry_v | okay will consider that when doing my site suveys |
| 21:59.20 | [TK]D-Fender | harry_v: quick answer is hardly worth thinking about either |
| 21:59.26 | harry_v | true |
| 21:59.54 | uwe | hello, the digium AEX2406E supports up to 24 lines, but only 6 modules, what does this mean ? can i have 24 concurrent calls ?? or only 6 ? |
| 22:00.46 | [TK]D-Fender | uwe: Each module supports 4 ports |
| 22:00.58 | drmessano | Linksys has PoE |
| 22:01.24 | harry_v | Steve Jobs to take leave till end of june. |
| 22:01.31 | [TK]D-Fender | drmessano: I know, just that Polcyom doesn't come with the brick if you aren't planning on PoE, and the Linksys come with it regardless. |
| 22:01.40 | drmessano | Gotcha |
| 22:01.44 | [TK]D-Fender | drmessano: So non PoE swings a few more $ in Linksys favour |
| 22:02.02 | uwe | thank you [TK]D-Fender ! |
| 22:02.30 | drmessano | yeah |
| 22:05.28 | [TK]D-Fender | drmessano: Linksys SPA941 = $86 at TD. So for non PoE, thats 4 lines fairly decent phone w/ switch port. |
| 22:05.49 | harry_v | snoms are not half bad looking phones |
| 22:06.02 | drmessano | 941 = No switch port |
| 22:06.06 | drmessano | Same for 921 |
| 22:06.16 | drmessano | Only the PoE phones have the extra port |
| 22:06.37 | [TK]D-Fender | harry_v: Shit use of display, more costly for the same level of lines, a notch below on audio quality, and a history of iffy firmware |
| 22:06.46 | [TK]D-Fender | drmessano: 941 had a passthrough... |
| 22:06.58 | drmessano | Not the 10 I have lol |
| 22:07.00 | [TK]D-Fender | IIRC |
| 22:07.13 | [TK]D-Fender | drmessano: Ok, a point against :) |
| 22:07.14 | harry_v | Tk, I do not want reliability issues with any of the phones |
| 22:07.17 | drmessano | Theres a knockout there with nothing behind it |
| 22:07.42 | [TK]D-Fender | drmessano: I see it now... its been a while... I don't particularly miss my old one :) |
| 22:08.04 | drmessano | lol |
| 22:08.17 | drmessano | They're not bad phones.. and the cheaper they get, the more the value increases for them |
| 22:08.29 | drmessano | Especially the like-new ones I have gotten for $35 ea |
| 22:08.48 | drmessano | They're not $300 phones by an means |
| 22:08.50 | drmessano | any* |
| 22:09.01 | uwe | um, how well will using 2 such cards (AEX2406E) work ? i mean is there any info about cpu interruption , or hints on hardware specifications ? |
| 22:09.17 | rickross | is there something about MeetMe that requires specifically to be configured to use dahdi on 1.6.x? |
| 22:09.37 | lowtek | Just in: Steve Jobs taking medical leave, quick sell your Apple Stock. |
| 22:09.42 | [TK]D-Fender | rickross: No. |
| 22:09.51 | [TK]D-Fender | uwe: EW |
| 22:10.01 | drmessano | lowtek: KNOCK IT OFF.. EVERYONE IS GONNA SELL IT ALREADY |
| 22:10.03 | [TK]D-Fender | uwe: WTF are are you looking at that for? |
| 22:10.07 | drmessano | heh |
| 22:10.13 | lowtek | LOL, not me, buy buy buy |
| 22:10.19 | rickross | I've core dumped two separate 1.6.0.3 installations by trying to use MeetMe :( |
| 22:10.31 | rickross | it dumps when the second party connects to the bridge |
| 22:10.48 | [TK]D-Fender | rickross: I've heard many crash issues with 1.6.0.3 this amongst them IIRC. Try rolling back a version |
| 22:10.56 | uwe | [TK]D-Fender, 48 analog lines :) |
| 22:10.58 | drmessano | Same here |
| 22:11.02 | drmessano | 48 WHAT |
| 22:11.05 | [TK]D-Fender | uwe: PRi <- |
| 22:11.21 | uwe | no , POTS |
| 22:11.29 | uwe | err |
| 22:11.43 | uwe | [TK]D-Fender, for the internal side |
| 22:11.55 | eppigy | hello |
| 22:12.36 | harry_v | rickross, some times it is just easier to downgrade if you absolutly cannot find a answer. |
| 22:12.43 | rickross | :) |
| 22:13.02 | uwe | i know this is stuped, i should be using Ip telephony internally , but im trying to do comparison with "reusing" existing infrastructure |
| 22:13.19 | rickross | yep, or maybe just slit my wrists and go to the latest svn version :) |
| 22:13.24 | harry_v | existing in what context? |
| 22:13.25 | rickross | to see if it fixes |
| 22:13.36 | drmessano | rickross: Pointless, 1.6.0.2 works |
| 22:13.49 | [TK]D-Fender | uwe: Screw analog, that is retarded |
| 22:13.51 | rickross | ok, I'll give 1.6.0.2 a shot |
| 22:13.57 | [TK]D-Fender | uwe: and PRI != VoIP |
| 22:14.09 | uwe | i know ! |
| 22:14.10 | harry_v | Rick, I gave up on some of the wierd issues of 1.6 and just rolled back. |
| 22:14.14 | [TK]D-Fender | uwe: 48 analog = crazy |
| 22:14.15 | rickross | we hadn't moed to 1.6 until the 1.6.0.3, but maybe its better not to be on this one |
| 22:14.53 | rickross | harry, we had a similar experience, and went back to 1.4.x - but we really like the G.722 and would like to use it for internal comm |
| 22:15.14 | rickross | we had backported it to 1.4.x, but that's a pain, too |
| 22:15.17 | harry_v | guess if you like that codec |
| 22:15.21 | [TK]D-Fender | rickross: What phones? |
| 22:15.24 | drmessano | 1.6 is fine |
| 22:15.27 | rickross | Polycom 550 |
| 22:15.29 | drmessano | You jsut got a bad version |
| 22:15.39 | rickross | drmessano - I'll build 1.6.0.2 |
| 22:15.43 | [TK]D-Fender | rickross: Good phones... pricy and hard to recommend though. |
| 22:15.44 | drmessano | good |
| 22:15.56 | drmessano | and stop listening to "Well, 1.6 blah.. so I am on 1.4 now" |
| 22:16.01 | drmessano | Those people JUST upgraded from 1.2 |
| 22:16.06 | *** join/#asterisk propellerhead (n=yogurt2u@host90.190-136-214.telecom.net.ar) |
| 22:16.30 | clive- | I am proudly on 1.2 still :) |
| 22:16.53 | Alric | Someone mocking 1.2? ;) |
| 22:16.58 | uwe | ok :), so its not applicable :), [TK]D-Fender , i basically trying to estimate prise on diffrent combination ... but i dont want to put "unapplicable" combinations, reusing existing stuff is not my idea in the first place |
| 22:17.00 | drmessano | Yes, it is I |
| 22:17.03 | [TK]D-Fender | clive-: And an OLD version of that series at that. |
| 22:17.29 | drmessano | 1.2 is good for like, running an oldies radio station, or a drive in theatre.. or like an old peoples home |
| 22:17.33 | drmessano | Get the PATTERN? |
| 22:18.09 | eppigy | fractal patterns |
| 22:18.37 | *** join/#asterisk path_ (n=path@pc-15-190-86-200.cm.vtr.net) |
| 22:18.47 | harry_v | at what point did DigitTimeout change on the versions? |
| 22:19.01 | Alric | Technically 1.4, afaik. |
| 22:19.13 | *** join/#asterisk strk (n=strk@ip-123-78.static.adsl.cheapnet.it) |
| 22:19.25 | drmessano | I noticed that using rocks and telegraph wire was deprecated in 1.2 |
| 22:19.28 | drmessano | :( |
| 22:19.33 | drmessano | Guess I wont be upgrading |
| 22:19.42 | Alric | Most of my dialplans are still from 1.0 or 1.2... |
| 22:20.36 | clive- | D-Fender, . the new version of 1.2 they messed up some iax2 trunking stuff, and I was told that they are not fixing any 1.2 bugs anymore ..:( |
| 22:21.02 | drmessano | Nope |
| 22:21.14 | [TK]D-Fender | clive-: its been that way for like 2 YEARS now. |
| 22:21.15 | harry_v | drmessano :) I did that as a kid |
| 22:21.43 | [TK]D-Fender | harry_v: 1.2 <- |
| 22:21.47 | drmessano | harry_v: I built a crystal set PBX using a 1N34A and Asterisk 1.2 |
| 22:21.47 | harry_v | :) |
| 22:21.53 | drmessano | Oh the good ole days |
| 22:21.58 | rue_mohr | [TK]D-Fender, I have the 6th line key with a keypad icon, its from the default configs, I cant find any reference to it in the current configs, know where its comming from? |
| 22:22.06 | [TK]D-Fender | drmessano: Nostalgia isn't what it used to be... |
| 22:22.08 | rue_mohr | cant find anything about it in the manual |
| 22:22.18 | harry_v | In the 70s, played with radio shack 200 in one project kit |
| 22:22.27 | drmessano | I still have mine |
| 22:22.31 | [TK]D-Fender | rue_mohr: Contat directory entry spilling onto an unused line key |
| 22:22.34 | drmessano | I got mine when I was 4 |
| 22:22.36 | harry_v | dam nice |
| 22:22.37 | drmessano | lol |
| 22:22.54 | harry_v | 70s was a great time for evolving technoligy. |
| 22:23.00 | clive- | d-fender , which version handles iax2 trunking nicely? |
| 22:23.03 | rue_mohr | ok, if its not int eh contact dir anymore its comming from the phone? can I tell it to forgot it? |
| 22:23.13 | drmessano | My dad stripped hundreds of feet of old telephone wire so I could have jumpers for the projects |
| 22:23.20 | harry_v | Mr tv repair man would always visit to replace the tubes. |
| 22:23.25 | [TK]D-Fender | rue_mohr: Go into the directory and DELETE it |
| 22:23.36 | drmessano | The Wheel of Fortune with the spinning 4 LEDs on the LED display got me |
| 22:23.37 | rue_mohr | 000000000-directory.xml? |
| 22:23.39 | drmessano | Like 70 steps |
| 22:23.41 | rue_mohr | its not there anymore |
| 22:23.51 | rue_mohr | hasn't been for many reboots |
| 22:23.52 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) |
| 22:24.15 | [TK]D-Fender | rue_mohr: ON THE DAMN PHONE |
| 22:24.20 | rue_mohr | hmm |
| 22:24.31 | rue_mohr | still going over the manuals again :) |
| 22:24.31 | harry_v | ohh and the good one, whole familly including uncles would suround the kitchen table watching my dad solder the very first of radar detectors. It was such exotic technoligy then. |
| 22:24.39 | [TK]D-Fender | rue_mohr: And your phone isn't LOOKING at 000000-directory.xml <- |
| 22:25.16 | rue_mohr | aha! I found the delete key for them :) |
| 22:25.29 | rue_mohr | thanks for confirming it was stuck on the phone |
| 22:25.43 | harry_v | listening to cross talk |
| 22:26.19 | harry_v | as a kid I always wondered why I would hear neighboors talking and thay would not hear me :) |
| 22:27.10 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.static.bezeqint.net) |
| 22:27.17 | harry_v | btw, is there a loop back number I can dial? dont know where my cell is at the moment and need to make a inbound call to test my *. |
| 22:30.21 | clive- | are many people using 1.4 in production boxes? |
| 22:30.35 | Alric | raises hand. |
| 22:30.46 | rue_mohr | I'm trying |
| 22:30.59 | clive- | Alric, how is the crash/stability on that ? |
| 22:31.02 | [TK]D-Fender | clive-: Probably just about the majority of * users... |
| 22:31.13 | rue_mohr | so far all my problems have been my own |
| 22:31.26 | rue_mohr | ask [TK]D-Fender |
| 22:31.27 | Alric | I can't remember the last time I had a 1.4 box crash |
| 22:31.31 | [TK]D-Fender | clive-: Like running Windows 3.11 much? |
| 22:31.43 | clive- | D-Fender:), how did you guess? |
| 22:31.57 | Alric | I guess it was sometime last September... when the box blew up and fried its MB, HDs, and probably everything else inside :) |
| 22:32.01 | [TK]D-Fender | clive-: sigh kick |
| 22:33.05 | clive- | my issue is that I have a sangoma card, and I am not sure how to get that that play with 1.4... and I dont have much time to play, besides like at 2am |
| 22:34.04 | Alric | I've never messed with Sangoma hardware. |
| 22:34.29 | [TK]D-Fender | clive-: So really... there is nothing wrong. |
| 22:34.35 | drmessano | I will have to agree with rue_mohr here |
| 22:34.36 | [TK]D-Fender | clive-: www.drphil.com |
| 22:34.42 | drmessano | He is indeed an idiot and messes crap up |
| 22:35.11 | rue_mohr | I'm getting the hang of it |
| 22:35.38 | clive- | D-Fender, no there is a serious memory leak that is causing wacky iax2 trunking behaviour on 1.2.30 |
| 22:36.02 | rue_mohr | tries to reech the rope tangled around the leg hes hanging by... |
| 22:36.38 | [TK]D-Fender | clive-: ... |
| 22:37.17 | rue_mohr | hahaha the config file specifies line1 3 times, that explains things |
| 22:38.39 | clive- | D-Fender , so you are trying to tell me 1.4 will work fine, without giving false busy signals and all the other funny wacko stuff thats been hapenning |
| 22:39.04 | rickross | ok, whatever is crashing MeetMe in 1.6.0.3 is still crashing it in the latest svn |
| 22:39.09 | rickross | now to test 1.6.0.2 |
| 22:40.10 | rickross | byw, is it best to roll back on dahdi and asterisk-addons, too? Or is it sufficient to just get 1.6.0.2, build it, and recompile the 1.6.0.1 addons? |
| 22:40.31 | *** join/#asterisk angryuser (n=mirggi@90-156-167-83.reverse.alphalink.fr) |
| 22:41.43 | [TK]D-Fender | clive-: Feel free to sit paranoid in the past... |
| 22:42.04 | drmessano | 1.6 is even better |
| 22:42.18 | drmessano | Despite 1.6.0.3 seemingly a bad build |
| 22:42.27 | drmessano | This far the others have been solid |
| 22:42.46 | *** join/#asterisk TOrrIeri (n=torrieri@nelug/crew/torrieri) |
| 22:45.22 | clive- | hi D-fender,...lets just jump in here.... Whats this dhadi stuff?,, do I need the "tools" , or the dhadi linux, or both? |
| 22:45.35 | [TK]D-Fender | ~dahdi |
| 22:45.36 | jbot | [~dahdi] Digium/Asterisk Hardware Device Interface (DAhdi). The new name of zaptel More info at http://www.asterisk.org/zaptel-to-dahdi , and is pronounced "dah-dee" with a short A, or pronounced like http://www.russellbryant.net/dahdi.wav |
| 22:45.40 | *** join/#asterisk C4away (n=DJpyro@66.185.111.33) |
| 22:46.02 | C4away | how do I retain the src field in the CDR while blocking outbound callerid? |
| 22:46.16 | C4away | there is some header, is that the P-Asserted-Identity header? |
| 22:46.48 | C4away | the final device that shoves the call onto the PSTN is a large audiocodes media gateway |
| 22:46.51 | rue_mohr | hmm wonder why aastra defaults rtp to 3000 |
| 22:47.10 | C4away | I wonder moreso why asterisk defaults to 10000-20000 |
| 22:47.52 | rue_mohr | I wonder what aastra deems a basic codec, and if I shoudl be worried its default off |
| 22:48.30 | [TK]D-Fender | rue_mohr: I wonder if I actually need to keep my eyes open when I get onto the highway... think anything bad might happen? |
| 22:49.01 | rue_mohr | well I didn't see what they consider a basic codec, is that ulaw/alaw!? |
| 22:49.20 | rue_mohr | or is it something obscure we never hear about |
| 22:49.39 | [TK]D-Fender | rue_mohr: rue_mohr wtf does "basic codec" mean? What is a "non-basic codec"? What is the functional difference of this inventive little term of yours? |
| 22:49.46 | rue_mohr | exactly |
| 22:50.08 | rue_mohr | thats what they say in the 33i manual "enable basic codecs" |
| 22:50.11 | rue_mohr | I dunno |
| 22:51.12 | C4away | rue_mohr I belive they mean "u/alaw and g.729" according to the web gui |
| 22:51.16 | C4away | if I remember correctly |
| 22:51.28 | rue_mohr | is that a baisc codec? they say more on the ui? |
| 22:51.31 | *** join/#asterisk ecret (n=ecret@CPE001e9002348e-CM001225d8ab30.cpe.net.cable.rogers.com) |
| 22:51.32 | C4away | which is interesting because I don't think they do any other codecs |
| 22:51.47 | [TK]D-Fender | 33i? EW |
| 22:52.25 | rue_mohr | [TK]D-Fender, well, for 2 lines we thought we would save money and not go with the ip601 |
| 22:52.31 | C4away | Basic Codecs(G.711 u-Law, G.711 a-Law, G.729) |
| 22:52.38 | C4away | that's copy/paste right off the web gui |
| 22:52.42 | [TK]D-Fender | rue_mohr: 33i = $150. IP320 = $87 |
| 22:52.45 | rue_mohr | basides, gives us a chance to check out different brands |
| 22:53.03 | C4away | which i think is also the "advanced" codecs and the "complete list of codecs" at the same time |
| 22:53.10 | rue_mohr | we paid $180~ for the 601's |
| 22:53.30 | rue_mohr | and ~$80 for the 33i |
| 22:53.46 | rue_mohr | er 120, dont recall |
| 22:57.35 | C4away | yea, the 33i is pretty damn basic |
| 23:00.20 | *** join/#asterisk etfonhomey (n=chatzill@74-143-192-75.static.insightbb.com) |
| 23:00.55 | C4away | to be honest the 9133i is a much better phone, the only reason the 33i would ever be used in my opinion is if you want the break room or waiting room phone to look consistant with the other phones |
| 23:01.20 | C4away | I think it's pretty much a waste of Aastra's R&D and marketing budget |
| 23:01.53 | harry_v | I have a really strange issue going on with fire recording not existing in /var/lib/asterisk/spounds. Seems when I have somone call to make a test call the cli goes into a rapid closed loop saying that a sound file does not exist while the call is being processed by the incomming start ivr. www.pastebin.ca/1308494. I looked at the gsm sound file and it does exist and it does have a byte count on it. |
| 23:10.06 | *** join/#asterisk andy-x_l (i=4009f600@gateway/web/ajax/mibbit.com/x-21a8b326768746bb) |
| 23:10.31 | [TK]D-Fender | harry_v: Nothing wierd at all... |
| 23:11.03 | [TK]D-Fender | harry_v: -- Executing [s@default:4] BackGround("DAHDI/1-1", "ivr-greeting.gsm") in new stack <- bad |
| 23:11.19 | rickross | ugh - can't build 1.6.0.2 - http://www.mail-archive.com/asterisk-users@lists.digium.com/msg216222.html |
| 23:11.20 | [TK]D-Fender | harry_v: -- Executing [8501@internal:5] Playback("SIP/200-08a59d70", "/tmp/ivr-greeting") in new stack <- Good |
| 23:12.01 | drmessano | 1.6.0.1? |
| 23:13.13 | rickross | drm - did you build 1.6.0.2 successfully? |
| 23:13.21 | rickross | I wonder if anyone did? |
| 23:13.33 | rickross | seems to have a 1.4.xx regression |
| 23:14.14 | rickross | ok, 1.6.0.1 - here I come |
| 23:14.28 | harry_v | yea i see |
| 23:15.06 | harry_v | by default Background(thissound.gsm) reads from /tmp/ |
| 23:15.38 | harry_v | unless otherwise, i put in a diferent path such as /var/lib/asterisk/sounds |
| 23:15.56 | *** part/#asterisk kb3ien (n=kb3ien@isl177-max1.accesshighway.net) |
| 23:16.20 | harry_v | wait yea I see |
| 23:16.27 | harry_v | no path specified. |
| 23:16.54 | rickross | ahh - the PATCH! |
| 23:16.56 | rickross | http://bugs.digium.com/bug_view_page.php?bug_id=14010&history=1 |
| 23:17.10 | [TK]D-Fender | harry_v: No, you have forgotten the all-important "don't add the f-ing extension no the end" rule <- |
| 23:17.15 | [TK]D-Fender | on* |
| 23:18.05 | [TK]D-Fender | And on that note... time to head home.... |
| 23:18.17 | [TK]D-Fender | bbiab |
| 23:19.22 | harry_v | he must live close to his house |
| 23:19.23 | harry_v | :) |
| 23:20.22 | eppigy | WHAT |
| 23:20.30 | rue_mohr | something about the dialplan causes it to lock up |
| 23:20.35 | rue_mohr | the aastra 33i |
| 23:20.56 | rue_mohr | I can reboot it, thats it |
| 23:21.44 | C4away | remote or local provisioning? |
| 23:21.56 | C4away | and is it connected to the internets? |
| 23:21.56 | rue_mohr | ftp |
| 23:22.01 | rue_mohr | local |
| 23:22.07 | rue_mohr | local * |
| 23:22.12 | C4away | is it ftp or local? |
| 23:22.15 | C4away | or local network? |
| 23:22.25 | C4away | one word doesn't really answer my questions unabiguously |
| 23:22.25 | rue_mohr | heh local network, config via ftp |
| 23:22.35 | C4away | there is no access to the internet at all? |
| 23:22.46 | rue_mohr | there is why? |
| 23:23.21 | C4away | so the phone is able to connect to the internet? |
| 23:23.28 | C4away | once it gets an IP |
| 23:23.59 | rue_mohr | its on the network, it downloads the config files, and then proceeds to ignore them |
| 23:24.16 | drmessano | Bad format |
| 23:24.20 | drmessano | Bad XML |
| 23:24.22 | rue_mohr | well, it also says no format |
| 23:24.24 | C4away | this is unconfirmed but someone said if the phone can't ping aastra on the first bootup (default config) it will hang |
| 23:24.31 | rue_mohr | the aastra config isn't xml |
| 23:24.32 | C4away | aastra isn't xml |
| 23:24.41 | drmessano | Ok, bad config period |
| 23:24.43 | C4away | but same concept applies |
| 23:24.47 | drmessano | If it download and does nothing, bad config |
| 23:25.06 | rue_mohr | it can reech the * server, it downloads the configs form it |
| 23:25.20 | drmessano | Yes, and the configs are bad |
| 23:25.40 | C4away | check for typos |
| 23:25.48 | rue_mohr | I'm gonna have to set up a log server to find out :( |
| 23:25.52 | C4away | and provisioning keys that are not valid for that model |
| 23:26.16 | C4away | aastra has different parameters that are available in different models |
| 23:26.29 | C4away | prgkey vs softkey, etc |
| 23:26.34 | rue_mohr | I'v had to build up the config from scratch |
| 23:26.36 | C4away | not all models share these |
| 23:26.46 | C4away | even more chance of missing something |
| 23:26.48 | rue_mohr | I'v got the manual, all 1200 pages of it |
| 23:26.53 | rue_mohr | its been working |
| 23:27.16 | strk | is reading the asterisk book and would like to tank anyone involved (in both the book and the tech) |
| 23:27.17 | rue_mohr | with debug 10 verbose 10 I should see it log in to * right? |
| 23:27.45 | rue_mohr | strk, the oriley one? |
| 23:27.58 | frogonwheels | strk: presumably you meant thank.. ;) |
| 23:28.47 | rue_mohr | I think he meant tank |
| 23:28.54 | rue_mohr | aka clobber |
| 23:30.10 | strk | rue_mohr: yes, and THANKS :) |
| 23:30.30 | strk | I'm feeling this revolution need very highly |
| 23:30.39 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 23:30.45 | rue_mohr | I'm amused that the mac address for polycom is lowercase, aastra is uppercase, and linux is case sensitive |
| 23:31.03 | rickross | OK!!! Yes! 1.6.0.2 compiled, installed and NOT CRASHING MeetMe! |
| 23:31.06 | rue_mohr | good thing I have an ftp log |
| 23:31.06 | strk | far too many friends keeps boring me with "skype" while I know what SIP is and I also keep fighing with my sip-based-but-commercially-closed provider |
| 23:31.16 | strk | I'll be likely setting up an asterisk instance soon |
| 23:31.17 | rickross | good call, drm - thx |
| 23:31.39 | rue_mohr | ok, shoudl I see the aastra phone log in when it boots? |
| 23:31.59 | *** join/#asterisk arpu (n=arpu@chello062178159144.10.14.univie.teleweb.at) |
| 23:32.46 | drmessano | oh? |
| 23:36.14 | rue_mohr | k, I'm tryingout sip set debug 192.168... |
| 23:36.31 | rue_mohr | I really really shoudl see th phone talk to * shouln't I? |
| 23:37.10 | rue_mohr | and why is it saying retrying config download |
| 23:37.11 | rue_mohr | ugh |
| 23:37.19 | rue_mohr | how th hell am I supposed to debug this!? |
| 23:39.17 | bmoraca | rue_mohr: does the phone download from TFTP? |
| 23:39.20 | rue_mohr | ftp |
| 23:39.27 | rue_mohr | its downloading them |
| 23:39.33 | rue_mohr | so says the ftp server |
| 23:39.36 | bmoraca | you've verified that in the FTP log? |
| 23:39.38 | bmoraca | ok |
| 23:39.41 | rue_mohr | yea |
| 23:39.43 | bmoraca | what kind of phone is it? |
| 23:39.49 | rue_mohr | aatra i33 |
| 23:39.58 | rue_mohr | 9143 if you like |
| 23:39.59 | bmoraca | i've never liked aastra phones. |
| 23:40.11 | rue_mohr | well I ahve to say the documentation is really good |
| 23:40.21 | rue_mohr | if I had a log server I could point it at it |
| 23:40.22 | bmoraca | the phones still suck :) |
| 23:40.54 | bmoraca | does the phone ever boot completely? |
| 23:41.21 | bmoraca | also, have you tried logging in to the phone's web interface and setting it up manually to verify connectivity? |
| 23:41.27 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194) |
| 23:42.25 | rue_mohr | yes, but it also says -no service- |
| 23:42.57 | rue_mohr | I want to stay away from the web interfaces cause they tend to cause settings to get stored in the phone and cause "magic" |
| 23:43.04 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 23:43.06 | drmessano | .........ROFL |
| 23:43.17 | C4away | rue_mohr how many phones are you configuring? |
| 23:43.18 | rue_mohr | what I WANT is a config file with every paramiter for the phone set to its default value |
| 23:43.20 | drmessano | ZOMG A UNICORN |
| 23:43.27 | rue_mohr | 2 of the aastras |
| 23:43.50 | C4away | my rule is if I'm configuring less than 10 phones I don't waste time with the remote provisioning |
| 23:44.09 | C4away | if I'm configuring more than 20 I don't waste time programming each phone through the web gui |
| 23:44.16 | C4away | between 10-20 I use my discretion |
| 23:44.23 | *** part/#asterisk clive- (i=ident@dsl-242-146-16.telkomadsl.co.za) |
| 23:44.23 | rue_mohr | I'd like to be able to make changes remotly |
| 23:44.28 | C4away | you can |
| 23:44.37 | C4away | just forward the ports to the web gui and change the password |
| 23:44.57 | rue_mohr | gets complex via ssh without setting up a tunnel |
| 23:45.04 | bmoraca | rue_mohr: it would be best to make sure that everything is set up properly before wasting all this time. your config files could be right and your asterisk wrong. |
| 23:45.04 | rue_mohr | which I cant do from just anywhere |
| 23:45.23 | C4away | I usually set the extenesion number for the port forwarding ... e.g. http://publicip:1234 => 192.168.1.55:80 |
| 23:45.25 | C4away | or whatever |
| 23:45.34 | rue_mohr | I say the asterisk config is right cause its working on antoher phone just fine |
| 23:45.38 | C4away | then I just browse to the site's public ip:exten |
| 23:45.45 | strk | a stupid question: if you want to use asterisk as a SIP register, is it still true that you need 400Mhz of processor for up to 5 contemporary calls as the book states ? |
| 23:45.56 | C4away | rue_mohr it is just rebooting over and over? |
| 23:46.03 | rue_mohr | c4 no |
| 23:46.04 | strk | I mean, shouldn't asterisk NOT be involved in actual voice from ACCEPT on ? |
| 23:46.11 | C4away | it is ignoring the config params? |
| 23:46.41 | rue_mohr | seems to be, letme change something definite and see if it takes, its definitly saying no service |
| 23:46.48 | C4away | strk: contemporary? |
| 23:46.56 | C4away | concurrent? |
| 23:47.07 | strk | concurrent, yes |
| 23:47.26 | C4away | 400mhz is pretty low in the pile of scrap computers over here |
| 23:47.39 | rue_mohr | huh, the admin password is gone... odd |
| 23:47.49 | C4away | I think the last 400mhz or slower computer we had was an OLD email server that was decommissoned somewhere around 2001 |
| 23:47.54 | strk | I know, but is a general question: does actual voice (or video) need to pass trough asterisk ? |
| 23:48.05 | C4away | technically, no |
| 23:48.09 | C4away | practically, yes |
| 23:48.24 | [TK]D-Fender | strk: And that isn't the load to watch out for. |
| 23:48.39 | strk | gah |
| 23:48.44 | strk | I liked the techical part :) |
| 23:48.45 | [TK]D-Fender | strk: You need to keep an eye on transcoding, call recording, SWEC, etc |
| 23:49.23 | strk | oh, you mean once you start you will eventually *want* voice to step by * to do those things ? |
| 23:49.25 | rue_mohr | no it is getting atleast part of the config files |
| 23:49.29 | C4away | if every device is on the same network you can probably set "canreinvite=yes" ... but if you have audio problems you need to disable that |
| 23:49.51 | rue_mohr | cause I was just able to change the name for line1 that it puts where it shoudl be putting the users name |
| 23:49.52 | C4away | many SIP gateways, such as audiocodes gxw-xxxx devices, cannot reinvite to themselves |
| 23:50.08 | strk | [TK]D-Fender: you shouldn't need transcoding if both clients share at least one codec, that's part of the SIP handshake isn't it ? |
| 23:50.11 | *** join/#asterisk Failrar (n=Failrar@fsm.xs4all.nl) |
| 23:50.26 | C4away | rue_mohr: the "teleephone number" field should be the SIP username |
| 23:50.34 | C4away | aastra's require that in order to register |
| 23:51.06 | C4away | if the username is the same as the extension that's fine, but if you have usernames for devices like phone1234 then you need to put phone1234 as the phone number |
| 23:51.08 | rue_mohr | sip user name: 14 |
| 23:51.27 | frogonwheels | strk: yes indeed, that's correct. |
| 23:52.03 | frogonwheels | strk: On my system I simplify things further by only having one codec I use :) |
| 23:52.05 | [TK]D-Fender | strk: something like that |
| 23:52.34 | rue_mohr | C4away, when was the last time you wrote an aastra config file? |
| 23:52.35 | *** join/#asterisk idrew (i=zero@73.sub-70-220-162.myvzw.com) |
| 23:52.52 | C4away | strk: well that's not exactly the whole story, but the server should have a list of codecs that it prefers, while usually this means that the devices will end up going with the server's preference, that doesn't mean that they will always end up that way. |
| 23:52.56 | C4away | rue_mohr: for what model? |
| 23:53.05 | rue_mohr | 9143 |
| 23:53.23 | *** join/#asterisk ki4lzk (n=jjones@ip24-255-222-124.ks.ks.cox.net) |
| 23:53.27 | ki4lzk | hello all |
| 23:53.33 | C4away | haven't written one for the 43 model, but 33 model I wrote about 50 of them a few months ago |
| 23:53.46 | C4away | and had to make changes as recently as 4 days ago |
| 23:53.54 | rue_mohr | odd cause you or the manual are verry wrong |
| 23:54.03 | C4away | in what respect? |
| 23:54.16 | rue_mohr | for the 9143 I have the i33 manual, as they seem to be the same |
| 23:54.18 | ki4lzk | I am currently installing asterisk and it says i need to install wget which I already have any suggestions? |
| 23:54.22 | idrew | have a quick question for someone that might be able to help. I have a centos box that I have on my home network with ssh enabled for a person to login and install and configure asterisk for me for my cisco ip phones that I plan on using asterisk with. however, I'm not really that good with linux. can someone tell me how to check through ssh to see if this guy has actually been doing anything? I see that he's logged into the server, but I'm not |
| 23:54.39 | rue_mohr | C4away, THERE is no config paramiter called telephone number |
| 23:55.03 | C4away | rue_mohr: [16:26] C4away: aastra has different parameters that are available in different models |
| 23:55.20 | C4away | I would be surprised if the 9143 and 33i are identical |
| 23:55.25 | rue_mohr | aka why I want a config file of defaults |
| 23:55.47 | C4away | there is an example config file in the back of the pdf |
| 23:55.56 | rue_mohr | C4away, could you prove me crazy and find a manual for the 9134 thats not an i33 manual |
| 23:56.14 | C4away | i have never purchased the 9143 nor the 33i |
| 23:56.27 | rue_mohr | fine, its not like mine came with a disk |
| 23:56.32 | rue_mohr | its downloaded from aastra |
| 23:56.34 | C4away | now that the 9133i is being phased out I will be buying the 9143i for my next project that requires low-end phones |
| 23:57.04 | rue_mohr | ok, canyou find the download fora manual |
| 23:57.04 | idrew | can someone please answer my question? |
| 23:57.36 | *** join/#asterisk tobias (n=tobias@user-0ce2hu8.cable.mindspring.com) |
| 23:57.40 | *** join/#asterisk Micc (n=Michael@c-76-121-255-52.hsd1.wa.comcast.net) |
| 23:58.10 | C4away | idrew: type w |
| 23:58.21 | C4away | at the command prompt on the server |
| 23:58.23 | rue_mohr | ok, this manualbetter not be different than what I'm using or I'm gonna flip |
| 23:58.28 | Micc | Can I use a voip adapter and take the lines out of that and punch down in a punch down block and will it work with all existing phones hooked up to that punch down block line? |
| 23:58.38 | C4away | under "what" is what they are doing |
| 23:58.39 | Micc | Its the same thing as what the phone company provides, right? |
| 23:58.47 | C4away | if you want to watch it type "watch w" |
| 23:58.47 | [TK]D-Fender | Micc: Yes |
| 23:59.06 | rue_mohr | Micc, yes |
| 23:59.17 | ki4lzk | has anyone had problems with asterisk and wget? |
| 23:59.22 | Micc | So if they have an NEC phone system that currently has regular phone lines coming into it, then I can just replace that with voip adapter lines and the pbx should continue to run the current phones the same, right? |
| 23:59.34 | [TK]D-Fender | Micc: Yes |
| 23:59.40 | Micc | Ok, great. :) |
| 23:59.48 | drmessano | and wget? |
| 23:59.49 | idrew | C4away: thanks. I just want to see if he's actually done any installing. does asterisk usually default to a specific directory that I could check via ssh? |
| 23:59.51 | rue_mohr | no I ahve the right manual 9143i |
| 23:59.56 | Micc | wget has nothing to do with asterisk. |
