| 00:07.04 | rudholm | my mobile carrier wouldn't change CID even to something sensible (like my landline number) |
| 00:08.08 | crimethinker | "","6169802426","s","UnwantedCall","""IMPORTANT "" <6169802426>","Zap/8-1" |
| 00:12.59 | *** join/#asterisk lowlevel (n=Stuart@CPE0017f2e2a3e8-CM000f9f7d6742.cpe.net.cable.rogers.com) |
| 00:15.35 | obnauticus | K |
| 00:15.45 | obnauticus | Sprint Supposidly changed it to Emergency <911> |
| 00:16.15 | JT | wait, you had to ring a person and ask them to change it, and they changed it to that? |
| 00:16.22 | obnauticus | yes |
| 00:16.24 | obnauticus | I hope so |
| 00:16.26 | rudholm | have you tested it? |
| 00:16.32 | obnauticus | 24 to 73hr |
| 00:16.37 | obnauticus | 74hr* |
| 00:16.40 | obnauticus | damnit |
| 00:16.41 | rudholm | you mean 72? |
| 00:16.42 | obnauticus | 72 |
| 00:16.43 | obnauticus | yes |
| 00:16.44 | obnauticus | LOL |
| 00:17.13 | rudholm | I'm sure if I called my carrier and asked them to change my outbound CID they'd just give me a blank stare |
| 00:17.22 | obnauticus | they like space out for a second |
| 00:17.23 | obnauticus | like wtf |
| 00:17.27 | Strom_C | doesn't know how many hours are in three days...can't spell "supposedly"... |
| 00:17.54 | obnauticus | I know how many hours there are in a day. I was constantly type-o'ing it |
| 00:17.57 | obnauticus | and i can't spell supposedly |
| 00:17.58 | obnauticus | lol. |
| 00:18.08 | Qwell | and you can't spell typo |
| 00:18.20 | Qwell | just pointing that out |
| 00:18.24 | obnauticus | be glad i can spell You and Are |
| 00:18.43 | rudholm | U R |
| 00:18.47 | Qwell | at least you didn't spell supposably |
| 00:18.51 | rudholm | hahaha |
| 00:18.54 | obnauticus | lol |
| 00:18.55 | Qwell | strom would've had a fit |
| 00:19.04 | obnauticus | Word. |
| 00:19.04 | rudholm | he should of spelled it that way |
| 00:19.14 | rudholm | but I could care less |
| 00:19.15 | Qwell | rudholm: now you're just trolling, aren't you? ;) |
| 00:19.19 | rudholm | hehe |
| 00:19.26 | Strom_C | qwell: I knew someone who thought there was a word called "oblivously" |
| 00:19.28 | Qwell | "should of", eh? |
| 00:19.37 | Strom_C | it stood in for both "obviously" and "obliviously" |
| 00:20.01 | Qwell | Strom_C: they've oblivously the same word |
| 00:20.08 | Qwell | You're just oblivous |
| 00:20.12 | rudholm | haha |
| 00:20.18 | Strom_C | catsexiciously. |
| 00:20.29 | rudholm | it's a doggie dog world. |
| 00:20.42 | Strom_C | forever in a day |
| 00:20.47 | rudholm | hahaha |
| 00:20.56 | Qwell | wait, what should it be? |
| 00:20.59 | Qwell | oh, nm |
| 00:21.01 | rudholm | the proof is in the pudding |
| 00:21.18 | rob0 | ewww |
| 00:21.26 | rudholm | that *should* be "the proof of the pudding is in the eating" |
| 00:21.38 | rudholm | yeah, it would be messy if the proof was in the pudding itself. |
| 00:21.59 | *** join/#asterisk tengulre (n=tengulre@222.90.66.10) |
| 00:22.01 | Strom_C | "is the proof in the pudding suppose to be a free toy or something?" |
| 00:22.09 | rudholm | yes |
| 00:22.20 | rudholm | it's 100-proof pudding |
| 00:22.22 | Qwell | it's the proof of purchase - they didn't have room for the barcode |
| 00:22.46 | Strom_C | UPC or EAN?????!!! |
| 00:22.52 | Qwell | UPC |
| 00:22.53 | rudholm | when does the cellphone driving ban take effect in CA? is it today? |
| 00:23.02 | Qwell | what? that's silly |
| 00:23.06 | Strom_C | oh, did they do that finally? |
| 00:23.10 | rudholm | yeah, you have to wear a headset |
| 00:23.15 | Qwell | pfft |
| 00:23.18 | rudholm | yeah, the Governator signed it a while back |
| 00:23.20 | Qwell | like that'll happen |
| 00:23.26 | Strom_C | they did it in new york |
| 00:23.28 | rudholm | no really, it's law |
| 00:23.31 | JT | been law here for years |
| 00:23.36 | rudholm | where? |
| 00:23.37 | JT | you'll get used to it |
| 00:23.47 | JT | either use a headset, or don't let a cop see you using the phone :P |
| 00:23.55 | Strom_C | JT: oh, i probably shouldn't have used my mobile while driving on the M1 then :) |
| 00:23.55 | rudholm | I don't think it'll have any effect really. headset != less distracting |
| 00:24.12 | JT | Strom_C: M1? |
| 00:24.15 | Strom_C | rudholm: also, fuck you; i want pudding now |
| 00:24.24 | JT | rudholm: err it makes a huge difference if driving a manual car |
| 00:24.24 | rudholm | mmm, pudding... |
| 00:24.26 | Strom_C | and i have none :( |
| 00:24.35 | Qwell | eh, people will find new distractions |
| 00:24.39 | JT | it is VERY hard to safely drive a manual car with a handheld mobile phone |
| 00:24.39 | Strom_C | JT: the freeway that runs southeast of melbourne |
| 00:24.43 | Qwell | laptops on the drivers lap aren't currently illegal |
| 00:24.44 | JT | Strom_C: ah ok |
| 00:24.52 | Strom_C | qwell: actually |
| 00:24.53 | rudholm | my phone is big enough that I can cradle it with my shoulder |
| 00:24.56 | Qwell | oh |
| 00:25.02 | Strom_C | california has a law about displays in view of the driver |
| 00:25.03 | Qwell | well then |
| 00:25.11 | Qwell | ahh, I guess that would count |
| 00:25.12 | rudholm | but also, just cruising on the freeway you don't need to shift |
| 00:25.13 | JT | you can't shoulder check properly if you are using it to hold your phone |
| 00:25.17 | rudholm | and most people in CA have automatics, anyway. |
| 00:25.18 | Strom_C | there's an exemption for navigaion systems and so on, but a laptop on the front seat is illegal |
| 00:25.26 | Qwell | JT: people in CA don't do that anyways |
| 00:25.29 | JT | haha |
| 00:25.31 | Qwell | so, the law is useless |
| 00:25.43 | Qwell | blinker == I'm getting over - MOVE |
| 00:25.57 | rudholm | yeah, I think the law won't really have much effect |
| 00:26.03 | Strom_C | even the blinker is only used half the time |
| 00:26.06 | Qwell | heh |
| 00:26.08 | rudholm | but whatever, legislators can say they "did something about the problem" |
| 00:26.11 | Strom_C | HOW TO CHANGE LANES LIKE A CALIFORNIAN |
| 00:26.13 | Qwell | my wife always used to say |
| 00:26.15 | Strom_C | 1: start moving over |
| 00:26.23 | Strom_C | 2: if you hear honking, start moving back |
| 00:26.27 | Qwell | "If you're gonna make an illegal turn, don't use your blinker - it becomes intent." |
| 00:26.37 | rudholm | yeah, and I have to say, the BMW driver stereotype is true, they seem to be the worst offenders. |
| 00:26.52 | Qwell | Strom_C: so true.. |
| 00:26.56 | Qwell | NJ is better |
| 00:27.01 | Qwell | they have a 3rd step |
| 00:27.08 | Qwell | flip off the driver - regardless of step 2 |
| 00:27.11 | Strom_C | rudholm: yeah, i saw a crazy driver on the harbor freeway last night, and of course it was like a riced out 1987 3-series |
| 00:27.15 | JT | "if you don't hear honking, move over faster" |
| 00:28.12 | Strom_C | qwell: so have they got you saying "the interstate" yet, or do you say "the 565" like a good californian? |
| 00:28.18 | *** join/#asterisk HaMYaI (i=HaMYaI@125-25-128-130.adsl.totbb.net) |
| 00:28.34 | *** part/#asterisk HaMYaI (i=HaMYaI@125-25-128-130.adsl.totbb.net) |
| 00:28.34 | Strom_C | s/cal/southern cal/ |
| 00:28.54 | *** join/#asterisk saftsack (n=saftsack@pD9E0514B.dip.t-dialin.net) |
| 00:28.54 | Qwell | man...I asked where the freeway was once |
| 00:28.55 | Qwell | BIG mistake |
| 00:29.13 | Strom_C | "boy, do you mean the INTERSTATE?!?" |
| 00:29.21 | Qwell | no, they led me to the 65 :p |
| 00:29.24 | Strom_C | oh heh |
| 00:29.34 | Strom_C | the 565 doesn't count as a freeway? |
| 00:29.37 | Qwell | no |
| 00:29.40 | Strom_C | wtf |
| 00:30.51 | rudholm | "the interstate"? |
| 00:30.57 | rudholm | which one? |
| 00:31.04 | rudholm | and what about state highways? |
| 00:31.11 | rudholm | like the 101 |
| 00:31.18 | Strom_C | the 101 is a US route |
| 00:31.21 | Strom_C | not a state highway |
| 00:31.21 | rob0 | 565 sounds like Huntsville. |
| 00:31.32 | rudholm | ok, but still not an "interstate" |
| 00:31.41 | Strom_C | PCH is a state highway...parts of the foothill freeway are state route 210 |
| 00:31.49 | rudholm | yeah, Topanga Canyon Blvd |
| 00:31.54 | rudholm | Rosemead Blvd |
| 00:31.58 | rudholm | Santa Monica Blvd |
| 00:32.02 | rudholm | all State Highways |
| 00:32.05 | Strom_C | haha yes |
| 00:32.23 | Qwell | Beach Blvd? |
| 00:32.36 | rudholm | freeway is a good generic term for a limited access roadway. |
| 00:32.42 | rudholm | yeah, I think Beach is as well |
| 00:32.58 | rudholm | but that's down in Orange County |
| 00:33.05 | rudholm | we don't care about Orange County |
| 00:33.15 | Strom_C | it's fantasyville down there |
| 00:33.17 | rob0 | The "free" part is to distinguish from a "toll" road. |
| 00:33.22 | rudholm | right |
| 00:33.23 | Strom_C | rob0: uh no |
| 00:33.35 | rudholm | it was, at least at some point. |
| 00:33.38 | Strom_C | oh o |
| 00:33.38 | Strom_C | k |
| 00:33.47 | rudholm | although these days it's not, since there are toll roads that people call freeways |
| 00:33.54 | Strom_C | I have a book on road traffic engineering from the early 1960s |
| 00:33.59 | rob0 | Those people are wrong. :) |
| 00:34.30 | rudholm | here in Los Angeles we call them all "turnpikes" :) |
| 00:34.33 | JT | they engineer roads in california? |
| 00:34.37 | rudholm | "expressways" |
| 00:34.39 | JT | don't they just lay them? |
| 00:35.08 | Strom_C | the Hollywood / Santa Ana Turnpike |
| 00:35.20 | Qwell | wtf, turnpike? |
| 00:35.33 | Strom_C | oh ok, this is the third edition from 1965 |
| 00:35.40 | Strom_C | previous editions are 1950 and 1941 |
| 00:36.13 | rudholm | oh, it does into effect July 2008, not July 2007 |
| 00:37.35 | Strom_C | apparently, the technical term for that device at the crosswalk is "pedestrian push button" |
| 00:38.59 | Strom_C | and the great sweeping ramp that you take rather than making a hard right turn is called a "jughandle" |
| 00:39.15 | rudholm | I thought it was an interchange road |
| 00:39.29 | JT | and a roundabout is a boobie |
| 00:39.59 | Strom_C | remember kids, don't drive on the nipple |
| 00:40.25 | JT | unless it's not very prnounced |
| 00:40.28 | JT | pronounced |
| 00:41.11 | rudholm | pron-ounced |
| 00:41.25 | JT | heh |
| 00:43.33 | *** join/#asterisk withnall (n=dave@60-242-0-60.static.tpgi.com.au) |
| 00:44.39 | withnall | greetings. I'm wondering if someone here can help me. I've got asterisk running fine. but I've just found that when I call a number that has a menu system (eg a bank) the key presses don't get sent through and I'm left looping the menu announcements |
| 00:45.08 | *** join/#asterisk mardum_ (n=mardum@71-81-73-42.dhcp.stls.mo.charter.com) |
| 00:45.10 | withnall | I'm not sure what I'm looking for so I haven't been able to find anything on the internet |
| 00:49.48 | JT | withnall: more info about the setup? |
| 00:51.34 | *** join/#asterisk lowlevel (n=Stuart@CPE0017f2e2a3e8-CM000f9f7d6742.cpe.net.cable.rogers.com) |
| 00:51.40 | *** join/#asterisk WindBack (n=Administ@host189.190-31-206.telecom.net.ar) |
| 00:51.55 | flenders | withnall: are you testing it using SIP? |
| 00:53.00 | WindBack | [TK]D-Fender, if I'm in a sip client and I want to call another extension and then transfer the call to another extension. Can I do that?? |
| 00:53.32 | WindBack | [TK]D-Fender, I know how to transfer when some extension call me but not when I call it |
| 00:53.43 | withnall | flenders: yes i'm calling out using sip |
| 00:53.57 | flenders | withnall: try a different dtmf mode on sip.conf |
| 00:54.15 | tengulre | http://voip.zrtek.com/upload/1182759633.JPG |
| 00:54.40 | tengulre | http://voip.zrtek.com/upload/1182759703.JPG |
| 00:55.19 | WindBack | if I'm in a sip client and I want to call another extension and then transfer the call to another extension. Can I do that?? |
| 00:55.29 | WindBack | I know how to transfer when some extension call me but not when I call it |
| 00:55.48 | WindBack | can somebody help me?' |
| 00:57.14 | flenders | I thought trasfering a call would be the same no matter who called who |
| 00:58.35 | rob0 | Let's not ARGUE and BICKER about WHO killed WHO. |
| 00:59.20 | WindBack | flenders, Yes I thought the same, but not |
| 01:01.12 | tengulre | Asterisk Manager 2.0 , Platform : windows xp/2000, |
| 01:01.58 | *** join/#asterisk HaMYaI (i=HaMYaI@125-25-128-130.adsl.totbb.net) |
| 01:01.58 | tengulre | main funcitons: 1. monitor channel, 2. custom IVR rules, 3. Add/edit/del Iax2,SIP and agent users. |
| 01:02.20 | tengulre | all AMI functions is in ! |
| 01:02.31 | obnauticus | tengulre what is that? |
| 01:02.32 | tengulre | s/is// |
| 01:03.14 | tengulre | sorry , my english skill is low ! ;( |
| 01:07.22 | *** part/#asterisk Here_And_There (n=Here_And@pool-68-238-252-162.phlapa.fios.verizon.net) |
| 01:08.34 | [TK]D-Fender | WindBack, "show application dial" , and read your phones manual |
| 01:12.36 | *** join/#asterisk JerJer[mobile] (n=jj@24-236-251-147.dhcp.aldl.mi.charter.com) |
| 01:22.28 | *** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.itb.ac.id) |
| 01:28.43 | *** join/#asterisk Defraz (n=t0tal@fw.fuzecore.com) |
| 01:42.58 | BSD_Tech | ok 1.4.6 now driving me insane the dial plan is fixed and no users info has chanegd but i get this |
| 01:43.05 | BSD_Tech | [Jul 1 18:41:27] WARNING[2630]: chan_sip.c:8190 check_auth: username mismatch, have <1002>, digest has <1000> |
| 01:43.05 | BSD_Tech | [Jul 1 18:41:27] NOTICE[2630]: chan_sip.c:13486 handle_request_invite: Failed to authenticate user "1000" <sip:1000@10.0.0.15>;tag=5DD21C1A-E678E4E9 |
| 01:43.31 | BSD_Tech | I have 2 diff extens on my phone |
| 01:43.39 | BSD_Tech | 1000 ans 1002 |
| 01:43.55 | BSD_Tech | diff passwords also |
| 01:44.11 | BSD_Tech | and one on port 5060 and on on 5061 |
| 02:08.15 | *** join/#asterisk Sedorox (n=brandon@smartserv/cna/Sedorox) |
| 02:11.16 | *** join/#asterisk Avochelm (n=damo@gw-morphett.koalatelecom.com.au) |
| 02:13.08 | *** join/#asterisk marchon (n=marchon@c-75-68-31-152.hsd1.nh.comcast.net) |
| 02:13.25 | marchon | hello.... |
| 02:13.52 | marchon | was hoping to find some help on why I cant register my asterisk for incoming calls from sipnumber.net |
| 02:14.58 | *** join/#asterisk cnet2 (n=nada@196.40.38.130) |
| 02:15.45 | marchon | I have logs. |
| 02:16.12 | cnet2 | hi guys.. I wanted to ask, what do you guys use for QoS, in/out of a small business? I use tcng.. but obviously it won't better my in selection of packets. |
| 02:16.30 | cnet2 | is there a opensource thing like pfsense on freebsd or similar? for QoS"? |
| 02:17.04 | *** join/#asterisk VOiCi (n=philip@132-199.sh.cgocable.ca) |
| 02:17.12 | VOiCi | hi |
| 02:17.15 | *** join/#asterisk IPmonger (n=ipmonger@c-71-224-31-246.hsd1.pa.comcast.net) |
| 02:17.18 | cnet2 | offices with something like 1 meg download stream, with emule on some computers will destroy my voice quality |
| 02:17.29 | cnet2 | i can't seem to contain that problem.. |
| 02:17.38 | JT | imagine that |
| 02:17.45 | JT | it's easy to contrain |
| 02:17.48 | JT | switch off emule? |
| 02:17.57 | JT | but yes, QoS is possible |
| 02:18.01 | JT | if your router supports it |
| 02:18.11 | cnet2 | but what about inbound traffic.. |
| 02:18.57 | cnet2 | i can't QoS that, since big downloads will compete in the same way as voice.. (the ISP doesn't give QoS before sendingme packets) |
| 02:19.43 | JT | i thought you might be able to shape that, but i'm not sure |
| 02:19.46 | IPmonger | you need to shape the traffic |
| 02:19.56 | flenders | cnet2: lartc.org |
| 02:20.29 | flenders | or get a packet shaper |
| 02:20.58 | flenders | http://www.packeteer.com/ |
| 02:21.48 | cnet2 | great.. i-ll look at that |
| 02:21.54 | *** join/#asterisk Tako-san (n=Tako-san@216.232.147.102) |
| 02:23.03 | JT | i don't run production VoIP traffic over the same link as web browsing, and especially not p2p |
| 02:23.24 | flenders | that's the smartest thing to do |
| 02:23.26 | flenders | I think |
| 02:24.07 | flenders | cnet2: get a dedicated link for voice... might be a lot cheaper than a decent QoS solution |
| 02:24.36 | cnet2 | that-s a good choice.. but in small business sometimes is not posible |
| 02:26.21 | marchon | in a small business - you might want to try firmware with traffic shaping on a linksys wrt54g with a good firmware |
| 02:26.29 | marchon | I use openwrt |
| 02:27.01 | flenders | I was testing it once and I bricked my router running dd-wrt |
| 02:27.17 | flenders | but I agree, it might be a good option |
| 02:28.11 | cnet2 | ok.. that-s a very good option. :P |
| 02:30.32 | flenders | how many users do you have? |
| 02:31.50 | cnet2 | one of the most problematic business is a hostel, that has 8 computers plus wireless. |
| 02:32.02 | cnet2 | with a 1 meg dsl. |
| 02:32.58 | flenders | how many users on wireless? |
| 02:33.12 | cnet2 | normally between 1-3 users |
| 02:33.42 | flenders | I don't think emule should be allowed at all |
| 02:33.44 | *** join/#asterisk kvit123 (n=kvit123@203.209.56.65) |
| 02:34.00 | cnet2 | the thing is they sometimes have things like limewire or emule. |
| 02:34.31 | flenders | is it a linux gateway? |
| 02:34.34 | cnet2 | even if i don-t let emule pass through to users.. probably the manager will want it |
| 02:34.49 | flenders | then, the manager shold get a bigger pipe |
| 02:34.58 | cnet2 | i have a linksys voip that has QoS.. but it don-t work so well |
| 02:34.59 | flenders | s/shold/should |
| 02:35.27 | flenders | WRT54GP2? |
| 02:35.58 | cnet2 | its like a pap2 but with fxo+fxs and in/out ethernet |
| 02:36.10 | cnet2 | pretty nice.. but not so good with QoS |
| 02:36.37 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 02:36.39 | cnet2 | spa3000 |
| 02:37.02 | flenders | well, it doesn't do QoS on your network |
| 02:37.06 | cnet2 | sorry spa3102 |
| 02:37.16 | flenders | it's not a gateway |
| 02:37.24 | flenders | it's just an ATA |
| 02:37.59 | cnet2 | indeed is a gateway and it has QoS options on the GUI.. |
| 02:38.23 | cnet2 | http://voipstore.atacomm.com/Shops/ViewItem.aspx/27934028032-51706688256.htm |
| 02:40.45 | flenders | are you saying that because of this line: |
| 02:40.46 | flenders | QoS - Voice Packet Prioritization over Other Packet Types |
| 02:41.22 | snuff-work | mm.. in * 1.4.x show transcoders doesn't exist.. how can i be sure my tc400b is working? |
| 02:41.50 | *** join/#asterisk nohop (n=root@cc501678-a.hgv1.dr.home.nl) |
| 02:44.37 | kvit123 | hi, i'm getting problem with the receiver cannot hear, but the caller can hear |
| 02:45.04 | kvit123 | when call come from pstn |
| 02:46.06 | *** join/#asterisk eAi2k (n=cow@81-86-205-45.dsl.pipex.com) |
| 02:46.07 | *** join/#asterisk IPmonger (n=ipmonger@c-71-224-31-246.hsd1.pa.comcast.net) |
| 02:48.12 | VOiCi | anyone got an ekiga account working atm so i can try mine. |
| 02:52.25 | JT | kvit123: using what access technology? |
| 02:53.38 | kvit123 | i'm using sip |
| 02:54.01 | JT | to what, an itsp? |
| 02:54.11 | kvit123 | for the extension, and pstn connect to the sangoma analog card |
| 02:54.50 | JT | see you are using POTS to connect to the pstn |
| 02:56.21 | Corydon76-home | sounds like a NAT and/or firewall problem |
| 02:56.48 | kvit123 | yes, i also config in the sip.conf to be able nat=yes |
| 02:56.58 | Corydon76-home | It usually is when you get one-way audio with SIP |
| 02:57.03 | JT | kvit123: nat=no |
| 02:57.17 | JT | if the phone is on the same lan as asterisk |
| 02:57.33 | Corydon76-home | That only works if the phone is behind a NAT and not Asterisk |
| 02:58.01 | kvit123 | the phone and asterisk are on the same lan |
| 02:58.11 | JT | kvit123: nat=no then |
| 02:58.59 | Corydon76-home | What phone model are you using? |
| 02:59.13 | kvit123 | i's using welltech ip302 |
| 02:59.30 | Corydon76-home | Not one I've run into |
| 02:59.49 | JT | neither |
| 03:00.02 | Corydon76-home | nat=yes is a technical violation of the SIP spec, so it may or may not work |
| 03:00.04 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 03:01.03 | JT | in any case, he's using it where it's not designed to be used :) |
| 03:01.13 | kvit123 | aha |
| 03:01.15 | kvit123 | :) |
| 03:01.51 | kvit123 | what model suitable for asterisk ? |
| 03:02.33 | Corydon76-home | Dunno, it could be suitable, but neither of us have seen one before |
| 03:04.26 | cnet2 | flenders: sorry i went out for asec, yes for the line. Voice Packet Prioritization over Other Packet Types, and because in the gui says some QoS options like incomming bandwith set.. |
| 03:05.49 | JT | kvit123: so have you tried nat=no yet? |
| 03:06.00 | flenders | cnet2: give it a go then |
| 03:06.46 | flenders | I'm not sure how you would set it up though. I wouldn't trust an ATA to do QoS and gateway for me, but that's me. |
| 03:06.55 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@cpe-75-84-232-194.socal.res.rr.com) |
| 03:07.05 | flenders | cnet2: are you even using asterisk? |
| 03:08.06 | kvit123 | JT:i have changed, but in the case as i told you. it happen sometime |
| 03:08.24 | JT | kvit123: might be the phone |
| 03:08.42 | JT | kvit123: next thing to do may be to hit it with a packet sniffer :) |
| 03:08.54 | JT | or at least a SIP debug |
| 03:09.11 | kvit123 | JT: i see |
| 03:11.34 | kvit123 | JT: it maybe the problem is ipphone which sometime i got the noise interfere during the conversation |
| 03:11.58 | JT | sorry, i did not understand that |
| 03:13.21 | kvit123 | JT: during the call between sip --> sip, it has noise interfering while i talking on the phone |
| 03:13.33 | *** join/#asterisk luckyone (n=hidden@CPE-65-28-7-102.kc.res.rr.com) |
| 03:15.13 | cnet2 | flenders: yes.. i always use asterisk |
| 03:15.18 | luckyone | can someone help me figure out how to reconnect with my * process? |
| 03:15.22 | luckyone | I get Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) |
| 03:15.23 | cnet2 | :) more than happy with it |
| 03:15.43 | JT | luckyone: asterisk needs to be running before you can reattach a console to it |
| 03:15.51 | luckyone | and it is right... /var/run/asterisk/asterisk.ctl doesn't exist |
| 03:15.53 | luckyone | it is running |
| 03:16.15 | JT | a bit erroneous that there is no .ctl file |
| 03:16.24 | cnet2 | i think i'll tell my customers to get a separate internet line as requisite |
| 03:16.52 | JT | i hope they're not doing VoIP over Internet only |
| 03:17.04 | luckyone | JT: asterisk 7636 7630 4 22:16 pts/3 00:00:00 /usr/sbin/asterisk -U asterisk -G asterisk -vvvg -c |
| 03:26.21 | JT | luckyone: does user asterisk have permission to write to /var/run/asterisk? |
| 03:26.28 | JT | J4k3: you there? |
| 03:36.24 | *** join/#asterisk Cabal (n=Cabal@unaffiliated/cabal) |
| 03:36.55 | rue_mohr | http://eds.dyndns.org:81/~ircjunk/shop/dscn9996.jpg <- done! (minus the incomming line) the 66 isn't so bad... |
| 03:39.25 | Strom_C | where's the other block? |
| 03:39.33 | Strom_C | you're supposed to have more than one, you know :) |
| 03:40.47 | Strom_C | and, while i'm being nitpicky, you mounted the block upside-down |
| 03:42.06 | russellb | picky picky |
| 03:43.04 | Corydon76-home | You don't necessarily need more than one... but he should have bridge clips |
| 03:48.38 | *** join/#asterisk stkn__ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 03:49.45 | rue_mohr | did I? I started from the botom |
| 03:49.51 | rue_mohr | your said this was easy? |
| 03:50.09 | rue_mohr | bridge clips? |
| 03:50.23 | *** join/#asterisk lisandropm (n=lisandro@cpe-22-76.bvconline.com.ar) |
| 03:51.02 | rue_mohr | the right side is telco, the left side is distribution |
| 03:51.09 | rue_mohr | :/ |
| 03:52.02 | rue_mohr | oh well, it'll do for now |
| 03:52.28 | rue_mohr | its only 4 pairs in and 9 pairs out |
| 03:52.52 | rue_mohr | oh, hmm |
| 03:53.07 | rue_mohr | mayeb i will install a jack in there, its in a good place for the wireless phone |
| 03:54.24 | *** join/#asterisk bmg505 (n=leon@196.209.179.122) |
| 03:58.41 | Strom_C | rue_mohr: you start at the top |
| 03:58.52 | Strom_C | and the vertical screw slot goes on the bottom |
| 04:02.02 | rue_mohr | ok |
| 04:02.09 | rue_mohr | I'll try that next time |
| 04:02.27 | rue_mohr | I didn't start athte top cause I wanted to leave as much wire as I could |
| 04:03.28 | Strom_C | i know I should have dragged out my scanner and given you the BSP :) |
| 04:03.34 | Strom_C | s/know/knew/ |
| 04:04.39 | *** join/#asterisk |dennis| (n=dennis@shc.edu.bz) |
| 04:10.27 | J4k3 | JT: I'm here |
| 04:13.52 | JT | J4k3: how hard is it to install Linux onto those routerboards? |
| 04:14.57 | J4k3 | JT: not very, the 500-series has CF slots |
| 04:16.03 | JT | so it definitely works on all routerboards? i guess routeros is just customised linux? |
| 04:16.11 | JT | 100 series? |
| 04:16.13 | J4k3 | JT: yep |
| 04:16.20 | J4k3 | the 100 series, you can load it up via serial+ethernet |
| 04:16.29 | J4k3 | serial console, tftp it over |
| 04:16.38 | JT | what is the ethernet chipset? and what is the network performance like in general? |
| 04:16.46 | J4k3 | hrm |
| 04:16.48 | J4k3 | lets see |
| 04:17.58 | J4k3 | I'm getting 60mbit udp one-way off a 532 |
| 04:18.02 | J4k3 | 63. |
| 04:18.11 | JT | what sort of udp traffic? |
| 04:18.19 | J4k3 | bandwidth test, non-random |
| 04:18.23 | JT | ah ok |
| 04:18.29 | J4k3 | 27mbit via tcp |
| 04:18.44 | J4k3 | 8mbit tcp from a 112 |
| 04:18.47 | JT | i'm wondering how well it will handle lots of RTP and SIP, including possibly some firewalling and port forwarding |
| 04:19.19 | J4k3 | oh shit, theres something wrong with an ethernet cable outside :| |
| 04:19.27 | J4k3 | this 112's bandwidth test just went *donk* |
| 04:20.01 | JT | damn |
| 04:20.07 | J4k3 | 46 udp from the other 112 outside |
| 04:20.18 | J4k3 | 17 tcp |
| 04:20.55 | J4k3 | hrm wow, that one just *quit* |
| 04:21.19 | J4k3 | freaky... but it actually runs on top of the ground, and its been driven over twice by the propane delivery truck :| |
| 04:21.31 | J4k3 | (it doesn't have any customer traffic on it yet) |
| 04:22.02 | JT | quality backhaul ;) |
| 04:22.42 | J4k3 | hehe |
| 04:23.06 | J4k3 | everything else runs through some pcv conduit I buried |
| 04:24.17 | JT | heh |
| 04:24.26 | JT | J4k3: so what chipset is the ethernet? |
| 04:26.19 | *** join/#asterisk LakeSolon (n=blake@12-208-207-85.client.mchsi.com) |
| 04:29.18 | J4k3 | theres an integrated ethernet, and two via rhine 3s (eww) on the 532 |
| 04:29.50 | J4k3 | integrated = 'korina' |
| 04:30.19 | JT | ah ok, i assume the 1xx has the same? |
| 04:30.41 | JT | looks like the ethernet chipsets are pretty crappy on most embedded boards |
| 04:30.45 | *** join/#asterisk alrs (n=lars@pozug.com) |
| 04:30.52 | J4k3 | the 112's onboard is the internal admtek adm5120 SOC ethernet... thats all its listing in here as the device name |
| 04:31.08 | J4k3 | yep, which is pretty craptacular really |
| 04:31.11 | JT | ah ok |
| 04:31.16 | JT | looking at the 150 |
| 04:31.39 | J4k3 | I'm not sure whats up with the 150 |
| 04:31.51 | J4k3 | I assume its just a 5 port switch with vlans |
| 04:31.59 | J4k3 | like a wrt54g |
| 04:32.16 | JT | well, i assume all the ports are accessible under linux |
| 04:34.45 | J4k3 | I'm sure they can be |
| 04:35.03 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 04:35.41 | J4k3 | JT: what application are you thinking of? |
| 04:35.55 | J4k3 | if its *, I'd guess a decent xscale board might be the better deal, but it might cost more. |
| 04:36.19 | JT | who makes decent xscale boards? :) |
| 04:36.32 | J4k3 | compex's looked pretty good |
| 04:36.36 | JT | was thinking more a load balancer for the routerboard |
| 04:36.39 | J4k3 | but I didn't investigate them THAT deeply. |
| 04:36.52 | JT | or a traffic accounting unit |
| 04:55.21 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 05:14.08 | tzafrir_laptop | speaking of via: don't make the mistake I did and use a board with a via on-board display adapter for your desktop. Their linux drivers are really bad |
| 05:14.39 | JT | heh |
| 05:14.40 | JT | hmm |
| 05:17.26 | *** join/#asterisk CaT[tm] (n=cat@58.172.242.75) |
| 05:18.02 | CaT[tm] | hey. is there an agi dos and don'ts list somewhere? Just want to be sure these scripts I've inherited are coded right as they are turning into zombies. |
| 05:18.34 | JT | tzafrir_laptop: re your A500 question, the drivers aren't out yet |
| 05:18.55 | tzafrir_laptop | thanks |
| 05:18.57 | [TK]D-Fender | I've probably got the biggest display of anytone here on their * server.... 120" ;) I wouldn't want that magled by poor drivers |
| 05:19.16 | JT | tzafrir_laptop: their wiki says the BETA ones will be released this month |
| 05:46.11 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 05:47.27 | *** join/#asterisk ManxPower (n=manxpowe@dpc67142183150.direcpc.com) |
| 05:47.48 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@cpe-75-84-232-194.socal.res.rr.com) |
| 05:48.00 | *** join/#asterisk jacq (n=jal@203.187.143.130) |
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| 05:57.56 | *** join/#asterisk zeeesh (i=zeeesh@14-237-154-202.wol.net.pk) |
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| 06:41.41 | CaT[tm] | does asterisk daemonise on startup by default? or should I start it with & ? |
| 06:42.05 | Krooks | I don't think it daemonise |
| 06:42.18 | Krooks | 'what distro you on ? |
| 06:42.21 | CaT[tm] | yeah. that's what I'm seeing. just want to be sure. |
| 06:42.35 | CaT[tm] | fc3 (yeah it's old) |
| 06:42.36 | Krooks | 'what distro you on ? |
| 06:42.43 | Krooks | oh boy |
| 06:43.09 | Krooks | chkconfig --list |
| 06:43.28 | CaT[tm] | better yet. it's started out of rc.local |
| 06:43.38 | CaT[tm] | don't blame me. I didn't build it. |
| 06:43.48 | CaT[tm] | just trying to cure a zombie problem relating to agis |
| 06:43.58 | Krooks | well you can chkconfig if it start on startup with chkconfig --list |
| 06:44.09 | Krooks | fc3 ? |
| 06:44.26 | CaT[tm] | ja. |
| 06:44.45 | CaT[tm] | it's started from rc.local. wont show up on that command |
| 06:45.17 | Krooks | I guess chkconfig not around when fc3 came out |
| 06:45.23 | CaT[tm] | it is |
| 06:45.32 | CaT[tm] | it's just not started the way you're assuming it is |
| 06:45.49 | Krooks | I see. honest mistake. sorry. |
| 06:46.00 | CaT[tm] | no worries. :) |
| 06:51.20 | tzafrir | CaT[tm], asterisk will daemonise on startup by ddefault. |
| 06:51.43 | tzafrir | However if you use safe_asterisk it doesn't. Yet another reason not to use safe_asterisk |
| 06:51.54 | CaT[tm] | :) |
| 06:52.14 | CaT[tm] | I wasn't using it. I guess it doesn't daemonise if it's not told to be quiet? |
| 06:52.31 | tzafrir | asterisk will daemonise, unless you use -c (or -d / -v which imply it) |
| 06:52.49 | CaT[tm] | aha |
| 06:52.52 | CaT[tm] | that'd do it |
| 06:54.04 | *** join/#asterisk stoffell (n=stoffell@fw.catsanddogs.com) |
| 07:13.00 | *** join/#asterisk mbranca (n=matteo@host-210-mi.linuxserver.it) |
| 07:20.19 | *** join/#asterisk redax (n=redax@mail.caracom.hu) |
| 07:20.21 | redax | hi, |
| 07:25.17 | creativx | hi2u |
| 07:28.56 | *** join/#asterisk vgster (n=vgster@host217-45-221-53.in-addr.btopenworld.com) |
| 07:40.57 | *** join/#asterisk qdk (n=qdk@213.150.62.32) |
| 07:46.54 | *** join/#asterisk syneus (n=syneus@syneus.aemcom.net) |
| 07:59.16 | *** join/#asterisk maerlin_ (i=merlin@dk0td.afthd.tu-darmstadt.de) |
| 07:59.18 | maerlin_ | hi |
| 08:04.19 | *** join/#asterisk remmo (n=junk@203.62.147.3) |
| 08:05.29 | *** join/#asterisk vgster (n=vgster@host217-45-221-53.in-addr.btopenworld.com) |
| 08:14.05 | *** join/#asterisk dharrigan (n=dharriga@dsl-217-155-228-129.zen.co.uk) |
| 08:19.09 | jeremy_g | hi |
| 08:22.05 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 08:36.52 | *** join/#asterisk berktr (n=cn@teknopet.com) |
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| 08:55.24 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
| 09:04.33 | *** join/#asterisk achu (n=achu@122.167.64.126) |
| 09:05.10 | *** join/#asterisk guillote_GNU (n=guillote@190.7.23.224) |
| 09:08.40 | *** join/#asterisk ghenry (n=ghenry@suretecsystems.plus.com) |
| 09:09.24 | *** join/#asterisk Nobbie (n=anony@fwb003.fw.is.co.za) |
| 09:14.11 | *** join/#asterisk Defraz (n=t0tal@24-116-152-177.cpe.cableone.net) |
| 09:15.55 | jmls | jeremy_g: hi |
| 09:21.19 | *** join/#asterisk yassaccan (n=yassacca@admin164.hgo.se) |
| 09:24.27 | Nobbie | <PROTECTED> |
| 09:24.31 | Nobbie | Where do i start to debug ? |
| 09:26.01 | Nobbie | the logs show that the call is answered though |
| 09:32.03 | *** join/#asterisk currach (n=currach@89.16.90.180) |
| 09:41.28 | *** join/#asterisk Paul_UK (n=foo@email.seatwave.com) |
| 09:42.01 | Paul_UK | hi, anyone who maintains the site here. I've registered but havent recieved a confirmation and also request my password and I dont recieve an email. :( |
| 09:42.23 | Strom_C | which site? |
| 09:43.04 | Paul_UK | http://www.asterisk.org |
| 09:43.19 | Strom_C | no one who maintains it is going to be awake right now |
| 09:43.37 | Strom_C | don't forget that the UK is 6 hours ahead of alabama... |
| 09:43.59 | Paul_UK | yeah I know.. Im chatting to some folks in the US on IM now, they are still awake lol. |
| 09:44.15 | Strom_C | yeah, i'm in los angeles, but i'm about to crash |
| 09:44.37 | Paul_UK | sure, no worries I will wait :) |
| 09:45.50 | *** join/#asterisk krdian_ (i=krdian@killer.radom.net) |
| 09:46.13 | krdian_ | hello |
| 09:47.59 | *** join/#asterisk magikxx (n=tawandax@196.15.164.11) |
| 09:54.30 | *** join/#asterisk remmo (n=junk@203.62.147.3) |
| 10:00.12 | Uatec | why do google have to obfuscate their googlemaps code? i can't fracking debug is cos i don't know what the bug is, cos it's all gibberish |
| 10:03.07 | J4k3 | google is not your friend. |
| 10:03.12 | *** join/#asterisk achu (n=achu@122.167.56.209) |
| 10:03.38 | J4k3 | I especially love when google maps and google earth randomly quit responding because you've used too much. |
| 10:04.13 | creativx | obfuscation is sweet |
| 10:04.56 | jeremy_g | what do you mean by offuscation |
| 10:05.04 | jeremy_g | obfuscation ? uhhh |
| 10:07.12 | *** join/#asterisk O_Zone (n=hasta@host44-201-static.30-87-b.business.telecomitalia.it) |
| 10:07.13 | O_Zone | mmm |
| 10:07.17 | O_Zone | i'm back |
| 10:07.51 | O_Zone | ok now work |
| 10:07.59 | O_Zone | i receive a Congestion |
| 10:09.35 | Uatec | jeremy_g, obfusctation is where you make code purposefully complicated so that it can't be reverse engineered |
| 10:09.56 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 10:10.02 | jeremy_g | I see. |
| 10:10.06 | jeremy_g | Thanks Uatec |
| 10:14.57 | O_Zone | i receive the caller id |
| 10:15.13 | O_Zone | i want to set an incoming trunk for all sip calls incoming |
| 10:15.18 | O_Zone | how ? |
| 10:20.21 | Uatec | jeremy_g, software exists which will obfuscate your code, for you |
| 10:20.27 | Uatec | only paranoid people use it |
| 10:20.41 | Uatec | and using it makes your code really hard to work with to other developers |
| 10:22.21 | O_Zone | can i use FOP with ztdummy ? |
| 10:24.25 | *** part/#asterisk achu (n=achu@122.167.56.209) |
| 10:33.37 | jeremy_g | when i do make install i get make[1]: *** No rule to make target `/usr/include/sys/types.h', needed by `localtime.o'. Stop. |
| 10:33.41 | jeremy_g | what am i missing? |
| 10:45.26 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 10:48.32 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 10:49.22 | JT | jeremy_g: this asterisk? |
| 10:49.52 | *** join/#asterisk CVirus (n=GoD@196.218.186.252) |
| 10:57.54 | *** join/#asterisk eAi2k (n=cow@81-86-205-45.dsl.pipex.com) |
| 11:02.33 | *** join/#asterisk xezz (n=xxx-4011@83.235.189.2) |
| 11:02.50 | *** join/#asterisk saftsack (n=saftsack@pD9E04177.dip.t-dialin.net) |
| 11:03.05 | Paul_UK | hi guys, does anyone know of asterisk developers in the UK? |
| 11:03.16 | xezz | hello, anybody heard of a panel that allows me to know what number i've called from a "to be called list" |
| 11:04.04 | *** join/#asterisk negativeduck (n=ducky@kitty.darkwing.net) |
| 11:06.10 | Uatec | i develop USING asterisk in the uk |
| 11:06.13 | Uatec | but i'm not very good |
| 11:09.37 | eAi2k | has anyone used the SPA-3000 with Asterisk? |
| 11:11.09 | drako | eAi2k, yes |
| 11:12.02 | eAi2k | I'd ideally like to connect 5 POTS phones to 2 PSTN lines through Asterisk |
| 11:12.09 | eAi2k | without investing too much ;) |
| 11:12.10 | *** join/#asterisk Phuntom (n=Phuntom@80.233.159.254) |
| 11:12.16 | Phuntom | hi ya! |
| 11:12.21 | eAi2k | is using the linksys systems a good solution? |
| 11:13.44 | eAi2k | the alternative seems to be to use hardware pci cards like the A200, but thats pretty expensive and clearly requires a lot of wiring |
| 11:16.54 | Phuntom | any alive? i need a presence information on asterisk, it is possible? |
| 11:20.31 | eAi2k | Phuntom, I've no idea, but does http://www.voip-info.org/wiki/view/Asterisk+presence help? |
| 11:20.32 | drako | eAi2k, spa-3000 you can only connect 1 POTS and 1 PSTN |
| 11:20.43 | eAi2k | yeah, I was thinking of a number of them |
| 11:21.03 | eAi2k | i.e. two SPA-3000s and two SPA-2002s |
| 11:21.12 | Phuntom | i need asterisk send notifies to subscribers |
| 11:21.17 | drako | eAi2k, I use one SPA-3000 and 2 SPA-2002 for 5 POTS and one PSTN |
| 11:21.18 | eAi2k | which should provide 6 POTS all in all |
| 11:21.23 | *** join/#asterisk mkl1525 (n=qwertz@pD9532BED.dip0.t-ipconnect.de) |
| 11:21.32 | Phuntom | about presence of all users on it |
| 11:21.34 | eAi2k | similar then |
| 11:21.49 | eAi2k | drako: is it relatively easy to set up? |
| 11:22.00 | eAi2k | I saw a guide and it looked easy |
| 11:22.50 | eAi2k | just setting up the extensions in Asterisk, then logging onto them from the SPA-* boxes |
| 11:23.34 | mkl1525 | Hi, I've got a problem with phones that should be normally reachable but also member of a queue. normal procedure is to ring the phone 15 sek and then got to voicemail. When such a phone is used in a queue the caller would be transfered to one of the agents voicemail boxes. so is there a way to distinguish if a caller comes from a queue or calls direct? |
| 11:24.15 | drako | eAi2k, yes, it is, but i still have some problem hanging calls |
| 11:25.33 | eAi2k | :| |
| 11:26.09 | eAi2k | if I just got one SPA-3000 for a start, would it play nice with other phones on the same line? |
| 11:26.23 | berktr | -- AGI Script Executing Application: (STOPMONITOR) Options: ((null)) |
| 11:26.23 | berktr | [Jul 2 14:24:25] WARNING[787]: res_monitor.c:318 ast_monitor_stop: Execute of ( nice -n 19 soxmix "/var/spool/asterisk/monitor/1183375383.351-in.gsm" "/var/spool/asterisk/monitor/1183375383.351-out.gsm" "/var/spool/asterisk/monitor/1183375383.351.gsm" && rm -f "/var/spool/asterisk/monitor/1183375383.351-"* ) & failed |
| 11:26.27 | *** join/#asterisk oej (n=olle@apollo.webway.se) |
| 11:26.27 | berktr | why do you think is this? |
| 11:26.30 | berktr | this sometimes happens |
| 11:26.58 | drako | eAi2k, other phones on the same line? |
| 11:27.15 | eAi2k | well, I've currently got 3 phones on one line |
| 11:27.28 | eAi2k | would putting the SPA-3000 on the line affect them? |
| 11:28.55 | drako | it will answer. |
| 11:29.09 | eAi2k | ah ok |
| 11:29.20 | drako | you can set up the timing tho. |
| 11:31.58 | eAi2k | drako, so does it provide a fake ringtone to the caller? |
| 11:32.36 | drako | eAi2k, what? |
| 11:33.10 | eAi2k | well, if you call it, and nobody has actually answered it, what happens? |
| 11:33.21 | eAi2k | I guess this is more of an Asterisk question |
| 11:33.39 | drako | eAi2k, i only have the spa3k connected to the line |
| 11:33.52 | drako | so it will always answer the phone at first ring |
| 11:33.59 | drako | and the call is recieved for asterisk. |
| 11:34.00 | eAi2k | and what does the caller hear? |
| 11:34.14 | drako | one ring then is answered by my asterisk IVR |
| 11:34.45 | eAi2k | ah ok |
| 11:35.02 | eAi2k | but if you didn't have an IVR, and just directed it to a call group |
| 11:35.19 | eAi2k | would Asterisk be generating a fake ring? |
| 11:35.21 | eAi2k | (for the caller) |
| 11:35.24 | drako | is a ring |
| 11:35.26 | drako | not fake. |
| 11:35.39 | drako | but yes, it will generate a ringtone |
| 11:35.45 | eAi2k | ok |
| 11:36.00 | eAi2k | I'll just buy one and experiement I guess |
| 11:36.17 | drako | best practice, yes. |
| 11:41.11 | *** join/#asterisk lisandropm (n=lisandro@cpe-22-76.bvconline.com.ar) |
| 11:44.34 | *** join/#asterisk kombi_ (n=kombi@213.160.14.18) |
| 11:45.41 | kombi_ | did I get this right: a jitterbuffer is now (as of 1.4) also available for sip channels? true or false? |
| 11:47.25 | kombi_ | anyone? |
| 11:49.33 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 11:50.15 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
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| 12:07.20 | jeremy_g | JT:yeah i am performing make install on asterisk |
| 12:16.52 | jeremy_g | i am kinda pissed off now. how can i perform equivalent of ./configure --prefix=/myasterisk |
| 12:17.11 | jeremy_g | i really need to get all asterisk binaries and modules and logs blah blah under that /myasterisk |
| 12:19.19 | *** join/#asterisk currach (n=currach@89.16.90.180) |
| 12:20.33 | *** join/#asterisk stimpie (n=michiel@ip565faf27.direct-adsl.nl) |
| 12:21.05 | jeremy_g | Problem solved. vi MakeFile, INSTALL_PREFIX=/myasterisk :) |
| 12:25.14 | stimpie | Does someone knows if the TE412P works in a Dell poweredge 1950? |
| 12:27.25 | cpm | Sounds like a question for digium/dell |
| 12:27.51 | stimpie | dell says they cant say anything |
| 12:28.07 | stimpie | it *might* work :-( |
| 12:28.54 | stimpie | and its a little early to call digium |
| 12:32.36 | *** join/#asterisk pigpen (n=mark@fw.seamans.cc) |
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| 12:44.54 | pj_ | Does anyone have experience configuring multiline on siemens phones ? Or multiline at all ? |
| 12:44.54 | *** join/#asterisk tcseke (n=chatzill@22-36.adsl.etel.hu) |
| 12:45.43 | *** join/#asterisk mardum (n=mardum@71-81-73-42.dhcp.stls.mo.charter.com) |
| 12:45.53 | kombi_ | after going from 1.2 to 1.4 Playback does not work anymore, any hints why that might be? |
| 12:52.06 | *** join/#asterisk MrWup (i=root@80-46-38-124.static.dsl.as9105.com) |
| 12:52.18 | MrWup | hey guys im trying to install the b410p |
| 12:52.28 | MrWup | with BT's ISDN2E lines now installed |
| 12:52.38 | MrWup | ive gone through all the setup stuff and the card is detected etc |
| 12:52.44 | MrWup | but i get the error No channel type registered for 'misdn' |
| 12:52.46 | MrWup | in asterisk |
| 12:52.51 | MrWup | anyone know what might be wrong? |
| 12:52.57 | MrWup | any help greatly appreciated |
| 12:53.15 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 12:54.46 | *** join/#asterisk af_ (n=getsmart@81-174-8-1.dynamic.ngi.it) |
| 12:55.10 | *** join/#asterisk mindCrime (n=chatzill@cpe-065-190-188-124.nc.res.rr.com) |
| 12:57.44 | MrWup | does asterisk need to be reinstalled/reomcpiled? |
| 12:59.54 | *** join/#asterisk javar (n=javar@69.79.134.24) |
| 13:01.02 | *** join/#asterisk Dovid (n=Dovid@79.178.24.155) |
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| 13:14.58 | nDuff | MrWup: You shouldn't need to recompile the whole thing -- at most, whichever module provides the misdn channel type. |
| 13:15.11 | MrWup | well |
| 13:15.17 | MrWup | ive now got asterisk 1.4.4 |
| 13:15.20 | MrWup | and the error is gone |
| 13:15.24 | MrWup | but now a new errror |
| 13:15.24 | MrWup | Could not Get the L2 up after 3 Attemps |
| 13:15.57 | msetim | I installed Kubuntu AMD 64 bit, but I'm have problems with flash 9.0, I was looking for solution and many site says to use nspluginwrapper -i to convert the plugin. But when the nspluginwrapper says that libflash isn't NAPI compability |
| 13:16.05 | Uatec | hey, i'm using a load of spa922s |
| 13:16.09 | Uatec | in the user guide it says: |
| 13:16.10 | Uatec | Ring1 Name Name in an INVITE’s Alert-Info Header to pick |
| 13:16.10 | Uatec | distinctive ring/CWT 1 for the inbound call |
| 13:16.10 | Uatec | Bellcore-r1 |
| 13:17.14 | Uatec | so i've got exten => 55,1,SIPAddHeader(Alert-Info: n=Bellcore-r1) |
| 13:17.20 | Uatec | but it does not change the way it rings |
| 13:17.53 | Uatec | am i using SIPAddHeader() wrong or what? |
| 13:20.56 | msetim | Sorry :-D Wrong list |
| 13:22.17 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 13:22.40 | *** part/#asterisk msetim (n=msetim@200.195.161.164) |
| 13:24.16 | *** join/#asterisk mwalling_ (n=mwalling@unaffiliated/mwalling/x-000000001) |
| 13:26.59 | creativx | Uatec: then the phone is misbehaving |
| 13:27.11 | creativx | or your headers are malformed |
| 13:27.12 | creativx | :) |
| 13:27.39 | *** join/#asterisk __DAW (n=_DAW@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 13:27.49 | Uatec | well i just showed you my headers |
| 13:28.23 | Uatec | i tried using Alert-Info: n=Simple-4 |
| 13:28.25 | Uatec | but that didn't work either |
| 13:30.15 | creativx | bad phone |
| 13:30.16 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:30.25 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 13:33.08 | *** join/#asterisk ToyMan (n=Stuart@user-12lcqol.cable.mindspring.com) |
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| 13:40.43 | jeremy_g | where can i get libssl to compile asterisk |
| 13:40.48 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 13:40.48 | *** mode/#asterisk [+o anthm] by ChanServ |
| 13:41.20 | *** part/#asterisk BSD_Tech (n=BSDTech@adsl-69-234-211-33.dsl.irvnca.pacbell.net) |
| 13:41.34 | berktr | is it possible to learn which carrier my provider uses without asking to provider? |
| 13:41.48 | *** join/#asterisk CristianoVOffice (n=cristian@201-24-89-249.fnsce702.dsl.brasiltelecom.net.br) |
| 13:44.04 | *** join/#asterisk bts3685|work (n=bts3685|@pool-71-244-104-93.phlapa.fios.verizon.net) |
| 13:45.20 | [TK]D-Fender | berktr, Yes, its called "industrial espionage" |
| 13:45.34 | stimpie | berktr, are we talking ip? |
| 13:45.34 | [TK]D-Fender | jeremy_g, depends on your distro |
| 13:46.03 | bts3685|work | i have a trixbox up and running...can i VoIP into asterisk from another box and make calls through the asterisk system? in other words, me ->VoIP (desktop) -> internet -> trixbox (server) -> asterisk -> telephone system |
| 13:46.25 | bts3685|work | receiving calls isn't necessary, but it'd be nice as well |
| 13:48.11 | ez` | whats that mean exactly ? Got SIP response 481 "Subcription Does Not Exist" back from 10.10.10.20 |
| 13:48.16 | [TK]D-Fender | bts3685|work, yes, but don't expect any help here. |
| 13:48.19 | [TK]D-Fender | ~trixbox |
| 13:48.20 | jbot | rumour has it, trixbox is a full linux distro that includes FreePBX, and other 3rd party add-ons. It is all this extra stuff which makes trixbox seriously painful to support and hence you will find little help here for it. Try asking in #trixbox or on their forums & WIKI at http://www.trixbox.org |
| 13:48.24 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^ |
| 13:49.38 | bts3685|work | thanks jbot_, [TK]D-Fender. i'm asking the guys in #trixbox too, i just wanted a quick-and-dirty answer if asterisk supports VoIP interfacing into the actual telephony. :) |
| 13:49.49 | *** join/#asterisk IPmonger (n=ipmonger@c-71-224-31-246.hsd1.pa.comcast.net) |
| 13:50.19 | bts3685|work | oops... |
| 13:50.29 | bts3685|work | just realized jbot is the bot. roflz. |
| 13:50.39 | [TK]D-Fender | bts3685|work, That didn't really add up right. Basically * can use several VoIP protocols as well as hardware interfaces tot he TDM. |
| 13:51.16 | [TK]D-Fender | bts3685|work, So you can quite naturally setup 2 * boxes to pass calls between each other any way you want them to. |
| 13:51.37 | [TK]D-Fender | bts3685|work, What amount of the TRIXBOX lets you do is a completely different matter. |
| 13:51.58 | bts3685|work | [TK]D-Fender: ohhhh. see, right now our * is only using hardware interfaces (SIP phones and such) and i'm only 30 pages into the MASSIVE o'reilly * book. so. :) |
| 13:52.27 | bts3685|work | but thanks, that was the answer i needed. i just wanted to know if it was possible. still new to this whole telephony thing |
| 13:52.32 | [TK]D-Fender | bts3685|work, So yes, ASTERISK can, but can TRIXBOX do what you want with it? to some degree I have seen articles. I hope that covers exactly what you're looking for. |
| 13:52.46 | [TK]D-Fender | bts3685|work, SIP is NOT a hardware interface. |
| 13:53.13 | [TK]D-Fender | bts3685|work, a TDM or T1 card would be a hardware interface to the PSTN |
| 13:54.14 | bts3685|work | oh i see what you mean. well, yeah, i mean we have it set up so we can plug "normal" phones in too... so basically, SIPs use VoIP of some codec or another? or is that an incorrect assumption? |
| 13:54.46 | *** join/#asterisk saftsack (n=saftsack@pD9E04177.dip.t-dialin.net) |
| 13:54.52 | JT | sip is a voip protocol |
| 13:54.53 | *** part/#asterisk mwalling_ (n=mwalling@unaffiliated/mwalling/x-000000001) |
| 13:55.04 | bts3685|work | OHHHH |
| 13:55.31 | bts3685|work | here i thought SIPs were the actual SIP phones. i didn't know it was a protocol in and of itself. |
| 13:55.32 | bts3685|work | wow. |
| 13:55.34 | [TK]D-Fender | SIP, H.323, MGCP, JABBER, etc |
| 13:55.41 | bts3685|work | i'm sorry guys, i'm a noob. |
| 13:55.41 | JT | IAX2 |
| 13:55.45 | [TK]D-Fender | bts3685|work, X-Lite is a SIP SOFT-phone. |
| 13:55.52 | [TK]D-Fender | JT : that too ;) |
| 13:56.01 | JT | SCCP ;) |
| 13:56.16 | [TK]D-Fender | bts3685|work, if it talks in IP.... its VoIP. Simple math ;) |
| 13:56.16 | *** join/#asterisk ai-a (n=jake2@cpc5-oxfd1-0-0-cust523.oxfd.cable.ntl.com) |
| 13:56.18 | bts3685|work | i can use jabber with *? (well, i mean, it makes sense...) know how the documentation is for setting that up in the wiki? |
| 13:56.34 | [TK]D-Fender | bts3685|work, Not with your setup, and not without pain IIRC. |
| 13:56.39 | ai-a | is there an issue where asterisk resets a lot ? ? in v1.4 ? a company is telling us to use 1.2 as its more stable ? ? |
| 13:57.17 | JT | something wrong with your question mark key, ai-a? :P |
| 13:57.20 | bts3685|work | oh jeez. sounds like a rainy-week project. well, thanks so much [TK]D-Fender you've been a huge help! |
| 13:57.29 | ai-a | JT: lol ;) |
| 13:57.33 | *** part/#asterisk bts3685|work (n=bts3685|@pool-71-244-104-93.phlapa.fios.verizon.net) |
| 13:57.49 | *** join/#asterisk VOiCi (n=o@132-199.sh.cgocable.ca) |
| 13:57.54 | VOiCi | anybody uses ekiga? |
| 13:58.37 | [TK]D-Fender | VOiCi, Plenty of people. Feel free to take a flying leap at your REAL question |
| 13:59.51 | VOiCi | :) |
| 14:00.02 | VOiCi | i only wanted to test it with someone, if anybody wanna try. |
| 14:01.08 | berktr | [Jul 2 14:24:25] WARNING[787]: res_monitor.c:318 ast_monitor_stop: Execute of ( nice -n 19 soxmix "/var/spool/asterisk/monitor/1183375383.351-in.gsm" "/var/spool/asterisk/monitor/1183375383.351-out.gsm" "/var/spool/asterisk/monitor/1183375383.351.gsm" && rm -f "/var/spool/asterisk/monitor/1183375383.351-"* ) & failed |
| 14:01.10 | berktr | this sometimes happens |
| 14:01.12 | berktr | why do you think is this? |
| 14:02.15 | [TK]D-Fender | VOiCi, test it with YOURSELF. you don't need anyone else |
| 14:02.48 | ai-a | whats the best Zap based ISDN card to use in Asterisk ? 30 channel PRI-30 ? thats easy detected and installed. |
| 14:03.02 | ai-a | im based in UK. |
| 14:04.43 | JT | both digium and sangoma make pri cards |
| 14:04.50 | Greenbox | berktr, do you have the command "soxmix" installed? |
| 14:05.50 | ai-a | JT: yes we're thinking of sangoma, but people were saying its not easy to set up. |
| 14:06.05 | ai-a | so we're wondering about Zap based idsn. |
| 14:06.11 | [TK]D-Fender | ai-a, A handful of extra little steps. Not something to base the decision on. |
| 14:06.14 | mvanbaak | I never had problems with sangoma cards |
| 14:06.36 | mvanbaak | and the sangoma cards are zap based in asterisk |
| 14:07.00 | mvanbaak | their driver will put a little patch in the zaptel module and recompile it for you |
| 14:08.13 | waKKu | hm.. is this normal when doing modprobe wctdm ? line 0: Unable to open master device '/dev/zap/ctl' ... even with this message, i get /dev/zap/ctl with major 196 minor 0 ... (correct, afaik) ... |
| 14:08.16 | Phuntom | wha can help me with presence in asterisk? |
| 14:08.25 | *** join/#asterisk irule (n=irule@189.164.43.194) |
| 14:09.10 | eAi2k | does anyone have any experience with receiving faxes with asterisk? |
| 14:09.39 | eAi2k | Ideally I'd like to be able to have asterisk go to voicemail after X rings and also accept faxes at that time |
| 14:09.52 | eAi2k | i.e. a line shared between fax and voice |
| 14:11.48 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 14:11.51 | JT | ai-a: err, it's easier to setup imho |
| 14:11.57 | JT | ai-a: has a text based gui |
| 14:12.14 | JT | ai-a: ignore the silly people |
| 14:12.36 | ai-a | JT: im a intermediate unix guy. |
| 14:12.52 | JT | i don't think you'll have any issues |
| 14:12.58 | ai-a | so we'll give it a go and buy one later when we get these perl scripts working on * |
| 14:13.04 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 14:13.39 | JT | go for hardware echo cancellation |
| 14:13.47 | JT | almost doubles the cost but is worth it |
| 14:14.12 | tzafrir_laptop | The problem with Sangoma IMHO is that extra setup is needed in the first place |
| 14:14.25 | JT | not really a problem |
| 14:14.49 | tzafrir_laptop | (note: the same could be said on the usage of Zaptel: an out-of-tree kernel module - in the first place) |
| 14:15.37 | tzafrir_laptop | the problems stem from there. You have to use the specific setup procedure and thus need to take extra steps to adapt it to your custom environment |
| 14:16.02 | JT | still not seeing an issue |
| 14:16.14 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 14:16.59 | tzafrir_laptop | waKKu, the error essage is from ztcfg . that is: the module is probably loaded, but you get an error message from the automatic run of ztcfg |
| 14:18.03 | tzafrir_laptop | do you have the file /etc/modprobe.d/zaptel ? |
| 14:19.00 | *** join/#asterisk kFuQ (n=somedude@c-67-185-123-34.hsd1.wa.comcast.net) |
| 14:19.14 | *** join/#asterisk __DAW (n=_DAW@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 14:19.49 | irule | si there a complete list of sound files used by the voicemail application? thanks! |
| 14:20.32 | [TK]D-Fender | tzafrir_laptop, I wouldn't call it a problem. Sangoma supports so much MORE than just *. Its only natural. |
| 14:20.48 | [TK]D-Fender | irule, You have to source, grep it out! |
| 14:21.08 | Qwell | berktr: I would very highly recommend changing your rm -rf line in the above dialplan... |
| 14:21.16 | tzafrir_laptop | [TK]D-Fender, when a setup scripts insists on patching my kernel source, it is a problem |
| 14:21.24 | Qwell | berktr: It appears that you could very easily rm -rf *, if some variable fails to be set |
| 14:21.39 | nDuff | eAi2k: I do that. |
| 14:22.00 | eAi2k | nDuff: how do you do it? AsterFax? |
| 14:22.47 | nDuff | eAi2k: iaxmodem+hylafax, and NVFaxDetect. |
| 14:23.37 | *** join/#asterisk Zaggynl (n=az@145.89.182.114) |
| 14:23.40 | Zaggynl | ?book |
| 14:23.51 | Krooks | I keep getting wrong password error when I want to register a client. But I check and doube-check, the password is right.What else could be the problem ? |
| 14:24.04 | Zaggynl | !book |
| 14:24.05 | Zaggynl | hmm |
| 14:24.12 | __DAW | ~book |
| 14:24.12 | jbot | somebody said book was a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 14:24.15 | Zaggynl | cheers |
| 14:24.42 | [TK]D-Fender | Krooks, until you show us your configs and tell us what client you're using we can't help much. |
| 14:24.56 | [TK]D-Fender | Krooks, beyond telling you that it is NOT lying |
| 14:24.58 | nDuff | eAi2k: HylaFAX has probably the most robust Class 1 implementation of any Free faxing solution I'm aware of -- and there are shiny Windows clients available to provide print-to-fax support and such on Windows. |
| 14:26.51 | eAi2k | nDuff, looks good, I'll read up on it |
| 14:27.34 | eAi2k | currently I've got a hardware fax machine that "listens" to the line for a fax, will this work like that? |
| 14:28.09 | nDuff | eAi2k: NVFaxDetect provides a Playback equivalent that also listens for fax tones on a channel. |
| 14:28.24 | nDuff | eAi2k: you can use that to transfer the call to your iaxmodem line. |
| 14:29.01 | nDuff | eAi2k: fax tone detection is also available on Digium's cards, if you use it; I don't. |
| 14:29.11 | *** join/#asterisk Keltus (n=Keltus@about/cooking/nakedchef/beefstew/Keltus) |
| 14:29.32 | eAi2k | I won't be, I'm thinking of using the Sipura/Linksys SPA system |
| 14:30.07 | nDuff | eAi2k: be sure your network is good (consistent and low latency, no dropped packets); I've had major problems faxing over those things in the past. |
| 14:30.36 | [TK]D-Fender | eAi2k, So what does this have to do with *? |
| 14:30.39 | eAi2k | yeah, I've read some stuff about that |
| 14:31.13 | nDuff | [TK]D-Fender: eh? He's inquiring about setting up a faxing solution working together with Asterisk; it seems pretty on-topic to me. |
| 14:31.23 | eAi2k | I thought so |
| 14:31.30 | Krooks | [TK]D-Fender: well ummsorry my mistake. typo |
| 14:31.35 | eAi2k | I'm new to this, so maybe I'm wrong |
| 14:31.40 | eAi2k | but it seemed on-topic |
| 14:32.09 | *** join/#asterisk demiv (n=demiv@134.42.128.66.PPPoECali.dynamic.telesat.net.co) |
| 14:32.26 | nDuff | eAi2k: how many lines are you running? If you have enough to justify a PRI, I strongly recommend going that route. |
| 14:33.06 | nDuff | eAi2k: the SPAs aren't bad hardware, but faxing can be finicky. |
| 14:33.21 | eAi2k | I might have to move the asterisk box |
| 14:33.24 | ai-a | is it easy to set up accepting of incomming fax by just using rxfax ? would love to get it to email the fax to the ext's email address. |
| 14:33.32 | Krooks | [TK]D-Fender: the other day you told I can run ekiga on the same machine I run asterisk. but I have to change the port from 50606 to 5061, right ? |
| 14:33.36 | eAi2k | currently theres a switch and a router between it and where the SPA will be |
| 14:34.13 | eAi2k | ai-a: http://www.voip-info.org/wiki/view/Asterisk+IAXmodem might help |
| 14:34.17 | eAi2k | I'm reading that |
| 14:36.12 | ai-a | eAi2k: lol, that guys box for recieving faxes "Athlon 64 running CentOS with a terabyte RAID5 array" is a bit of an overkill. |
| 14:36.21 | nDuff | ...indeed. |
| 14:36.38 | ai-a | whats the point of a terabyte raid if hes only relaying tif's to email accounts on another server ? |
| 14:36.53 | ai-a | unless he expects 2000 faxes a minute. |
| 14:38.40 | nDuff | Relaying TIFs doesn't work that well -- most folks don't have appropriate clients. Converting to PDF first is a Good Thing. |
| 14:38.57 | Zaggynl | I'm trying to make the sample voice menu from the AsteriskTFOT.pdf document, like this:http://pastebin.ca/599911 |
| 14:39.01 | *** join/#asterisk rmayorga_ (i=rmyorg@unaffiliated/rmayorga) |
| 14:39.18 | Zaggynl | But after saying the 'enter-ext-of-person' sound, it hangs me up |
| 14:40.20 | nDuff | Zaggynl: you're probably using a dialplan that was written pre-1.4 |
| 14:40.41 | Zaggynl | ah, the book is a bit outdated |
| 14:40.54 | Zaggynl | Do you happen to know more recent documents? |
| 14:41.09 | nDuff | Zaggynl: use WaitExten to wait for data entry at the end of your dialplan. |
| 14:41.17 | [TK]D-Fender | Krooks, basically you can't have 2 apps fighting to LISTEN on a given port. |
| 14:41.30 | Zaggynl | nDuff, how would that line look? |
| 14:41.41 | Krooks | but you told me it was possible |
| 14:41.50 | *** join/#asterisk SuPrSluG (n=SuPrSluG@pool-71-241-164-154.buff.east.verizon.net) |
| 14:41.56 | Zaggynl | just 'WaitExten' ? |
| 14:42.22 | [TK]D-Fender | ai-a, Considering 500g HD's go for about 100$ and I can get a MB w/ RAID 5 onboard for under 200$.... thats not so impressive anymore ;) |
| 14:42.28 | nDuff | Zaggynl: voip-info.org is your friend; read the fine wiki. that said, exten => s,n,WaitExten(10) |
| 14:42.35 | Zaggynl | thank you! |
| 14:43.09 | [TK]D-Fender | Zaggynl, thats one way, but not the reason it hung up |
| 14:43.31 | Zaggynl | Hmm okay |
| 14:43.46 | Zaggynl | it should return me to the same menu again, but it doesn't |
| 14:43.49 | [TK]D-Fender | Zaggynl, under [general] add "autofallthrough=no" |
| 14:43.52 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 14:44.23 | [TK]D-Fender | Zaggynl, no, it shouldn't . look at the context of that GOTO. |
| 14:44.37 | [TK]D-Fender | Zaggynl, And you should get your ass out of that GUI before you mess yourself up. |
| 14:44.47 | Zaggynl | I'm not using the gui ;-) |
| 14:44.51 | Zaggynl | Gave up on that already |
| 14:44.55 | Zaggynl | messed up the order badly |
| 14:45.30 | [TK]D-Fender | Zaggynl, And you have no "i" or "t" extens in your mean so unless they enter a valid choice fast the call will bomb out. |
| 14:45.53 | Zaggynl | hmm okay |
| 14:46.24 | [TK]D-Fender | Zaggynl, But good to haer that you're on your way to learning this thre right way |
| 14:46.40 | [TK]D-Fender | Zaggynl, Give a GOOD read to the section on *'s Standard Extensions. |
| 14:47.04 | Zaggynl | Those already in the extensions.conf file? |
| 14:47.51 | [TK]D-Fender | Zaggynl, Standard Extensions are o,s,h,i,t,a,fax ( Oh shit... a FAX! ;)) |
| 14:47.59 | Zaggynl | :D |
| 14:48.17 | VOiCi | good way to remember them heh. |
| 14:48.48 | [TK]D-Fender | Zaggynl, In there you use "s" (Start). When a normal context runs's out of "s" priorities * checks if autofallthrough is set. if NOT, it then awaits INPUT as an IVR. |
| 14:49.10 | [TK]D-Fender | Zaggynl, So the book is right, but you need autofallthrough disabled. |
| 14:49.11 | VOiCi | which distro is considered good to run a dedicated asterisk pbx? |
| 14:49.24 | [TK]D-Fender | VOiCi, Wheichever you feel most comfortable administering |
| 14:49.39 | [TK]D-Fender | VOiCi, As long as you can satisfy *'s dependencies, its all good. |
| 14:49.39 | Corydon76-work | [TK]D-Fender: Don't forget 'A' |
| 14:49.51 | [TK]D-Fender | Corydon76-work, its in there.... |
| 14:49.56 | VOiCi | thanks |
| 14:50.11 | Zaggynl | capital A then I would say |
| 14:50.48 | Corydon76-work | Oh, not 'A', 'T' |
| 14:51.01 | [TK]D-Fender | Corydon76-work, I don't recall a "capital T" being of any significance. |
| 14:51.14 | Corydon76-work | AbsoluteTimeout extension |
| 14:51.38 | [TK]D-Fender | Corydon76-work, really.... hmmm never used that. |
| 14:51.47 | Corydon76-work | It fires when you run out of time |
| 14:52.03 | [TK]D-Fender | Corydon76-work, standard "t" seems to do it jsut fine... |
| 14:52.25 | Corydon76-work | No, "t" is for an entry timeout |
| 14:53.09 | [TK]D-Fender | Corydon76-work, Then again since I don't do Pre-paid type stuff I suppose its only natural that I never needed it |
| 14:53.10 | Corydon76-work | If AbsoluteTimeout times out your call, then "T" will fire, followed by "h" |
| 14:54.57 | Corydon76-work | I used to use AbsoluteTimeout all the time back when I was still using analog circuits and I needed to ensure a call never got stuck in an infinite loop when a hangup was not detected |
| 14:55.31 | Corydon76-work | in the early days, before variables were in common use |
| 14:56.52 | *** join/#asterisk lwh (n=lwh192@66.212.165.24.tor.pathcom.com) |
| 14:58.26 | Krooks | hmm it seems that ekiga does not support iax. |
| 14:58.31 | Krooks | I thought it does |
| 14:59.24 | [TK]D-Fender | Krooks, very little does. |
| 14:59.47 | Krooks | only idefisk so far that I know |
| 14:59.48 | [TK]D-Fender | Krooks, IAX is not a "standard" the world at large cares much about. |
| 14:59.54 | *** join/#asterisk EricL (n=eric@clydesdale.linkexperts.com) |
| 15:00.22 | [TK]D-Fender | Krooks, idefisk is quite decent for a free phone and multi-protocol |
| 15:00.35 | EricL | Is there a more elegant solution for providing *69 than adding every extensions last received call to the DB and then pulling it up again if they dial *69? |
| 15:00.52 | *** join/#asterisk heison (n=heison@CPE000a01d49e6f-CM00111a59bec2.cpe.net.cable.rogers.com) |
| 15:00.52 | [TK]D-Fender | EricL, No. |
| 15:01.00 | [TK]D-Fender | EricL, That about covers it :) |
| 15:01.14 | EricL | [TK]D-Fender: Excellent. Not quite the answer I was looking for, but ok. |
| 15:01.18 | EricL | :) |
| 15:01.29 | EricL | Thanks. |
| 15:01.32 | [TK]D-Fender | EricL, You could do something absurd like cruise through CRD for that, but ICK, tell me that isn't 1000000% worse :) |
| 15:01.39 | [TK]D-Fender | CDR* |
| 15:02.11 | [TK]D-Fender | EricL, Thats the only place that remembers anything about what was called, so there really sin't anything else. |
| 15:04.11 | EricL | I don't understand why just dialing *69 on the outbound context doesn't work, but ok. |
| 15:04.46 | [TK]D-Fender | EricL, because * does only exactly what you TELL it to. |
| 15:05.04 | [TK]D-Fender | EricL, * is NOT a PBX! It is a telephony toolkit that lets you BUILD a PBX or other things |
| 15:06.28 | EricL | [TK]D-Fender: I just meant that *69 works on house phone without a pbx. |
| 15:07.10 | [TK]D-Fender | EricL, because your TELCO *IS* a PBX. |
| 15:07.20 | EricL | Ah, got it. |
| 15:07.58 | [TK]D-Fender | EricL, THEY make those "special" numbers you dial mean something. |
| 15:08.21 | EricL | [TK]D-Fender: Welp, then I guess its off to go code *69 into the dialplan. |
| 15:08.25 | [TK]D-Fender | EricL, Today I feel like having *69 call YOU back 100 time and play tt-weasels! |
| 15:08.40 | [TK]D-Fender | EricL, *69 is dead-easy. |
| 15:09.10 | EricL | [TK]D-Fender: I am going to Google it until I find an easy implementation. I am not a fan of re-inventing the wheel. |
| 15:09.38 | [TK]D-Fender | EricL, this is a PUNY use of "show function DB". |
| 15:09.52 | [TK]D-Fender | EricL, you'll waste more time looking that I would WRITING ;) |
| 15:10.07 | *** join/#asterisk Nephfl (n=nephfl@adsl-070-147-105-151.sip.gnv.bellsouth.net) |
| 15:10.33 | EricL | [TK]D-Fender: Dont' I have to store (DBPut) extension + callerID(num) everytime there is an incoming call. |
| 15:11.20 | *** join/#asterisk Xen^ (n=linux@unaffiliated/lnux/x-10290) |
| 15:11.50 | [TK]D-Fender | EricL, indeed you do. but that app is DEPRECATED. You only need the FUNCTION I referenced you on |
| 15:12.18 | *** join/#asterisk jiboneus (n=jiboneus@60.50.86.203) |
| 15:12.28 | *** join/#asterisk kippi (n=none@untrust-gct.equinoxit.net) |
| 15:12.30 | kippi | hey |
| 15:12.31 | Nephfl | im having trouble getting the settings to work in the polycom boot server, anyone else set one of these things up that can help me out? |
| 15:12.36 | EricL | [TK]D-Fender: Ah yes, I keep forgetting all the 1.4 changes. |
| 15:12.42 | jiboneus | hello |
| 15:12.57 | [TK]D-Fender | EricL, .... that was **1.2*** |
| 15:13.11 | EricL | [TK]D-Fender: I mean the deprecations. |
| 15:13.15 | kippi | has anyone had bt tell them that they have a frame fault? and does anyone have anyidea what might be causing this on asterisk? |
| 15:13.17 | EricL | (if that's even a work). |
| 15:13.23 | *** join/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
| 15:13.58 | [TK]D-Fender | EricL, the function came in in 1.2 and you got WARNINGS that the carpet was going to be pulled out from under you in 1.4 |
| 15:14.06 | [TK]D-Fender | EricL, HEED THE SIGNS AND CHANGE NOW! |
| 15:14.12 | *** join/#asterisk alrs (n=lars@pozug.com) |
| 15:14.18 | EricL | [TK]D-Fender: I never learned to read. |
| 15:18.44 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 15:18.45 | *** mode/#asterisk [+o blitzrage] by ChanServ |
| 15:19.40 | EricL | [TK]D-Fender: There really isn't much documentation on how to use DB(). I am Googling and not finding much. |
| 15:20.05 | EricL | Can you point me to some more thorough examples aside from the one on voip-info (not very thorough). |
| 15:20.40 | [TK]D-Fender | EricL, ...... I told you rather up front... "show function DB", and on the WIKI lookup "asterisk functions" |
| 15:21.01 | EricL | [TK]D-Fender: I did both of those alreayd. |
| 15:21.13 | [TK]D-Fender | EricL, So what aren't you getting? |
| 15:21.50 | EricL | [TK]D-Fender: It says it will read or write to the DB depending on if the value exists or not. I want to force it to read in one spot and force it to write in another. |
| 15:23.25 | EricL | [TK]D-Fender: Forget it...I think I understand now. |
| 15:23.34 | [TK]D-Fender | EricL, if there's nothing where you're reading it will return a null, if you write to one that doesn't exist it will create |
| 15:25.40 | kippi | if I was having porblems with ehco etc, would people recommend updating zaptel |
| 15:26.03 | polerin | btw, D... thanks for the canreinvite() help the other night... even though I read it wrong initially. for some reason it doesn't like careinvite() ;) |
| 15:27.12 | [TK]D-Fender | polerin, no.... you don't CARE to INVITE ;) |
| 15:27.42 | [TK]D-Fender | kippi, I'd recommend buying a REAL card. |
| 15:27.55 | [TK]D-Fender | kippi, Thogh I'm told the later zaptel's are better |
| 15:28.11 | *** join/#asterisk SuPrSluG (n=SuPrSluG@pool-71-241-164-154.buff.east.verizon.net) |
| 15:28.40 | *** join/#asterisk docelmo (n=vircuser@c-76-99-157-112.hsd1.de.comcast.net) |
| 15:29.26 | kippi | [TK]D-Fender: I have a digium card |
| 15:29.36 | docelmo | Does anyone know if the polycom sip application is the same for all versions of the phones |
| 15:29.42 | [TK]D-Fender | kippi, Like I said, buy a REAL card ;) |
| 15:29.54 | kippi | what is a real card? |
| 15:29.57 | nDuff | kippi: exactly. [TK]D-Fender is suggesting that you buy Sangoma. Echo cancellation is in large part their big selling point. |
| 15:30.10 | [TK]D-Fender | docelmo, everything except the 400 can take the latest SIP & BR, yes. |
| 15:30.24 | docelmo | thanks tk |
| 15:30.25 | kippi | ah |
| 15:30.33 | docelmo | you're just full of useful information |
| 15:30.59 | [TK]D-Fender | docelmo, np |
| 15:31.44 | kippi | how can I check what version of zaptel i am runing? |
| 15:31.54 | [TK]D-Fender | nDuff, I didn't say that...... I'm not sdisputing it either though ;) |
| 15:32.09 | [TK]D-Fender | kippi, How do you NOT know what you're running? |
| 15:33.05 | nDuff | [TK]D-Fender: it's consistent with your history. |
| 15:33.12 | nDuff | [TK]D-Fender: granted, I was extrapolating a bit. |
| 15:34.28 | [TK]D-Fender | nDuff, my best statements are the ones I don't even have to make :) |
| 15:34.41 | *** join/#asterisk skyphyr (n=alanj@135.196.58.222) |
| 15:36.07 | skyphyr | hi all - I was wondering if anyone had experience with using ldap for sip accounts in asterisk? There seems to be a couple of ldap related tools in Asterisk, but trying to decipher how they each work. |
| 15:37.12 | *** join/#asterisk ELBunce (n=erik@kde/developer/bunce) |
| 15:40.32 | *** join/#asterisk Lawbringer (n=Lawbring@84-45-215-247.no-dns-yet.enta.net) |
| 15:40.50 | *** part/#asterisk luckyone (n=hidden@CPE-65-28-7-102.kc.res.rr.com) |
| 15:45.21 | *** join/#asterisk qdk (n=qdk@0x50c627be.bynxx11.adsl-dhcp.tele.dk) |
| 15:45.31 | *** join/#asterisk Tako-san (n=Tako-san@216.232.147.102) |
| 15:45.46 | irule | I dial 8500 into voicemailmain, then 0 for voicemail options, 1 to record unavailable message, I speak the message after the beep, then I get 3 options, 1 to accept, 2 to listen to the recorded message and 3 to record again, I press 2, and I dont hear the message, I am just given the same 3 options again! am I missing something? thanks! |
| 15:45.57 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 15:46.27 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 15:49.31 | blitzrage | irule: sounds like your audio either isn't making it to the far end, or Asterisk can't write the file to the HD |
| 15:50.03 | stoffell | anyone knows specific settings to configure kirk sip 600v3 with asterisk? (keep getting failed sip registrations) |
| 15:50.27 | *** join/#asterisk Voicemeup (n=VoiceMeU@modemcable159.131-56-74.mc.videotron.ca) |
| 15:53.07 | irule | blitzrage thanks, so, what is the directory it wants to write to? |
| 15:53.47 | *** join/#asterisk lisandropm (n=lisandro@cpe-22-76.bvconline.com.ar) |
| 15:53.47 | blitzrage | /var/spool/voicemail/... I think |
| 15:53.48 | blitzrage | errr |
| 15:53.48 | blitzrage | /var/spool/asterisk/voicemail |
| 15:53.48 | Krooks | so if I use iax , all I have to do is to forward port 5060, then I'm good ? |
| 15:53.48 | irule | thanks |
| 15:54.02 | blitzrage | Krooks: ummmm... 5060 is the signalling port for SIP |
| 15:54.14 | blitzrage | you shouldn't have to forward anything |
| 15:54.21 | blitzrage | IAX2 uses 4569 |
| 15:54.21 | Krooks | what port do I forwaord for iax ? |
| 15:54.43 | Krooks | so I forward 4569, I'm all good ? |
| 15:55.20 | *** join/#asterisk inspired (n=mikael@ti211310a341-3993.bb.online.no) |
| 15:55.43 | *** part/#asterisk inspired (n=mikael@ti211310a341-3993.bb.online.no) |
| 15:56.02 | blitzrage | <blitzrage> you shouldn't have to forward anything |
| 15:56.12 | blitzrage | is *Asterisk* behind NAT? |
| 15:56.17 | Krooks | yes |
| 15:56.31 | Krooks | thats why I want to forward |
| 15:56.34 | *** join/#asterisk Zeeek (n=IceChat7@pdpc/supporter/active/Zeeek) |
| 15:56.44 | blitzrage | Krooks: try it |
| 15:56.57 | Zeeek | 'sup blitzrage |
| 15:57.05 | blitzrage | Zeeek: nada much muchachos! |
| 15:57.13 | Krooks | nada much hehe |
| 15:57.15 | blitzrage | I'm thinking of making breakfast (it IS noon afterall) |
| 15:57.19 | Zeeek | nachos cheese |
| 15:57.29 | blitzrage | "What kind of cheese is not yours?" |
| 15:57.44 | Zeeek | you are planning to be at the Asterisk TV party, right? |
| 15:57.49 | blitzrage | eh? |
| 15:58.01 | blitzrage | I've been in B.C. for a week |
| 15:58.06 | Corydon76-work | Nor have I |
| 15:58.12 | Zeeek | because if you're not, your preferred membership will be pre-expired |
| 15:58.18 | blitzrage | o i c |
| 15:58.32 | blitzrage | The kind of cheese that is not yours is.... NACHO CHEESE! |
| 15:58.35 | Zeeek | and we all know that preferred membership has its privileges |
| 15:58.50 | blitzrage | (not cho cheese..... not your cheese.... ha. ha......) |
| 15:58.56 | Zeeek | yes,w e got it |
| 15:58.59 | blitzrage | :D |
| 15:59.06 | blitzrage | I like to be obvious |
| 15:59.07 | Zeeek | having lived in various inner cities... |
| 15:59.41 | Zeeek | although the correct form is "aintcho cheese" |
| 15:59.41 | blitzrage | ok, breakfast is a great idea... back in a bit |
| 15:59.51 | Zeeek | watch the cholesterol |
| 16:00.18 | blitzrage | I'm having eggs.... so I'm having LDL |
| 16:00.27 | Zeeek | ~AsteriskTV |
| 16:00.27 | blitzrage | (the good cholesterol) |
| 16:00.36 | irule | <PROTECTED> |
| 16:00.48 | Zeeek | ~AsteriskTV Live video about asterisk |
| 16:02.03 | *** part/#asterisk Phuntom (n=Phuntom@80.233.159.254) |
| 16:03.23 | Krooks | blitzrage: 4569 tcp or udp ? |
| 16:03.34 | blitzrage | always UDP with Asterisk |
| 16:03.45 | blitzrage | Zeeek: links.... etc... ? |
| 16:03.54 | Krooks | ok |
| 16:04.38 | Hmmhesays | ok I have one customer asking about infoline, something about calling in and hearing news. Is this possible? |
| 16:05.02 | Zeeek | blitzrage well, for now it'll be announced on http://Talkathon.org - it's at the same time as the COnference this Friday |
| 16:05.20 | af_ | where could I fond declaration of conformity for Eu of Digium adapters? |
| 16:05.22 | af_ | find |
| 16:05.39 | blitzrage | Hmmhesays: everything is possible with scripting and Asterisk |
| 16:06.03 | Zeeek | cepestral would do that nicely |
| 16:06.37 | Hmmhesays | I'm wondering if there is a service set up that reads the news off of for you |
| 16:06.47 | Zeeek | the iht does it |
| 16:06.53 | Zeeek | iht.com |
| 16:07.21 | Zeeek | http://audionews.iht.com/ |
| 16:08.23 | *** join/#asterisk VOiCi (n=o@132-199.sh.cgocable.ca) |
| 16:08.49 | *** join/#asterisk saftsack (n=saftsack@pD9E04177.dip.t-dialin.net) |
| 16:09.37 | Krooks | Anyone can help me check if my asterisk behind a nat is really working. IP=61.6.47.42 username=user1 and password=1234 extension=1111 . Call me on extension 2222. |
| 16:10.34 | Krooks | IAX |
| 16:10.45 | Krooks | must be iax |
| 16:13.06 | Zeeek | I'll try it |
| 16:13.33 | Krooks | thanks |
| 16:15.57 | Krooks | Zeeek : Can you connect ? You must use iax. |
| 16:15.58 | Zeeek | doesnt appear to work |
| 16:16.07 | Krooks | you using idefisk ? |
| 16:16.10 | Zeeek | yeah |
| 16:17.34 | *** join/#asterisk nighty^ (n=nighty@sushi.rural-networks.com) |
| 16:17.42 | Nephfl | anybody here who is good with provisioning polycom phones |
| 16:17.58 | Zeeek | ask the question |
| 16:19.22 | *** join/#asterisk Andri[DK] (n=andri@hydrogen.calidris.com) |
| 16:20.10 | Andri[DK] | I need some help. I may be too optimistic but I'm trying to link two Asterisk boxes (1.0 and 1.4) together with IAX2 but the old one keeps "hanging up" on the new one. Any advise? |
| 16:20.13 | Zeeek | Krooks what have you done with asterisk to make it work behind NAT? Enumerate the steps if you will |
| 16:20.32 | Krooks | Zeeek : using iax. no nat issues |
| 16:21.01 | rob0 | Andri[DK]: Peace of mind and persistence are your friends. Possibly also "set verbose 9". |
| 16:21.01 | shido6 | lol |
| 16:21.05 | Zeeek | How does an unsolicited packet on 4569 get to the rught box? |
| 16:21.27 | Zeeek | Krooks I couldn't register on your box |
| 16:21.49 | Zeeek | you may need to forward |
| 16:21.57 | shido6 | 4569 UDP |
| 16:22.06 | shido6 | to the internal ip of your ast box |
| 16:22.15 | Zeeek | das what I'da thought |
| 16:22.19 | nohop | hello all :) |
| 16:23.51 | *** join/#asterisk Acidcrawl (n=Miranda@12.168.96.254) |
| 16:24.38 | Acidcrawl | For some reason i'm not getting AGENTLOGIN and AGENTLOGOFF events in my queue_log, is there some configuration file that I need to edit to turn these events on? |
| 16:26.02 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 16:27.01 | *** join/#asterisk linex (n=Blahme@60.52.15.183) |
| 16:27.22 | linex | I'm Krooks |
| 16:27.32 | linex | Krooks a zombie |
| 16:27.39 | linex | Krooks a zombie |
| 16:27.45 | Nephfl | just seems like when i try to change the configuration throught he phone.cfg file, it never takes my settins |
| 16:27.49 | Nephfl | settings |
| 16:28.26 | Nephfl | i have gotten the basic provisioning done that way, but i am trying to get 1 button voicemail and some other settings |
| 16:28.43 | Zeeek | Nephfl how are you setting? Web ? |
| 16:28.51 | Zeeek | oh through the phone.cfg |
| 16:28.57 | Nephfl | yes |
| 16:29.04 | Zeeek | it's on your server and you reboot the phone? |
| 16:29.10 | Nephfl | yes |
| 16:29.54 | Zeeek | I seem to remember some chicken-sacrifice routing between the differentfiles |
| 16:30.12 | rob0 | mmmmmm fried chicken |
| 16:30.12 | Zeeek | but you'll need the help of [TK]D-Fender the resident (evil) expert |
| 16:30.17 | Nephfl | i also need to see if i can get this thing to display the status of the analog trunks |
| 16:30.28 | Zeeek | walk, don't run |
| 16:30.58 | Zeeek | Manxpower would know but he isn't even logged in |
| 16:32.04 | linex | 60.52.15.183 |
| 16:32.12 | linex | Zeeek : 60.52.15.183 |
| 16:32.24 | Zeeek | what? |
| 16:32.35 | Zeeek | is this the magic Polycom incantation? |
| 16:32.44 | linex | what ? |
| 16:32.49 | linex | Its me. Krooks |
| 16:33.05 | Zeeek | oh. load chan_mindreader.so |
| 16:33.14 | linex | I had to reboot the Dlink router because of the port-forward |
| 16:33.39 | Zeeek | you want me to deprogram idefisk again and try this new ip |
| 16:33.56 | linex | everything is the same. Just the IP |
| 16:34.40 | Zeeek | ok looks registered |
| 16:34.43 | linex | kool |
| 16:34.45 | Zeeek | take a look |
| 16:34.46 | linex | I can see you |
| 16:34.51 | linex | call me. |
| 16:34.54 | Zeeek | wow, I better get dressed |
| 16:34.58 | linex | hey wait I call you |
| 16:35.13 | Zeeek | sure try it |
| 16:35.32 | Andri[DK] | So, nobody here has been brave enough to try to link two Asterisk boxes (v1.0 and v1.4) together? :) |
| 16:35.34 | Zeeek | might now work |
| 16:35.49 | Zeeek | Andri[DK] I have 1.2 + 1.4 |
| 16:36.06 | Zeeek | not only did it work, but the wrong machine was answering |
| 16:36.11 | Zeeek | my fault |
| 16:36.13 | Andri[DK] | I've tried verbose, sip and iax debugging but somehow it just looks like the 1.0 box is hanging up on the 1.4 box |
| 16:36.25 | Andri[DK] | Zeeek: hehe |
| 16:36.39 | linex | Zeeek |
| 16:36.44 | Zeeek | I see no reason why it wont work |
| 16:36.48 | Andri[DK] | <PROTECTED> |
| 16:36.48 | Andri[DK] | <PROTECTED> |
| 16:36.48 | linex | Try and call me. |
| 16:36.53 | Zeeek | linex I did |
| 16:36.57 | Zeeek | no route |
| 16:37.01 | Andri[DK] | this is what the 1.4 box says :( |
| 16:37.33 | *** join/#asterisk ygguh2 (n=admin@ool-44c5e3c2.dyn.optonline.net) |
| 16:37.34 | Andri[DK] | could it be because I'm not using any authentication between the boxes? |
| 16:37.41 | Andri[DK] | (just lazy) |
| 16:37.48 | Barmal | who can sell me latest sip firmware for cisco 7970G? |
| 16:38.00 | Zeeek | Andri[DK] maybe you better set up peers or friends |
| 16:38.19 | Andri[DK] | Zeeek: both set up as friends |
| 16:38.33 | Zeeek | gsm |
| 16:38.49 | linex | Zeeek : I nneed to post something |
| 16:38.56 | linex | Can I query you ? |
| 16:38.59 | Zeeek | linex I just called myself and it worked |
| 16:39.04 | Zeeek | But I can't call you |
| 16:39.06 | *** join/#asterisk nDuff (n=ccd@fw2.isgenesis.com) |
| 16:39.08 | nDuff | gah. |
| 16:39.13 | Zeeek | It neeeded to nbe GSM only |
| 16:39.54 | nDuff | ...so, as I was asking... is Festival the best available free-for-commercial-use speech synthesis engine, or does anyone suggest others? |
| 16:40.37 | Zeeek | shit... free it may well be |
| 16:40.57 | Zeeek | linex |
| 16:40.58 | Qwell[] | Barmal: Cisco can. |
| 16:41.18 | Zeeek | I can't leave this like this. Looks like it works, you have a problem with codec |
| 16:41.23 | Barmal | Qwell: Yes but I not have the right account with them |
| 16:41.26 | *** join/#asterisk Krooks (n=Blahme@60.52.15.183) |
| 16:41.29 | Qwell[] | Barmal: Buy one. |
| 16:41.58 | Barmal | Qwell: buy what account? do you have a link? |
| 16:42.15 | Qwell[] | No, Cisco doesn't either. I believe it's pretty much impossible to get. |
| 16:43.38 | Zeeek | speaking of crisco: http://one.revver.com/watch/317970 |
| 16:46.51 | Barmal | qwell: I have 8 phones here as I understand I need to buy 8 firmwares or 1 is enough? |
| 16:46.59 | Qwell[] | You need 8 licenses. |
| 16:47.51 | nohop | is there a way to make asterisk 'try' to Dial another phone if one's not answering ? |
| 16:48.02 | Barmal | Qwell: understood. the problem is where can I buy it :)... |
| 16:49.36 | Krooks | Can anyone help me check out if my asterisk works behind my firewall ? IP=60.52.15.183 , username=user1 password=1234 extension=1111. Call me at 2222. |
| 16:49.38 | nDuff | nohop: sure. |
| 16:49.43 | Krooks | IAX |
| 16:49.47 | nDuff | nohop: read the Dial app's documentation. |
| 16:50.02 | Krooks | Can anyone help me check out if my asterisk works behind my firewall ? IP=60.52.15.183 , username=user1 password=1234 extension=1111. Call me at 2222. Protocol =iax. Must be iax. |
| 16:50.15 | Krooks | Please |
| 16:50.25 | nohop | nDuff: that's a specific document ? where would i find it ? |
| 16:50.46 | Zeeek | show application dial |
| 16:52.58 | nohop | ahh... i think i already found it.... |
| 16:54.26 | *** join/#asterisk tecnico (n=tecnico@24.96.146.69) |
| 16:55.28 | Krooks | If an asterisk is in Japan. An guy from california registers to it and another guy in canada registers to it. Does the conversation goes thru Japan or will it be direct US to Canada ? |
| 16:55.29 | Voicemeup | <-- SIP read from 192.168.1.80:50758: but the contact says port 5060 .. that 1.2.20. |
| 16:55.42 | Voicemeup | nat issue ? or i need to specify port now ? |
| 16:56.05 | Krooks | Voicemeup: use IAX |
| 16:56.09 | Voicemeup | ahah |
| 16:56.10 | Voicemeup | i cant |
| 16:56.12 | *** join/#asterisk Qwell_ (i=qwell@pdpc/sponsor/digium/Qwell) |
| 16:56.12 | *** mode/#asterisk [+o Qwell_] by ChanServ |
| 16:56.16 | Voicemeup | phone is sip |
| 16:56.22 | Voicemeup | its on lan |
| 16:56.34 | Voicemeup | .80 to .7 |
| 16:56.57 | Voicemeup | copying the .17 config direclty didnt work no audio , so im restarting configs from the samples to see |
| 16:57.00 | Krooks | no iax ? |
| 16:57.40 | Krooks | Can anyone help me check out if my asterisk works behind my firewall ? IP=60.52.15.183 , username=user1 password=1234 extension=1111. Call me at 2222. Protocol =iax.Not SIP. Please |
| 16:59.30 | *** join/#asterisk tsurko (n=tsurko@77.70.24.142) |
| 16:59.49 | Voicemeup | Testing 192.168.1.80 with 192.168.0.0 ok so that hmm |
| 17:00.21 | Voicemeup | sip show peer shows 5060 .. but anyteply is renatted for soemrreason |
| 17:00.25 | Voicemeup | its a cisco phone |
| 17:01.40 | ygguh2 | Krooks, calling now |
| 17:03.53 | Voicemeup | ok how can i debug htis more ? |
| 17:04.03 | Voicemeup | anyoen want a pastebin ? |
| 17:04.18 | ygguh2 | Krooks, iaxcomm |
| 17:04.58 | Voicemeup | fixed |
| 17:05.21 | Voicemeup | was a flaw in /.17 that when you had a intenral phone on lan. .you could set nat=yes.. where now it needs to be no |
| 17:07.04 | Krooks | ygguh2: thanks |
| 17:07.16 | ygguh2 | I need some help with server to server IAX communication. Im receiving a regisrtation rejected on serverA and on serverB Im receiving serverA is not dynamic. |
| 17:07.27 | ygguh2 | Krooks, no problem, glad I could help. |
| 17:07.35 | Krooks | Finally sure it works good. |
| 17:07.42 | ygguh2 | besides, you are the first person I've talked to in your country. |
| 17:07.42 | *** join/#asterisk bkruse (i=bkruse@nat/digium/x-73776c90aa5b0547) |
| 17:09.47 | nohop | exten => _X.,1,SetCallerID(00xxx) <=- is that a correct line, of did the syntax change ? (got that from some tutorial) |
| 17:09.49 | Zeeek | Krooks are you in Japan? |
| 17:09.51 | Strom__ | ygguh2: if you know the IP addresses of both servers, you don't need registrations |
| 17:10.04 | Strom__ | nohop: setcallerid is deprecated |
| 17:10.09 | Strom__ | please read upgrade.txt |
| 17:10.26 | Krooks | Can't say the same but never spoken to someone in that Big city |
| 17:10.29 | nohop | i installed asterisk with apt-get... i have no idea where i'd find that readme :) |
| 17:10.52 | Krooks | Zeeek: nope |
| 17:10.52 | Strom__ | nohop: in the source tarball |
| 17:10.54 | nohop | Set(CALLERID(num)=) is what i should use ? |
| 17:11.02 | Strom__ | nohop: yes |
| 17:11.07 | nohop | ok, thanks |
| 17:13.16 | Strom__ | i'd still recommend you read the upgrade.txt for the version you installed |
| 17:14.06 | *** join/#asterisk EmleyMoor (i=phil@topdeck.tinsleyviaduct.com) |
| 17:14.19 | *** part/#asterisk Zeeek (n=IceChat7@pdpc/supporter/active/Zeeek) |
| 17:15.31 | ygguh2 | Strom__, actually, I currently have three asterisk servers around the US. Since they are all on different networks, without a VPN or tunnel, I thought I needed to reg each other. |
| 17:16.12 | Strom__ | no |
| 17:16.27 | Strom__ | registration is only for situations where one is on a dynamic IP |
| 17:16.49 | Strom__ | registration does nothing more than say "HEY THERE! HERE'S MY IP ADDRESS!" |
| 17:17.01 | ygguh2 | okay. let me work on it without the reg and set the host to the IP address in each iax.conf. |
| 17:17.10 | ygguh2 | lol, nice. |
| 17:18.32 | *** join/#asterisk jm|laptop (n=jm|home@cpc1-papw3-0-0-cust17.cmbg.cable.ntl.com) |
| 17:19.45 | ygguh2 | in the iax.conf, should the type= peer,friend,user or what? |
| 17:20.19 | Strom__ | peer is for calls outbound from your server |
| 17:20.23 | Strom__ | user is for calls inbound |
| 17:20.30 | Strom__ | friend is for both |
| 17:21.04 | ygguh2 | so Ill use type=friend |
| 17:21.05 | ygguh2 | thanks |
| 17:21.43 | nohop | strom: i'll check that one out sometime :) |
| 17:21.46 | ygguh2 | SIP is easy compared to this ! |
| 17:21.48 | Strom__ | this is all documented in the sample file, you know ;) |
| 17:21.50 | nohop | gotto go again now though... thanks anyways :) |
| 17:22.03 | Strom__ | ygguh2: iax is easier than sip, TBH |
| 17:22.40 | Strom__ | iax tends to just work while sip tends to misbehave like an annoying child |
| 17:23.27 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-88-153-47-145.red.bezeqint.net) |
| 17:23.31 | ygguh2 | lol, I'be dealing with SIP being annoying for over a year. while Im completely new to IAX. I guess Im so used to issues that I cant figure out a simple config anymore. |
| 17:24.30 | Strom__ | read the comments in the sample file |
| 17:25.04 | *** join/#asterisk Jingles (n=dfbarth@39.183.dowl.anc.borealisbroadband.net) |
| 17:26.39 | *** join/#asterisk matsk (n=mk@83.233.97.210) |
| 17:26.53 | *** join/#asterisk falz (n=falz@proxy.supranet.net) |
| 17:26.58 | *** join/#asterisk Mercestes (n=Merceste@rrcs-71-41-157-70.sw.biz.rr.com) |
| 17:27.17 | ygguh2 | believe it or not, which I dont think you will, I have read the comments in the iax.conf.sample. |
| 17:27.48 | ygguh2 | but, anywho, your advise appears to work, and Im communicating between two of the three server now. |
| 17:28.30 | Strom__ | :) |
| 17:28.56 | *** join/#asterisk eliyahud (n=eliyahud@ool-182f9fe7.dyn.optonline.net) |
| 17:30.54 | EmleyMoor | Anyone else having problems with echo on BT should try rxgain=3.0 and txgain=-9.0 on the fxo port |
| 17:31.52 | *** join/#asterisk agile (n=mike@68-189-222-201.dhcp.ftwo.tx.charter.com) |
| 17:31.58 | *** join/#asterisk zael (n=zael@unaffiliated/zael) |
| 17:32.20 | *** part/#asterisk zael (n=zael@unaffiliated/zael) |
| 17:34.30 | falz | I'm beating my head. am I understanding zapata.conf correctly- I've got multiple FXO's in zapata.conf I have areas defined for them, where channel=FOO is assigned to a specific context. |
| 17:35.03 | falz | in one case, I have strange behavior where if someone calls on that #, it goes to that context, but somehow "forks" the call to also go to another, completely unrelated context that is also in zapata.conf |
| 17:35.05 | ygguh2 | beautiful, nice, works great now. |
| 17:35.12 | ygguh2 | Strom__, thanks for you help. |
| 17:35.26 | ygguh2 | now, can you convince my boss to puchase three new servers? |
| 17:36.01 | bkruse | yes, Strom__ can |
| 17:36.31 | ygguh2 | bkruse, thansk. Strom__ let me get you his name and number |
| 17:36.34 | ygguh2 | hehehehe |
| 17:36.42 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@cpe-75-84-232-194.socal.res.rr.com) |
| 17:36.42 | *** join/#asterisk kclaussen (n=kclausse@204.13.224.242) |
| 17:37.03 | Strom__ | i'm superstro |
| 17:37.04 | Strom__ | m |
| 17:37.21 | bkruse | jbot_: is this really you? |
| 17:38.02 | EmleyMoor | Next job for me - and it will probably have to wait for an upgrade, is to change over to using dring to detect faxes, rather than just seeing if the fax machine answers. |
| 17:38.06 | Corydon76-work | bkruse: see the ident. It's really the bot |
| 17:38.41 | Mercestes | ~mercestes |
| 17:38.42 | jbot | mercestes is definitely a total nub |
| 17:38.49 | Mercestes | Yup, it's definately him. |
| 17:39.05 | Strom__ | there's on "a" in "definitely" |
| 17:39.39 | Qwell[] | huh? |
| 17:39.41 | Mercestes | let me put on my dyslexia translation glasses. |
| 17:40.12 | Qwell[] | http://d-e-f-i-n-i-t-e-l-y.com/ |
| 17:40.44 | Mercestes | L M A O |
| 17:40.51 | Mercestes | zomg, I love it. |
| 17:41.36 | *** join/#asterisk anonymouz666 (n=anonymou@re366.compuland.com.br) |
| 17:42.11 | anonymouz666 | [TK]D-Fender! |
| 17:42.55 | eliyahud | how do I determine where bad call quality is coming from? |
| 17:43.26 | EmleyMoor | eliyahud: What kind of bad call quality? |
| 17:43.48 | eliyahud | sounds like bubbles or underwater |
| 17:45.29 | *** join/#asterisk naitram (n=ttech@216.77.58.40) |
| 17:45.31 | *** join/#asterisk kirberich (n=robert@i538704D4.versanet.de) |
| 17:45.48 | *** part/#asterisk mardum (n=mardum@71-81-73-42.dhcp.stls.mo.charter.com) |
| 17:47.48 | falz | so, has anyone seen two FXO's pick up a second or so apart when someone calls? somehow zap channel 12 picks up, which is supposed to, but something "forks" and sends the caller to channel 5 |
| 17:48.19 | Strom__ | falz: pastebin zapata.conf and extensions.conf |
| 17:48.36 | *** part/#asterisk naitram (n=ttech@216.77.58.40) |
| 17:49.36 | falz | here's zapata.conf: http://pastebin.ca/600173 |
| 17:50.33 | *** join/#asterisk tdonahue-laptop (n=tdonahue@vonmail.vonworldwide.com) |
| 17:50.39 | tdonahue-laptop | hi all\ |
| 17:50.56 | tdonahue-laptop | i'm trying to figure out what codec a channel is using using show channel <channel> |
| 17:50.58 | falz | however, the problem arizes before it even hits extensions.conf, or so it seems. it shows -- Starting simple switch on 'Zap/12-1' && -- Starting simple switch on 'Zap/5-1' before it even shows me debug info on the extensions.conf stuff |
| 17:50.58 | Strom__ | you don't need to keep re-specifying signalling if they're all fxs_ks |
| 17:51.10 | tdonahue-laptop | is 'NativeFormat' the codec being used? |
| 17:51.22 | falz | Strom__: yea, I realized that a few minutes ago re-reading docs. but that shouldn't hurt anything like this |
| 17:51.27 | Strom__ | falz: these are all analog lines? |
| 17:51.34 | falz | Strom__: yep. |
| 17:51.37 | Strom__ | what kind of card are you using? |
| 17:51.40 | Strom__ | and who is your provider? |
| 17:51.46 | falz | dtm400p and tdm800p |
| 17:52.04 | falz | provider, as in analog? centurytel, but that shouldn't matter. |
| 17:52.10 | falz | I'm trying to determine if somehow there's a wiring problem |
| 17:52.25 | falz | although the wiring hasn't been touched for a year |
| 17:52.31 | Strom__ | as in you wired two of the circuits together somehow? |
| 17:52.40 | falz | that's how it's acting, but I don't see how it's possible |
| 17:52.49 | Strom__ | well, try this - unplug channel 5 and see if the problem persists |
| 17:52.58 | falz | yea, that's probably smart :) |
| 17:53.11 | Strom__ | if it doesn't, then attach your buttset to channel 5 and see what happens when you call channel 12 |
| 17:53.31 | Strom__ | er, attach your buttset to the circuit, not to the card :) |
| 17:55.04 | eliyahud | when my termination provider returns "Everyone is busy/congested at this time', that means its a problem on their end, right? |
| 17:55.52 | Strom__ | eliyahud: depends |
| 17:56.08 | falz | Strom__: happens even w/ channel 5 unplugged, so at least it's not a wiring problem. |
| 17:56.10 | Strom__ | who's your provider, what kind of circuit are you using, and under what conditions are you receiving that message |
| 17:56.10 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 17:56.17 | drako | asterisk 1.4 does not compile on FreeBSD |
| 17:56.27 | falz | drako: ports? |
| 17:56.39 | Strom__ | falz: ok, so pastebin extensions.conf |
| 17:56.39 | drako | falz, no |
| 17:56.53 | falz | drako: ports is not 1.4.x? |
| 17:57.07 | eliyahud | i have an iax provider, i'm using it for an outbound call center |
| 17:57.15 | falz | Strom__: this will be fun. (I'll put all of the split up files back together) |
| 17:57.31 | drako | falz, wtf... didn't know they upgraded to 1.4 |
| 17:57.33 | EmleyMoor | eliyahud: Has it ever worked? |
| 17:58.11 | eliyahud | yes, its just today that it's been very erratic |
| 17:58.11 | Strom__ | eliyahud: now the third part of my question: under what conditions are you receiving that message? |
| 17:58.11 | EmleyMoor | It probably is their problem then - especially if the numbers are reachable in other ways |
| 17:58.37 | Andri[DK] | if you have multiple ways of making outside calls, is there any way of queuing them in a priority order ... like a fancy way without all the getchannel thingy |
| 17:58.47 | eliyahud | as far as I can tell, under completely normal conditions of server load in terms of bandwith and cpu |
| 17:58.59 | Strom__ | eliyahud: uh, no, forget server load |
| 17:59.04 | Strom__ | think in terms of call traffic |
| 17:59.15 | Strom__ | specific destinations you're trying to reach |
| 17:59.19 | Strom__ | number of concurrent calls |
| 17:59.21 | Strom__ | etc etc |
| 17:59.27 | EmleyMoor | Andri[DK]: You could use ChanIsAvail and find the first available channel in the order you want |
| 17:59.27 | Strom__ | etc |
| 17:59.29 | eliyahud | about 50 simultaneous outbound calls |
| 17:59.37 | EmleyMoor | (I do - cheapest first for the call situation) |
| 18:00.04 | EmleyMoor | (It even came into play the other week when we both phoned our mothers at the same time) |
| 18:00.06 | Strom__ | you don't even need to mess with ChanIsAvail; just have cascading Dial() commands |
| 18:00.26 | Strom__ | eliyahud: and only when you have 50ish calls up? |
| 18:00.53 | EmleyMoor | Strom__: Do you mean if a Dial command finds its channel busy it moves on automatically to the next priority, or to the n+101th? |
| 18:01.00 | eliyahud | no, actually it normally handles this many calls no problem.. it seemed to just start doing this today |
| 18:01.15 | Strom__ | EmleyMoor: that n+101 shit was deprecated ages ago |
| 18:01.30 | Strom__ | EmleyMoor: you do conditional branching based on the value of DIALSTATUS |
| 18:01.43 | EmleyMoor | I will try and work that out |
| 18:01.48 | Barmal | Shit! IS IT somehow possible for a regular guy to get or buy 7970 sip firmware... Called cisco, called other ciso $, called 4 partners nobody can help... Got to buy some service agreement from parner then they can tell me how i can get a firmware... some kind of BS or I do not understand anything??? |
| 18:02.15 | Strom__ | Barmal: the reseller /should/ sell you just a license |
| 18:02.29 | Nugget | nobody wants to sell you a service agreement on a single phone. it's hours of headache and cisco paperwork and their commission on the $8 sale doesn't cover the grief. |
| 18:03.10 | *** join/#asterisk ToyMan (n=Stuart@24-148-108-186.ip.mhcable.com) |
| 18:03.17 | Barmal | so on other words it is immpossible to get sip firmware if you a regular guy not a company and got a phone from ebay? |
| 18:03.30 | Strom__ | Barmal: no, it's not impossible. stop panicking |
| 18:03.31 | Nugget | pretty much, yes. |
| 18:03.40 | EmleyMoor | Strom__: Is there a good example available for a "use first available out of three in given order" using it? |
| 18:03.59 | Strom__ | EmleyMoor: no, but it's not exactly rocket science to code |
| 18:04.01 | Strom__ | Dial() |
| 18:04.18 | Strom__ | Gotoif(DIALSTATUS = CONGESTION) |
| 18:04.19 | Strom__ | etc |
| 18:04.21 | falz | Strom__: ok, extensions.conf ( http://pastebin.ca/600198 ) to go along with my zapata.conf ( http://pastebin.ca/600173 ) |
| 18:04.33 | Barmal | strom: can you recommend a resseler? |
| 18:04.36 | falz | so calls coming into channel 12 somehow set zap5/1 off at the same time |
| 18:04.42 | EmleyMoor | If you've gone to s-CHANUNAVAIL and you do a Dial in that also, what happens? |
| 18:05.07 | Strom__ | falz: jeeeez, man, you know you can just number your priorities with "n" and not have to individually have sequential numbers, right? |
| 18:05.18 | falz | apparently not! |
| 18:05.20 | EmleyMoor | Does s-CHANUNAVAIL become the extension? |
| 18:05.24 | Strom__ | EmleyMoor: you're not required to do that |
| 18:05.25 | falz | that's a good reason to rewrite. |
| 18:05.41 | falz | perhaps that "feature" is newer than asterisk 1.0.x? this dialplan has been ported since that |
| 18:05.44 | Strom__ | EmleyMoor: you can goto anywhere |
| 18:05.49 | Strom__ | falz: yes |
| 18:05.50 | EmleyMoor | Strom__: But whatever is gone to becomes the extension, yes? |
| 18:05.52 | falz | that would actually save lots of headaches if a # is missed. |
| 18:05.55 | Strom__ | EmleyMoor: yes |
| 18:06.39 | *** join/#asterisk [[blah]asfd (n=ckwall@c-71-195-196-198.hsd1.ut.comcast.net) |
| 18:07.05 | *** join/#asterisk adorah (n=Michael@87.69.57.73.cable.012.net.il) |
| 18:07.33 | [[blah]asfd | what could possibly cause an error like this: http://pastebin.ca/600202 when it happens I drop all calls. |
| 18:08.06 | Strom__ | falz: hmm, i'm honestly not sure what the problem might be |
| 18:08.40 | Strom__ | you're /sure/ the line you unplugged was the one plugged into channel 5? |
| 18:08.57 | falz | Strom__: yea, it shows zap/12-1 and zap /5-1 answering before it even executes anything in extensions.conf, unless debug output is way lagged. |
| 18:08.59 | EmleyMoor | Strom__: If I want to implement this, presumably I set priorityjumping=no? |
| 18:09.07 | *** join/#asterisk gardo (n=gardo@203.82.42.106) |
| 18:09.07 | Strom__ | EmleyMoor: yes |
| 18:09.23 | falz | yes, it was in #5 in the card, and I had put a stickie in it when it was plugged in as #5, so about as sure as can be. |
| 18:09.27 | Strom__ | hmm |
| 18:09.35 | Strom__ | i'd consider calling digium support |
| 18:09.39 | falz | yea. |
| 18:09.45 | falz | any additional zap debugging you can think of? |
| 18:09.47 | Strom__ | no |
| 18:09.48 | falz | doesnt seem to be any really |
| 18:10.02 | tdonahue-laptop | in the output from show channel <channel name> is 'NativeFormat' the codec being used? |
| 18:10.22 | falz | that exten=> s,n,XXX will be helpful, but I still have to port some goto commands to their own contexts before I can do that |
| 18:10.40 | Strom__ | falz: you can also label your priorities |
| 18:11.05 | Strom__ | so you can have s,n(label),NoOp() and then s,n,Goto(label) |
| 18:11.12 | falz | interesting |
| 18:11.23 | EmleyMoor | Are labels case-sensitive? |
| 18:11.37 | falz | I think I'll just use contexts for that anyhow. the one I pasted is really in 4 files, so it's easier to read |
| 18:11.39 | Strom__ | i dont recall, but i've always treated them as such |
| 18:11.49 | falz | yea, no reason to consider something insensitive |
| 18:11.53 | falz | will just bork you later. |
| 18:12.31 | falz | I'm going to re-verify that extension 5 is what I think it is by caalling it while unplugged. brb. |
| 18:13.28 | *** join/#asterisk __DAW (n=_DAW@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 18:16.11 | *** join/#asterisk Strom_M (n=pocketir@m150e36d0.tmodns.net) |
| 18:16.15 | falz | Strom__: bingo. my labels were wrong! what I had labelled as '5' was actually '1' on the 8 port card. |
| 18:16.32 | Strom_M | heh |
| 18:16.45 | falz | so, unplugging zap/5-1 DOES keep it on zap/12-1, although it's still acting strange. wiring wiring wiring. |
| 18:16.49 | Strom_M | now its time to hunt down the wiring fault |
| 18:16.52 | falz | telco must have f'd something up somehow |
| 18:17.03 | falz | yea. we ordered a new pots line for another customer last week, must be related to that somehow |
| 18:18.13 | blitzrage | EmleyMoor: congrats! |
| 18:18.17 | blitzrage | welcome to 2003 :) |
| 18:18.26 | Strom_M | hahahahaha |
| 18:18.40 | blitzrage | actually.... probably more like 2004.... early 2005 |
| 18:19.08 | *** join/#asterisk Capps- (n=andrew@67-67-242-2.ded.swbell.net) |
| 18:20.19 | Capps- | is it possible to configure asterisk with a gxp2000 to make it so that if person A's phone rings, person B can grab the line? if so, does anyone know of good documentation that could explain how to set this up? |
| 18:21.29 | __DAW | Capps-: http://www.voip-info.org/wiki/view/Asterisk+callgroups+and+pickupgroups |
| 18:26.38 | *** join/#asterisk Defraz (n=t0tal@24-116-152-177.cpe.cableone.net) |
| 18:28.34 | EmleyMoor | Has anyone written an emergency call routine that clears non-emergency calls from a channel if needed and doesn't use a "+101"? |
| 18:29.54 | EmleyMoor | I have one that does use a "+101" but it may take me a while to work out a way round it - so I may have to live with that one |
| 18:30.46 | rob0 | Captain Oveur, I have an emergency call for you from a Mr. Hamm on line 5. |
| 18:31.04 | rob0 | Okay, give me Hamm on five, hold the Mayo. |
| 18:33.31 | *** join/#asterisk maduxi (n=skdbjf@89.129.156.169) |
| 18:34.38 | EmleyMoor | When Dial exits due to a hangup, what happens next in the dialplan? |
| 18:35.49 | maduxi | hi! im configuring an asterisknow beta6 machine, how can i add a simple voicemail, not attached to any extension? just hi, there my enterprise, let ur message! |
| 18:36.19 | *** join/#asterisk oej (n=olle@apollo.webway.se) |
| 18:38.46 | Strom__ | EmleyMoor: depends on who hangs up |
| 18:39.01 | EmleyMoor | Strom: OK - if the person inside hangs up? |
| 18:39.09 | EmleyMoor | (recipient) |
| 18:39.09 | Strom__ | "inside"? |
| 18:39.24 | Strom__ | it depends on who the calling party is |
| 18:39.34 | EmleyMoor | If the called party hangs up, what happens? |
| 18:39.38 | Strom__ | if the calling party hangs up, execution either dies or goes to the h extension |
| 18:39.39 | *** join/#asterisk dmorgga (n=dmorgga@c-24-99-16-65.hsd1.ga.comcast.net) |
| 18:40.00 | Strom__ | if the called party hangs up, execution dies, goes to h, or can continue based on options passed to Dial() |
| 18:41.23 | *** join/#asterisk Voicemeup (n=VoiceMeU@modemcable159.131-56-74.mc.videotron.ca) |
| 18:41.26 | Voicemeup | j #openser |
| 18:41.58 | Strom__ | you fail at IRC |
| 18:42.05 | bkruse | gg |
| 18:42.08 | EmleyMoor | Strom: So, if called party hangs up, it dies if the "g" option was not used? |
| 18:42.26 | *** join/#asterisk oej (n=olle@apollo.webway.se) |
| 18:42.26 | Strom__ | EmleyMoor: or it goes to the h extension if such an extension exists in that context |
| 18:46.34 | MrWup | im confused |
| 18:46.49 | *** join/#asterisk Spiked (n=spiked@spiked.nl) |
| 18:47.16 | MrWup | im using misdn. normally the callerid works fine, if a callerid is available, but say the caller withholds their callerid by typing 141 before dialling my number... it shows up as unknown in asterisk |
| 18:47.44 | MrWup | this is ok, because when i see it says unknown, my asterisk should set callerid(name) to ID-XXX where XXX is an id number |
| 18:47.46 | MrWup | thing is... |
| 18:48.12 | MrWup | even if i set CALLERID(name) = "ID-123" in the dialplan, before dialling one of the SIP extensions, it still shows up as "unknown" on the SIP's display screen |
| 18:48.22 | MrWup | its like when it is unknown nothing can set it to anything else |
| 18:48.28 | MrWup | anyone know what might be wrong? |
| 18:49.42 | *** join/#asterisk martin[ug] (n=martin@gewus.de) |
| 18:50.01 | martin[ug] | hi |
| 18:51.01 | martin[ug] | is there a way to keep the original callernum with external call forwarding? |
| 18:51.02 | EmleyMoor | MrWup: SetCallerPres(allowed) |
| 18:51.34 | EmleyMoor | Had that problem myself |
| 18:52.52 | *** join/#asterisk yonahw (n=yonahw@IGLD-83-130-49-41.inter.net.il) |
| 18:55.01 | MrWup | thanks emleymoor works fine =] |
| 18:55.11 | falz | Strom__: my telco was forwaridng calls from one # to another, with my issue earlier (zap5+zap12 being mixed). strangest thing ever. |
| 18:55.20 | *** join/#asterisk koosroos (n=koosroos@196.209.244.207) |
| 18:55.28 | Barmal | Storm_: you said it is not impossible... I called 17 cisco partners so far nobody can help. Lookes like it is impossible... Nobody wants to fool with selling service agreement and a license is $250 but it is for call manager and nobody knows if it will give me ability to download the file... |
| 18:55.35 | ez` | how to register a sip phone manual ? |
| 18:56.14 | koosroos | hi storm i see you are a regular here, third time around and your still doing your thing! |
| 18:58.02 | ThoMe | hello.! |
| 18:58.10 | ThoMe | martin[ug]: hi. kennst duu dich mit snom aus? ;) |
| 18:58.13 | koosroos | hi ThoME |
| 18:58.19 | ThoMe | koosroos: hiho |
| 18:58.32 | koosroos | what's up |
| 18:59.07 | Krooks | If an asterisk box is in Japan. An guy from california registers to it and another guy in canada registers to it. Does the conversation goes thru Japan or will it be direct US to Canada ? |
| 18:59.26 | ThoMe | how i can use the led of my snom phone? if i recieve a call? |
| 19:01.27 | *** part/#asterisk Spiked (n=spiked@spiked.nl) |
| 19:01.30 | koosroos | wiill go thru japan |
| 19:01.47 | koosroos | what phone are you using ThoMe |
| 19:01.49 | koosroos | ? |
| 19:02.01 | JerJer | what is the Asterisk 1.4 command to turn on DTMF digit debugging ? |
| 19:02.02 | ThoMe | koosroos: snom 360/370 |
| 19:02.30 | *** join/#asterisk marchon (n=marchon@dsl092-071-002.bos1.dsl.speakeasy.net) |
| 19:02.32 | Krooks | koosroos: so the asterisk at Japan could record the conversation if it wanted to, yes ? |
| 19:02.49 | koosroos | ThoMe: you can program your phones led individualy |
| 19:02.56 | koosroos | Krooks :yes |
| 19:03.24 | ThoMe | koosroos: yes. bit how i can use this subscrib function? |
| 19:03.28 | Krooks | Wow, the technology today |
| 19:03.32 | Innatech | has anyone else experienced a ~.5 second latency in using the * paging system? |
| 19:03.36 | koosroos | ThoMe: do you have the phone working with asterisk? |
| 19:04.04 | ThoMe | koosroos: jepp. i can calling with them. |
| 19:04.31 | Krooks | how do I keep asterisk running and get out of the CLI ? |
| 19:04.41 | Innatech | krooks: quit |
| 19:05.01 | Krooks | oh . thnaks |
| 19:05.05 | Innatech | np :) |
| 19:05.15 | koosroos | go to internet explorer and tipe in the phone IP into the http://XXX.XXX.XXX.XXX then ling to th ephone there you can set the phone up |
| 19:06.31 | tzafrir_laptop | JerJer, something new and specific to 1.4? |
| 19:06.36 | Krooks | Ok, if user1 and user2 are talking. How can user3 join in the conversation ? |
| 19:07.04 | ThoMe | koosroos: oh, no its works! |
| 19:07.04 | EmleyMoor | It all works :-) |
| 19:07.19 | ThoMe | huihui |
| 19:07.21 | ThoMe | :-) |
| 19:07.21 | tzafrir_laptop | JerJer, log debug and dtmf to a log file, set debug 0, and tail -f that file | grep DTMF |
| 19:09.43 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 19:10.22 | *** join/#asterisk DJ_Kit (n=lamass@83.149.52.8) |
| 19:10.26 | *** part/#asterisk marchon (n=marchon@dsl092-071-002.bos1.dsl.speakeasy.net) |
| 19:10.29 | DJ_Kit | Hi guys |
| 19:10.38 | DJ_Kit | could anybody test my server |
| 19:10.38 | Strom__ | whee |
| 19:10.48 | DJ_Kit | I can't connect with him... |
| 19:10.51 | Strom__ | ...what do you mean "test my server"? |
| 19:10.55 | JerJer | tzafrir_laptop: ok that |
| 19:11.01 | DJ_Kit | it wrotes: connection timout |
| 19:11.05 | DJ_Kit | :( |
| 19:11.23 | DJ_Kit | i wrote sip.conf, ext..conf and other |
| 19:11.37 | DJ_Kit | but my xlite could'nt connect to him |
| 19:11.40 | DJ_Kit | :( |
| 19:11.57 | DJ_Kit | can anybody help me |
| 19:12.03 | Strom__ | DJ_Kit: well...ask a specific question and give us all the information we need to tell you what the problem is and how to go about fixing it |
| 19:12.10 | DJ_Kit | (default installation under freeBSD) |
| 19:12.11 | *** part/#asterisk [[blah]asfd (n=ckwall@c-71-195-196-198.hsd1.ut.comcast.net) |
| 19:12.24 | Jingles | connection timeouts usually mean 'firewalls' |
| 19:13.46 | DJ_Kit | :( |
| 19:14.25 | DJ_Kit | any other reasons? |
| 19:14.39 | *** join/#asterisk Jabroni (n=asskick@red-corp-201.143.143.138.telnor.net) |
| 19:14.45 | Jingles | asterisk isn't running, so there's nothing listening on the port the xlite is trying to reach. |
| 19:14.45 | DJ_Kit | just try to connect him test.bgteam.ru:5060 |
| 19:14.49 | DJ_Kit | username: 200 |
| 19:15.17 | DJ_Kit | secret=user1_password |
| 19:15.35 | DJ_Kit | Jingles, no it's running |
| 19:15.45 | Krooks | how do i tell asterisk to record every conversation ? |
| 19:15.47 | Jingles | fair enough. you asked for other reasons - I gave one. |
| 19:15.58 | DJ_Kit | when i send message (via peer to peer mode) it say that there is no user |
| 19:15.59 | DJ_Kit | ;) |
| 19:17.44 | nDuff | Krooks: see the Monitor application. |
| 19:18.29 | Krooks | the cli ? |
| 19:18.49 | nDuff | Krooks: no. |
| 19:19.23 | nDuff | Krooks: but in the cli, you can run "show application Monitor" to see the relevant documentation. |
| 19:20.03 | Krooks | oh thanks |
| 19:20.09 | Krooks | lemme try it now |
| 19:21.26 | *** join/#asterisk marchon (n=marchon@dsl092-071-002.bos1.dsl.speakeasy.net) |
| 19:22.22 | marchon | Question: Is there a problem in older versions of asterisk (1.0-rc2) in detecting dtmf tones in machines without a timing source? |
| 19:22.50 | marchon | or does a timing source have nothing to do with it? |
| 19:22.56 | Strom__ | 1.0 RC2?! |
| 19:23.00 | marchon | or is it just a problem with asterisk. |
| 19:23.01 | Strom__ | that's....three years old! |
| 19:23.21 | marchon | yes - know it is 3 years old. |
| 19:23.28 | Strom__ | please please please please please please please upgrade |
| 19:23.42 | marchon | that is what I keep saying.... |
| 19:23.57 | Mercestes | I wouldn't go much past 1.0.9. |
| 19:24.00 | Mercestes | The rest is just trash. |
| 19:24.01 | marchon | have 1.4 on another server - but |
| 19:24.04 | Mercestes | Clean on up to 1.4.5 |
| 19:24.06 | nohop | is there anyone who has got callerID working on outbound calls in asterisk using sip.voipbustrer.com ? |
| 19:24.25 | Mercestes | and then I hear 1.4.5 is happiness again. |
| 19:24.34 | Mercestes | I should try it one of these days |
| 19:27.27 | nDuff | DJ_Kit: If I were you, I'd try going through the generic diagnostic steps appropriate for any kind of network connection failure. Use netstat --listen to make sure the port is open on your server; make sure there aren't any firewall rules blocking it; try using tcpdump/ethereal to see what's going on. |
| 19:27.48 | *** join/#asterisk ToyMan (n=Stuart@user-12lcqol.cable.mindspring.com) |
| 19:29.58 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 19:31.56 | *** part/#asterisk EricL (n=eric@clydesdale.linkexperts.com) |
| 19:32.55 | EmleyMoor | Is there a definitive list of what's deprecated in 1.2? How about 1.4? |
| 19:33.07 | EmleyMoor | (Apps/variables/functions) |
| 19:33.22 | Strom__ | EmleyMoor: yes |
| 19:33.25 | Strom__ | UPGRADE.TXT |
| 19:33.40 | Strom__ | and all of the other documentation that comes with the asterisk source |
| 19:36.56 | irule | how may I react to an applications' return value? most aps say they return 0 or 1 depending on status |
| 19:38.24 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 19:38.24 | *** mode/#asterisk [+o blitzrage] by ChanServ |
| 19:39.03 | *** join/#asterisk marchon (n=marchon@dsl092-071-002.bos1.dsl.speakeasy.net) |
| 19:39.07 | *** join/#asterisk rooskoos (i=koosroos@196.211.34.2) |
| 19:42.14 | *** join/#asterisk yonahw (n=yonahw@IGLD-83-130-49-41.inter.net.il) |
| 19:43.05 | Mercestes | irule: Depends on if the return value is what you want or not. I usually curse, swear, and shake my fist. |
| 19:47.57 | rob0 | offtopic, but perhaps of interest to some, Slackware 12.0 is released today. |
| 19:48.29 | Strom_M | ooh |
| 19:48.36 | Strom_M | is this the best 1999 has to offer now?? |
| 19:48.37 | Mercestes | Slackware is a gateway drug |
| 19:48.50 | Qwell[] | 1999 hell |
| 19:49.09 | Strom_M | I remember slackware 11 was all the best 1997 had to offer |
| 19:49.21 | Qwell[] | I hear they're just getting X11r4 |
| 19:49.32 | rob0 | haha |
| 19:49.52 | Mercestes | now now, remember the tortoise and the hare. Slow and steady wins the race. |
| 19:49.54 | bkruse | dont make fun of pari like that |
| 19:51.10 | *** join/#asterisk eatmypiano (n=eatmypia@host86-141-133-193.range86-141.btcentralplus.com) |
| 19:51.53 | Mercestes | The important thing is it's *stable*. |
| 19:52.16 | Mercestes | they had to test it throughly before they released it |
| 19:52.22 | nDuff | Mercestes: There's a rock sitting on my desk. It's pretty stable too. |
| 19:52.35 | Strom_M | like asterisk 1.0? |
| 19:52.50 | Mercestes | Strom_M: I'd say that's a good analogy. :P |
| 19:53.37 | Mercestes | In about 5 years they'll introduce beryl and aiglxgl |
| 19:54.21 | marchon | was that asterisk 1.0 comment aimed at me? |
| 19:54.31 | Mercestes | No, it was aimed at me. |
| 19:54.43 | *** join/#asterisk PierreY (n=pierre@86.66.233.142) |
| 19:54.55 | PierreY | Hi |
| 19:55.21 | Capps- | is there a good pre-made documentation template to document settings that have been applied to ones asterisk server? |
| 19:55.41 | Capps- | or does anyone in here have one they have created that they feel is useful and is willing to allow me to use? |
| 19:56.09 | Corydon76-work | Capps-: that sounds fully buzzword compliant. I have no idea what you just said. |
| 19:57.23 | Capps- | Corydon76-work: looking for a pre-made word document to put all of my asterisk settings into for future use. |
| 19:57.40 | Corydon76-work | Capps-: uh, why?!! |
| 19:57.42 | *** join/#asterisk marchon (n=marchon@dsl092-071-002.bos1.dsl.speakeasy.net) |
| 19:57.44 | marchon | hi - |
| 19:57.48 | Strom_M | he wants a form to fill out |
| 19:57.49 | Corydon76-work | MS Word does you no good at all |
| 19:57.56 | nDuff | ...a RCS and the config files themselves seem fully adequate to track and describe changes. |
| 19:57.58 | Capps- | Corydon76-work: the company i work for likes documentation. |
| 19:57.59 | marchon | I was trying to ask - before I got disconnected --- |
| 19:58.00 | PierreY | Do you know if I can use a Firebird DBMS backend with Asterisk ? |
| 19:58.01 | Corydon76-work | I'd suggest Subversion |
| 19:58.09 | Qwell[] | PierreY: if there are odbc connectors for it, sure |
| 19:58.09 | nDuff | ugh. |
| 19:58.17 | Corydon76-work | source-control your config files |
| 19:58.24 | marchon | if I am using asterisk 1.0rc2 - and having agi dtmf decoding issues - will a 1.4 upgrade help |
| 19:58.26 | PierreY | I would prefer a native connector |
| 19:58.27 | *** join/#asterisk voipnet-tech (n=voipnet-@rrcs-24-97-250-50.nys.biz.rr.com) |
| 19:58.31 | Corydon76-work | nDuff: bizarre |
| 19:58.35 | Qwell[] | marchon: umm |
| 19:58.38 | Qwell[] | marchon: seriously? |
| 19:58.45 | marchon | yes - seriously. |
| 19:58.50 | Strom_M | Qwell[]: it's time machine day |
| 19:58.50 | marchon | I have a demo tomorrow |
| 19:58.55 | PierreY | and... I'm ready to write or pay for this connector ;-) |
| 19:59.00 | Qwell[] | marchon: start from scratch |
| 19:59.08 | Qwell[] | 1.0 > 1.4 WILL NOT go well |
| 19:59.29 | marchon | and I do not know |
| 19:59.35 | nDuff | Corydon76-home: I like to be able to do things like renaming files twice between commits, or being able to commit from my laptop disconnected, or (for open source projects) allowing people to start their own branches without giving them write access to my server. |
| 19:59.35 | voipnet-tech | hi all... I can make asterisk run a PHP script when a call is placed, then route the call to a specific trunk based on what the PHP script returns? |
| 19:59.42 | Qwell[] | PierreY: I'd be surprised if one didn't exist |
| 19:59.54 | PierreY | that don't exists :-( |
| 20:00.05 | nDuff | Corydon76-home: Subversion is a helluva lot better than CVS, but for folks who cut their teeth on BitKeeper and Arch, it's severely limiting. |
| 20:00.08 | Corydon76-work | nDuff: uh, what's the point of committing if it doesn't hit your server? |
| 20:00.44 | Qwell[] | svn limiting? |
| 20:00.48 | nDuff | Corydon76-home: well, for 3rd party branches, it can hit *their* server, and I can pull their changes from that server when I want to review them. |
| 20:01.25 | Corydon76-work | nDuff: if you're disconnected, it's not going to hit ANY server |
| 20:01.42 | nDuff | Corydon76-home: right; in that case, you do a local commit, and push your changesets up when you're connected. |
| 20:02.04 | Corydon76-work | What's the difference then with just waiting? |
| 20:02.08 | nDuff | Corydon76-home: ...but that way, you can have a tight resolution on your changesets (keep your separate changes separate, rather than jumbling them all together) |
| 20:02.31 | nDuff | Corydon76-home: ...and that way, people can merge ("cherrypick") only the changes they want between branches. |
| 20:02.53 | Corydon76-work | nDuff: and then you can have conflicts on top of conflicts when your first offline commit fails. Great. |
| 20:03.23 | Strom_M | ZZZZZZZZZZZZZZZZZZZ GNU/BORING |
| 20:03.31 | nDuff | Corydon76-home: I'm not going to go into details here, but suffice to say that all major distributed SCMs have much better conflict resolution than you're used to. |
| 20:04.23 | Corydon76-work | nDuff: "It's too advanced; you wouldn't understand." |
| 20:04.42 | nDuff | Corydon76-home: not that, but we're off-topic enough already. I'd be happy to discuss it out-of-channel. |
| 20:05.21 | Corydon76-work | You sidetracked an off-the-cuff suggestion |
| 20:05.26 | nDuff | granted. |
| 20:05.46 | *** join/#asterisk Ad-Hoc (n=nimbus@ppp55-76.adsl.forthnet.gr) |
| 20:05.56 | Ad-Hoc | aloha |
| 20:06.05 | Strom_M | ahola |
| 20:06.29 | Corydon76-work | For that matter, he's welcome to use perforce or some other repo. All of which are superior to sticking your changes in a Word document. |
| 20:06.36 | nDuff | yup. |
| 20:11.48 | *** join/#asterisk terrymr (n=terrymr@192.220.217.189) |
| 20:12.17 | PierreY | /notice Qwell[] Can I talk to you in private ? |
| 20:12.20 | PierreY | oup |
| 20:12.22 | PierreY | s |
| 20:12.24 | PierreY | :-) |
| 20:12.49 | file | he's gone temporarily |
| 20:13.05 | PierreY | thank you |
| 20:13.31 | Strom_M | has been temporarily disconnected |
| 20:13.50 | Strom_M | calls are being taken by |
| 20:13.54 | Strom_M | 555-2368 |
| 20:14.00 | Strom_M | in area code 311 |
| 20:14.02 | Strom_M | please make a note of it |
| 20:17.06 | PierreY | so... can I compile Asterisk and install it to /opt/asterisk (with zaptel and libpri, seems that ./configure --prefix don't work with zaptel) |
| 20:20.36 | Mercestes | PierreY, May i ask why you wish to install it to /opt/asterisk? |
| 20:20.54 | PierreY | cause my gentoo already havec an asterisk package |
| 20:21.00 | PierreY | -c |
| 20:22.06 | ixela | just make a new directory in /usr/src/ |
| 20:22.32 | Mercestes | ixela: I don't think that's the point. |
| 20:22.47 | Mercestes | PierreY, Are you trying to run two asterisk servers on one box? Or have multiple version source codes installed? |
| 20:22.53 | Mercestes | PierreY, Might I suggest chroots? |
| 20:23.15 | PierreY | seems a nice idea but I don't know how to do. |
| 20:23.15 | *** join/#asterisk adorah (n=Michael@87.69.57.73.cable.012.net.il) |
| 20:23.20 | Mercestes | I'm not 100% certain that asterisk will succesfully install in a filepath other than ...where it does now. |
| 20:23.38 | blitzrage | FYI: http://arstechnica.com/news.ars/post/20070702-voip-must-support-disabled-users-pay-fees.html |
| 20:25.05 | Mercestes | Forgive my language, but if deaf people want TTY VoIP why don't they just use fscking IM or email? RETARDED! |
| 20:25.05 | PierreY | I want to have my own asterisk build tree. I'm looking on how to create a firebird dbms connector for asterisk. I never played with such existing softwares... |
| 20:25.41 | blitzrage | Mercestes: yes, I agree you're retarded |
| 20:25.58 | Mercestes | blitzrage, Turd. |
| 20:26.08 | blitzrage | I was thinking the same thing |
| 20:26.19 | Mercestes | Sorry, didn't realize you were deaf. :P |
| 20:26.24 | blitzrage | Turd Fergusan |
| 20:27.49 | blitzrage | Mercestes: I'm not, but I have deaf friends -- plus ignorant people piss me off |
| 20:29.16 | *** join/#asterisk csmiga (n=chsmiga@68-114-9-62.dhcp.gsvl.ga.charter.com) |
| 20:29.20 | Mercestes | People who campaign against the worthless tend to piss me off. |
| 20:29.33 | Mercestes | If they are using VoIP they are already on the Inet...use Inet tty |
| 20:29.34 | nDuff | Mercestes: there's something to be said for being able to call people and companies who are only available via land line. I fully agree that there are more reasonable ways to provide that service than what the FCC is doing, but telling folks to stick to IM and email exclusively is unreasonable. |
| 20:30.01 | Mercestes | nDuff, There is still Inet TTY |
| 20:30.19 | *** join/#asterisk Paul_UK (n=foo@cpc1-acto7-0-0-cust402.brnt.cable.ntl.com) |
| 20:30.20 | blitzrage | PierreY: ./configure --prefix=$HOME/asterisk-bin --sysconfdir=$HOME/asterisk-bin --localstatedir=$HOME/asterisk- |
| 20:30.30 | blitzrage | for example |
| 20:30.37 | nDuff | Mercestes: yes -- and suggesting that is a lot more reasonable than "just use fscking IM or email". |
| 20:30.38 | Paul_UK | hey guys, anyone know if the snom 320 is IAX2 compatible? |
| 20:30.39 | blitzrage | if you want to install Asterisk into an alternative location |
| 20:30.46 | nDuff | Paul_UK: I believe it isn't. |
| 20:30.48 | Mercestes | nDuff: Well, that's what I meant. |
| 20:30.48 | *** join/#asterisk Remowylliams (n=Mare@71.16.217.178) |
| 20:31.04 | Paul_UK | nDuff, are there any decent IAX2 phones around? |
| 20:31.36 | ThoMe | what is a "stun"? |
| 20:31.55 | bkruse | stun server right? |
| 20:32.03 | ThoMe | jep |
| 20:32.05 | ThoMe | what is this? |
| 20:32.05 | falz | it's the same as 'whatismyip.com' effectively. |
| 20:32.10 | bkruse | right |
| 20:32.15 | ThoMe | how i can "make" a own stun-server? |
| 20:32.16 | bkruse | its a server in the sky that can see both hosts |
| 20:32.18 | Remowylliams | Hello everyone, can anyone explain to me why someone would want to build asterisk for a Nokia 770 internet tablet? I'm wanting a IAX2 softphone to run on it and most of my searches are turning up stuff about people running asterisk on the Nokia. |
| 20:32.19 | bkruse | google |
| 20:32.36 | bkruse | Remowylliams: what os? |
| 20:32.41 | *** join/#asterisk OloBola (n=not@74.95.13.57) |
| 20:32.53 | nDuff | Paul_UK: none that I know of. |
| 20:32.59 | Remowylliams | bkruse: it's a embedded version of linux. |
| 20:32.59 | Paul_UK | is anyone here running SIP with asterisk and its behind a firewall (nat) |
| 20:33.08 | bkruse | Paul_UK: hard phones? of soft |
| 20:33.13 | Paul_UK | bkruse: both |
| 20:33.21 | bkruse | Paul_UK: soft phones are more abundant |
| 20:33.31 | bkruse | Remowylliams: does it have X? or somethign similar? |
| 20:33.37 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 20:33.37 | *** mode/#asterisk [+o blitzrage] by ChanServ |
| 20:33.51 | Paul_UK | bkruse: sorry more abundant? |
| 20:33.58 | Remowylliams | bkruse yes it has something x-ish. |
| 20:34.00 | bkruse | Paul_UK: there are more soft phones than hard phones |
| 20:34.06 | bkruse | Remowylliams: do you know if its gnome or kde? |
| 20:34.19 | bkruse | can you build apps from source? |
| 20:34.47 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 20:34.47 | *** mode/#asterisk [+o blitzrage] by ChanServ |
| 20:34.50 | Paul_UK | bkruse, but has has nothing to do with my question. If I setup asterisk behind a firewall, will it connect to my carrier and then allow me to make a call with a hard phone, should the hardphone connect to the asterisk box? |
| 20:35.05 | bkruse | Paul_UK: oh! |
| 20:35.05 | bkruse | yes |
| 20:35.07 | Remowylliams | it's neither gnome (death to it) or kde (double death to it.) this is a small pocket pc as it were that runs a tiny version of linux. |
| 20:35.09 | bkruse | then set nat=yes |
| 20:35.16 | bkruse | theres TONS of docs on it |
| 20:35.19 | *** join/#asterisk scurb (n=scurb@c-25aae355.14-16-64736c13.cust.bredbandsbolaget.se) |
| 20:35.26 | Paul_UK | bkruse: ok cool.. sorry im having a hard day |
| 20:35.28 | bkruse | Remowylliams: I am not sure, can you build apps from source? |
| 20:35.48 | bkruse | Paul_UK: no problem, yes, do that, forward ports 5060, and 10000-20000 if you use all rtp ports ( just do it for the sake of argueing :P ) |
| 20:35.49 | blitzrage | if Asterisk is behind NAT, make sure you have externip and localnet setup |
| 20:35.59 | bkruse | blitzrage: yep |
| 20:36.06 | bkruse | blitzrage: DID YOU PUT THAT IN THE BOOK!? |
| 20:36.08 | blitzrage | nat=yes is for devices which are behind NAT |
| 20:36.11 | Paul_UK | its my first implementation and people keep on telling me differnet things and one is saying sip is ok and another that iax2 is better |
| 20:36.11 | blitzrage | bkruse: I think so |
| 20:36.14 | bkruse | blitzrage: good ;] |
| 20:36.21 | bkruse | blitzrage: hows all that now? good? |
| 20:36.22 | Remowylliams | bkruse: I might be able to find the stuff to do so. but I'd rather try to find a iax client. Do you know of a solid linux iax client ? |
| 20:36.23 | blitzrage | Paul_UK: both work fine (I prefer SIP) |
| 20:36.32 | blitzrage | bkruse: gotta do my copyedit review tonight... ugh |
| 20:36.34 | bkruse | Paul_UK: iax is better, because its only 1 port |
| 20:36.39 | bkruse | sip is more widely adopted, obviously |
| 20:36.43 | blitzrage | Remowylliams: idefisk |
| 20:36.48 | bkruse | blitzrage: i but thats tasking :[ |
| 20:36.54 | PierreY | can you advise a devlopment tool to play with asterisk sources ? |
| 20:36.54 | bkruse | taxing* |
| 20:36.58 | blitzrage | "better" is a relative term -- it doesn't exist |
| 20:37.01 | bkruse | PierreY: gcc? lol |
| 20:37.05 | PierreY | yes |
| 20:37.05 | PierreY | sure |
| 20:37.06 | blitzrage | PierreY: vim? |
| 20:37.07 | Paul_UK | bkruse thats the thing, im trying to find a decent iax2 phone |
| 20:37.16 | bkruse | Paul_UK: soft or hard |
| 20:37.21 | PierreY | something like kdevelop ? vim ... ? |
| 20:37.26 | bkruse | PierreY: vim |
| 20:37.29 | Remowylliams | blitzrage: eh.. I've had mixed interactions with idefisk.. but maybe |
| 20:37.30 | bkruse | :syntax on |
| 20:37.40 | bkruse | kdevelop helps me for javascript related things, but thats about it |
| 20:37.41 | Paul_UK | bkruse: hard |
| 20:37.47 | blitzrage | Remowylliams: best IAX2 softphone in my opinion |
| 20:37.51 | bkruse | better to just use vim and get used to the source tree and where things are at |
| 20:38.01 | bkruse | Paul_UK: there are not many iax2 hard phones yet |
| 20:38.13 | bkruse | you could buy an ATA from digium, for an analog phone, or buy one from china |
| 20:38.14 | blitzrage | PierreY: none of the core Asterisk developers use an IDE if that means anything |
| 20:38.16 | Remowylliams | Paul_UK: best softphone I know for IAX2 was firefly back when it allowed for third party |
| 20:38.18 | Paul_UK | bkruse: out of interest, can you have both on 1 server? |
| 20:38.19 | blitzrage | (which of course it doesn't) |
| 20:38.23 | bkruse | Paul_UK: yes |
| 20:38.33 | Paul_UK | bkruse: hmm interesting |
| 20:38.44 | PierreY | bkruse, blitzrage : thank you both |
| 20:39.03 | bkruse | Paul_UK: yes, its not as widely adopted as SIP. Sip has been around for awhile, even though iax will progressivly make its way into the hard phone world, maybe, sorta kinda, eventually |
| 20:39.22 | PierreY | emerging vim... |
| 20:39.26 | Paul_UK | bkruse: thanks.. I will ask my carrier if I can do that. As then I can have SIp for my hardphones and the rest IAX softphones |
| 20:39.45 | bkruse | Paul_UK: yep |
| 20:39.54 | bkruse | but this does not matter for your outbound/inbound trunk |
| 20:39.58 | bkruse | that can be whatever you want |
| 20:40.12 | bkruse | asterisk does all the bridging and transcoding (if necessary) |
| 20:40.14 | bkruse | good luck! |
| 20:40.43 | Paul_UK | bkruse: lol I need it. Im having to setup a small user base about 25 users initally but its a business environment and has to work! |
| 20:41.31 | bkruse | Paul_UK: awesome, let us know if you need any more help |
| 20:43.36 | Paul_UK | bkruse: sure I will do.. just lost my head there for a sec and needed clarification. The setup should be next week lol. |
| 20:44.34 | *** join/#asterisk tako-san (n=Tako-san@24.68.129.29) |
| 20:44.38 | *** join/#asterisk Nephfl (n=nephfl@adsl-070-147-105-151.sip.gnv.bellsouth.net) |
| 20:45.01 | Nephfl | anyone know how to get a polycom phone to show the status of a zap channel? |
| 20:45.07 | bkruse | Paul_UK: sounds good! have fun with it :] |
| 20:45.42 | Remowylliams | Hmm looks like there are SIP clients but short of installing asterisk as a pbx on this little thing I'm not finding much else. |
| 20:48.08 | Paul_UK | thanks bkruse :) |
| 20:49.28 | irule | find / -name mysql.sock finds nothing, what should I do in this case? this is debian sarge and I already created the db table with phpmyadmin |
| 20:50.53 | *** join/#asterisk gardo (n=gardo@121.97.211.162) |
| 20:53.21 | *** join/#asterisk waygel (n=peter@dslb-088-064-027-181.pools.arcor-ip.net) |
| 20:53.43 | *** join/#asterisk ovnicraft (n=logan@190.10.193.4) |
| 20:53.58 | ovnicraft | anyone know how to change audio lang |
| 20:57.36 | *** part/#asterisk waygel (n=peter@dslb-088-064-027-181.pools.arcor-ip.net) |
| 20:58.05 | *** part/#asterisk Remowylliams (n=Mare@71.16.217.178) |
| 20:58.31 | Mercestes | ovnicraft, of the prompts? |
| 20:59.37 | ovnicraft | yes |
| 21:00.03 | Mercestes | ovnicraft, http://www.voip-info.org/wiki-Asterisk+sound+files+international |
| 21:00.05 | Mercestes | That maybe? |
| 21:01.08 | ovnicraft | i'll tell you if it works... |
| 21:01.16 | ovnicraft | or not ;-) |
| 21:02.35 | *** join/#asterisk UnixDog (n=UnixDog@adsl-69-234-211-33.dsl.irvnca.pacbell.net) |
| 21:02.50 | UnixDog | hey guys anyone know whats going on wiht nufone ? |
| 21:03.20 | UnixDog | I tried to signup for account but it wont let me make apayment says to contact thier accounting dept |
| 21:03.40 | *** join/#asterisk justdave (n=dave@unaffiliated/justdave) |
| 21:03.45 | UnixDog | but all I get when I call is that exten 3000 is unavaible |
| 21:04.13 | OloBola | you're FIRED! |
| 21:04.20 | agile | ok |
| 21:04.52 | OloBola | that's how I feel. |
| 21:05.04 | UnixDog | ok |
| 21:08.07 | *** join/#asterisk zoa (n=zoa@pirus.securax.be) |
| 21:08.08 | zoa | yo yo |
| 21:08.39 | OloBola | how do I disconnect from asterisk CL without shutting it down? |
| 21:08.55 | nDuff | OloBola: exit |
| 21:09.03 | OloBola | thanks |
| 21:09.51 | OloBola | hmmmm... ? |
| 21:10.02 | OloBola | No such command. |
| 21:10.03 | Strom_C | OloBola: did you start asterisk as a background process? |
| 21:10.12 | Strom_C | or did you start it by typing "asterisk -cvvvvvg" |
| 21:10.43 | OloBola | the answer is B: "asterisk -cvvvvvg" |
| 21:10.47 | ThoMe | <PROTECTED> |
| 21:10.47 | Strom_C | wrong |
| 21:10.47 | ThoMe | <PROTECTED> |
| 21:10.47 | ThoMe | <PROTECTED> |
| 21:10.47 | ThoMe | <PROTECTED> |
| 21:10.47 | ThoMe | <PROTECTED> |
| 21:10.51 | Strom_C | ThoMe: don't spam |
| 21:10.54 | *** join/#asterisk javar (n=javar@69.79.134.24) |
| 21:10.56 | ThoMe | Strom_C: sorry |
| 21:10.56 | agile | is it just me or is it difficult to find a decent iax/sip provider? |
| 21:11.04 | Strom_C | agile: i like teliax |
| 21:11.12 | ThoMe | Strom_C: over WHICH port send asterisk the notify? |
| 21:11.36 | Strom_C | OloBola: if you start asterisk like that, then you can't disconnect from the console |
| 21:12.00 | zoa | anybody online who has the private cellphone number for marko ? |
| 21:12.02 | Strom_C | OloBola: stop asterisk, then run "safe_asterisk" or just "asterisk", then connect to the console by typing "asterisk -r" |
| 21:12.08 | UnixDog | ok what is a good provider that has byod thats not and arm and leg to setup and allows multi devices |
| 21:12.15 | Strom_C | zoa: now why would any of us give that out? |
| 21:12.25 | zoa | well i have it |
| 21:12.31 | zoa | in my phone that is out for a repair |
| 21:12.32 | zoa | hmm |
| 21:12.34 | zoa | wait a sec |
| 21:12.34 | ThoMe | Strom_C: hm? |
| 21:12.37 | zoa | i have a printout |
| 21:12.38 | zoa | :) |
| 21:12.43 | zoa | nevermind |
| 21:12.43 | zoa | :) |
| 21:12.59 | OloBola | Strom_C: thank you very much. |
| 21:14.37 | OloBola | how do I set verbosity? |
| 21:14.50 | UnixDog | -vvvvvvvvvvvvvvvvv |
| 21:14.51 | tako-san | asterisk -rvvvvvvv (or as many v's as you want) |
| 21:14.59 | tako-san | each "v" is one level of verbosity |
| 21:15.09 | OloBola | thank you. |
| 21:15.12 | bkruse | tako-san: only applicable up to 32,768 :/ |
| 21:15.24 | tako-san | hahaha |
| 21:15.30 | tako-san | ok thanks for letting me know ;) |
| 21:15.34 | bkruse | lol |
| 21:15.49 | bkruse | im not sure how much truth is in that statement, but i had to say it :P |
| 21:16.05 | tako-san | it sounded pretty impressive |
| 21:16.08 | tako-san | 8-) |
| 21:16.12 | bkruse | yay |
| 21:18.05 | OloBola | do you know if lumenvox starter kit will run on FC4? Their docs state 4 in some places and 5/6 in most places. |
| 21:18.47 | *** join/#asterisk koosroos (n=koosroos@196.209.244.207) |
| 21:18.47 | file | FC4 is being dropped in the next release |
| 21:18.55 | *** join/#asterisk asdx (n=asdx@adsl-131-233.click.com.py) |
| 21:18.58 | asdx | hi |
| 21:19.01 | OloBola | I see. |
| 21:19.01 | koosroos | any sip experts around |
| 21:19.09 | asdx | are fxs/fxo devices different to a modem? |
| 21:19.19 | asdx | and is possible to use asterisk with a modem? |
| 21:19.26 | koosroos | no they are lines od extensions |
| 21:19.54 | asdx | oh |
| 21:20.48 | Strom_C | koosroos: better to just ask your question |
| 21:21.05 | koosroos | hi Strom_C how are u |
| 21:21.06 | koosroos | ? |
| 21:21.17 | drako | lol |
| 21:21.22 | drako | good question |
| 21:21.23 | *** join/#asterisk bjohnson (n=bjohnson@i209-195-110-172.cia.com) |
| 21:21.37 | drako | asdx, buy a fxo/fxs device. |
| 21:21.46 | koosroos | can someone please place more light on caninvite=yes in the sip contect? please |
| 21:22.05 | asdx | drako: i wil |
| 21:22.07 | asdx | will* |
| 21:22.12 | Strom_C | koosroos: yes |
| 21:22.35 | Strom_C | koosroos: reinvites are when the various sip endpoints send INVITE messages to each other after the establishment of a call |
| 21:22.57 | Strom_C | you can permit or deny that behavior by setting canreinvte=yes or no, respectively |
| 21:23.22 | koosroos | why would you use it :strom_c? |
| 21:23.59 | Strom_C | when you want the call to go direct between endpoints and not through the asterisk server |
| 21:24.00 | Innatech | koosroos: to handoff the call to the endpoints so they communicate directly. |
| 21:24.51 | koosroos | does this then free up the asterisk servers resources? |
| 21:25.09 | Strom_C | a single sip call doesn't use much in the way of server resources |
| 21:25.14 | *** part/#asterisk terrymr (n=terrymr@192.220.217.189) |
| 21:25.16 | Strom_C | bandwidth is a greater concern |
| 21:25.35 | koosroos | so must use for iax connections? |
| 21:26.48 | Strom_C | ....what? |
| 21:26.51 | Strom_C | that question makes no sense |
| 21:26.54 | Innatech | koosroos: DOES NOT COMPUTE. |
| 21:29.04 | koosroos | so if you use caninvite=yes you only gain some resources by utilizing this in your sip.conf file IAX(remote asterisk server connections(linking more than one asterisk bax over a WAN)),does this make sence?> |
| 21:29.24 | Strom_C | sip has nothing to do with IAX |
| 21:29.36 | Strom_C | and no, that still makes almost no sense |
| 21:29.40 | koosroos | k you lost me |
| 21:29.51 | Strom_C | where did I lose you? |
| 21:30.27 | koosroos | hour ago |
| 21:30.53 | UnixDog | man I cant find a good provider thats byod and allows multi device |
| 21:31.05 | Strom_C | UnixDog: what do you mean "multi device" |
| 21:31.06 | koosroos | so what you syaing you can set caninvete =yes or no does not effect your dial plan for making sip calls |
| 21:31.14 | UnixDog | nufone says they do but cant reach anyone |
| 21:31.16 | Strom_C | koosroos: no, it does not affect the fialplan |
| 21:31.17 | Strom_C | er, dialplan |
| 21:31.30 | UnixDog | 2 devices diff locations |
| 21:31.30 | koosroos | thx |
| 21:31.40 | Strom_C | UnixDog: for the same DID number? |
| 21:31.59 | UnixDog | yes |
| 21:32.15 | Strom_C | ...why not just run an asterisk box and do the multi-device yourself? |
| 21:33.02 | UnixDog | my cousin and I want to split a connection but we have a issue with the net between us |
| 21:33.17 | Strom_C | and what issue would that be |
| 21:33.30 | UnixDog | savvis |
| 21:33.38 | Strom_C | ...? |
| 21:34.11 | UnixDog | seems the I hit the savvis backbone when it hits denver and it starts loosing major packets |
| 21:34.38 | Strom_C | SIP calls, or IAX calls? |
| 21:34.46 | Strom_C | or is this any type of traffic |
| 21:34.52 | stimpie | Does someone knows if the TE412P works in a Dell poweredge 1950? |
| 21:34.53 | UnixDog | I have reported it to my isp and its been 2 months now |
| 21:35.01 | UnixDog | any traffic |
| 21:35.29 | Strom_C | UnixDog: why must it be multi-device and not, say, two individual accounts? |
| 21:36.16 | UnixDog | because my cousin wants to foot the bill and most places we have tried said he cant have multi accounts on 1 creditcard |
| 21:36.43 | Strom_C | have you talked to teliax? |
| 21:36.58 | UnixDog | <=== not a fan of teliax |
| 21:37.05 | UnixDog | long story |
| 21:37.05 | Strom_C | por que? |
| 21:37.12 | Strom_C | well, give me the short short version |
| 21:37.18 | UnixDog | got screwed over |
| 21:37.40 | Strom_C | that was the short short short short version |
| 21:37.47 | *** join/#asterisk EricL (n=eric@clydesdale.linkexperts.com) |
| 21:38.04 | UnixDog | their system trippled billed me |
| 21:38.19 | EricL | Is this going to send the progression to command 5 if ${internal_exten} is 5131? |
| 21:38.21 | EricL | exten =>_1NXXNXXXXXX,2,GotoIf(${internal_exten}:("513[0-9]")?5) |
| 21:38.49 | UnixDog | then I had to wait 30 days to get the money back and they did not bill me for 2 months and then sent me a bill and I paid it but then the disabled my account saying I abused them |
| 21:39.12 | EmleyMoor | EricL: That doesn't look like a valid expression to me |
| 21:39.44 | EricL | EmleyMoor: It doesn't look valid to me either, but I am not sure what to do to fix it. |
| 21:40.12 | EmleyMoor | EricL: If I had some better idea what you were trying to achieve, it would help |
| 21:41.04 | EricL | I have ${internal_exten} set properly...it will be something between 5130-5139. |
| 21:41.14 | EmleyMoor | ... |
| 21:41.24 | EricL | Sorry, if it is between those 2, I want it to go to step 5. |
| 21:41.39 | EricL | If not, I want it to continue through the progression. |
| 21:42.23 | EmleyMoor | I don't know if you can match patterns, but anyway, the expression you have used is invalid |
| 21:42.59 | EricL | I dont' mind doing it longhand since its only a few numbers. |
| 21:43.08 | EmleyMoor | If it was $["${internal_exten}"="5131"], that would match for the case you mentioned |
| 21:43.10 | Corydon76-work | EricL: GotoIf($[${internal_exten:0:3} = 513]?... |
| 21:43.13 | EricL | How would I do it typing it out 5131 5132 etc. |
| 21:43.34 | EricL | Corydon76-work: Ah, that's what I am looking for, thanks. |
| 21:44.04 | Corydon76-work | Don't think of it as just a number; think of it also as a string |
| 21:45.43 | EricL | Gotchya...I am trying to make it so that certain extensions dialout with their DID as their callerid and others dial out with the main number. |
| 21:46.14 | EmleyMoor | Ah, yes, understood |
| 21:47.38 | Strom_C | why bother using ${internal_exten} in that case? why not just use ${CALLERID(num)} ? |
| 21:47.40 | Corydon76-work | EricL: we do a similar thing here... each DID is mapped into a database. We then set the DID based upon what is in the DB and fallback to the main number if nothing is set |
| 21:47.53 | koosroos | why does this jump from s,2 to s,102? exten => s,2,Voicemail(u100) exten => s,102,Voicemail(b100) |
| 21:48.07 | Strom_C | koosroos: old example. don't do that anymore. |
| 21:48.17 | EmleyMoor | I have default caller ID set in zapata.conf/sip.conf/iax.conf for mine, and ways to manually change it if needed |
| 21:48.26 | koosroos | what do you meen by old? |
| 21:48.45 | Corydon76-work | EricL: i.e. Set(${IF($["${CALLERID(num)" = ""]?CALLERID(num):foo)}=6015551212) |
| 21:48.52 | EmleyMoor | koosroos: Under the old example, if busy, the next priority is 101 more than the dial command |
| 21:49.25 | EmleyMoor | However, a better way exists now - you can stop that jump and process the result of the Dial app |
| 21:50.13 | koosroos | so you saying if i dial an extension and it is busy it wil add a 10 ? |
| 21:50.15 | Corydon76-work | koosroos: standard behavior on 1.0, deprecated in 1.2, removed after 1.4 was released |
| 21:50.31 | *** join/#asterisk fujin (n=fujin@unaffiliated/fujin) |
| 21:50.52 | fujin | hi, I'm not getting any colour output on my * console, can't work out what the flag is |
| 21:50.56 | fujin | is there an option which needs to be set? |
| 21:51.01 | EmleyMoor | koosroos: No... under that behaviour, now obsolete, it goes to (priority of Dial app line plus 101). However, it's obsolete now. |
| 21:51.02 | fujin | we went from * 1.4 to 1.2 |
| 21:51.19 | fujin | I have TERM=linux, too. |
| 21:51.20 | koosroos | so if i wan to transfer a call to exten and user is busy, how can i route it to a nother extension? |
| 21:51.21 | fujin | same as the old box |
| 21:51.23 | Strom_C | from 1.4 TO 1.2? |
| 21:51.28 | fujin | Strom_C: yes |
| 21:51.33 | Strom_C | ....why? |
| 21:51.49 | fujin | because the distrobution that *we* run here has 1.2 as the maintained package |
| 21:51.56 | EricL | Strom_M: Because that references the incoming callerid. |
| 21:52.20 | EricL | Corydon76-work: What is foo referencing there? |
| 21:52.31 | EmleyMoor | EricL: Only if the extension is triggered by incoming calls, not someone dialing |
| 21:52.36 | Strom_C | EricL: outside callers are allowed to dial back out of your dialplan? |
| 21:52.41 | Corydon76-work | EricL: it's a placeholder, for when you already have a CALLERID set |
| 21:53.10 | Corydon76-work | EricL: in other words, if you already have a callerid, then you DON'T want to set callerid |
| 21:53.20 | UnixDog | brb |
| 21:53.26 | EricL | Strom_M: Sorry, brain fart. I do use CALLERID(num), but I set it with that gotoif statement. |
| 21:54.38 | EricL | Corydon76-work: I always want to set a callerid, but I want it to be, if the dialing extension is one of about 10 extensions, then the callerid should be their did and no callerid(name), other set the default outgoing callerid(name) and callerid(num). |
| 21:55.00 | Corydon76-work | EricL: oh, and just before that we do: Set(CALLERID(num)=${ODBC_LOOKUPCID(${CUT(CHANNEL,-,1)})}) |
| 21:55.44 | Corydon76-work | so if they have a DID in the database, that's their CID. If they don't, they get the main number with the second statement. Make sense? |
| 21:56.34 | *** join/#asterisk af_ (n=getsmart@81-174-8-1.dynamic.ngi.it) |
| 21:57.14 | EricL | Yes, the problem I am having is that if their DID is to be the CID, then I don't want a CID(name) to be set. |
| 21:57.14 | *** part/#asterisk foug (n=foug@cpe-24-28-152-69.satx.res.rr.com) |
| 21:57.21 | EricL | The GotoIf() isn't working here, I have evaluate the same statement twice I think. |
| 21:59.08 | *** join/#asterisk ghenry (n=ghenry@212.159.59.85) |
| 22:04.34 | *** join/#asterisk FuriousGeorge (n=brian@ool-43536ea8.dyn.optonline.net) |
| 22:09.58 | *** join/#asterisk __DAW (n=_DAW@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 22:18.26 | *** join/#asterisk _VoicePulse (n=contact@unaffiliated/voicepulse) |
| 22:22.24 | Nephfl | anyone know where to get the bmp files that polycom 601s are looking for on boot? |
| 22:22.42 | Nephfl | or how to add a softbutton tothese things? |
| 22:30.02 | *** join/#asterisk Bu5hm4nn (n=Miranda@pD9E0D676.dip0.t-ipconnect.de) |
| 22:32.53 | *** part/#asterisk csmiga (n=chsmiga@68-114-9-62.dhcp.gsvl.ga.charter.com) |
| 22:33.15 | Nephfl | anybody in here? |
| 22:33.42 | Bu5hm4nn | just a n00b ;-) |
| 22:33.48 | Nephfl | ah |
| 22:34.52 | *** join/#asterisk MoutaPT (n=Blink@a213-22-40-195.cpe.netcabo.pt) |
| 22:36.09 | *** join/#asterisk remmo (n=junk@203.62.147.3) |
| 22:36.50 | MoutaPT | hi guys, i have a polycom 330 that when in queue on asterisk after exactly 60 seg waitingdrops the call sending SIP CANCEL, I believe it's the parameter call.ringbacktimeout, but i couldn't find out yet how to change the parameter via webportal of the phone, can any one experienced with polycom help me on this? |
| 22:37.50 | *** part/#asterisk MoutaPT (n=Blink@a213-22-40-195.cpe.netcabo.pt) |
| 22:38.39 | *** join/#asterisk ToyMan (n=Stuart@user-12lcqol.cable.mindspring.com) |
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| 22:42.10 | FuriousGeorge | ~lastspoke russellb |
| 22:42.31 | shmaltz | FuriousGeorge, may I msg you? |
| 22:42.40 | FuriousGeorge | of course, buddy |
| 22:43.06 | ThoMe | so |
| 22:43.10 | ThoMe | I'm tird... |
| 22:43.12 | ThoMe | tred |
| 22:43.13 | ThoMe | tired! |
| 22:43.15 | ThoMe | good night ;) |
| 22:43.37 | MoutaPT | hi Guys, I've two polycoms 330 and 430 with same problem, I call into an Asterisk Queue and always after waiting exactly 60 seconds the phones send a SIP CANCEL and the call is dropped, i believe this could be related with the parameter call.ringbacktimeout, but i can't change it on the GUI at least I don't know how....Can any one help me on this? |
| 22:44.14 | MoutaPT | I'm talking about GUI of Polycom Phones... |
| 22:47.29 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 22:48.31 | Innatech | mouta: webconfig or provision by FTP/TFTP/HTTP etc. |
| 22:48.45 | MoutaPT | <PROTECTED> |
| 22:49.01 | Innatech | mouta: should be there somewhere in the webconfig pages. |
| 22:49.37 | Innatech | mouta: if not, I know you can set it by using server side provisioning. |
| 22:49.44 | MoutaPT | i believe... i just can't find... |
| 22:49.57 | MoutaPT | i've been looking deeply |
| 22:49.59 | Innatech | I have some 430s. I'll take a look, BRB. |
| 22:50.05 | MoutaPT | thaks! |
| 22:50.07 | Strom_M | MoutaPT: does the call supervise before going into the queue? |
| 22:50.42 | MoutaPT | what u mean with call supervise?transfer? NO i just dial directly to the queue |
| 22:51.09 | Strom_M | supervise as in "does the terminating end of the call go off-hook" |
| 22:52.01 | MoutaPT | who terminates the call is polycom |
| 22:52.06 | MoutaPT | sending a SIP CANCEL |
| 22:52.16 | Strom_M | that's not what i'm asking you |
| 22:52.27 | Strom_M | does asterisk answer the call before dumping it into the queue |
| 22:52.41 | MoutaPT | I can listen MOH |
| 22:52.49 | MoutaPT | so i believe the call is answered |
| 22:52.52 | Strom_M | no |
| 22:52.56 | Strom_M | thats not a good indication |
| 22:53.04 | Strom_M | does the clock on the phone start counting up? |
| 22:53.05 | MoutaPT | you mean in dialplan |
| 22:53.08 | MoutaPT | answer |
| 22:53.14 | MoutaPT | no |
| 22:53.23 | MoutaPT | i understand what u said now:) |
| 22:53.28 | MoutaPT | on another phone |
| 22:53.31 | MoutaPT | Thomson 2030 |
| 22:53.37 | MoutaPT | i could notice msg Ringing |
| 22:53.45 | MoutaPT | while listenning MOH |
| 22:53.46 | Strom_M | if the clock doesn't start counting, then the call is not being answered; therefore the phone terminates the call |
| 22:54.00 | Strom_M | so in your dialplan, Answer() before you Queue() |
| 22:54.04 | MoutaPT | so dialplan bug |
| 22:54.41 | MoutaPT | unfortunately i'm out of the office now, but i remember from CDRs |
| 22:54.48 | MoutaPT | ringing took 60seg |
| 22:54.53 | MoutaPT | and duration 0sec |
| 22:54.54 | Innatech | FYI, strange as it seems, I think that setting might have to be provisioned by config file. I can't find it in the webconfig either. |
| 22:55.27 | Innatech | but yes, Answer() before Queue() |
| 22:55.30 | MoutaPT | but with the Answer parameter this can be solved! |
| 22:55.39 | MoutaPT | i will check it out tomorrow |
| 22:55.46 | MoutaPT | thks guys |
| 22:55.59 | tc3driver | Hello, I have a few general questions about asterisk. 1. can it be used as a tapi interface to a standard pbx (I know that asterisk is a pbx itself), and can tapi commands be sent to asteris? |
| 22:56.24 | tc3driver | asterisk* |
| 22:57.06 | MoutaPT | <PROTECTED> |
| 22:57.13 | MoutaPT | i'm not using DHCP... |
| 22:58.41 | *** join/#asterisk fujin (n=fujin@unaffiliated/fujin) |
| 22:59.05 | *** join/#asterisk __DAW (n=_DAW@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 22:59.58 | Innatech | nah, not really. |
| 23:00.40 | *** join/#asterisk nohop (n=root@cc501678-a.hgv1.dr.home.nl) |
| 23:01.27 | Innatech | adduser ProvisioningUserName , passwd ProvisioningUserName provisionpassword, wget the config files into the new user's home directory, customize the basic config file and then cp it once to each MAC you want to provision, adjusting the extensions. |
| 23:01.48 | remmo | sigh |
| 23:02.16 | *** join/#asterisk DrCron (n=rszasz@c-67-174-231-152.hsd1.ca.comcast.net) |
| 23:02.23 | MoutaPT | <PROTECTED> |
| 23:02.25 | fujin | Hi guys, I'm not getting any color on my term - I know this isn't an important issue but wondering if it's an easy fix |
| 23:02.37 | fujin | type adduser in your console - it's part of the shadow package |
| 23:02.37 | Nephfl | anybody know if you can get a polycom phone to ring while you are on a call? |
| 23:03.27 | fujin | hrm |
| 23:04.02 | fujin | term=linux, no color in my asterisk console - it's handy for debugging, anyone know how to turn it on? |
| 23:04.27 | Innatech | MoutaPT: Asterisktutorials.com has a video screencast that covers bulk Polycom provisioning. There's also a good guide linked to from the Sureteq trixbox 2.2 "complete setup guide." |
| 23:04.38 | MoutaPT | thks!! |
| 23:04.53 | Bu5hm4nn | fujin: do you mean a standard linux console? |
| 23:04.54 | Innatech | no prob. : ) |
| 23:05.03 | fujin | Bu5hm4nn: well, I'm sshed in, but yes. a "standard linux console". |
| 23:05.56 | fujin | ls --color=auto is working correctly |
| 23:06.01 | fujin | but if I asterisk -r it isn't coloured. |
| 23:06.03 | [TK]D-Fender | Nephfl, No. |
| 23:06.21 | Bu5hm4nn | fujin: ok, thats beyond my experience |
| 23:07.06 | Nephfl | damn, that sucks, is there a phone that does support that? |
| 23:07.27 | JT | fujin: it depends upon how the daemon was started as to whether there's colour or not |
| 23:07.44 | fujin | JT: oh - what's the flag I'm looking for? |
| 23:07.49 | fujin | it's being started with -p -U asterisk |
| 23:08.47 | tzafrir_laptop | JT, actually my packages have colors even when daemonized |
| 23:08.57 | Nephfl | can you add custom soft buttons to the polycom 601? |
| 23:09.06 | tzafrir_laptop | All it takes is a very little patch |
| 23:09.13 | JT | tzafrir_laptop: sure, it just depends upon the daemon being started right |
| 23:10.11 | Innatech | fujin: If you google for Asterisk CLI color, you'll see some mailing list complaints about this. They don't have solutions, though. Odd. Same symptoms: ls --color works but the * CLI is b&w. |
| 23:10.14 | tzafrir_laptop | (a patch that fakes TERM=linux when starting daemonized) |
| 23:10.45 | tzafrir_laptop | asterisk -rn works regardless |
| 23:10.50 | JT | Innatech: imho, ls --color has nothing to do with it |
| 23:10.54 | tzafrir_laptop | to restore sanity |
| 23:11.10 | Innatech | JT: ls --color demonstrates that color output is working over the terminal connection. |
| 23:11.42 | JT | Innatech: it doesn't demonstrate that asterisk will output colour |
| 23:12.01 | Innatech | JT: No one said that it did. |
| 23:12.04 | ThoMe | hi JT |
| 23:12.10 | JT | so it has no relevance :) |
| 23:12.13 | ThoMe | is it posibiel more as one in the contexte? |
| 23:12.13 | ThoMe | subscribecontext=eingehend |
| 23:12.15 | JT | hi ThoMe |
| 23:12.23 | ThoMe | example, eingehend1, eingehend2 ? |
| 23:12.36 | ThoMe | subscribecontext=bla,bla2,bla3 ? |
| 23:13.09 | Nephfl | the documentation for these polycom phones doesnt seem very good |
| 23:13.16 | ThoMe | JT: do u have a idea? |
| 23:13.31 | tzafrir_laptop | JT: http://bugs.digium.com/view.php?id=9048 |
| 23:13.40 | ThoMe | tzafrir_laptop: or u have a idea? |
| 23:13.49 | Innatech | JT: The relevance is that it tests functionality outside of *. If you can't turn on your stove, its a good idea to make sure your gas connection is one before you take the thing apart. Poor metahpor, but you get the idea. |
| 23:13.58 | Innatech | *metaphor |
| 23:14.10 | JT | ThoMe: can you stop harrassing random people for answers? if people know, they'd say |
| 23:14.28 | *** join/#asterisk lullabud (n=lullabud@12.24.42.67) |
| 23:14.28 | ThoMe | JT: jep |
| 23:14.32 | JT | Innatech: no really, i don't, colours usually work in consoles these days, asterisk is just finicky |
| 23:14.49 | Nephfl | Maybe I should take a job as a ditch digger |
| 23:14.56 | lullabud | Looking for recommendations here... I'm looking for phones that have about the same functionality as the Polycom Soundpoint 501, but that is more geared towards remote use (eg: NAT friendly) |
| 23:15.14 | lullabud | something with speaker phone, conference, transfer, etc.. |
| 23:15.15 | JT | the polycoms work fine through nat |
| 23:15.24 | lullabud | jt: i've had a lot of problems... |
| 23:15.42 | JT | maybe your nat device is rubbish, or some setting is incorrect |
| 23:16.35 | *** part/#asterisk MoutaPT (n=Blink@a213-22-40-195.cpe.netcabo.pt) |
| 23:16.44 | ThoMe | JT: ok |
| 23:16.51 | ThoMe | JT: what mean hi with "create one that includes those 3" ? |
| 23:17.06 | JT | ThoMe: that made no sense |
| 23:17.16 | *** join/#asterisk bkruse (i=bkruse@nat/digium/x-760b2cf91caa7947) |
| 23:17.20 | Innatech | JT: I take your point, but there's nothing wrong with verifying the environment as a general troubleshooting step. Dismissing that step as irrelevant might miselad noobs. |
| 23:18.02 | Innatech | nephfl: Polycom is finicky about providing access to full documentation and software to non-partners. |
| 23:18.09 | ThoMe | russellb: oeeh. |
| 23:18.23 | ThoMe | russellb: u mean a context in the extentions.conf ? or? |
| 23:18.43 | lullabud | so, anybody else have any recommendations on network friendly phones that doesn't involve replacing routers? |
| 23:18.57 | Innatech | It's not the phones, its the protocol. |
| 23:19.07 | Innatech | If you can't deal with routing SIP, use IAX. |
| 23:19.12 | lullabud | well, soft-phones work fine, the polycom doesn't, so i'm willing to bet the phone has something to do with it. |
| 23:19.35 | Innatech | oh, there's NAT between your phones and your *? |
| 23:19.40 | lullabud | exactly. |
| 23:19.43 | lullabud | remote employees. |
| 23:19.51 | Innatech | mm. |
| 23:21.04 | Innatech | <PROTECTED> |
| 23:21.18 | Innatech | Kinda skimpy on features for most office workers, tho. |
| 23:21.25 | *** join/#asterisk fujin (n=fujin@unaffiliated/fujin) |
| 23:21.28 | fujin | o hai |
| 23:21.29 | JT | lullabud: it must be setup wrong. |
| 23:21.34 | fujin | sorry, ctrl+c'd my window |
| 23:21.37 | JT | polycomes work FINE behind NAT |
| 23:21.40 | JT | polycoms |
| 23:21.50 | Innatech | yeah, I would expect they would. |
| 23:21.57 | Innatech | Polycoms generally *work.* |
| 23:22.01 | Innatech | if complicated to config... |
| 23:22.02 | JT | randomly spending money on new phones will not solve your problem |
| 23:22.20 | Innatech | JT speaks the truth. |
| 23:22.37 | JT | you should diagnose the issue first |
| 23:24.06 | lullabud | JT: i'm not sure what you're talking about. everything i've read says the polycom's have known NAT issues. |
| 23:24.11 | fujin | JT: solved all mine going from Mitel->Linksys |
| 23:24.13 | lullabud | JT: it's in the Asterisk@home handbook |
| 23:24.19 | JT | lullabud: asterisk@home ROFL |
| 23:24.24 | JT | asterisk@home is a joke |
| 23:24.25 | lullabud | if you search for "polycom nat" in google it brings up lots of stuff about problems |
| 23:24.27 | fujin | yuck, if there's anything I can recommend NOT using, it's Mitel phones for SIP. |
| 23:24.40 | fujin | a@h is stupid. Asterisk is so easy to configure |
| 23:24.41 | fujin | ;\ |
| 23:25.07 | lullabud | dude, that's not the point. i'm not using A@H, i'm just saying that it's a known problem with polycom. |
| 23:25.09 | JT | lullabud: ok, instead of telling me to google random outdated stuff, tell me what exact issues YOU are having, and what you have diagnosed as the cause in each case |
| 23:25.10 | Innatech | Relax people. Yes, the Polycoms can be temperamental with NAT. At the same time, yes, they will work. |
| 23:25.42 | JT | lullabud: it's not a problem i'm having, some people just must be configuring their equipment incorrectly |
| 23:25.46 | lullabud | JT: nevermind man, it's cool, i'll get help from somebody who's not busting my balls. |
| 23:25.55 | fujin | lol |
| 23:25.56 | fujin | rofl |
| 23:25.59 | JT | lullabud: ok, so you can't say what a single issue is? |
| 23:26.09 | JT | lullabud: are you having any real issues with them? |
| 23:26.16 | lullabud | JT: i'd rather not perpetuate an argument. |
| 23:26.24 | JT | lullabud: no, i was hoping to help you |
| 23:26.37 | lullabud | JT: well, i'm sorry then. |
| 23:26.53 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 23:26.59 | fujin | anyone familiar with polycom's SIP conference units? which model is best? I'm lookin at an IP4000 on their site |
| 23:27.04 | JT | sometimes polycom config is not that intuitive, but when setup right, should work through nat |
| 23:27.16 | fujin | get a local SIP proxy lol |
| 23:27.20 | [TK]D-Fender | fujin, SoundStation 2W + SPA-2101 |
| 23:27.21 | fujin | solve all your probs |
| 23:27.31 | fujin | [TK]D-Fender: the spa-2101 is an ATA right? |
| 23:27.55 | [TK]D-Fender | fujin, Correct. 2W = Wireless + ATA = Anytime, anywhere fun. |
| 23:28.08 | *** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox) |
| 23:28.10 | fujin | hrm |
| 23:28.16 | Innatech | lullabud: I concur with fujin. The easiest solution is to run a SIP proxy on your firewall/bastion host. |
| 23:28.19 | fujin | I don't really have much reason to move it - it will be sitting in our board room |
| 23:28.46 | JT | gar, you don't need a sip proxy unless your nat box/sip server is buster in the first place :) |
| 23:28.51 | JT | busted |
| 23:29.08 | fujin | generally you should be able to negotiate sip through NAT |
| 23:29.10 | [TK]D-Fender | fujin, Your call. Being analog + ATA means its more reusable. Depends on your deployment methodology. |
| 23:29.12 | Innatech | It eliminates having to set extern ip and manually forward ports to each Polycom. |
| 23:29.24 | [TK]D-Fender | lullabud, Polycom |
| 23:29.30 | fujin | [TK]D-Fender: I see. Deployment currently is "replace all pabx phones with IP phones". |
| 23:29.33 | [TK]D-Fender | lullabud, Polycom's usually run fine behind NAT |
| 23:29.34 | fujin | funtimes ;) |
| 23:29.35 | JT | Innatech: that is not at all necessary |
| 23:29.46 | JT | Innatech: NO port forwarding is required on the phone side |
| 23:29.49 | lullabud | Innatech: the problem is that I am running a SIP proxy, but calls aren't delivered to the remote phone consistently. sometimes they go through every other time, sometimes not at all. |
| 23:29.59 | fujin | learn2rtp |
| 23:30.01 | lullabud | [TK]D-Fender: the problem is with the far-end NAT. |
| 23:30.12 | fujin | tell them to configure their NAT correctly |
| 23:30.17 | lullabud | [TK]D-Fender: many remote employees don't have problems at all, some do. |
| 23:30.21 | Innatech | lullabud: ah. |
| 23:30.24 | [TK]D-Fender | lullabud, Depends on your proxy, and if it fights NAT for ownership of the same SIP port. |
| 23:30.25 | JT | lullabud: where is the sip proxy, client side or server side? |
| 23:30.39 | lullabud | JT: Server |
| 23:30.44 | [TK]D-Fender | lullabud, But typically those who have those issues jsut didn't think things through :) |
| 23:30.51 | JT | what proxy is it, and is asterisk behind it? |
| 23:31.11 | lullabud | [TK]D-Fender: yeah, they're sales people. it's not their job to think these through, nor to buy hardware that actually works. |
| 23:31.24 | fujin | polycoms work fine |
| 23:31.28 | lullabud | [TK]D-Fender: the thing is, sjphone and eyebeam both work fine, polycom does not. |
| 23:31.29 | [TK]D-Fender | lullabud, if the phone is directly behind NAT, then each should have its own port, and possibly RTP range. That can be easily arranged in provisioning. |
| 23:31.29 | fujin | fix your network |
| 23:31.42 | fujin | eyebeam has nat traversal via STUN, iirc |
| 23:32.02 | JT | is stun enabled in sjphone/eyebeam? |
| 23:32.02 | Innatech | yeah, its STUN that's making the difference between the softphones and the Polycoms. |
| 23:32.19 | fujin | seriously, setup a local asterisk proxy box |
| 23:32.28 | Innatech | ^agree |
| 23:32.33 | JT | Innatech: it makes more sense to buy a consumer router that is known to work for problem sites, than to buy new but inferior phones |
| 23:32.42 | fujin | forward sip and rtp ranges to that box from your router/firewall |
| 23:32.47 | lullabud | fujin: i HAVE a proxy here, the problem is the far-end NAT> |
| 23:33.02 | lullabud | fujin: and not all of these people are in the same location. |
| 23:33.06 | JT | fujin: i don't think all these hacks are necessary |
| 23:33.08 | fujin | so wait |
| 23:33.14 | lullabud | fujin: one of them is in philly, one of them is on a boat in miami, one of them is in texas. |
| 23:33.16 | fujin | what's the issue? |
| 23:33.20 | Innatech | JT: He *specifically* asked if anyone had a non-Polycom hardphone working behind NAT. I'm here to help answer questions, not tell people how to live. |
| 23:33.38 | lullabud | Innatech: thank you. |
| 23:33.58 | JT | Innatech: i'm specifically asking for info so we can help solve the issue instead of wasting money on useless crap |
| 23:34.14 | JT | i bet other hardphones would have the same issues |
| 23:34.15 | Innatech | That said, I don't see your average sales exec being pleased with something as modest as a Snom300 being dropped on them. |
| 23:34.42 | *** join/#asterisk ToyMan (n=Stuart@user-12lcqol.cable.mindspring.com) |
| 23:35.06 | Innatech | JT: Look, we all understand you like Polycom. That doesn't mean everything else is shit. |
| 23:35.16 | fujin | all 50 of my Linksys spa942s are behind NAT, with a * box and an as5400 with an E1 connected to it. It works fine |
| 23:35.26 | Mercestes | Doesn't he already have a polycom? |
| 23:35.32 | fujin | but I do not do sip->sip calls :) |
| 23:35.39 | JT | Innatech: actually, everything else is at least inferior, but my main point is that everything else will perform NO BETTER |
| 23:35.54 | shido6 | ZzZZ |
| 23:35.54 | Innatech | JT: Do you own anything else, OOC? |
| 23:35.54 | JT | especially for nat |
| 23:35.56 | shido6 | whats up |
| 23:36.18 | Innatech | JT: You might consider that a phone that supports STUN......well, supports STUN. QED. |
| 23:36.19 | [TK]D-Fender | I've hade Linksys ATA's/phones, CounterPath's full lineup of SIP soft phones, idefisk (SIP / IAX2), and a few others working behind NAT just fine.... |
| 23:36.28 | [TK]D-Fender | and of course.... Polycom's ;) |
| 23:36.33 | JT | Innatech: stop being defensive, i am only trying to solve his problem, it'd be a lot easier if he was forthcoming with info, and you two weren't both so defensive |
| 23:36.33 | lullabud | fujin: the problem isn't the local phones though. we have over 100 polycom phones here running fine. |
| 23:36.37 | lullabud | fujin: it's the remote phones. |
| 23:36.44 | [TK]D-Fender | Anyways, enough for my controibution... must be off! |
| 23:36.45 | [TK]D-Fender | TTFN |
| 23:36.50 | Mercestes | laters |
| 23:36.52 | shido6 | heh |
| 23:36.55 | JT | Innatech: stun is an unnecessary hack imho |
| 23:36.56 | fujin | so how the hell are we supposed to help fix your remote phones? |
| 23:37.29 | fujin | lullabud: setup a VPN! ;] |
| 23:37.29 | shido6 | do you have a standard set of routers these phones are behind? |
| 23:37.29 | JT | i have never used stun to get phones working behind nat, soft, hard or otherwise |
| 23:37.33 | Innatech | JT: I'm not feeling defensive. Just a little exasperated that your advice amounts to insiting that something that doesnt work should, and that all other known working solutions are unacceptable. That's nice in a classroom, but not @ work with people breathing down your neck. |
| 23:37.40 | lullabud | fujin: i would, if polycom's supported vpn... |
| 23:37.52 | lullabud | fujin: doing it at the router requires replacing the routers, which i don't want to do. |
| 23:38.01 | JT | Innatech: i'm suggesting that more DETAILS provided so we can make something work :) |
| 23:38.05 | fujin | lullabud: details |
| 23:38.10 | Innatech | Details rule. |
| 23:38.11 | fujin | what parts of your call are nothappening |
| 23:38.15 | lullabud | i gave all the details. i'll give them again.... |
| 23:38.20 | fujin | do you have RTP both ways |
| 23:38.26 | lullabud | call is made to an employee who is remotely. it isn't delivered. |
| 23:38.27 | fujin | have you tcpdumped on your proxy |
| 23:38.30 | lullabud | straight to voicemail. |
| 23:38.37 | shido6 | is the remote employee reigstered ? |
| 23:38.38 | fujin | is the remote employee registered to the proxy |
| 23:38.41 | lullabud | it works fine with sjphone and eyebeam, but not with polycom. |
| 23:38.47 | shido6 | can u see their peer in sip show peers at the CLI ? |
| 23:38.49 | lullabud | shido6: yes. |
| 23:38.58 | fujin | can you dial xxx from * console |
| 23:39.00 | shido6 | whats your sip.conf look like |
| 23:39.01 | Mercestes | do you have rtp both ways? eh? that's kind of an advanced question. |
| 23:39.05 | JT | lullabud: is qualify=yes set on the relevant sip.conf entry? |
| 23:39.06 | Mercestes | How abuot an easier one... |
| 23:39.09 | shido6 | include [general] and that particular phones sip stanza info |
| 23:39.13 | shido6 | at pastebin.ca |
| 23:39.16 | Mercestes | IP address of * server, IP address of REMOTE polycom phone, pastebin of SIP.conf |
| 23:39.21 | lullabud | holy hell |
| 23:39.31 | fujin | lullabud: can you ping the remote phone from * |
| 23:39.37 | fujin | have you done any layer3 network diagnostics |
| 23:39.47 | Mercestes | fujin: ....that is unnecessary at this point. |
| 23:39.49 | lullabud | fujin: remember, this works fine from all soft-phones.... |
| 23:40.06 | lullabud | this is simply a problem of the phone not talking to the sip proxy correctly. |
| 23:40.10 | shido6 | different phones may need different tweaks, pastebin lullabud, pastebin! :) |
| 23:40.27 | JT | lullabud: no, that's your assumption, let's get some actual details instead of assumptions :) |
| 23:40.28 | k31th | yo |
| 23:40.43 | JT | you may be right |
| 23:40.50 | fujin | yeah, show relevant sip.conf for the phone which CANNOT RECEIVE CALLS |
| 23:40.52 | fujin | and the general section |
| 23:40.55 | Nephfl | anybody know how to or have a link to set up a soft button on a polycom 601 |
| 23:40.57 | JT | but until we rule things in or out, we don't know what's going on |
| 23:41.17 | shido6 | ... |
| 23:41.25 | lullabud | Nephfl: directory.xml |
| 23:41.50 | Innatech | I wonder if that LJ article on diagnosing */SIP problems with Wireshark is up on their site yet. That'd be a good link to keep around for people in this position. |
| 23:42.19 | lullabud | Nephfl: search for 8500 in here: http://www.voip-info.org/wiki-Polycom+Phones |
| 23:42.22 | Innatech | it was in the March issue... |
| 23:42.24 | JT | Innatech: it sounds like he's not in a position to packet sniff at the remote site |
| 23:42.30 | k31th | wats a decent distro for running asterisk on, i want it minimal im thinking about building it on a flash drive. |
| 23:42.35 | *** join/#asterisk Here_And_There (n=Here_And@pool-68-238-252-162.phlapa.fios.verizon.net) |
| 23:42.36 | JT | astlinux |
| 23:42.49 | Innatech | I kinda figured he could SSH in, or at least get a capture sent to him. |
| 23:42.49 | shido6 | gumstix and astlinux I may have spelled gumstix wrong |
| 23:43.05 | JT | you spelled it correctly |
| 23:43.12 | shido6 | ya |
| 23:43.14 | shido6 | y |
| 23:43.37 | JT | Innatech: they're random consumer routers at non-techs' homes, so i dunno ;) |
| 23:43.43 | k31th | astlinux... |
| 23:43.52 | Innatech | oh. Homes. Scary. |
| 23:43.57 | Innatech | <---runs for the hills |
| 23:44.30 | Innatech | I use ATAs for that. If the user breaks it, they can drop it into a box and send it to me. |
| 23:44.38 | shido6 | heh |
| 23:44.48 | *** join/#asterisk Hymie (n=Hymie@l8r.net) |
| 23:45.02 | JT | lullabud: so do you think you can put sip.conf minus passwords into pastebin.ca? |
| 23:45.31 | Hymie | does anyone know why the asterisk 1.4 can't reboot unidens, but 1.2 can?? |
| 23:45.48 | JT | uniden... make sip phones? :o |
| 23:45.59 | Hymie | yes, very crappy ones.. but yes |
| 23:46.20 | Innatech | Maybe I'm thinking too simplistically, but would disabling reinvite help? |
| 23:46.35 | mihinomenest | they sell them direct to packet8/vongag? |
| 23:46.47 | JT | Innatech: perhaps, but we really haven't got that far yet |
| 23:46.54 | Innatech | yeah. |
| 23:46.58 | JT | i want an idea of what may be causing the problem |
| 23:47.16 | mihinomenest | anyone know of an affordable tool I can use to reduce the volume of my MoH files? |
| 23:47.25 | Hymie | sox |
| 23:47.51 | Hymie | audacity if you must have a gui |
| 23:47.54 | mihinomenest | any of those windows-admin friendly? |
| 23:48.26 | Hymie | well, if you ssh to your asterisk box, and type 'sox', how is that not friendly ;) regardless, audacity has a windows version, I belive |
| 23:52.04 | dc3aes | ive got an iax2 trunk w/ les.net and it just "stopped working".. nothing ever shows up in asterisk console, no firewall reject/drop messages, my FWD IAX2 incoming terminations work just fine.. im baffled |
| 23:52.40 | JT | dc3aes: in/out/what? |
| 23:52.41 | Nephfl | >dc3aes did you pay the bill? |
| 23:52.44 | dc3aes | haha |
| 23:53.15 | dc3aes | ya but funny you mention that, they screwed up my account hardcore for some time.. it took about 10 emails back and forth over 2 months to fix.. the balance is "now" $10+ USD |
| 23:53.30 | dc3aes | so maybe it is something to do with all the chaos earlier |
| 23:53.50 | JT | inbound or outbound calls are an issue? |
| 23:54.01 | dc3aes | JT, they are incoming IAX2 connections.. my asterisk box registers to LES.NET just fine along with my 2 other IAX2 trunks I have |
| 23:54.11 | dc3aes | JT, inbound from a 1800 number to my * |
| 23:54.32 | JT | btw they're not trunks unless trunking mode is enabled in iax.conf |
| 23:54.34 | JT | ok |
| 23:54.41 | dc3aes | true, misusing the term I beleive |
| 23:54.51 | JT | so what happens when you call the number? |
| 23:55.16 | dc3aes | it just rings and rings, I do not receive any noise in the asterisk console at all.. the context is never entered, nothing |
| 23:55.32 | JT | switch on iax2 debug |
| 23:55.45 | dc3aes | les.net has an "error log" but it has always been completely empty.. the CDR shows all my connections in the last month as unanswered (naturally) |
| 23:56.18 | dc3aes | JT, will try.. forgot about different debugging levels.. was really expecting to see _something_ .. my other IAX connectsion have create and teardown noise in the console.. |
| 23:56.42 | JT | dc3aes: hmm |
| 23:56.43 | JT | well |
| 23:56.51 | JT | you can restrict it to an ip |
| 23:56.57 | JT | as long as you know the ip |
| 23:57.45 | dc3aes | JT, theres a bunch now in the console.. will take a moment to go through it |
| 23:57.51 | dc3aes | I do see the les.net IP :D |
| 23:59.45 | *** join/#asterisk Here_And_There (n=Here_And@pool-68-238-252-162.phlapa.fios.verizon.net) |
| 23:59.54 | *** part/#asterisk Trevor_b (n=tbenson@69.12.220.201) |
