| 00:00.16 | fujin | bit late now |
| 00:00.18 | fujin | but thanks for that |
| 00:00.53 | nichtwirklich | flenders: inband |
| 00:01.14 | mjmarrio | does asterisk 1.4 use the new version of sox (sox -m infile outfile mergedfile) when the 'm' option is used in the monitor command? |
| 00:03.35 | nichtwirklich | flenders: interesting would be to test it with an external tone generator (I think people in the '80s used it ;-) ), but I don't have one here right now |
| 00:04.14 | grimsy | fujin: yeah, I run the spa942s |
| 00:04.55 | *** join/#asterisk hyphen (n=user@m7c5e36d0.tmodns.net) |
| 00:05.21 | fujin | grimsy: I don't suppose you've got the phones working with multiple mailboxes & display? |
| 00:05.29 | grimsy | nah, sorry |
| 00:05.34 | fujin | righto ta |
| 00:05.37 | grimsy | np |
| 00:05.43 | grimsy | only set them up with single lines |
| 00:08.10 | *** join/#asterisk gardo (n=gardo@121.97.192.1) |
| 00:08.55 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 00:10.01 | *** join/#asterisk Galeras (n=Galeras@190.84.206.174) |
| 00:10.44 | *** join/#asterisk Qwell (n=north@pdpc/sponsor/digium/Qwell) |
| 00:10.44 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 00:11.23 | fujin | russellb: looks like I got your func_devstate working here, the inaccuracy issues I was seeing where due to me using 'ringinuse=no' when ringing Local channels, AND trying to use func_devstate at the same time |
| 00:20.25 | nichtwirklich | flenders: okay its dmtf=info .... |
| 00:20.39 | nichtwirklich | thanks a lot |
| 00:22.35 | *** part/#asterisk Rospo (i=Geo@202.189.78.66) |
| 00:22.39 | nichtwirklich | f*, from sip phone I cannot send dtmf to the asterisk mailbox anymore |
| 00:24.12 | nephfl | i still cant find a way to get paging to work |
| 00:24.41 | [TK]D-Fender | nephfl, You you still haven't told us what you've done so far, or better yet, SHOWN US. |
| 00:26.05 | nephfl | i have polycom 601s , i just need to get auto answer to work... the ref in the wiki is for older versions the files it refers to are no longer in use |
| 00:26.34 | nephfl | i am using sip 2.2 |
| 00:27.10 | nephfl | i have been semi-randomly plaing with sip.cfg and the per phone configs |
| 00:28.03 | [TK]D-Fender | nephfl, Everything the guides say about provisioning is every bit as applicable to 2.2.0 as it was 1.5.2 |
| 00:28.21 | [TK]D-Fender | nephfl, so, pastebin is your friend.... |
| 00:29.12 | nephfl | the wiki says to edit ipmid.cfg |
| 00:29.23 | CrashSys | ipmid.cfg is now part of sip.cfg |
| 00:29.31 | CrashSys | so same settings, but look in sip.cfg |
| 00:30.04 | nephfl | thats what ive been doing...ill probably get it eventually... i would just say it should be a bit simpler |
| 00:30.47 | *** join/#asterisk Krurst (n=me@eth244.wa.adsl.internode.on.net) |
| 00:31.37 | [TK]D-Fender | nephfl, Ah, THAT part, well yes, ipmid was gone with 1.5. The Asterisk @ Home guide to provisinoning on the WIKI is more modern. |
| 00:31.55 | [TK]D-Fender | nephfl, but the hearders are actually the same. So sho me what you've done with everything |
| 00:32.19 | CrashSys | And if you want a faster-booting polycom, just have the settings that you dont want to be default in your sip.cfg and phone.cfg :) |
| 00:32.41 | *** part/#asterisk cirgal (n=robert@wsip-70-169-190-173.sb.sd.cox.net) |
| 00:32.52 | Krurst | can someone tell me whats wrong with this - " testrecord => #1,self/both,MixMonitor, ${TIMESTAMP} - '${CALLERID(all)}' - "${EXTEN}".wav|b| " in features.conf. It's recording to a file called "${TIMESTAMP} - '${CALLERID(all)}' - ${EXTEN}.wav" |
| 00:32.57 | CrashSys | go from like 2 minutes to boot to around 1 :) |
| 00:34.13 | *** join/#asterisk coppice (n=chatzill@83.155.17.210.dyn.pacific.net.hk) |
| 00:35.46 | [TK]D-Fender | CrashSys, Considering Mine only reboot on power failures & upgrades, who cares about 1 minute more? :) |
| 00:36.31 | coppice | people who have power failures every 50 seconds? |
| 00:36.46 | nephfl | i tried to get the buddylist thing to work forever...until the reseller i have finally got sip 2.2... |
| 00:38.02 | *** join/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-571cb35a36fc75ed) |
| 00:39.09 | [TK]D-Fender | nephfl, presence works fine on every version I've tried since 1.5.2 |
| 00:39.20 | [TK]D-Fender | nephfl, so get to it! show us what you've done! |
| 00:40.31 | [TK]D-Fender | coppice, That would indeed be a very perturbed demographic! You are quite adept at showcasing how "Nothing is so bad in life that it can't get WORSE" :p |
| 00:41.34 | coppice | actually, it reminds me a little of india |
| 00:41.51 | nephfl | presence didnt work for me at all..then in installed 2.2 and it did...who knows maybe it was something random i did when upgrading |
| 00:43.21 | Qwell | Krurst: How does that differ from the name you requested? |
| 00:44.51 | Krurst | I was hoping to use variables |
| 00:45.58 | Qwell | I would've expected it to work, but its probably just not implemented |
| 00:47.37 | Krurst | bugger. I need at least something that'll change the name for each recording. |
| 00:47.49 | Inez | hi |
| 00:49.17 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@216-110-94-230.static.twtelecom.net) |
| 00:49.32 | [TK]D-Fender | Krurst, You should use the dynamic monitoring feature that's already included. You can set the name in CASE it gets called upon. |
| 00:49.54 | [TK]D-Fender | Krurst, Thats sort of the purpose of "wW" for Dial you know... |
| 00:50.00 | Krurst | automon doesn't work after parking a call, as it resets the dial options when it re bridges. |
| 00:50.06 | [TK]D-Fender | Krurst, But I'm sure your wheel will be ROUNDER. |
| 00:50.27 | [TK]D-Fender | Krurst, ok, that would suck. |
| 00:51.02 | [TK]D-Fender | Krurst, I think I see.... so you dial your parked exten and you lose it right? I have a hack for you... ;) |
| 00:51.11 | Krurst | nono, |
| 00:51.20 | Krurst | I lose the dial options on the call |
| 00:52.02 | Krurst | theres some hack around it for 1.6 but they only apply to tT and kK |
| 00:52.45 | Krurst | re: http://bugs.digium.com/print_bug_page.php?bug_id=9267 |
| 00:53.00 | [TK]D-Fender | Krurst, don't "include => parkedcalls" instead include another context that will have "exten => _7XX,1,Dial(Local/${EXTEN}@parkedcalls,,Ww) |
| 00:53.11 | [TK]D-Fender | Krurst, ;) |
| 00:53.33 | Krurst | ah ok |
| 00:53.43 | Krurst | I'll give that a go |
| 00:53.48 | [TK]D-Fender | <- ub3r ch34t4r |
| 00:54.55 | fujin | heh, Local channels for the win. |
| 00:57.26 | Yourname` | Hey fujin, [TK]D-Fender |
| 00:57.29 | Yourname` | What's up |
| 00:58.00 | fujin | hey dude |
| 00:58.02 | fujin | same ol same ol |
| 01:00.39 | Yourname` | heh nice |
| 01:01.17 | *** join/#asterisk dfriend (n=dfriend@69.89.168.17) |
| 01:02.41 | fujin | what about you |
| 01:02.42 | fujin | mane |
| 01:03.59 | fujin | Yourname`: hows your asterisk going |
| 01:04.02 | fujin | you got everything runningyet? |
| 01:07.20 | Krurst | oh god, it works. [TK]D-Fender, thankyou so very very much. |
| 01:07.59 | [TK]D-Fender | Krurst, you're welcome :) |
| 01:07.59 | [TK]D-Fender | Krurst, And in case you're wondering, I came up with that on the spot :p |
| 01:08.05 | Krurst | genius |
| 01:08.09 | [TK]D-Fender | improv++ |
| 01:08.25 | [TK]D-Fender | "Hey Rocky, watch me pull a rabbit out of my hat!" |
| 01:08.39 | [TK]D-Fender | "That trick AGAIN, that'll NEVER work!" |
| 01:08.46 | [TK]D-Fender | "Nothing up my sleeve...." |
| 01:08.49 | [TK]D-Fender | "PRESTO!" |
| 01:09.48 | tzanger | http://tinyurl.com/33suug |
| 01:09.52 | tzanger | paris hilton screwing by the pool |
| 01:09.55 | Krurst | now to fix up all the other things I've done to this asterisk install :) |
| 01:10.23 | fujin | oh god, that's terrible tza |
| 01:10.26 | tzanger | fujin: :-) |
| 01:10.27 | fujin | nger |
| 01:10.37 | Yourname` | fujin: Same man. It's going well. Trying to come up with something that'll help me control the number of channels we're dialing by looking at the number of callers holding in the queue. What do you think? |
| 01:11.12 | fujin | oh, so you can make sure that there are always free channels? |
| 01:12.25 | Yourname` | fujin: Naw, so I make sure there are less abandons. |
| 01:12.38 | fujin | eh? |
| 01:12.42 | fujin | I'm not following you |
| 01:12.59 | Yourname` | fujin: So callers don't have to wait in the queue for too long, and hang up because agents are still busy. |
| 01:13.22 | [TK]D-Fender | tzanger, You're ruining the only thing she's good for with that misleading non-smut! |
| 01:13.31 | fujin | eh, dunno if there is anything you can do about that |
| 01:14.09 | fujin | Yourname`: so what do you want it to do? go somewhere else when all agents are busy? |
| 01:14.15 | fujin | not sure if I really understand what you're trying to build |
| 01:14.44 | Yourname` | fujin: Well, for example. There are 10 agents, and two dialers dialing on 400 channels each. 800 channels total dialout. ALL 10 agents are busy, and we have about 5 callers still waiting in the queue to talk to the agents. HAD there been something that could control the speed of dialing in retrospect with the agents on call/callers waiting, those 5 callers wouldn't be holding, or better yet,... |
| 01:14.46 | Yourname` | ...would be holding for a lesser time. |
| 01:15.20 | fujin | so you want to make it so that agents can't dial out when there are calls waiting? |
| 01:15.57 | Yourname` | LOL, you're going everywhere BUT the point. I want to make sure that when agents are busy, the dialers slow down preventing MORE callers to come in the queue. |
| 01:16.12 | fujin | isn't that what a queue is for? :p |
| 01:16.12 | Qwell | hire more agents |
| 01:16.15 | [TK]D-Fender | Yourname`, have your dialer check their state first then |
| 01:16.37 | Qwell | or, stop telemarketing ;) |
| 01:16.41 | Yourname` | fujin: lol yeah, but I hate looking at people wait. |
| 01:17.05 | Yourname` | Qwell: 40 so far, unfortunately response is better than I thought.. even during this 'test campaign' :( |
| 01:17.27 | Yourname` | [TK]D-Fender: What do you mean? Meaning dialers check the state of the queue all the time, right? |
| 01:17.49 | fujin | the state of the individual available agents |
| 01:17.50 | [TK]D-Fender | Yourname`, before dialing out. |
| 01:18.01 | fujin | which would be hard to enumerate using dynamic members ** |
| 01:18.10 | fujin | mmmm. funtimes. |
| 01:18.19 | Yourname` | Hmm |
| 01:20.10 | Yourname` | I was thinking about something like queue status dumped on an XML file, or something like that. And the dialers "pinging" this XML feed and checking how many callers are in the queue waiting. Each dialer goes by different args, like dialer 1 will STOP dialing (subtracting about 400 channels) when it sees more than 3 and less than 5 callers holding. Dialer2 = More than 5, less than 8..... |
| 01:20.11 | Yourname` | ...Dialer3=More than 8, less than 10.. and so on. |
| 01:21.31 | Qwell | why less than? |
| 01:21.42 | Qwell | that would only ever stop one dialer at a time |
| 01:22.39 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 01:22.51 | Yourname` | Because then dialing will actually be slowing down than completely stopping. For example if I'm running the dialing on 1600 channels.. 400 channels each box will be slowing down. |
| 01:24.36 | *** part/#asterisk lirakis (n=lirakis@cpe-68-175-38-65.nyc.res.rr.com) |
| 01:26.33 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@216-110-94-230.static.twtelecom.net) |
| 01:27.51 | *** join/#asterisk chendy (n=chendy@218.242.110.26) |
| 01:37.04 | Yourname` | Am I lagging or did no one really respond? ! |
| 01:37.31 | *** join/#asterisk aub (n=Universa@99.165.188.72.cfl.res.rr.com) |
| 01:37.31 | Mavvie | Is there any other fax software which can be incorperated into Asterisk besides app-rxfax with spandsp? |
| 01:37.39 | aub | i miss nick. |
| 01:37.41 | aub | :D |
| 01:37.47 | *** join/#asterisk |dennis| (n=dennis@200.32.236.10) |
| 01:38.53 | *** join/#asterisk Iamnach0 (i=Iamnacho@ip68-103-153-140.ks.ok.cox.net) |
| 01:42.04 | Yourname` | I was looking for something like that a few days earlier, Mavvie. People suggested Hylafax... which looked a little too much for what I was looking for. |
| 01:43.50 | *** part/#asterisk Mavvie (n=edwin@ppp121-44-14-82.lns4.syd7.internode.on.net) |
| 01:43.57 | *** join/#asterisk Mavvie (n=edwin@ppp121-44-14-82.lns4.syd7.internode.on.net) |
| 01:43.58 | Mavvie | woops |
| 01:44.15 | Mavvie | Yourname`: inside asterisk? |
| 01:44.47 | Yourname` | I think so. |
| 01:45.00 | Yourname` | I liked app_txfax/rx_fax and spandsp myself.. |
| 01:45.24 | Mavvie | I prefer them, but too many people complain about halfly received faxes. |
| 01:45.29 | WilliamK | Yourname`, actually I liked it as well |
| 01:46.17 | Mavvie | we have hylafax here for years, but we want to get rid of the (redhat 7.1) box. |
| 01:46.26 | Yourname` | Yeah, but I think it's not the apps that are at fault. It's just that fax is being sent on something that was meant for voice. It's a tradeoff, IMHO. |
| 01:49.27 | *** join/#asterisk yxa (n=lonari@58.185.90.101) |
| 01:50.02 | yxa | how much does canreinvite=no impact CPU usage? |
| 01:51.44 | tzanger | [TK]D-Fender: haha |
| 01:51.49 | tzanger | she's not even good for that, IMO |
| 01:52.32 | *** join/#asterisk PioneerVM2 (n=IceChat7@ool-45779466.dyn.optonline.net) |
| 01:52.38 | PioneerVM2 | hi -- does asterisk have a built in option to clear blocked caller id? when i get calls from blocked caller ID they are being asked to say there name and it makes me verify before connecting, not sure if its the voip provider or asterisk option |
| 01:53.09 | *** join/#asterisk mtaht4 (n=m@200.62.111.228) |
| 01:53.15 | aub | PioneerVM2, what is the voice sound like. |
| 01:53.25 | PioneerVM2 | hmmm i dont know off hand actually :) |
| 01:53.28 | PioneerVM2 | i think its female |
| 01:53.33 | aub | PioneerVM2, is it the common sounding asterisk lady? |
| 01:53.42 | PioneerVM2 | i dont remember its only happened a few times |
| 01:53.45 | aub | PioneerVM2, what viop provider do you have? |
| 01:53.49 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@216-110-94-230.static.twtelecom.net) |
| 01:53.55 | PioneerVM2 | teliax & voicepulse |
| 01:54.01 | aub | Ahh.. |
| 01:54.03 | PioneerVM2 | i couldnt find an option in their systems |
| 01:54.06 | aub | No clue. :D |
| 01:54.17 | PioneerVM2 | is there an option in asterisk? |
| 01:54.23 | PioneerVM2 | i can just search for it and see if its set |
| 01:54.37 | aub | My best guess is it is asterisk, or their special system. But they probably developed an asterisk function themselves if it is them. |
| 01:54.46 | aub | I have never seen or heard of that function before? |
| 01:54.57 | PioneerVM2 | yea dont know if i turned it on by accident |
| 01:55.07 | aub | But, it has been about 3 months since I have worked with an asterisk system. Sorry! |
| 01:55.21 | aub | Sounds like an interesting function to have. |
| 01:55.40 | *** join/#asterisk Igbothom_III (n=Hilton@office.quarkgroup.com.au) |
| 01:55.59 | aub | PioneerVM2, are you planning on running your own asterisk system? |
| 01:56.12 | fujin | You can probably script it in the dialplan. |
| 01:56.28 | *** join/#asterisk NightMonkey (n=NightMon@pdpc/supporter/sustaining/NightMonkey) |
| 01:56.32 | PioneerVM2 | yes i am |
| 01:56.36 | aub | fujin is right there, you can do pretty much anything with your dialplans |
| 01:56.39 | PioneerVM2 | im looking to deactivate it not activate it :) |
| 01:56.50 | PioneerVM2 | it's happening automatically, it must be the voip provider |
| 01:56.55 | PioneerVM2 | ill have to call them |
| 01:57.02 | aub | PioneerVM2, are you running your asterisk on like a dedicated server or a predefined PBX system that you pay for, or get for free like pbxboxes.com |
| 01:57.07 | PioneerVM2 | dedicated |
| 01:57.16 | PioneerVM2 | my own |
| 01:57.27 | aub | PioneerVM2, well when you run your own asterisk, i gurantee it wont do that, unless you set it to do so. |
| 01:57.32 | PioneerVM2 | ok |
| 01:57.39 | PioneerVM2 | i didnt know if there was an option that turned it on |
| 01:57.46 | aub | PioneerVM2, as for voicepulse I can look really fast if it is them? |
| 01:57.49 | PioneerVM2 | sounds like no, in which case i did not "implement it" |
| 01:57.55 | PioneerVM2 | i think it would be teliax |
| 01:58.01 | PioneerVM2 | as they are handling the main lines |
| 01:58.47 | PioneerVM2 | ill have to check into it further here |
| 01:58.49 | aub | PioneerVM2, http://www.voicepulse.com/features/advanced/AnonymousCallBlock.aspx |
| 01:59.16 | aub | PioneerVM2, Dial *160 on your phone to turn the option on/off <-- I have a feeling that is the solution to your problem. |
| 01:59.45 | PioneerVM2 | yea i dont know if that will work here |
| 01:59.52 | PioneerVM2 | since im running it thru asterisk not just a direct phone |
| 01:59.56 | PioneerVM2 | so they will never see *160 |
| 02:00.04 | aub | PioneerVM2, what do you mean? |
| 02:00.10 | PioneerVM2 | im using connect.voicepulse.com but it may be an option in my settings for the DIDs |
| 02:00.25 | aub | PioneerVM2, you run your voicepulse through a asterisk dedicated server, then to your phone? |
| 02:00.26 | PioneerVM2 | i have a PAP2t box connected to my own asterisk server |
| 02:00.35 | aub | O? |
| 02:00.35 | PioneerVM2 | asterisk server to voicepulse |
| 02:00.40 | aub | Why? |
| 02:00.41 | PioneerVM2 | so if i type *160 they will never get it |
| 02:00.48 | PioneerVM2 | because i manage all my phone lines |
| 02:01.01 | aub | Yes they will, just dial the call as if you are dialing out of the system. |
| 02:01.03 | PioneerVM2 | i have an elaborate multi-ring setup with special voice mail |
| 02:01.12 | PioneerVM2 | hmm. |
| 02:01.21 | PioneerVM2 | well the problem is i use different outgoing lines than incoming |
| 02:01.24 | aub | If you have it setup to dial 9 first, then dial 9 *160 or *161 |
| 02:01.28 | PioneerVM2 | i get incoming from one svc and outgiong on another |
| 02:01.36 | aub | Well the other option is to call them? |
| 02:01.43 | PioneerVM2 | ill figure it out, there is prob an option in the menus somewhere |
| 02:01.44 | aub | or login to your Account Center and turn the option on/off |
| 02:01.49 | PioneerVM2 | yes thats what im thinking |
| 02:01.56 | aub | That should be easy if you just login? |
| 02:01.59 | PioneerVM2 | i dont know why calls are coming in thru voicepulse though tahts confusing |
| 02:02.09 | aub | It seems voicepulse is the problem, I don't see asterisk doing it :] |
| 02:02.15 | PioneerVM2 | they should be coming thru teliax, might have something routed thru voicepulse incoming leftover from a while back |
| 02:02.20 | PioneerVM2 | ok next question |
| 02:02.24 | aub | Sure. |
| 02:02.28 | PioneerVM2 | how do i get *67 to work thru asterisk |
| 02:02.40 | PioneerVM2 | i have PAP2T box, i want to dial *67 to do caller ID blocking |
| 02:02.57 | PioneerVM2 | but im setting the outgoing caller ID # in dialplan when creating the call |
| 02:03.05 | aub | What is your method of outgoing calls? |
| 02:03.07 | PioneerVM2 | so i somehow need asterisk to tell me *67 was dialed |
| 02:03.12 | PioneerVM2 | PAP2t -> asterisk |
| 02:03.22 | luke-jr | PioneerVM2: you know how to unlock PAP2-NAs?? |
| 02:03.24 | PioneerVM2 | asterisk dialplan that sets callerID depending on what box is doing the call |
| 02:03.32 | PioneerVM2 | i have unlocked ones i purchased |
| 02:03.40 | aub | luke-jr, your long off. |
| 02:03.41 | PioneerVM2 | just bought them unlocked, real ones |
| 02:03.48 | luke-jr | I purchased one, but it's locked :( |
| 02:03.53 | luke-jr | aub: ? |
| 02:03.53 | aub | PioneerVM2, well if you set them in the dialplan you wont be able to do it? |
| 02:04.01 | PioneerVM2 | i dont know |
| 02:04.03 | aub | luke-jr, you want help unlocking it? |
| 02:04.04 | PioneerVM2 | there must be a way |
| 02:04.14 | aub | luke-jr, I cant help you here, but I can help you in pm probably. |
| 02:04.28 | PioneerVM2 | if i can see that *67 was typed on teh asterisk box in teh dialplan i can skip the setting |
| 02:04.32 | PioneerVM2 | or set it to junk |
| 02:04.42 | aub | PioneerVM2, If I think like im coding php, not that its relevant. I can do an if statement example if *67 dont set dialplan. |
| 02:05.11 | luke-jr | AEL shares a bit in common with PHP ;) |
| 02:05.26 | *** join/#asterisk n00dle (n=ccraft@ip-249-27.springsips.com) |
| 02:05.37 | PioneerVM2 | i dont think asterisk sees "*67" |
| 02:05.42 | PioneerVM2 | unless its a code set somewhere |
| 02:05.47 | n00dle | Good evening... anyone know how to unlock a Lingo/Primus K10L016 ATA? |
| 02:06.02 | aub | PioneerVM2, it does see it. Just like any other number :D |
| 02:06.10 | aub | n00dle, we cant give illegal advice in this room. |
| 02:06.26 | *** join/#asterisk Defraz (n=t0tal@24-116-152-177.cpe.cableone.net) |
| 02:06.34 | fujin | AEL FOR THE WIN! |
| 02:06.51 | n00dle | I own the ata, had been subscribed for a couple of years. (at least Lingo don't want it back...) |
| 02:07.06 | luke-jr | aub: how exactly is it illegal? O.o |
| 02:07.21 | aub | It is illegal technically to unlock a unit locked to a specific company. |
| 02:07.35 | aub | n00dle, I can link you to some documents in pm. |
| 02:08.07 | CrashSys | Documents to unlock my old vonage paperweight? I'm in... |
| 02:08.09 | Qwell | I'd think that the DMCA would say that it is quite legal |
| 02:08.12 | PioneerVM2 | can i set a caller ID setting in the handset config |
| 02:10.36 | aub | I have voipbuster, gizmo and grandcentral |
| 02:10.40 | aub | thats all i need free calling :D |
| 02:12.25 | luke-jr | heck, I'm not even trying to get free calling |
| 02:12.35 | luke-jr | just going through my * before hitting the ITSP |
| 02:17.25 | aub | Anyways i need advice myself.. I want to get a free call in number like grand central and ipkall. |
| 02:17.34 | aub | What other services are out there that are free? |
| 02:17.57 | Qwell | why not use grand central or ipkall/ |
| 02:19.38 | blitzrage | hehehe |
| 02:19.42 | aub | Becuase I have yet to get my completed grand central number. |
| 02:19.53 | blitzrage | FWD? |
| 02:20.00 | blitzrage | IAXtel |
| 02:22.05 | aub | FWD dosnt do what I need :D i need just like ipkall, but I cant use ipcall becuase sadly my voipbuster username the part before @ is longer then 10 characters |
| 02:22.25 | aub | i would register a new one, except I have already placed money on my voip buster account |
| 02:25.04 | *** join/#asterisk weasel00 (n=snowball@pencomsf.com) |
| 02:25.13 | PioneerVM2 | anyone here use a linksys pap box with asterisk? |
| 02:25.24 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@216-110-94-230.static.twtelecom.net) |
| 02:25.35 | flenders | PioneerVM2: a lot of people do |
| 02:25.39 | aub | PioneerVM2, I do. |
| 02:25.43 | PioneerVM2 | i know that :() |
| 02:25.43 | weasel00 | was 'make linux26 |
| 02:25.44 | PioneerVM2 | :) |
| 02:25.47 | aub | Did. |
| 02:25.49 | PioneerVM2 | im looking for someone that has experience with it |
| 02:25.57 | weasel00 | was 'make linux26' depreciated with the *1.4 trunk? |
| 02:26.03 | PioneerVM2 | im trying to get *67 caller ID block to work and proper way to config |
| 02:26.47 | aub | Well im off to sleep. |
| 02:26.52 | aub | School sucks ass. |
| 02:27.08 | aub | Unless your going for what I am :D |
| 02:27.15 | weasel00 | whats that? |
| 02:27.22 | aub | Zoologiist / Curator |
| 02:28.06 | weasel00 | nice :) |
| 02:28.46 | Krurst | That?s what I love about these high school girls, man. I keep getting older, they stay the same age. |
| 02:29.01 | Nugget | all right all right all right |
| 02:30.10 | aub | lol. |
| 02:30.19 | aub | are you saying im a high school girl? |
| 02:30.41 | weasel00 | Krurst, omg lmfao |
| 02:30.51 | Qwell | aub: Why? Are you? :p |
| 02:31.16 | *** join/#asterisk compumike (n=compumik@MAXIMUM-ENTROPY.MIT.EDU) |
| 02:32.05 | Nugget | be a lot cooler if you were |
| 02:32.25 | *** join/#asterisk cirgal (n=foo@wsip-70-169-190-173.sb.sd.cox.net) |
| 02:33.06 | *** join/#asterisk nephfl (n=no@wsip-68-110-130-57.ga.at.cox.net) |
| 02:33.10 | aub | Lol. |
| 02:33.16 | nephfl | ok...now i broke hud... |
| 02:33.19 | aub | why would I answer that here. |
| 02:33.36 | aub | you are all guys.. im pretty sure |
| 02:33.43 | Nugget | http://youtube.com/watch?v=vo4kDrWBa6c |
| 02:33.53 | *** join/#asterisk lovely2 (n=tylerj@fluoride.crm114.net) |
| 02:34.04 | lovely2 | hello all. |
| 02:34.15 | lovely2 | i've got a n00b question |
| 02:34.38 | lovely2 | Dial(SIP/041xxxxxxx@MainTrunk, 60, tmg) |
| 02:34.38 | nephfl | me too...i broke it can someone help me fix it? |
| 02:35.17 | lovely2 | why does that try the extension multiple times, eg if gives back busy, it just keeps trying. |
| 02:35.33 | lovely2 | until the timeout is finished. |
| 02:36.13 | nephfl | anyone? |
| 02:37.35 | flenders | lovely2: add another priority after it with: exten => X,n,Congestion |
| 02:37.48 | aub | night. |
| 02:37.54 | lovely2 | exten => s,3,Dial(SIP/0412025103@MainTrunk, 60, tmg) |
| 02:37.54 | lovely2 | exten => s,4,Goto(s-${DIALSTATUS},1) |
| 02:38.05 | lovely2 | should that work. |
| 02:38.16 | lovely2 | or is there something specal about n or Congestion |
| 02:38.30 | Nugget | only if you have valid s-BUSY and s-NOANSWER extensions and all that. |
| 02:38.41 | flenders | lovely2: dunno, what's your s-${DIALSTATUS} like? |
| 02:39.02 | aub | exten => s,800000000000000(GCALL/ui.uli.899992@gov.ca, 120) |
| 02:40.16 | *** join/#asterisk |dennis| (n=dennis@200.32.236.10) |
| 02:54.50 | PioneerVM2 | anyone here using PAP2T and their own asterisk server that can assist with caller ID? |
| 02:55.02 | cirgal | PioneerVM2: possibly |
| 02:55.11 | PioneerVM2 | im trying to get *67 blocking to work |
| 02:55.11 | cirgal | what's the question? |
| 02:55.19 | PioneerVM2 | im trying to understand whats involved |
| 02:55.26 | cirgal | PioneerVM2: already beyond my knowledge :) |
| 02:55.29 | PioneerVM2 | :( |
| 02:55.57 | PioneerVM2 | i need to figure out how asterisk receives knowledge that *67 was entered on pap2t |
| 02:57.00 | PioneerVM2 | D-Fender u there? |
| 02:57.25 | *** join/#asterisk Qwell (n=north@pdpc/sponsor/digium/Qwell) |
| 02:57.25 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 03:07.38 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 03:09.29 | *** join/#asterisk tengulre (n=tengulre@222.90.66.10) |
| 03:13.12 | *** join/#asterisk _ShrikE (n=ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 03:13.29 | *** join/#asterisk bkw_ (n=brian@adsl-70-142-57-147.dsl.tul2ok.sbcglobal.net) |
| 03:17.04 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 03:26.23 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 03:30.15 | fujin | PioneerVM2: "asterisk receives knowledge" that *62 was entered by sending any keys entered into the context specified in sip.conf |
| 03:30.23 | fujin | and in that context you should handle *62 |
| 03:31.32 | *** join/#asterisk PioneerVM2 (n=IceChat7@ool-45779466.dyn.optonline.net) |
| 03:40.00 | PioneerVM2 | hi sorry i was disconnected |
| 03:40.05 | PioneerVM2 | fujin just saw your msg |
| 03:40.19 | PioneerVM2 | fujin -- thats not correct in this case |
| 03:40.34 | PioneerVM2 | the linksys pap2t intercepts *67 (or *62 in your example) and plays a stutter tone |
| 03:40.39 | *** join/#asterisk johann8384_home (n=johann83@71-81-221-188.dhcp.stls.mo.charter.com) |
| 03:40.49 | PioneerVM2 | the asterisk server is not seeing *67 |
| 03:44.08 | fujin | then you need to configure the dialplan in the pap2t. |
| 03:45.00 | fujin | like this |
| 03:45.00 | fujin | <Dial_Plan_1_ group="Ext_1/Dial_Plan">(xxxx|1x.|2x.)</Dial_Plan_1_> |
| 03:45.04 | fujin | not sure what it's like in a pap2t |
| 03:45.07 | fujin | that's for a spa942 ^^. |
| 03:45.54 | PioneerVM2 | yea but then i wont get stutter tone in pap2 |
| 03:46.04 | PioneerVM2 | im guessing if i dial *67 555-1212 |
| 03:46.17 | PioneerVM2 | i won't hear anything and asterisk will see *675551212 |
| 03:46.18 | PioneerVM2 | right? |
| 03:48.46 | fujin | don't know |
| 03:48.48 | fujin | depends |
| 03:48.52 | fujin | you can make the dialplan strip off |
| 03:48.55 | fujin | and prepend extra stuff |
| 03:49.01 | fujin | fully depends on your pap2t's configuration. |
| 03:52.51 | *** join/#asterisk Nuitari (n=Nuitari@mail.nuitari.net) |
| 03:52.57 | *** join/#asterisk bmg505 (n=leon@196.209.183.188) |
| 03:53.59 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
| 03:55.27 | *** part/#asterisk cirgal (n=foo@wsip-70-169-190-173.sb.sd.cox.net) |
| 03:58.33 | *** join/#asterisk antimoof (n=dogcow@netbsd/developer/antimoof) |
| 04:00.43 | *** join/#asterisk [T]ank (n=ckwall@c-71-195-196-198.hsd1.ut.comcast.net) |
| 04:01.07 | craigk | i hope that this is not a silly question, but what is the difference between asterisk 1.2 and 1.4 ? |
| 04:01.15 | Nuitari | .2 |
| 04:01.33 | Nuitari | seriously, there are a ton of changes everywhere |
| 04:02.03 | Nuitari | craigk: www.asterisk.org will have some info |
| 04:02.55 | craigk | i am reading that - but mostly what i see is that both versions/streams are quite active and have on going releases ... i would assume that 1.4 is the latest, but 1.2 seems to still be getting updates/release so was wondering why |
| 04:03.10 | *** join/#asterisk IgorG (n=FeedomPa@195.162.32.126) |
| 04:03.39 | craigk | ah - just found it "Version 1.2.24 is the final 1.2 release that contains normal bug fixes. The 1.2 branch will only be maintained with security fix releases from now until it is completely deprecated." |
| 04:03.51 | craigk | sorry to ask a question that is documented ... i just had trouble finding it :) |
| 04:14.21 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
| 04:15.24 | *** join/#asterisk TheGuy- (n=None@c-67-188-71-166.hsd1.ca.comcast.net) |
| 04:16.12 | TheGuy- | Has anyone heard of VoiceRD? |
| 04:29.26 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 04:39.25 | weasel00 | if i an running linux2.6 kernels and load the ztdumy driver, would that cause issues with the conferencing audio? |
| 04:45.23 | Krurst | it would cause issues if it wasn't loaded, wouldn't it? |
| 04:45.23 | *** join/#asterisk cirgal (n=cirgal@pool-71-102-137-33.snloca.dsl-w.verizon.net) |
| 04:45.50 | weasel00 | there are docs saying that it isnt nessesary as a timing device with 2.6 kernel |
| 04:46.43 | *** part/#asterisk cirgal (n=cirgal@pool-71-102-137-33.snloca.dsl-w.verizon.net) |
| 04:51.51 | Krurst | no, the usb driver isn't nessesary |
| 04:52.06 | Krurst | a zaptel driver of some sort is still nessesary |
| 04:52.38 | tzafrir_laptop | In 2.6 it's not a USB driver |
| 04:53.04 | tzafrir_laptop | weasel00, run zttest for a minute or so |
| 04:53.11 | tzafrir_laptop | (press ctrl-c to stop) |
| 04:53.17 | tzafrir_laptop | what do you get? |
| 04:57.05 | weasel00 | http://pastebin.ca/675463 |
| 04:57.45 | weasel00 | no zap interface without loading ztdummy... |
| 05:00.07 | Krurst | those results don't look very nice. |
| 05:01.03 | Krurst | have you tried recompiling? |
| 05:01.05 | weasel00 | what should it look like? |
| 05:01.12 | weasel00 | this is the recompile =) |
| 05:01.50 | weasel00 | the box isnt beefy enough to much more than route peers... this kind of confirms it |
| 05:01.57 | Krurst | you should be seeing between 98% and 100%. not -200% |
| 05:02.26 | Krurst | beefy shouldn't matter. have a look at your dmesg and see if there's any errors |
| 05:03.48 | Krurst | you might be getting some kind of interrupt problem |
| 05:05.00 | weasel00 | http://pastebin.ca/675466 |
| 05:07.03 | weasel00 | goal was to get this setup for easier communication between field techs as they bounce around through different offices across the country |
| 05:07.25 | weasel00 | so we scrounged parts for it lol |
| 05:07.42 | Krurst | hmm looks ok to me. you might still want to try disabling ACPI |
| 05:09.33 | *** join/#asterisk codefreeze (n=steve_mu@216.166.159.235) |
| 05:09.33 | *** mode/#asterisk [+o codefreeze] by ChanServ |
| 05:10.48 | Krurst | and you can always give up and use callweaver - it apparently doesn't need a timing device for it's conference. |
| 05:11.25 | bkw_ | FreeSWITCH doesn't either |
| 05:11.29 | bkw_ | its all software |
| 05:11.34 | bkw_ | and even can do wideband |
| 05:13.14 | Krurst | I wish wideband was more widespread. |
| 05:13.49 | weasel00 | the sounds come across fine on their own.. moh, ivr, vmail.. all sound perfect.. conference calls - omg choppy |
| 05:14.21 | Krurst | yeah, thats your dodgy timing device at work. |
| 05:14.59 | Krurst | you'll probably also get bad moh if you enable silence supression |
| 05:15.10 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
| 05:15.51 | Krurst | which ver of zaptel are you using? |
| 05:16.13 | weasel00 | 1.4 |
| 05:16.49 | weasel00 | moh was sounding fine.. until i used moh for the empty conference room.. then it too was choppy |
| 05:18.17 | Krurst | did you use "make clean; ./configure; make;make install"? |
| 05:18.47 | weasel00 | yep |
| 05:19.28 | Krurst | well, after you disable acpi, and maybe try the svn trunk, I'm out of ideas. |
| 05:19.51 | weasel00 | ya... we will know more after that.. |
| 05:21.10 | weasel00 | not going to be terribly heartbroken if all i cant do is conferencing =) |
| 05:23.45 | weasel00 | oh wow |
| 05:24.03 | weasel00 | 99.95 - 99.96 now |
| 05:24.59 | Krurst | hows your conference sound now? |
| 05:25.05 | weasel00 | --- Results after 56 passes --- |
| 05:25.05 | weasel00 | Best: 99.963379 -- Worst: 99.951172 -- Average: 99.952480 |
| 05:25.18 | weasel00 | no idea.. have to reset everything up |
| 05:25.38 | JT | still unacceptable |
| 05:25.57 | JT | zttest must always be 99.97% or above |
| 05:26.10 | Krurst | ahhh it's better than -200% |
| 05:30.12 | *** join/#asterisk jarod14 (n=jarod14@mail.viatelecom.com) |
| 05:34.23 | *** join/#asterisk [T]ank (n=ckwall@c-71-195-196-198.hsd1.ut.comcast.net) |
| 05:35.30 | [T]ank | is it normal that if I make an outbound call to a number and the dialstatus is noanswer that I should get a record in my cdr database not only for that call out, but also another one into extension s of my defined inbound context of my zap channel? |
| 05:36.04 | jarod14 | hi guys |
| 05:38.33 | [T]ank | here is what I am seeing... this is the cdr data from one call. I have two records. one for the outbound call, and one that is coming back in. http://pastebin.ca/675476 |
| 05:39.49 | [T]ank | it does not happen when the call is answered... only when it does not answer. |
| 05:40.20 | [T]ank | any ideas? |
| 05:43.26 | [T]ank | :'( everyone sleeping. |
| 05:44.13 | weasel00 | Krurst, killing the acpi was the problem =) woot!! |
| 05:44.38 | weasel00 | [T]ank, i would love to help but you are working with something beyond me :( |
| 05:50.16 | *** join/#asterisk toddejohnson (n=toddejoh@69.220.214.65) |
| 05:50.32 | *** join/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net) |
| 05:51.46 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@216-110-94-230.static.twtelecom.net) |
| 05:52.55 | *** join/#asterisk rodent|S (i=nobody@foster.stonedcoder.org) |
| 05:53.45 | jarod14 | [T]ank, both cdr do not have the same uniqueid, it seems there is two different calls, don't you have any loop in your extensions file? |
| 05:54.17 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@216-110-94-230.static.twtelecom.net) |
| 05:55.22 | jarod14 | or any fork making a second call ? |
| 05:55.40 | *** join/#asterisk mitcheloc (n=mitchel@adsl-67-126-140-84.dsl.irvnca.pacbell.net) |
| 06:02.10 | [T]ank | jarod14: no loop |
| 06:06.18 | *** join/#asterisk NightMonkey (n=NightMon@pdpc/supporter/sustaining/NightMonkey) |
| 06:06.50 | *** join/#asterisk Krurst (n=me@eth244.wa.adsl.internode.on.net) |
| 06:07.47 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) [NETSPLIT VICTIM] |
| 06:20.17 | *** join/#asterisk [hC] (n=hardcore@S01060016b6b53c0c.vc.shawcable.net) |
| 06:26.46 | *** join/#asterisk |dennis| (n=dennis@200.32.236.10) |
| 06:30.27 | *** join/#asterisk stimpie (n=michiel@ip565faf27.direct-adsl.nl) |
| 06:32.13 | *** join/#asterisk lbow (n=lbow@dsl-241-31-00.telkomadsl.co.za) |
| 06:32.44 | jarod14 | [T]ank, could you copy on pastebin, the corresponding section of your extensions.conf ? |
| 06:40.32 | *** join/#asterisk zapp-branigan (n=zapp-bra@9.218.216.87.static.jazztel.es) |
| 06:44.31 | McDouglas | why cant i make an outgoing call with this dialplan: http://pastebin.com/d4efd6b65 ? |
| 06:47.10 | Krurst | are you dialing 11 digit numbers? |
| 06:47.11 | many | probably because the numbers dont match |
| 06:48.20 | many | most people use a prefix to mark outgoing numbers, then you could use _0XX. or something as match |
| 06:49.14 | *** join/#asterisk tzafrir (n=tzafrir@62.90.10.53) |
| 06:51.13 | Krurst | well you're only using 2 digit extentions, so you could just use "_XXX." |
| 06:52.30 | McDouglas | i'm dialing 0630xxxxxxx |
| 06:52.34 | McDouglas | so its 10 nmber |
| 06:52.42 | McDouglas | oops |
| 06:52.42 | McDouglas | 11 |
| 06:52.43 | McDouglas | :P |
| 06:53.09 | McDouglas | is it a problem? |
| 06:54.23 | McDouglas | many: i dont want to use 0 to "get" a line |
| 06:57.19 | flenders | hey, how do I test the lenght of a variable? |
| 06:57.43 | *** join/#asterisk [hC] (n=hardcore@S01060016b6b53c0c.vc.shawcable.net) |
| 06:57.43 | flenders | for example, a callerid? |
| 06:57.53 | *** join/#asterisk kiscokid (n=Ron_Laut@adsl-216-101-109-187.dsl.snfc21.pacbell.net) |
| 06:58.32 | kiscokid | I got my first * into production today |
| 06:58.57 | *** join/#asterisk tsurko (n=tsurko@213.91.216.130) |
| 07:00.40 | *** join/#asterisk zeeesh (i=zeeesh@202.166.161.45) |
| 07:01.28 | McDouglas | hm |
| 07:01.45 | McDouglas | i can make an outgoing call with a SIP phone withthis dialplan: http://pastebin.com/d4efd6b65 |
| 07:01.57 | McDouglas | but cant make it with an analog phone connected to Zap/1 |
| 07:02.47 | kiscokid | what happens when you enter the command zap show status? |
| 07:03.00 | flenders | McDouglas: can you receive calls on zap/1? |
| 07:03.18 | jarod14 | flenders about the length : exten => _X.,n,Set(LENGTH=${LEN(${CALLERID})}) |
| 07:03.25 | *** join/#asterisk toddejohnson (n=toddejoh@69.220.214.65) |
| 07:03.37 | McDouglas | flenders: well, i can dial zap/1 from the sip phone |
| 07:03.42 | McDouglas | also can dial the ship phone from zap/1 |
| 07:04.02 | jarod14 | this set the varaible LENGTH as the length of digit of the CALLERID vairable |
| 07:04.32 | McDouglas | kiscokid: http://pastebin.com/d2f125d48 |
| 07:05.20 | kiscokid | looks ok to me |
| 07:05.46 | flenders | jarod14: thanks a lot |
| 07:06.36 | jarod14 | ;o) |
| 07:06.42 | kiscokid | McDouglas: zap/3 is an fxo? |
| 07:07.19 | McDouglas | kiscokid: of course |
| 07:07.26 | kiscokid | just checking |
| 07:08.03 | kiscokid | and I assume you get dialtone when you plug an analog phone into it |
| 07:08.27 | McDouglas | you mean plug into the asterisk or into the line? |
| 07:08.40 | McDouglas | but anyway, yes in both case |
| 07:09.27 | kiscokid | let me look at one of my dialplans |
| 07:10.49 | Krurst | can you pop something like http://pastebin.com/m24ccaaed in to make sure * is reading the digits properly? |
| 07:11.44 | *** join/#asterisk steliosk (n=Stelios@ipa226.211.tellas.gr) |
| 07:13.18 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 07:13.33 | Krurst | hay, anyone know if you can get moh to play instead of the dailtone? |
| 07:14.35 | *** join/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net) |
| 07:14.42 | kiscokid | McDouglas:: looks ok to me, I just don't have the congestion items cause I assumed you would get congestion automatically |
| 07:15.59 | McDouglas | Krurst: omg you are onto something |
| 07:16.15 | McDouglas | i entered: 06307476553 |
| 07:16.19 | McDouglas | asterisk said: User entered '060747655' |
| 07:16.45 | *** join/#asterisk denon (n=denon@208.122.43.201) |
| 07:16.45 | *** mode/#asterisk [+o denon] by ChanServ |
| 07:16.58 | FlatFoot | morning all , if anyone is using Snom 320 i have just found that version 6.5.8 does NOT work correctly . You need to upgrade to 6.5.10 |
| 07:17.00 | Krurst | ok, well now we know whats wrong, we need to fix it. |
| 07:17.15 | Krurst | I think theres a relax dtmf setting somewhere... |
| 07:17.34 | McDouglas | if i use SIP phone |
| 07:17.37 | McDouglas | asterisk says: -- User entered '06307476553' |
| 07:17.42 | McDouglas | the correct number |
| 07:18.18 | Krurst | put relaxdtmf=yes in your zapata.conf |
| 07:18.26 | Krurst | see if that helps |
| 07:19.26 | Krurst | it "may make the DTMF detector more likely to have "talkoff" where DTMF is detected when it shouldn't be"... |
| 07:19.29 | *** join/#asterisk Iamnacho (i=Iamnacho@ip68-103-153-140.ks.ok.cox.net) |
| 07:20.24 | McDouglas | Krurst: omg you are a god |
| 07:20.28 | McDouglas | working now ;) |
| 07:20.54 | kiscokid | cool |
| 07:21.19 | McDouglas | thanks |
| 07:21.22 | Krurst | glad I could help. |
| 07:21.33 | *** join/#asterisk Azam (n=azamzia@58-65-160-140.nayatel.pk) |
| 07:23.11 | Azam | Hello |
| 07:23.15 | Azam | is anyone there??? |
| 07:23.30 | JT | no. |
| 07:24.01 | Azam | i need help regarding asterisk |
| 07:25.13 | Azam | in function ast_app_getdata |
| 07:25.51 | Azam | can u tell me in timeout aurgument the time is in seconds or milliseconds or wat? |
| 07:26.28 | Azam | JT can u help me please |
| 07:26.57 | Azam | or anyone |
| 07:34.08 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 07:37.53 | CCFL_Man2 | which 2.6 kernel has a 1000Hz timebase by default? |
| 07:39.12 | JT | one that yoy compile. |
| 07:39.14 | JT | you |
| 07:39.18 | JT | the default is 250Hz |
| 07:40.15 | CCFL_Man2 | i hear the early 2.6 kernels have that timebase |
| 07:40.25 | CCFL_Man2 | like 2.6.11 |
| 07:40.25 | tzafrir | CCFL_Man2, it was 1000Hz from up to 2.6.13, so you may be confused by some old documentation |
| 07:40.58 | tzafrir | sorry: up to 2.6.12 . 2.6.13 made it configurable, and changed the defualt to 250 |
| 07:41.00 | CCFL_Man2 | tzafrir: i read your site, infact you were the one i wanted to ask |
| 07:41.25 | CCFL_Man2 | tzafrir: so get a 2.6.12 kernel then? |
| 07:41.32 | tzafrir | no |
| 07:41.40 | tzafrir | what kernel do you have? |
| 07:41.52 | tzafrir | what distro? |
| 07:42.03 | CCFL_Man2 | i had 2.6.18, debian sparc |
| 07:42.19 | JT | CCFL_Man2: so just compile the damn kernel with 1000Hz |
| 07:42.22 | JT | it's not terribly hard. |
| 07:42.30 | CCFL_Man2 | i tried to build my own but i really don't know what i need |
| 07:42.44 | CCFL_Man2 | JT: this is a sun netra |
| 07:42.52 | tzafrir | ah, so I guess that USE_RTC won't work. and a UHCI USB controller is not available |
| 07:43.10 | tzafrir | So I guess you'll need to rebuild the kernel, indeed |
| 07:43.29 | CCFL_Man2 | tzafrir: if 2.6.12 has a default time base of 1000Hz? |
| 07:43.32 | JT | CCFL_Man2: you'll have to learn |
| 07:43.43 | JT | CCFL_Man2: T200 or something else? |
| 07:43.55 | CCFL_Man2 | t1 200 |
| 07:44.03 | tzafrir | CCFL_Man2, downgrading the kernel just for that is not the best idea. And you'll still have to use a custom kernel |
| 07:44.16 | JT | AC200 then |
| 07:44.28 | CCFL_Man2 | JT: right |
| 07:44.29 | JT | CCFL_Man2: first question, why are you running linux? |
| 07:44.56 | CCFL_Man2 | JT: asterisk timing |
| 07:45.10 | tzafrir | any chance sparc supports high resolution timers of latest kernels? |
| 07:45.51 | JT | CCFL_Man2: moot point on suns anyway, zaptel only works for sure on x86 linux |
| 07:46.07 | CCFL_Man2 | tzafrir: why would i need to build a new one anyway? when i do the install it installs a 2.4 kernel |
| 07:46.22 | JT | CCFL_Man2: because it's good practise |
| 07:46.28 | JT | seriously |
| 07:46.41 | JT | admining linux without knowing how to recompile a kernel |
| 07:46.42 | JT | ... |
| 07:46.49 | CCFL_Man2 | JT: i have to ho through all kernel options |
| 07:46.59 | tzafrir | CCFL_Man2, you don't seem to be listening |
| 07:47.08 | JT | as opposed to spending days here whingeing about it |
| 07:47.08 | JT | right |
| 07:47.11 | JT | great logic there |
| 07:47.22 | tzafrir | apt-get install kernel-source-`uname -r` |
| 07:47.33 | tzafrir | reconfigure it |
| 07:47.35 | tzafrir | etc. |
| 07:47.46 | tzafrir | adjust ocnfiguration a bit |
| 07:47.53 | CCFL_Man2 | tzafrir: exactly what i did, the built kernel panics |
| 07:48.02 | tzafrir | HZ=1000, PEREMPT=yes, etc |
| 07:48.25 | tzafrir | CCFL_Man2, generally ask about that on #debian |
| 07:49.07 | *** join/#asterisk gerphimum (n=trekkie@cpe-70-125-148-108.satx.res.rr.com) |
| 07:50.12 | *** join/#asterisk tzafrir (n=tzafrir@62.90.10.53) |
| 07:50.29 | tzafrir | duh, used /quit to part from a channel :-( |
| 07:50.31 | CCFL_Man2 | there are over 100 kernel options, it takes me a half our to go through them all, and the defaults do not work on a sparc system |
| 07:50.49 | CCFL_Man2 | it panics |
| 07:51.04 | CCFL_Man2 | most options are set to "m" |
| 07:51.26 | tzafrir | CCFL_Man2, you did start from cp /boot/config-`uname -r` .config before make menuconfig, right? |
| 07:51.35 | *** part/#asterisk Nuitari (n=Nuitari@mail.nuitari.net) |
| 07:51.56 | CCFL_Man2 | i leave them at "m" and it doesn't work |
| 07:52.09 | CCFL_Man2 | tzafrir: you mean back it up? |
| 07:52.31 | CCFL_Man2 | i used make config |
| 07:52.34 | tzafrir | I mean: start from that config, as it is known to work. And change it as little as possible. |
| 07:53.42 | CCFL_Man2 | tzafrir: that'll take forever |
| 07:54.00 | tzafrir | What will take forever? |
| 07:54.20 | CCFL_Man2 | going through all that |
| 07:54.22 | *** join/#asterisk qdk (n=qdk@213.150.62.32) |
| 07:54.47 | CCFL_Man2 | you mean opening it in a text editor and doing it that way? |
| 07:57.30 | CCFL_Man2 | 2.6.12 isn't too far off, and etch doesn't even use the latest kernel from kernel.rg |
| 07:58.10 | CCFL_Man2 | or, just get an x100p card to provide timing |
| 07:58.30 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@216-110-94-230.static.twtelecom.net) |
| 07:58.43 | citats | CCFL_Man2: what he means is take that file and copy it over to .config in your kernel source dir before running make menuconfig, that will set your default settings to what your currently using |
| 08:00.05 | *** part/#asterisk kiscokid (n=Ron_Laut@adsl-216-101-109-187.dsl.snfc21.pacbell.net) |
| 08:01.56 | *** join/#asterisk Strom_M (n=strom@adsl-69-105-168-167.dsl.irvnca.pacbell.net) |
| 08:07.27 | *** join/#asterisk Delvar (n=Delvar@77.240.56.18) |
| 08:13.47 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@216-110-94-230.static.twtelecom.net) |
| 08:15.42 | *** join/#asterisk razu (n=razu@razu.data.ee) |
| 08:21.49 | *** join/#asterisk KermitTheFragger (n=siepkes@118-197.bbned.dsl.internl.net) |
| 08:23.39 | *** join/#asterisk snook3r (n=snook3r@bzq-219-46-202.isdn.bezeqint.net) |
| 08:30.44 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 08:34.13 | *** join/#asterisk Trentster (n=Trike@d58-104-33-175.dsl.vic.optusnet.com.au) |
| 08:35.59 | *** join/#asterisk engrxyz (i=1000@host81-150-247-250.in-addr.btopenworld.com) |
| 08:38.38 | Trentster | Hey all, I am busy setting up and iax trunk to a remote asterisk server, it does not seem to connect properly, I have opened udp 4569 incoming and out going and portforwarded to the asterisk box, on local server iax2 show peers shows " Colo/Mark (Unspecified) (D) 255.255.255.255 0 UNKNOWN " and on remote side looks ok "Mark/Colo 58.104.xx.xxx (D) 255.255.255.255 60427 OK (535 ms) " ex |
| 08:38.38 | Trentster | cept for showing the port as 60427 instead of 4569, any ideas? |
| 08:41.26 | *** join/#asterisk lbow (n=lbow@wbs-41-208-224-157.wbs.co.za) |
| 08:45.08 | *** join/#asterisk kclaussen (n=kclausse@204.13.224.242) |
| 08:47.17 | *** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust826.leed.cable.ntl.com) |
| 08:47.48 | *** join/#asterisk snook3r (n=snook3r@bzq-219-46-202.isdn.bezeqint.net) |
| 08:52.14 | kippi | is there away to load the a new codec without restarting asterisk? |
| 08:55.11 | *** join/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-ffbd722513cca42a) |
| 09:03.27 | *** join/#asterisk snook3r (n=snook3r@bzq-219-46-202.isdn.bezeqint.net) |
| 09:04.13 | JT | CCFL_Man2: just read about compiling kernels already |
| 09:04.24 | JT | of course it's hard to do something you don't know how to do. |
| 09:05.24 | So3kris | hello |
| 09:06.10 | So3kris | <PROTECTED> |
| 09:14.23 | *** join/#asterisk saftsack (n=oliver@p54A7BE15.dip.t-dialin.net) |
| 09:23.37 | s0ck | morning JT |
| 09:24.52 | *** join/#asterisk chendy (n=chendy@218.242.110.26) |
| 09:29.30 | gerphimum | morning all - does anyone know if there is a hardware device that can broadcast to cb radio frequencies (ie: walkie talkies) ? specifically, is it possible for asterisk to broadcast on a cb freq instead of, say, an intercom, assuming such hardware exists |
| 09:35.07 | ruskie | http://www.theregister.co.uk/2007/08/29/google_yahoo_amazon_sued_ove_email_patents <-- lol |
| 09:35.53 | mvanbaak | <PROTECTED> |
| 09:35.59 | mvanbaak | what can it be ? |
| 09:38.20 | *** join/#asterisk tengulre (n=tengulre@222.90.66.10) |
| 09:40.15 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
| 09:48.13 | mvanbaak | found it, never mind |
| 09:56.54 | *** join/#asterisk lbow (n=lbow@dsl-241-59-244.telkomadsl.co.za) |
| 09:58.14 | *** join/#asterisk Krurst (n=me@55.90.233.220.exetel.com.au) |
| 09:58.29 | *** join/#asterisk af_ (n=getsmart@81-174-47-128.dynamic.ngi.it) |
| 10:04.07 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 10:06.32 | *** join/#asterisk IgorG (n=FeedomPa@195.162.32.126) |
| 10:09.15 | *** join/#asterisk shinao1 (n=shinao1@217.20.242.51) |
| 10:19.22 | *** join/#asterisk alin` (n=user@193.226.173.50) |
| 10:19.28 | alin` | what is the diffenrrence between g723 and g723.1 ? |
| 10:21.56 | *** join/#asterisk Woifi1988 (n=anon@M1049P008.adsl.highway.telekom.at) |
| 10:24.55 | *** join/#asterisk saftsack (n=saftsack@p57A76966.dip.t-dialin.net) |
| 10:26.15 | *** join/#asterisk mitcheloc (n=mitchel@adsl-67-126-140-84.dsl.irvnca.pacbell.net) |
| 10:26.33 | McDouglas | is it possible to play music for a caller while asterisk dials the given extension? |
| 10:30.21 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 10:33.18 | *** join/#asterisk |dennis| (n=dennis@200.32.233.84) |
| 10:37.42 | JunK-Y | McDouglas: see option m in dial. |
| 10:47.44 | *** join/#asterisk mitcheloc (n=mitchel@adsl-67-126-140-84.dsl.irvnca.pacbell.net) |
| 10:48.51 | *** join/#asterisk hwt (n=hwt@curb.thorkildssen.com) |
| 10:49.21 | hwt | hey. is it possible to "tap into" a meetme channel? that is, only a spesific user in a meetme conference. |
| 10:49.31 | hwt | to find out who is making noise and kick them. |
| 10:51.34 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 10:56.12 | *** join/#asterisk imagitronics (n=tofarley@YRMfb-01p2-113.ppp11.odn.ad.jp) |
| 11:04.00 | mvanbaak | hwt: you can use /4 |
| 11:04.04 | mvanbaak | oops |
| 11:04.23 | *** join/#asterisk steliosk (n=Stelios@ipa226.211.tellas.gr) |
| 11:05.11 | FlatFoot | hello all , using 1.2 addons to get mysql support but a make errors on format_mp3 DO i need this or can it be hashed out of the make config ? if so any idea how to do . TIA |
| 11:08.22 | *** join/#asterisk shtoom (n=shtoom@123.252.144.92) |
| 11:08.52 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 11:09.09 | *** join/#asterisk RsaMan (n=aa@196.210.155.3) |
| 11:09.37 | *** join/#asterisk xtr-II (n=94752345@216.19.191.191.novuscom.net) |
| 11:11.22 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 11:13.16 | *** join/#asterisk J4k3 (n=jsuter@openwrt.us) |
| 11:14.18 | J4k3 | (telco's fiber went down for an hour, wasn't sure how long it was going to be down so I got prepared for a long term outage by reconfiguring our local POTS, which is normally forwarded to a local voip #) |
| 11:14.40 | J4k3 | suprisingly the audio is great now that I don't have any other phones on that line, plugged in |
| 11:14.50 | J4k3 | it buzzed like hell with 3 other phones attached. |
| 11:14.58 | J4k3 | shouldn't have but it did |
| 11:16.15 | RsaMan | http://www.voip-info.org/wiki/view/Asterisk+Connect+2+servers |
| 11:16.18 | *** join/#asterisk Daejeo1 (n=chatzill@211.177.189.25) |
| 11:16.29 | RsaMan | can someone verify method 3 |
| 11:16.34 | RsaMan | i dont think it is correct |
| 11:16.40 | Daejeo1 | hello friends |
| 11:16.44 | RsaMan | hello |
| 11:17.39 | Daejeo1 | anyone tried "VIDEO IP PHONE MOTOROLA® Ojo " with Asterisk server so far? |
| 11:18.16 | Daejeo1 | it is available on ebay now |
| 11:18.19 | Daejeo1 | http://cgi.ebay.com/VIDEO-PHONE-MOTOROLA-IP-Ojo-Caller-ID-Cordless-New-IBx_W0QQitemZ180065877941QQihZ008QQcategoryZ33932QQrdZ1QQssPageNameZWD1VQQcmdZViewItem |
| 11:18.29 | Daejeo1 | I want to buy one |
| 11:19.02 | Daejeo1 | I wonder if there is any issue |
| 11:19.12 | McDouglas | hmm, how can i tell asterisk which smtp server to use (to send the voicemail messgaes) |
| 11:19.55 | J4k3 | is it sip-based at all? |
| 11:20.04 | J4k3 | the writeup makes it sound proprietary to some degree |
| 11:21.00 | Daejeo1 | J4k3: I did not do much research on it |
| 11:22.17 | supers | McDouglas: there may be a setting, i'm not sure. it usally spawns /usr/sbin/sendmail though. |
| 11:23.03 | McDouglas | how do sendmail know what server to use then? |
| 11:23.47 | Daejeo1 | J4K3:http://blogs.zdnet.com/ip-telephony/?p=170 |
| 11:24.31 | *** join/#asterisk michael-i (n=michael-@141.41.40.55) |
| 11:25.18 | michael-i | Does anyone here speak russian? I don't and need someone to tell me what a sound file says. |
| 11:25.33 | Daejeo1 | J4k3: http://www.bestwebbuys.com/electronics/Motorola_Ojo_Personal_Video_Phone_PVP1000_-_IP_video_phone-4740802.html?isrc=e-search |
| 11:25.45 | J4k3 | Daejeo1: it should work fine, technically. |
| 11:26.08 | RsaMan | ~pb |
| 11:26.09 | jbot | A Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 11:26.13 | J4k3 | the video will be passed through fine from what I've been told about h.264 devices working with * |
| 11:26.49 | Daejeo1 | right. |
| 11:26.53 | J4k3 | but "it should" probably isn't what you were looking for ;) |
| 11:27.00 | RsaMan | does www.pastebin.ca work? |
| 11:27.30 | Daejeo1 | J4k3: :) |
| 11:32.15 | RsaMan | trying to connect to asterisk servers via iax,, |
| 11:32.23 | RsaMan | they connect when i check iax2 show peers |
| 11:32.30 | RsaMan | but when i try make a call i get this |
| 11:33.21 | RsaMan | [Aug 30 13:28:44] NOTICE[24749]: chan_iax2.c:7398 socket_process: Host xxxx0.155.3 failed to authenticate as 888 |
| 11:35.03 | supers | McDouglas: by default it will send it out via port 25, you can setup sendmail to relay to another smtp server. |
| 11:35.20 | McDouglas | how? |
| 11:35.45 | Daviey | dump sendmail - get exim |
| 11:36.01 | Daviey | razu: |
| 11:37.39 | *** join/#asterisk blinky42 (n=me@67.106.175.130.ptr.us.xo.net) |
| 11:38.07 | *** join/#asterisk snook3r (n=snook3r@bzq-219-46-202.isdn.bezeqint.net) |
| 11:38.57 | razu | Daviey ? |
| 11:39.31 | Daviey | razu: sorry, Tab error |
| 11:39.37 | razu | :) |
| 11:40.39 | razu | have anyone had problems with connecting cisco 7941g to asterisk ... mine show-s it's registering and stays that way ... |
| 11:41.23 | *** join/#asterisk Galeras (n=Galeras@190.84.206.174) |
| 11:42.04 | razu | as I saw somewhere then old asterisks and 79xx have communicating problems with registration ... is it possible that upgrading asterisk will solve the problem ? |
| 11:42.36 | Daejeo1 | razu :what is your sip firmware version? |
| 11:42.49 | razu | SIP41.8-2-2SR3S |
| 11:43.15 | Daejeo1 | asterisk version? |
| 11:43.27 | razu | CVS-v1-0-06/15/05-11:02:26 |
| 11:43.30 | razu | real old one :) |
| 11:44.03 | hwt | mvanbaak: /4? |
| 11:44.06 | hwt | mvanbaak: what do you mean? |
| 11:44.15 | razu | of course there might be something wrong in my config file |
| 11:44.25 | Daejeo1 | try sip firmware 8.6 |
| 11:44.32 | McDouglas | talking about cisco, anyone knows if the 7911g compatible with asterisk? |
| 11:45.15 | razu | Daejeo1, this can be found somewhere on cisco web ? |
| 11:45.57 | Daejeo1 | do you have a CCO account? |
| 11:46.29 | Daejeo1 | it is not free to download |
| 11:46.35 | Daejeo1 | but |
| 11:46.48 | Daejeo1 | you can get it for free from wiki |
| 11:47.58 | Daejeo1 | also learn more about how to upgrade |
| 11:48.01 | Daejeo1 | http://www.it4u2.com/asterisk2.htm |
| 11:48.03 | hwt | is it possible to "tap into" a meetme channel? that is, only a spesific user in a meetme conference. |
| 11:48.06 | hwt | to find out who is making noise and kick them. |
| 11:48.18 | McDouglas | http://www.cisco.com/cgi-bin/tablebuild.pl/sip-ip-phone7960 |
| 11:49.03 | Daejeo1 | you can download sip 8.6 http://www.voip-info.org/wiki/index.php?page=Asterisk+phone+cisco+79xx |
| 11:50.37 | *** join/#asterisk Daejeo1 (n=chatzill@211.177.189.25) |
| 11:50.51 | Daejeo1 | razu: ? |
| 11:52.19 | Daejeo1 | McDouglas: do you have CCO access on cisco website? |
| 11:52.31 | McDouglas | yes |
| 11:53.00 | mvanbaak | hwt: I was going to say: use the chanspy stuff for that |
| 11:53.09 | *** join/#asterisk shinao1 (n=shinao1@217.20.242.51) |
| 11:53.13 | mvanbaak | but halfway the line I decided to watch irc screen 4 |
| 11:53.17 | mvanbaak | sorry, my mistake |
| 11:53.27 | Daejeo1 | what cisco phone do you have? |
| 11:53.33 | puzzled | Daejeo1: do you know what the latest sip release is for the 7941/7961? |
| 11:53.37 | McDouglas | none |
| 11:53.41 | Daejeo1 | 8.7 |
| 11:53.53 | Daejeo1 | 8.7 is latest |
| 11:53.54 | puzzled | thanks |
| 11:55.18 | Daejeo1 | McDouglas: i sent pvt msg |
| 11:55.34 | RsaMan | hello guys |
| 11:55.41 | RsaMan | having probs connecting two asterisk servers |
| 11:55.45 | RsaMan | they register |
| 11:55.46 | RsaMan | but.. |
| 11:55.56 | puzzled | Daejeo1: I have 8.3 for my 7961 but I am still using the factory sccp 3.x release since it works fine |
| 11:56.11 | RsaMan | http://paste.debian.net/35785 |
| 11:56.20 | RsaMan | i get this error |
| 11:56.31 | *** join/#asterisk coppice (n=chatzill@140.196.17.210.dyn.pacific.net.hk) |
| 11:56.51 | RsaMan | failed to authenticate as 888 on "server" |
| 11:57.01 | RsaMan | failed to authenticate as 888 |
| 11:57.05 | RsaMan | on the server side |
| 11:57.26 | puzzled | RsaMan: try without a password first and see if that fixes it. maybe it's just a password typo |
| 11:58.01 | RsaMan | kk |
| 11:58.20 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 11:58.37 | *** join/#asterisk jarod14 (n=jarod14@mail.viatelecom.com) |
| 11:58.44 | RsaMan | but it registers |
| 12:00.10 | razu | Daejeo1, well ... I found SIP41.8-3-2S from CCO ... or should I use P003-08-6-00 |
| 12:01.00 | razu | Daejeo1, phone I have is CP-7941g |
| 12:01.21 | hwt | mvanbaak: i haven an exten => foo,1,Chanspy() |
| 12:01.39 | hwt | mvanbaak: but all it does is beeping, no matter what i press (nums, *, #) |
| 12:01.46 | puzzled | razu: I would get the latest release |
| 12:01.49 | hwt | mvanbaak: i want to be able to cycle through channels |
| 12:02.17 | hwt | mvanbaak: that is; chanspy(scan) |
| 12:04.58 | Daejeo1 | razu: 8.6 or 8.7 |
| 12:08.58 | Daejeo1 | razu : http://www.it4u2.com/asterisk2.htm |
| 12:09.52 | puzzled | Daejeo1: I thought the procedure to upgrade a 7941/7961 is different from a 7940/7960? |
| 12:09.57 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 12:11.42 | razu | I'm thinking so also at the moment : cause my app load ID is jar41sip.8-3-1-22 ... |
| 12:12.38 | *** join/#asterisk snook3r (n=snook3r@bzq-219-46-202.isdn.bezeqint.net) |
| 12:13.59 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 12:18.00 | *** join/#asterisk torch (n=logan@200.99.7.98) |
| 12:18.08 | torch | hi all ... |
| 12:19.35 | hwt | mvanbaak: ok, now it works, but i get audio in both directions, hence it will be just as listening to the conference |
| 12:19.54 | hwt | mvanbaak: and it seems like there is no way to just get one direction of the audio |
| 12:20.57 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 12:23.40 | *** join/#asterisk Corydon76-dig (i=orange@pdpc/supporter/sustaining/Corydon76-home) |
| 12:23.40 | *** mode/#asterisk [+o Corydon76-dig] by ChanServ |
| 12:27.45 | *** join/#asterisk lirakis (n=etamme@65.200.191.253) |
| 12:31.05 | RsaMan | aarg |
| 12:31.21 | RsaMan | my blind call transfers dont seem to work with my digium card |
| 12:31.33 | RsaMan | anyone have similar issues ? |
| 12:36.05 | *** join/#asterisk destructure (n=oyashiro@rrcs-24-173-126-174.se.biz.rr.com) |
| 12:41.07 | *** join/#asterisk flart (n=flart@atommuell.mum.jku.at) |
| 12:41.11 | flart | hi |
| 12:41.31 | destructure | hi |
| 12:44.46 | lirakis | hola |
| 12:47.20 | _x86_ | flart: morgen |
| 12:47.28 | _x86_ | flart: wie geht es Dir? |
| 12:47.54 | flart | gut gut |
| 12:48.47 | _x86_ | gut :) |
| 12:50.26 | flart | is it possible to send an other contact-address in the sip-header than the one which is configured on the local network-interface? |
| 12:50.32 | [TK]D-Fender | RsaMan: pastebin your configs, and the CLI output of the failed attempt at verbose 10 |
| 12:50.46 | *** join/#asterisk guillote_GNU (n=bancaria@host185.190-136-203.telecom.net.ar) |
| 12:50.56 | flart | example: local is 1.2.3.4 but i want to send 1.2.3.5 |
| 12:51.16 | [TK]D-Fender | flart: Yes, thats what externhost & externip are for, though this is the ONLY one you can set. Typically used for NAT scenarios |
| 12:51.35 | flart | ah, fine. then i will take a look on this. thx :) |
| 12:56.54 | *** join/#asterisk antimoof (n=dogcow@netbsd/developer/antimoof) |
| 12:56.56 | *** join/#asterisk af_ (n=getsmart@81-174-47-128.dynamic.ngi.it) |
| 12:58.43 | FlatFoot | anyone help ????? get this error when 'make' asterisk-addons ........ |
| 12:58.45 | FlatFoot | error: C compiler cannot create executables |
| 12:59.00 | FlatFoot | using gcc as compiler on debian |
| 12:59.48 | *** join/#asterisk blitzrage (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 12:59.48 | *** mode/#asterisk [+o blitzrage] by ChanServ |
| 13:00.14 | tzafrir | FlatFoot, apt-get install build-essential |
| 13:00.27 | [TK]D-Fender | FlatFoot: It appears you are missing G++ |
| 13:00.28 | FlatFoot | cheers i'll try that |
| 13:00.56 | FlatFoot | TK: G++ ????? |
| 13:01.13 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 13:01.25 | [TK]D-Fender | FlatFoot: Yes. its a GCC related package. |
| 13:01.27 | tzafrir | gnu compiler collection's C++ front end |
| 13:01.38 | tzafrir | isn't it simpler to just write g++? |
| 13:01.51 | FlatFoot | aha thanks , done the apt-get and i see it includes g++ |
| 13:02.12 | tzafrir | I wonder who has the nick 'c' or 'g' and gets such free karma boosts |
| 13:02.46 | [TK]D-Fender | FlatFoot: I found the same recommendation that tzafrir just gave you within the first 3 Google search results with your error + Debian as key. You really should try harder... |
| 13:03.06 | *** join/#asterisk Corydon76-home (i=ten@pdpc/supporter/sustaining/Corydon76-home) |
| 13:03.06 | *** mode/#asterisk [+o Corydon76-home] by ChanServ |
| 13:03.08 | *** join/#asterisk tc3driver (n=huh@dsl253-090-134.lax1.dsl.speakeasy.net) |
| 13:03.14 | FlatFoot | TK aha , i searched with the error message only |
| 13:03.30 | flart | hm, i configured externip correctly, set "nat = yes" an set "canreinvite = nonat". anything else i've to do on the serverside that voip works through a nat? |
| 13:04.04 | [TK]D-Fender | tzafrir: Doesn't work really. it always occurs later in the line :) |
| 13:04.04 | [TK]D-Fender | flart: read this : |
| 13:04.04 | [TK]D-Fender | ~sipnat |
| 13:04.05 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 13:04.05 | *** join/#asterisk keulin (n=cray@AMontpellier-152-1-83-152.w86-197.abo.wanadoo.fr) |
| 13:04.10 | flart | merci |
| 13:04.21 | [TK]D-Fender | bienvenue |
| 13:09.05 | Daejeo1 | [TK]D-Fender: :) |
| 13:11.15 | *** join/#asterisk SuPrSluG (n=SuPrSluG@pool-72-65-30-220.bflony.east.verizon.net) |
| 13:12.29 | Wonka | i got a queue in lastrecent strategy with four queue members. i then made eight parallel calls to them, connecting a Local/ extension only playing sounds to this queue. why does only one phone ring at any given time? |
| 13:12.41 | Wonka | there are four agents waiting for calls... |
| 13:14.12 | SuPrSluG | hi all |
| 13:14.54 | SuPrSluG | what are people here using for faxing purposes? asterfax or configure manually? |
| 13:15.08 | SuPrSluG | or something else? |
| 13:21.29 | *** join/#asterisk famicon (n=pastry@c51447ddc.cable.wanadoo.nl) |
| 13:22.55 | *** join/#asterisk _ShrikE (n=ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 13:25.35 | *** join/#asterisk Sliss (n=chatzill@213.219.170.193.adsl.static.edpnet.net) |
| 13:25.36 | [TK]D-Fender | Wonka: Your agent is never being auto-kicked or paused. Because of that, they are always considered a valid choice and is STILL the "last recent" so THEY just keep ringing. Either have the queue boot off or pause idle agents who aren't answering their calls |
| 13:26.16 | *** join/#asterisk bonanza83 (n=business@89.253.168.92) |
| 13:26.17 | [TK]D-Fender | SuPrSluG: Hylafax seems to be the preferred way, and recommended to be though its own modem on a seperate line. |
| 13:26.25 | *** join/#asterisk robl^ (n=robl@pdpc/supporter/monthlybyte/robl) |
| 13:26.46 | *** join/#asterisk anonymouz666 (n=anonymou@189.25.193.185) |
| 13:26.55 | bonanza83 | <PROTECTED> |
| 13:27.18 | SuPrSluG | Thanks. I'm using tdm400p card though. |
| 13:27.51 | [TK]D-Fender | SuPrSluG: This will risk some reliability, but is better than most of the alternatives. |
| 13:28.16 | coppice | SuPrSluG: the TDM400P is the best route to misery in faxing |
| 13:28.16 | [TK]D-Fender | bonanza83: Teliax & VoicePulse Connect |
| 13:28.52 | [TK]D-Fender | Nivex: I don't .... I just use something that looks just like it ;) |
| 13:29.22 | coppice | many fax machines sit gathering dust, but others are faxing all day long |
| 13:29.56 | SuPrSluG | bonanza83:i also use voicepulse connect for years and found them quite reliable. i also mix in vitelity for cheap outbound calls. voicepulse is nice because they give you 4 channels |
| 13:30.34 | SuPrSluG | Nivex: agreed. Faxing is soooooooo 80's |
| 13:31.46 | robl^ | I just have a fax - email gateway service provider for the 5 faxes I need to send or receive per month |
| 13:32.52 | cpm | SuPrSluG, yeah, it's another one of those horrible 80s things that just won't die |
| 13:33.10 | *** join/#asterisk Ebola (n=Ebola@host86-138-6-48.range86-138.btcentralplus.com) |
| 13:34.23 | CCFL_Man2 | oh dear |
| 13:34.33 | SuPrSluG | unfortunately lawyer still think it's cutting edge technology |
| 13:34.56 | coppice | no. lawyers think its cutting throat technology |
| 13:35.32 | Nivex | Those newfangled copiers with built-in e-mail modules... high tech shit! :) |
| 13:35.36 | *** join/#asterisk mindCrime (n=chatzill@66.83.208.218.nw.nuvox.net) |
| 13:36.10 | Nivex | sup mindCrime? |
| 13:37.18 | CCFL_Man2 | citats: i see what he means |
| 13:38.02 | mindCrime | Nivex: not much. You? |
| 13:38.24 | Nivex | mindCrime: as little as possible :) |
| 13:38.40 | CCFL_Man2 | citats: but when i install a kernel source it just downloads a .tar.bz2 of the kernel source, and there is no architecture specific config on my system |
| 13:39.32 | mindCrime | :-) |
| 13:39.48 | *** join/#asterisk bintut (n=bintut@cm24.gamma176.maxonline.com.sg) |
| 13:41.06 | [TK]D-Fender | CCFL_Man2: I think your aim is a bit off... |
| 13:41.49 | CCFL_Man2 | [TK]D-Fender: i've been having issues with building this kernel |
| 13:43.13 | *** part/#asterisk bintut (n=bintut@cm24.gamma176.maxonline.com.sg) |
| 13:44.47 | *** join/#asterisk jcims (n=jcims@oh-71-53-55-242.dhcp.embarqhsd.net) |
| 13:46.11 | jcims | gah, hate asking this here because it's a bit of a general voip question. is there a simple solution for building a 'virtual callcenter' that is manned by volunteers.. e.g. someone calls a PSTN number and something finds an available person and routes the call to them? preferrably a service provider as we have no infrastructure |
| 13:46.53 | tzafrir | CCFL_Man2, or get the debian kernel-source package, which is essentially that, butr you have a tested config for it |
| 13:47.51 | *** join/#asterisk Krurst (n=me@55.90.233.220.exetel.com.au) |
| 13:48.06 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 13:48.06 | *** mode/#asterisk [+o anthm] by ChanServ |
| 13:48.25 | CCFL_Man2 | tzafrir: i do, kernel-source-2.6.18 |
| 13:48.28 | jsmith | jcims: It wouldn't be hard at all to build something like that |
| 13:49.03 | CCFL_Man2 | it might have the debian specific configuration but nothing specific for sparc |
| 13:49.09 | jcims | i know (er. figured anyway)...this is for a small volunteer organization though...not sure they are going to have a place to host anything like this. |
| 13:49.34 | CCFL_Man2 | actually, it's linux-source-2.6.18 |
| 13:50.02 | CCFL_Man2 | and all that does is download a .tar.bz2 file of the jernel |
| 13:50.18 | CCFL_Man2 | in my /usr/src directory |
| 13:53.29 | *** join/#asterisk darviria (n=darviria@194-105-181-29.ifb.co.uk) |
| 13:55.00 | Sliss | I use the ESCAUX net.PBX and wonder if I can change the settings for number recognition? It shows now only the local country number without the starting "0". Any ideas? (wrong channel perhaps ;-)) |
| 13:58.06 | *** join/#asterisk mog (i=mog@nat/digium/x-62dc3b525d269c17) |
| 13:58.06 | *** mode/#asterisk [+o mog] by ChanServ |
| 14:00.57 | [TK]D-Fender | Sliss: "Perhaps"? |
| 14:01.35 | Sliss | well Esaux is astriks based this is why I ask here |
| 14:02.07 | Sliss | how would you do it in Asterisk? |
| 14:02.23 | *** join/#asterisk Ebola (n=Ebola@host86-138-6-48.range86-138.btcentralplus.com) |
| 14:02.57 | *** join/#asterisk RsaMan (n=aa@196.210.155.2) |
| 14:03.03 | RsaMan | hiya |
| 14:03.13 | RsaMan | its me , the guy with the blind call transfer issues |
| 14:03.22 | RsaMan | i have narrowed it down .. |
| 14:03.42 | RsaMan | i cannot transfer a call received on my fxs channel the digium card |
| 14:03.43 | Krurst | you'd edit extensions.conf |
| 14:03.54 | Krurst | http://www.voip-info.org/wiki/index.php?page=Asterisk+Dialplan+Patterns |
| 14:04.06 | RsaMan | it drops both calls after it says transfer |
| 14:04.26 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 14:04.41 | Krurst | RsaMan: stupid question, is transfer allowed by your zapata.conf? |
| 14:04.58 | RsaMan | Krurst : not sure :( i will check |
| 14:05.09 | RsaMan | I am a bit of a noob |
| 14:06.18 | RsaMan | Krurst : yes it is |
| 14:06.48 | Krurst | how are you transferring? |
| 14:06.57 | RsaMan | blind call transfer |
| 14:07.05 | Krurst | ##? |
| 14:07.09 | RsaMan | it puts the other party |
| 14:07.11 | RsaMan | # |
| 14:07.16 | lirakis | lol |
| 14:07.17 | RsaMan | on hold and says transfer |
| 14:07.40 | RsaMan | but then after i enter the extension it hangs up on both sides |
| 14:07.53 | RsaMan | the blind call transfer works between my iax clients |
| 14:08.22 | Krurst | any hints in a verbose log? |
| 14:08.32 | lirakis | RsaMan: for any help i think you are going to need to pastebin what you are doing.. explaining it verbally is really not clear |
| 14:08.51 | RsaMan | lirakis : ok give me a minute |
| 14:09.30 | Krurst | from what I can tell, he's using the features.conf blind transfer but its dropping zap channels. |
| 14:09.34 | *** join/#asterisk bkruse_home (n=kruz@thuroros.wca-hsv.org) |
| 14:09.56 | *** join/#asterisk Dan0maN_Work (n=dschuh@64.149.174.136) |
| 14:10.25 | coppice | why does it only transfer the blind? does it work for sighted people if they close their eyes? |
| 14:11.33 | Krurst | no, everones blind on the phone. unless its a video phone. then you use sighted transfers. |
| 14:12.39 | Krurst | But then the problem is of course what to do with a blind person using a video phone. |
| 14:12.46 | Wonka | [TK]D-Fender: that's not the problem... i have 4 agents ready to take calls, and 8 calls in the queue, but still only one phone rings. as soon as that one is picked up, the next phone rings. |
| 14:13.05 | coppice | especially if they called someone deaf |
| 14:13.17 | Nivex | "The tones that follow are for the deaf." |
| 14:13.34 | Wonka | [TK]D-Fender: if there were 8 agents ready to take calls and three callers, i'd like to have the three last recent agents to be rung at once, not one-by-one |
| 14:14.04 | Krurst | can you have a que ring a hunt group? |
| 14:14.22 | Wonka | you can have a queue ring a Local channel |
| 14:14.26 | [TK]D-Fender | Wonka: taht is something completely different. In 1.4 the option you're looking for is "autofill=yes". I'm not sure what version of 1.2 this was introduced in if any. |
| 14:14.35 | Wonka | aaah |
| 14:14.38 | Wonka | i'll try that |
| 14:14.53 | Wonka | seems the docs aren't written in my language ;) |
| 14:15.31 | [TK]D-Fender | Wonka: You seemt o handling english jsut fine NOW... |
| 14:15.37 | coppice | you want them translated to Oompah Lumpah? |
| 14:16.00 | Nivex | coppice: you need a Redneck language file a la Red Hat 4.x |
| 14:16.08 | Wonka | [TK]D-Fender: yes... some britons i visited were astonished about my good english too... |
| 14:16.43 | bkw_ | Wonka, so do you make chocolate? |
| 14:16.59 | Wonka | hrm. search on voip.org for autofill spits _only_ http://www.voip-info.org/wiki/view/Asterisk+v1.2+upgrade+to+v1.4+gotchas |
| 14:17.08 | Wonka | no wonder i didn't find it yet |
| 14:17.27 | coppice | bkw__ you mean like the oopah, lumpah, dumpadee do? |
| 14:17.33 | bkw_ | coppice, yep |
| 14:17.38 | bkw_ | Wonka, why would you need 1.4? |
| 14:17.45 | bkw_ | does it offer something 1.2 doesn't? |
| 14:17.53 | *** join/#asterisk geoff_k (n=geoff_k_@host81-152-90-185.range81-152.btcentralplus.com) |
| 14:19.25 | *** join/#asterisk qdk (n=qdk@0x50c627be.bynxx11.adsl-dhcp.tele.dk) |
| 14:19.26 | *** join/#asterisk [intra]lanman (n=lanman@va-76-6-213-8.dhcp.embarqhsd.net) |
| 14:19.50 | *** join/#asterisk defswork (n=andy@mailgate2.3gcomms.co.uk) |
| 14:20.43 | [TK]D-Fender | bkw_: he pointed out rather specifically a difference he cares about. |
| 14:20.58 | Wonka | thx [TK]D-Fender, works... |
| 14:21.00 | bkw_ | autofill what does that do? |
| 14:21.04 | [TK]D-Fender | bkw_: And as 1.2 is in maintenance mode only its looking to be time to move on... |
| 14:22.02 | [TK]D-Fender | bkw_ : instead of ringing 1 agent at a time with a single call, AUTOFILL fills the maximum number of agents with as many queued calls as fits |
| 14:22.26 | [TK]D-Fender | bkw_: so if you have 3 calls queued and 3 free agents (simultaneous), that all start rining immediately |
| 14:22.46 | bkw_ | [TK]D-Fender, i'm going to shut up about it.. its one of those subjects I would rather not comment on because I have nothing nice to say about that. |
| 14:22.53 | Krurst | that sounds logical. Why isn't it the default behavior |
| 14:23.24 | [TK]D-Fender | bkw_ : I'll sum it up for you : "Shit looks pretty good, when compared to CRAP" :p |
| 14:23.24 | jsmith | bkw_: I know, I know... app_queue needs to be rewritten |
| 14:23.43 | bkw_ | jsmith, :) |
| 14:23.50 | defswork | can anyone recommend a cordless SIP handset solution |
| 14:23.55 | coppice | it was, more or less, but nobody keep the new code rolling along |
| 14:24.08 | jsmith | Oh? |
| 14:24.15 | [TK]D-Fender | defswork:.... |
| 14:24.18 | [TK]D-Fender | ~wifisip |
| 14:24.18 | jbot | Wi-Fi SIP phones suck. All of them. HARD. Some only slightly less than others... |
| 14:24.29 | bkw_ | no they don't |
| 14:24.31 | defswork | [TK]D-Fender: what about DECT SIP phones ? |
| 14:24.33 | [TK]D-Fender | defswork: in an office environment you're best off with ATA + cordless analog |
| 14:24.33 | jsmith | defswork: I've had good luck with the cordless phones from Aastra |
| 14:24.36 | bkw_ | the Nokia e61i is a kick ass Wifi SIP phone |
| 14:24.52 | jsmith | bkw_: Amen... that's twice we've agreed this morning.... I'm starting to get nervous |
| 14:24.55 | [TK]D-Fender | defswork: DECT != SIP. if the base talks sip, then fine. |
| 14:24.58 | defswork | http://www.voipon.co.uk/voip-ip-telephones-siemens-dect-sip-phone-c-1_93.html - like this ? |
| 14:25.11 | bkw_ | jsmith, ;) |
| 14:25.16 | coppice | bkw_: its only kicks ass until you replace the matsushita battery :-) |
| 14:25.17 | [TK]D-Fender | bkw_ : hear it sucked with NAT and lacked a lot of features |
| 14:25.34 | bkw_ | [TK]D-Fender, thats not so much of an issue in the e61i |
| 14:25.48 | *** join/#asterisk Mavvie (n=edwin@ppp121-44-92-167.lns10.syd6.internode.on.net) |
| 14:25.50 | jsmith | [TK]D-Fender: The earlier E series phones had problems, but the new firmware on the E61i took care of most of that |
| 14:25.52 | bkw_ | the e61 did have the older voip client... the i has the 2.0 client |
| 14:25.52 | defswork | sip on mobile phones drains batteries too quick |
| 14:25.55 | [TK]D-Fender | bkw_Cool... I'm still waiting for MokoIAX personally ;) |
| 14:26.03 | bkw_ | defswork, no they don't |
| 14:26.03 | SuPrSluG | i'll be testing the nokia n80 soon as it get here |
| 14:26.13 | bkw_ | the e61i lasts 4-8 hours depending on many factors |
| 14:26.16 | bkw_ | on talk time that is |
| 14:26.20 | defswork | bkw_: well my n95 doesnt last 30 minutes when connected |
| 14:26.26 | bkw_ | SuPrSluG, the n80 sucks the battery dry |
| 14:26.36 | defswork | and thats just when connected to wifi - no calls |
| 14:26.38 | bkw_ | defswork, go into the settings and lower the power on the transmitter to 5mw |
| 14:27.00 | bkw_ | it should have 3 levels to set the power of the wifi |
| 14:27.07 | bkw_ | 5,10 and 50 or something similar |
| 14:27.08 | SuPrSluG | after how long? |
| 14:27.20 | bkw_ | SuPrSluG, 1 hour talk maybe |
| 14:27.31 | [TK]D-Fender | bkw_how long on idle? |
| 14:27.43 | bkw_ | a day or so |
| 14:27.53 | bkw_ | now if it gets stuck doing a wifi search |
| 14:27.54 | *** join/#asterisk slima (i=slima@unaffiliated/slima) |
| 14:27.56 | bkw_ | which can and does happen |
| 14:28.01 | [TK]D-Fender | bkw_Acceptable it you can charge it mid-day |
| 14:28.02 | bkw_ | its gone in less than an hour |
| 14:28.06 | defswork | I have a new install in a scrap metal reclaimation yard - they currently have dect phones and they work ok-ish. As long as I put something in that is similar then it will be ok |
| 14:28.06 | bkw_ | [TK]D-Fender, yah |
| 14:28.09 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 14:28.38 | [TK]D-Fender | defswork: Try one and see. Decent odds I guess |
| 14:28.56 | coppice | too many of these machines are dumb. put a lot of GSM phones in a screened room and the battery is dead in an hour or two |
| 14:28.58 | defswork | I'm not sure about wifi sip handsets - perhaps dect sip ones will be better |
| 14:29.24 | bkw_ | dect sip... I seen one of those.... now if I can recall where |
| 14:29.27 | defswork | hes talking about 20 of them though |
| 14:29.33 | coppice | DECT phones with SIP bases are indeed much better |
| 14:29.44 | coppice | bkw_ everyone in asia seems to make one |
| 14:30.05 | [TK]D-Fender | defswork: WIFI is the problem. DECT is a better basis, and the fact the BASE is doing the SIP, so much the better |
| 14:30.40 | bkw_ | so why not just use an ATA and a regular Dect phone? |
| 14:30.45 | bkw_ | :P |
| 14:30.47 | *** join/#asterisk snook3r (n=snook3r@bzq-219-46-202.isdn.bezeqint.net) |
| 14:30.53 | Wonka | i'd like an ISDN DECT base without a handset cradle and some DECT handsets with cradles without bases |
| 14:31.09 | jsmith | bkw_: The Aastra phone has a SIP handset with a DECT cordless phone that goes with it |
| 14:31.10 | coppice | bkw_ it avoid that nasty analogue hole :-) |
| 14:31.29 | bkw_ | IC |
| 14:31.31 | defswork | jsmith: I've installed Aastra before - which model is that ? |
| 14:31.32 | Wonka | i just don't trust SIP phones yes |
| 14:31.33 | Wonka | yet |
| 14:31.45 | bkw_ | Wonka, why not? they work great |
| 14:31.49 | bkw_ | its SIP you shouldn't trust |
| 14:31.51 | bkw_ | :) |
| 14:32.15 | jsmith | defswork: Anything with "CTi" in the model... the one I used was the 480 CTi... but there are newer models now |
| 14:32.37 | Wonka | a friend of mine breaks sip phone's sip stacks without breakfast... and he hasn't found one yet which he didn't break |
| 14:32.49 | bkw_ | Wonka, oh really? |
| 14:32.52 | jsmith | bkw_: No, it's most *implementations* of SIP you shouldn't trust... SIP is fine, in theory... In theory, there's no difference between theory and reality |
| 14:32.52 | bkw_ | how is he breaking them? |
| 14:32.59 | Wonka | i'd rather have IAX2 DECT bases |
| 14:33.03 | defswork | jsmith: hmm - I can't get 20 of those :) |
| 14:33.07 | coppice | blendtech? |
| 14:33.08 | [TK]D-Fender | jsmith: AASTRA one is a bad idea |
| 14:33.11 | jsmith | defswork: Why not? |
| 14:33.16 | bkw_ | Wonka, sorry but IAX2 has its issues also |
| 14:33.17 | defswork | jsmith: too expensive |
| 14:33.18 | Wonka | bkw_: sip attack stuff |
| 14:33.33 | bkw_ | Wonka, I know one you can attack with sipp and it survives quite well |
| 14:33.40 | Wonka | bkw_: but doesn't try to do RTP to the internet with RFC1918 addresses |
| 14:34.01 | JoseBravo | I have analog card, and asterisk didn't determine when the caller hangup. I can do something? |
| 14:34.08 | bkw_ | doing crazy shit to the poor stack and it seems to take it quite well |
| 14:34.32 | coppice | let a 4 year old loose on it. they'll break it |
| 14:34.34 | Krurst | JoseBravo: theres plenty of stuff to try |
| 14:34.35 | *** join/#asterisk putnopvut (i=putnopvu@nat/digium/x-5b7c15bdd7d26830) |
| 14:34.59 | [TK]D-Fender | jsmith: Aastra's DECT is an ACCESSORY to the base, and from what I've seen you can't use a single base with multiple INDEPENDANT handsets. Theye are all treated as the same. And at that price, its a waste |
| 14:34.59 | defswork | jsmith: hmm - it supports up to 4 handsets on one base - do you know if they are separate extensions ? |
| 14:35.25 | jsmith | [TK]D-Fender: Actually, I think that's been changed now. |
| 14:35.28 | [TK]D-Fender | defswork: NO, thats the problem. Seimens looks like a good deal |
| 14:35.35 | jsmith | [TK]D-Fender: with some of the newer models |
| 14:35.39 | *** join/#asterisk |dennis| (n=dennis@200.32.233.84) |
| 14:35.44 | [TK]D-Fender | jsmith: Firmware change? |
| 14:35.56 | [TK]D-Fender | jsmith: I had a 57i CT on my desk. I HATED it. |
| 14:36.05 | jsmith | [TK]D-Fender: Not sure if it was just firmware, or newer phones. I haven't got my grubby little paws on the newer models yet. |
| 14:36.19 | [TK]D-Fender | jsmith: My 57i CT blew |
| 14:36.25 | RsaMan | ok guys i have pastbined it, basically i receive a call on zap channel 4 , which is answered by zap channel 1 fxs, i want to then transfer that call using blind call transfer to an iax client |
| 14:36.28 | jsmith | [TK]D-Fender: Send it to me then :-) |
| 14:36.30 | RsaMan | http://pastebin.com/d60c82dc5 |
| 14:36.39 | RsaMan | but it seems to drop both calls |
| 14:37.04 | Krurst | JoseBravo: start here I guess : http://www.voip-info.org/wiki/index.php?page=Asterisk+Disconnect+Supervision |
| 14:37.09 | [TK]D-Fender | jsmith: I still need the DECT part (why I got it here), and since it RANG MY DESK when the hopefully independant DECT was called, it pissed me off even more. |
| 14:38.24 | *** join/#asterisk Defraz (n=t0tal@24-116-152-177.cpe.cableone.net) |
| 14:38.30 | [TK]D-Fender | RsaMan: Oh yeah, this problem. If you move that included context's extens DIRECTLY into the main, does it work? |
| 14:39.59 | *** join/#asterisk southtel (n=southtel@68-114-23-151.dhcp.gwnt.ga.charter.com) |
| 14:41.21 | defswork | I'll suggest the Siemens units then |
| 14:41.22 | MindTheGap | morning all, how does cdr recording is triggered? i mean, if i set nocdr on the first priority of a call then in the 2nd it dials i get no cdr and thats expected but if the call goes trough a gotoif before the actual dial command it is recorded. It traverses a Set just fine but not a gotoif... I couldnt find any pointers of what command should reset the Nocdr status... |
| 14:42.23 | *** join/#asterisk santiago (i=santiago@debian/developer/santiago) |
| 14:42.42 | *** join/#asterisk NOT_guru (n=chatzill@24-241-103-142.static.stls.mo.charter.com) |
| 14:43.12 | southtel | With respect to SER and OpenSER, is one considered "better"? |
| 14:43.29 | southtel | I realize that's a broad question, but for all I know it may have a simple answer. |
| 14:43.34 | *** join/#asterisk flujan (n=flujan@200-160-115-020.static.spo.ctbc.com.br) |
| 14:43.43 | jsmith | southtel: I prefer OpenSER, but there's really not much difference |
| 14:43.53 | *** join/#asterisk Zefk (n=Zefk@wsc-fo.b.astral.ro) |
| 14:44.49 | Wonka | OpenSER is Open ;) |
| 14:45.07 | *** join/#asterisk Kurin- (n=Kurin@lithium.delete.org) |
| 14:46.13 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 14:46.16 | Kurin- | Should I keep hold music in gsm files, or some other format? |
| 14:46.17 | Zefk | Hii. I have 2 SIP ip phones connected to * box. The phones are in 2 separate LANs. The phones cannot see each other. When dialing from one phone and the other answer I have the following message: Packet2Packet bridging SIP/200-b7c16860 and SIP/102-0830b890. People cannot heard each other. How can I disable this bridging ? thx |
| 14:46.29 | Kurin- | Because they all sound terrible as gsm, and I can't tell if I'm converting it wrong or not |
| 14:46.41 | southtel | Wonka: How do you quantify "Open"? |
| 14:46.43 | jsmith | Kurin-: I put mine in ulaw, to avoid transcoding |
| 14:46.51 | jsmith | Kurin-: (Since most of my calls are ulaw) |
| 14:47.00 | Wonka | southtel: as it was a joke, only by "it's in the name" |
| 14:47.25 | jsmith | Zefk: Which version of Asterisk? You're probably using 1.4.0... use 1.4.11 instead, and see if that fixes it |
| 14:47.29 | southtel | Wonka: Fair enough...one never can tell! |
| 14:47.51 | Zefk | jsmith: I'm running 1.4.9 |
| 14:48.32 | *** join/#asterisk xtr (n=94752345@216.19.191.191.novuscom.net) |
| 14:48.40 | Krurst | umm reinvite=no might do something |
| 14:48.43 | jsmith | Zefk: There's a setting in sip.conf to disable packet2packet bridging... but try upgrading to 1.4.11 first |
| 14:49.08 | Zefk | jsmith: I will try tonight .. |
| 14:49.18 | Wonka | packet2packet? didn't find... |
| 14:49.46 | southtel | It looks like SER is a little stagnant (no releases in a year and a half). That's the main reason for my quesitons. |
| 14:51.36 | *** join/#asterisk xachen (n=justin@pdpc/supporter/student/xachen) |
| 14:51.47 | blitzrage | OpenSER |
| 14:51.51 | RsaMan | [TK]D-Fender : sorry about that , was afk, i will try that |
| 14:52.50 | RsaMan | [TK]D-Fender : what do u mean move it to the main ? |
| 14:52.52 | Daviey | Is it possible just to build a ztdummy module from zaptel? |
| 14:52.54 | DrAk0 | how u guys do when you want a mailbox to go to multiples email recipents? |
| 14:53.03 | RsaMan | [TK]D-Fender : have u seen this problem before ? |
| 14:53.25 | blitzrage | Daviey: that's what the Zaptel package does -- it builds various modules for hardware, plus the ztdummy module |
| 14:53.39 | blitzrage | select just the ztdummy module from menuselect |
| 14:54.02 | Daviey | blitzrage: thought so.. but how? |
| 14:54.13 | blitzrage | ./configure ; make menuselect ; make install |
| 14:54.27 | Daviey | ta |
| 14:54.28 | blitzrage | this assumes 1.4 of course |
| 14:54.31 | Daviey | it is |
| 14:54.58 | *** join/#asterisk brodiem (n=brodiem@67.110.68.66.ptr.us.xo.net) |
| 14:55.01 | *** join/#asterisk WildPikachu (n=WildPika@about/linux/staff/wildpikachu) |
| 14:55.13 | WildPikachu | is it tough to get bristuff working with asterisk 1.4.11? |
| 14:55.40 | brodiem | Is there a "preferred" ATA for use with T.38 w/ an analog fax machine plugged in? |
| 14:56.09 | *** join/#asterisk lbow (n=lbow@falcon.clarotech.co.za) |
| 14:56.52 | coppice | it seems to be more a matter of preferred software revision than preferred ATA. |
| 14:57.05 | brodiem | the software on the ATA you mean? |
| 14:58.03 | coppice | yes. and newer doesn't always mean more debugged |
| 14:59.05 | brodiem | are you (or anyone else in here) using t.38 passing through Ast? |
| 14:59.54 | brodiem | I'm wondering how reliable it is |
| 14:59.58 | coppice | not through ast, but we've been doing a lot of investigation of different T.38 platforms lately |
| 15:00.33 | coppice | its a lot better than trying to send audio over IP, but its far from perfect |
| 15:01.31 | tzanger | coppice: what've you seen in terms of the better ones? |
| 15:02.17 | brodiem | I'm currently using iaxmodem+hylafax, but trying to get out of it due to a pbx move and needing everything coming in as g729 on our current sip vendor |
| 15:02.29 | coppice | they all suck with the wrong software. many work well when you find the sweet spot amongst the revisions. some, like mediatrix are just crappy implementations |
| 15:02.52 | brodiem | so I was hoping to use a 2nd vendor w/ t.38 support and start passing through to analog fax machines |
| 15:03.09 | *** join/#asterisk jfitzgibbon-away (n=NADT@64.72.237.130) |
| 15:03.50 | tzanger | hmm |
| 15:03.54 | coppice | we haven't done anything towards making iaxmodem work with T.38 |
| 15:04.08 | tzanger | so my idea of using t38modem with hylafax and the h323 stack for asterisk is not a good idea |
| 15:04.21 | brodiem | no I mean I want to stop using iaxmodem+hyla and use T.38 only w/ standard fax machines |
| 15:04.33 | coppice | * doesn't have any support for T.38 with H.323 |
| 15:04.37 | *** join/#asterisk wunderkin (n=wunderki@ip72-223-86-126.ph.ph.cox.net) |
| 15:04.51 | tzanger | coppice: I thought I could shuttle a fax call off to t38modem and have hylafax answer it |
| 15:05.06 | tzanger | i.e. PRI -> * -> h323 -> t38modem -> hylafax -> pdf |
| 15:05.12 | coppice | * it only has T.38 passthrough, and only with SIP |
| 15:05.43 | coppice | * has no ability to get from a PRI to T.38 at all |
| 15:06.07 | tzanger | aha |
| 15:06.56 | coppice | PRI -> * -> iaxmodem -> hylafax will work OK, though |
| 15:07.00 | [TK]D-Fender | RsaMan: I mean don't "include" that other context. COPY the extens into [internal] and test |
| 15:07.12 | brodiem | is the Grandstream HT502 decent at t.38? I'm not a huge fan of GS but according to the wiki is one of the only ATAs with a "yes" for NAT and "works" |
| 15:07.47 | coppice | some revisions of the grandstream code work well for T.38. others don't |
| 15:08.42 | *** join/#asterisk snook3r (n=snook3r@bzq-219-46-202.isdn.bezeqint.net) |
| 15:09.23 | brodiem | coppice: do you recommend something else? |
| 15:09.31 | *** join/#asterisk CtRiX (n=CtRiX@aretha.navynet.it) |
| 15:10.17 | *** join/#asterisk waKKu (n=eurulo@unaffiliated/wakku) |
| 15:11.29 | coppice | e-mail :-) |
| 15:12.22 | brodiem | coppice: so you would suspect iaxmodem+hyla as being used now would still be more reliable than passing as t.38? |
| 15:13.08 | coppice | this depends on your environment. if you want to bring in FAX from a VoIP provider its not much of an option, though |
| 15:13.25 | brodiem | coppice: it would be from a voip provider sending in t.38 |
| 15:13.26 | [TK]D-Fender | brodiem: Linksys works fine |
| 15:13.39 | coppice | rubbish |
| 15:14.04 | coppice | don't believe anything anyone says about T.38 working. you need to look *very* carefully at revisions. |
| 15:14.19 | jsmith | That's been my experience as well... |
| 15:14.55 | brodiem | [TK]D-Fender: the 2102? (iirc the pap2's don't support t38 right?) |
| 15:15.14 | [TK]D-Fender | brodiem: Thats my model of choice. |
| 15:15.20 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 15:16.14 | bkw_ | jsmith, since when couldn't you use vars in extensions |
| 15:16.19 | bkw_ | jsmith, it did used to work |
| 15:16.28 | bkw_ | but I think it was limited to using globals |
| 15:17.03 | brodiem | [TK]D-Fender: are you currently using t.38 on a 2102? |
| 15:17.18 | *** join/#asterisk azrishahril (n=azrishah@60.52.122.65) |
| 15:17.26 | jsmith | bkw_: Hmmmn... I swore I tested it like a year ago, and found that it didn't work... but then again, I've been known to be wrong |
| 15:17.34 | jsmith | (and even pig-headed about being wrong) |
| 15:17.46 | bkw_ | I tested it in 1.2 but I think you're right about the pattern macth part |
| 15:18.05 | bkw_ | you can use it for static extensions |
| 15:18.06 | bkw_ | but thats it |
| 15:18.18 | bkw_ | but then again that could have been changed since last I tested... |
| 15:18.19 | jsmith | Ah... |
| 15:18.29 | jsmith | I stand corrected |
| 15:18.40 | *** join/#asterisk Nix (n=Nix@opensuse/developer/nix) |
| 15:19.54 | *** join/#asterisk barros (n=barros@189-19-24-162.dsl.telesp.net.br) |
| 15:20.00 | *** join/#asterisk MattMuss (n=MattM@24.83.96.156) |
| 15:20.51 | barros | Hi. I'm having a problem with the frequency asterisk probes hosts with qualify=yes. for iax, there are the qualifyfreqok to control the interval. What about SIP peers, are there any option to control it? |
| 15:22.24 | jfitzgibbon-away | barros: I believe qualify can be yes|no|value in sip.conf |
| 15:23.11 | *** join/#asterisk pacneil_ (n=pacneil@68.15.17.81) |
| 15:23.17 | jfitzgibbon | barros: yeah, the value is in ms |
| 15:23.36 | *** join/#asterisk simond (n=simon@208.68.95.5) |
| 15:23.46 | barros | jfitzgibbon, yes, I know.. but I need to control how often asterisk probes the peer |
| 15:23.49 | jsmith | jfitzgibbon: That controls how much latency Asterisk will tolerate, *not* how often Asterisk will probe |
| 15:24.06 | jsmith | barros: As far as I know, you have to change the source code to change the probe interval... but I'm not 100% sure |
| 15:24.08 | barros | it probes once per minute for UP peers, and once per 10 seconds for down |
| 15:24.10 | jfitzgibbon | jsmith: ah, gotcha |
| 15:24.18 | *** join/#asterisk |dennis| (n=dennis@200.32.236.10) |
| 15:24.22 | barros | jsmith, hmm, that is what I don't want :( |
| 15:25.13 | barros | that is weird, as IAX has the option.. |
| 15:25.22 | barros | the main problem with the default time, is with ip_conntrack |
| 15:25.47 | *** join/#asterisk CunningPike (n=CunningP@dhcp-10-234.district.north-van.bc.ca) |
| 15:25.47 | jfitzgibbon | barros: it's trivial to change in source, and would be only slightly less trivial to make it a sip.conf option |
| 15:25.54 | jsmith | barros: Why's that? |
| 15:26.51 | defswork | anyone recommend a ISDN 30 PCI card ? I've used Sangoma previously but am open to suggestions |
| 15:27.03 | defswork | (T1/E1) |
| 15:27.10 | jfitzgibbon | barrros: look for DEFAULT_FREQ_OK in channels/chan_sip.c - defined once, used once |
| 15:27.16 | *** join/#asterisk hfb (n=hfb@pool-71-116-255-84.lsanca.dsl-w.verizon.net) |
| 15:27.17 | jsmith | defswork: Digium and Sangoma are the two major suppliers that work with Asterisk |
| 15:27.18 | bkw_ | defswork, why does sangoma not work? |
| 15:27.27 | bkw_ | jsmith, Pika works with asterisk too... |
| 15:27.28 | jsmith | defswork: And since I work for Digium, I'll plug their card :-) |
| 15:27.29 | *** join/#asterisk deeperror (n=deeperro@69-215-202-195.ded.ameritech.net) |
| 15:27.46 | barros | jfitzgibbon, nice, I'll check it |
| 15:27.46 | jsmith | bkw_: Sure, and so does Acculab... but most people get a Digium or Sangoma card |
| 15:27.52 | bkw_ | true |
| 15:27.52 | defswork | bkw_: it works fine but I don't necessarily need to stick with it if there are better price/performance options :) |
| 15:28.11 | *** join/#asterisk Cyon (n=cyon@216.179.31.170) |
| 15:28.22 | defswork | jsmith: last time I did an install I found the digium cards a bit expensive ? |
| 15:28.35 | coppice | defswork: make your own tormenta 2 cards. much cheaper :-) |
| 15:28.36 | defswork | I used a Sangoma A102 in the end |
| 15:28.49 | bkw_ | defswork, I find sangoma cards do perform very well |
| 15:28.49 | russellb | defswork: expensive? aren't they very similarly priced to sangoma's cards? |
| 15:29.18 | barros | jsmith, the problem is that when there are an SIP behind a NAT and you do DNAT to reach the peer |
| 15:29.31 | *** join/#asterisk apardo (n=apardo@bonemachine.ipv6.theprimusproject.com) |
| 15:29.31 | jsmith | barros: OK... |
| 15:29.34 | barros | jsmith, if the peer is down, ip_conntrack will mark the probe as UNREPLIED |
| 15:29.56 | barros | jsmith, and the timeout is 30s. But asterisk probes every 10sec, so this mark will NEVER be removed |
| 15:30.05 | jsmith | barros: That sounds like a problem with ip_conntrack, not Asterisk |
| 15:30.21 | barros | yes, it is with ipcontrack |
| 15:30.25 | barros | not exactly a problem |
| 15:30.36 | barros | but some kind of incompatibility |
| 15:31.06 | deeperror | fxs phone hears fast busy after callee over SIP channel hangs up. Any clues on silencing the fast busy after hangup? |
| 15:31.22 | jsmith | Definitely.... anyway, looks like your options are to change the value in Asterisk or fix ip_conntrack somehow |
| 15:31.28 | *** join/#asterisk Ciber311 (n=Ciber311@user-12ld42j.cable.mindspring.com) |
| 15:32.14 | barros | jsmith, yes.. I prefer change Asterisk, as I'm not sure about the consequences of changing the ip_conntrack. And it is easy to change ONE place, than various places :) |
| 15:32.21 | barros | anyway, thanks for the help |
| 15:32.32 | jsmith | No problem |
| 15:36.06 | Kurin- | Is there a way to define voicemail settings in LDAP or AGI? |
| 15:36.54 | Kurin- | users, email addresses, etc |
| 15:38.30 | jsmith | Kurin-: I think so, if you use res_config_ldap. |
| 15:38.37 | Kurin- | okay |
| 15:38.41 | Kurin- | thanks |
| 15:41.44 | Kurin- | Is that build separately? |
| 15:41.47 | Kurin- | built* |
| 15:42.21 | *** join/#asterisk darviria (n=darviria@194-105-181-29.ifb.co.uk) |
| 15:42.29 | jsmith | I think so... at least it was last time I played with it... maybe it's in asterisk-addons... I don't remember |
| 15:43.20 | russellb | it is in http://svn.digium.com/svn/asterisk/team/group/res_config_ldap |
| 15:43.28 | russellb | unknown whether it works or not |
| 15:43.43 | Kurin- | maybe I'll just do it manually for now then |
| 15:44.49 | Kurin- | But a PBX integrated with the company's auth and directory services would be really hot |
| 15:45.01 | jsmith | Yeah... I know Sun Labs is doing it that way |
| 15:45.34 | *** join/#asterisk harryr (n=harryr@cpc2-lamb3-0-0-cust255.bmly.cable.ntl.com) |
| 15:47.44 | harryr | coming soon to the public domain |
| 15:49.04 | *** part/#asterisk stimpie (n=michiel@ip565faf27.direct-adsl.nl) |
| 15:50.18 | *** join/#asterisk Defraz (n=t0tal@24-116-152-177.cpe.cableone.net) |
| 15:51.12 | *** part/#asterisk MattMuss (n=MattM@24.83.96.156) |
| 15:54.54 | *** join/#asterisk AsteriskProblems (n=pbarnsle@81.171.174.178) |
| 15:55.14 | AsteriskProblems | hey guys, when you set up a user as an agent, what do you dial on a phone to sign in as that agent? |
| 15:55.32 | *** join/#asterisk McJerry (n=bryanj02@wsip-24-249-32-170.mc.at.cox.net) |
| 15:55.46 | jsmith | AsteriskProblems: You dial whatever extension in your dialplan calls the AgentLogin() or AgentCallbackLogin() applications |
| 15:55.55 | AsteriskProblems | right thanks |
| 15:55.55 | Corydon76-dig | AsteriskProblems: You would need to set up a dialplan to allow the user to login |
| 15:56.10 | McJerry | using a sangoma a200 it seems i have a line disconnect problem as follows....if the caller hangs up after the ivr has played or if they choose an extension, asterisk does not detect the remote hangup and continues looping the ivr or dialing the sip extension. However, if the remote caller hangs up prior to the ivr loop, remote disconnect is detected.....any suggestions? |
| 15:56.14 | AsteriskProblems | ok :D |
| 15:56.33 | McJerry | asterisk version 1.2.23 |
| 16:02.50 | Corydon76-dig | McJerry: Use an AbsoluteTimeout to disconnect the call after a certain amount of time |
| 16:03.07 | *** join/#asterisk masus (n=tet@88.248.73.2) |
| 16:03.08 | Corydon76-dig | McJerry: there's not much we can do about bad analog signalling |
| 16:03.37 | masus | hi all , "No closing bracket found" can't authenticate one sip user doeas anyone has an idea |
| 16:04.14 | *** join/#asterisk ATXFormFactor (n=mark@h-68-165-93-154.nycmny83.covad.net) |
| 16:04.18 | McJerry | I have read about the AbsoluteTimeout feature, but am not understanding does it set a specific timeout from time of pickup or from time of no line traffic or what? |
| 16:04.52 | masus | here is mor information --> http://pastebin.ca/675880 |
| 16:04.56 | jsmith | McJerry: Yes, from the time the call is answered |
| 16:05.18 | Corydon76-dig | McJerry: It sets a timeout from the point at which you set it |
| 16:05.53 | Corydon76-dig | McJerry: if you get DTMF, you can cancel the impending timeout with a Set(TIMEOUT(absolute)=0) |
| 16:06.14 | Corydon76-dig | It's not the most elegant solution, but it will work in all cases |
| 16:06.32 | masus | nobody an idea ;( http://pastebin.ca/675880 |
| 16:07.21 | [TK]D-Fender | masus: That SIP UA you are using is sending malformed packets |
| 16:07.45 | masus | ok how to fix it |
| 16:07.54 | Corydon76-dig | masus: use a different phone |
| 16:08.04 | *** join/#asterisk crichardson (n=crichard@38.113.5.185) |
| 16:08.12 | masus | i have use one softphone and one bt100 grandstream |
| 16:08.47 | masus | but the oe is working at another location well |
| 16:08.51 | masus | phone |
| 16:08.56 | *** join/#asterisk nvzn (n=nvzn@dormir.dreaming.org) |
| 16:09.21 | Corydon76-dig | masus: complain to the developer |
| 16:09.48 | *** join/#asterisk datachomper (n=russ@ool-43509aa5.dyn.optonline.net) |
| 16:09.57 | masus | :S |
| 16:11.19 | *** join/#asterisk shinao1 (n=shinao1@217.20.242.51) |
| 16:11.45 | nichtwirklich | does anybody know, why asterisk 1.2.17, dtmfmode=info, sip phone doesn't work with the voicemailsystem? |
| 16:12.10 | [TK]D-Fender | nichtwirklich: because maybe the phone isn't USING that mode? |
| 16:16.22 | tzafrir | Is it possible to talk from asterisk (any version) behind NAT to gtalk users? |
| 16:16.59 | nichtwirklich | [TK]D-Fender: but I can use DTMF from this phone to the outside world (via ISDN) |
| 16:17.02 | *** join/#asterisk |dennis| (n=dennis@200.32.233.84) |
| 16:17.19 | tzafrir | any success stories there? |
| 16:17.24 | *** join/#asterisk FlyboySR22 (n=rsears@hq.fw.americanis.net) |
| 16:18.22 | masus | Corydon76-dig: and the phone is can register to another asterisk server |
| 16:18.48 | Corydon76-dig | Registration isn't the problem. Lack of SIP compliance is |
| 16:19.13 | tzafrir | Also: is it possible at all to pass some sort of text messages from various SIMPLE-capable clients through asterisk to jabber users? |
| 16:19.31 | tzafrir | or the other way around? |
| 16:19.42 | McJerry | @Corydon76-dig: suppose I set the absolute timeout to 0 after detecting dmtf, this still wouldn't resolve the issue if a caller hangs up after dialing a sip extension but hangs up before the ext picks up or goes to voicemail. What would you suggest in this case? |
| 16:20.49 | ATXFormFactor | Anyone here using AsteriskNow Beta 6? |
| 16:21.43 | *** part/#asterisk ATXFormFactor (n=mark@h-68-165-93-154.nycmny83.covad.net) |
| 16:24.16 | *** join/#asterisk implicit (n=implicit@210.16.55.38) |
| 16:25.16 | [TK]D-Fender | nichtwirklich: probably because its using INBAND as well |
| 16:25.50 | defswork | jsmith: I've just checked the prices again on the digium cards and they are similar after all - I'm not sure why I thought they were more expensive now |
| 16:26.29 | defswork | so now I have a choice of a Digium TE110P or a Sangoma A101 |
| 16:26.43 | deeperror | this is a good one |
| 16:26.44 | deeperror | BYE sip:+12485836000@209.247.16.221:5060;transport=udp SIP/2.0 |
| 16:26.56 | deeperror | this is the bad one |
| 16:26.57 | deeperror | BYE sip:s@69.215.202.195:1041 SIP/2.0 |
| 16:29.13 | jsmith | defswork: Get the TE120P instead of the 110P... |
| 16:30.19 | *** join/#asterisk tsurko (n=tsurko@78.90.100.214) |
| 16:31.10 | defswork | I cant see the differences - my internets just gone titsup |
| 16:33.36 | [TK]D-Fender | defswork: NEITHER. Sangoma A101d <-------- |
| 16:33.53 | defswork | [TK]D-Fender: aye thats the sangoma card I meant |
| 16:34.15 | defswork | I've no idea why on my last install I specced a 102d |
| 16:34.15 | *** join/#asterisk anonymouz666 (n=anonymou@189.25.193.185) |
| 16:34.30 | jsmith | defswork: Because two is better than one? |
| 16:34.43 | defswork | yeah but they only have one E1 |
| 16:34.49 | [TK]D-Fender | defswork: A101d SI a fair bit more expensive than the TE120P, but has hardware echo-cancellation, etc. |
| 16:35.01 | tzafrir | defswork, so get E2 |
| 16:35.34 | defswork | [TK]D-Fender: aah - thats not the one I meant then - I've didn't think I needed hw echo canc - I've not had a problem with the 102 install |
| 16:35.52 | [TK]D-Fender | defswork: you DO want HWEC. |
| 16:36.20 | defswork | [TK]D-Fender: it's been in 6 months and haven't missed it yet |
| 16:36.20 | *** join/#asterisk davixx (n=davixx@82.251.215.59) |
| 16:36.21 | *** part/#asterisk [intra]lanman (n=lanman@va-76-6-213-8.dhcp.embarqhsd.net) |
| 16:36.36 | [TK]D-Fender | defswork: If you're feeling LUCKY, sure... |
| 16:37.57 | defswork | [TK]D-Fender: ? |
| 16:41.33 | *** part/#asterisk deeperror (n=deeperro@69-215-202-195.ded.ameritech.net) |
| 16:41.42 | [TK]D-Fender | defswork: I take no chances with audio quality |
| 16:42.48 | defswork | When I did my first install I asked on the asterisk forum about h/w echo cancellation - I sure the consensus was that it wasn't needed on a E1 circuit |
| 16:43.20 | [TK]D-Fender | defswork: lol. |
| 16:43.46 | coppice | of course you don't need echo cancellation. the echo gives the calls a nice open spaces feel |
| 16:44.20 | defswork | if it was needed why do some cards not have it ? |
| 16:44.52 | coppice | strange question |
| 16:44.52 | Qwell[] | defswork: because it can also be done in software - just not as well. |
| 16:45.35 | NOT_guru | in the asterisk console... is there a way to have it time stamp the output? |
| 16:45.39 | defswork | coppice: sorry if I'm sounding dumb but I don't know where the echo comes from on a digital medium and I've not experienced any problems with echo so far |
| 16:45.43 | NOT_guru | I am looking at a timeing issue |
| 16:45.55 | defswork | Qwell[]: not as well but still adequately ? |
| 16:46.18 | Qwell[] | defswork: it really just depends on your lines |
| 16:46.31 | Qwell[] | sometimes, you just don't need it.. for instance, connecting two PBXes |
| 16:46.36 | Qwell[] | you aren't gonna get squat for echo |
| 16:46.45 | coppice | defswork: lots of products have lots of bits missing which are important. also lots of E1s are used for data, rather than voice |
| 16:46.55 | Strom_M | defswork: the echo usually comes from impedance mismatches at 2 to 4 wire conversion points |
| 16:47.05 | Qwell[] | oh, Strom_M.. |
| 16:47.09 | defswork | Qwell[]: right - but isn't it a digital medium ? wheres does echo come in ? |
| 16:47.25 | Qwell[] | I was gonna ask you, but forgot.. Do you know what the time resolution is on those AT&T numbers? |
| 16:47.30 | Qwell[] | is it like 5 seconds? |
| 16:47.39 | Strom_M | ten second cycle |
| 16:48.16 | coppice | defswork: so you make calls which are digital all the way to someone's brain? |
| 16:48.27 | Qwell[] | so, ring, answer, "at the tone ... xx:xx:x0/x5" <variable length pause> *beep*? |
| 16:48.29 | Strom_M | "Good morning. At the tone, Pacific Daylight Time will be nine forty-seven and fifty seconds. *beep* Good morning. At the tone, Pacific Daylight Time will be nine forty-eight exactly. *beep*" etc |
| 16:48.31 | *** join/#asterisk snook3r (n=snook3r@bzq-219-46-202.isdn.bezeqint.net) |
| 16:48.32 | Qwell[] | erm, 10.. |
| 16:48.38 | Qwell[] | so just x0 |
| 16:48.52 | *** part/#asterisk masus (n=tet@88.248.73.2) |
| 16:48.56 | *** join/#asterisk ATXFormFactor (n=mark@h-68-165-93-154.nycmny83.covad.net) |
| 16:49.06 | ATXFormFactor | hello |
| 16:49.13 | Qwell[] | Strom_M: I guess the question is, where is the pause at? |
| 16:49.19 | Qwell[] | and is it variable length |
| 16:49.24 | ATXFormFactor | I've got a question about AsteriskNow, but the channel appears to be dead |
| 16:49.25 | Strom_M | Qwell[]: http://www.stromcarlson.com/misc/timechange.wav |
| 16:49.28 | Strom_M | listen :) |
| 16:49.33 | jsmith | ATXFormFactor: Go ahead and ask |
| 16:49.35 | Qwell[] | awesome |
| 16:50.09 | ATXFormFactor | Just set up AsteriskNow Beta 6, went immediately into the rpath setup and fixed the IP address info and password and saved |
| 16:50.19 | Strom_M | Qwell[]: and yes, that recording is accurate; the los angeles machine is LOUD |
| 16:50.21 | defswork | ok - so (seeing as I'm not buying the hardware) I shuold always spec h/w echo canc cards - period ? |
| 16:50.30 | Qwell[] | ahh, so it just loops, and if you call in at xx:xx:05, you might catch the middle of the recording? |
| 16:50.32 | ATXFormFactor | now when I try to configure Asterisk through the web interface, it always refreshes to the login page for the setup wizard |
| 16:50.48 | Strom_M | Qwell[]: no, it rings to pacify the caller until the beginning of a cycle |
| 16:50.48 | ATXFormFactor | when I try to answer the wizard questions and hit next, it refreshes me to the login page for the setup wizard again |
| 16:50.53 | Qwell[] | neat |
| 16:50.54 | Strom_M | Qwell[]: the machine is crosstalk-y |
| 16:51.01 | Qwell[] | yeah, I hear that, heh |
| 16:51.12 | Qwell[] | at the end it's really quiet.. |
| 16:51.17 | coppice | defswork: echo cancellation just takes a lot of CPU power. a few analogue lines can be easily cancelled with host software. an E1 starts to become a heavy load |
| 16:51.20 | jsmith | ATXFormFactor: Very strange... |
| 16:51.34 | jsmith | bkruse: Got any ideas on this one? |
| 16:51.35 | *** join/#asterisk holiday_42 (n=no@spike.wcta.net) |
| 16:51.41 | Strom_M | Qwell[]: it only lets you listen to one minute of recordings |
| 16:51.45 | Strom_M | then it unsupervises |
| 16:52.20 | ATXFormFactor | It's a static IP address setup. When I went into rpath, the gateway wasn't showing there even though I put it in during the initial setup. I put the gateway address in. That, and the initial password change are the extent of my interaction with rpath |
| 16:52.41 | ATXFormFactor | I use FireFox on Linux |
| 16:52.50 | Qwell[] | Strom_M: murf and I were pondering recreating it, heh |
| 16:53.09 | Qwell[] | in an asterisk app... it'd be extremely tricky to do in dialplan |
| 16:53.12 | *** join/#asterisk kolian123 (n=kvirc@124.107.63.223) |
| 16:53.14 | kolian123 | hello |
| 16:53.45 | kolian123 | Was wondering if asterisk able to send different ANI and caller id on a PRI trunk |
| 16:54.06 | Qwell[] | Strom_M: man, that sounds so natural... I bet she recorded every single digit from 1 to 60 |
| 16:54.23 | kolian123 | i'm setting CALLERID(ANI) and CALLERID(NUM) to different values but doesn't seem like it's passed by correctly |
| 16:54.26 | Strom_M | well, the hours are 1-12 |
| 16:54.33 | defswork | coppice: ok - but I'm specifying E1 for all my installs but only using upto 10 channels so far |
| 16:54.37 | Strom_M | minutes one through fifty-nine inclusive |
| 16:54.44 | Strom_M | "and ten seconds" |
| 16:54.46 | Qwell[] | and "exactly" |
| 16:54.53 | Qwell[] | "1 o'clock, exactly" |
| 16:54.54 | Strom_M | "and twenty seconds" |
| 16:55.06 | Strom_M | Qwell[]: it's actually fairly easy to pull that off |
| 16:55.07 | defswork | coppice: and the server being used has never broken sweat - I think I over spec'd that completely :) |
| 16:55.09 | Strom_M | i've done it |
| 16:55.35 | Qwell[] | oh, ha, there's also variablity (..) in the "Good morning" |
| 16:55.55 | Strom_M | if you time all your digits right, you can do 1-99 with: |
| 16:56.24 | Qwell[] | sure, with about 20 recordings, but these sound like she actually recorded "42", "58", etc, all individually |
| 16:57.12 | Strom_M | one two three four five six seven eight nine ten eleven twelve thir- four- fif- six- seven- eight- nine- -teen twenty twenty- thirty thirty- forty forty- fifty fifty- sixty sixty- seventy seventy- eighty eighty- ninety ninety- -one -two -three -four -five -six -seven -eight -nine |
| 16:57.36 | Qwell[] | listen to it though...that'd be *very* difficult to make it sound as natural as it does, with only those recordings |
| 16:57.43 | Strom_M | Qwell[]: I've DONE that |
| 16:57.44 | robl^ | "*SIT* I'm sorry, but due to lack of demand, this is the end of time. Please hang up and look at your digital watch. *REPEAT* |
| 16:58.16 | Strom_M | Qwell[]: it's not difficult if you intonate them all exactly the same way and time them all at the exact same moment |
| 16:58.28 | Qwell[] | oh, I see what you're saying.. fifty and fifty- |
| 17:00.03 | Strom_M | you just need to have a good sense of rhythm and at least relative pitch so that you say every number exactly the same way :) |
| 17:00.13 | Qwell[] | yeah... she was definitely a pro |
| 17:00.23 | Strom_M | is |
| 17:00.26 | Strom_M | she's not dead yet |
| 17:00.39 | Qwell[] | yeah, but I'm sure she'll sound different 40 years later, heh |
| 17:00.43 | robl^ | we need someone with to set up a cron job and dial() & monitor() time every few seconds.. get a complete set of recordings ;-) |
| 17:01.02 | Qwell[] | she's 65 now, and these were probably recorded in the 60s |
| 17:01.17 | Strom_M | Qwell[]: no, they're from the mid to late 1970s |
| 17:01.28 | Qwell[] | 30 years then... |
| 17:01.42 | Strom_M | robl^: I'd much rather get a complete recordings of this time number: |
| 17:01.45 | Strom_M | 702-564-1111 |
| 17:01.46 | *** join/#asterisk flujan (n=flujan@200-160-115-020.static.spo.ctbc.com.br) |
| 17:01.51 | Strom_M | I like Jane's voice better :) |
| 17:02.04 | Strom_M | plus it's a cleaner-sounding machine |
| 17:02.14 | Strom_M | that Weatherchron thing really does sound like crap |
| 17:02.18 | robl^ | was she the one with the weird pronunciations ? |
| 17:02.21 | Qwell[] | the cross-talk gives it character |
| 17:03.31 | robl^ | Strom_M: well time is running out.. :( |
| 17:04.12 | Qwell[] | robl^: get me some clean recordings of the one linked above :D |
| 17:04.31 | Qwell[] | every permeation |
| 17:04.51 | Qwell[] | unfortunately, we'll never be able to get "Pacific Standard Time" |
| 17:04.55 | robl^ | Qwell[]: don't have time to set it up at the moment. I have 5 minutes left of lunch then I have to setup a meeting |
| 17:06.23 | robl^ | Strom_M: that number is a sprint number.. not subject to the AT&T termination is it? |
| 17:07.23 | Strom_M | Qwell[]: just wait till the machine goes to a museum :) |
| 17:07.30 | Strom_M | robl^: duh |
| 17:07.42 | Strom_M | actually the company is called "Embarq" now |
| 17:07.51 | robl^ | do you have an AT&T number? |
| 17:08.07 | Strom_M | an AT&T number where |
| 17:08.21 | *** join/#asterisk Givemelove (n=bozoo@216.70.173.176) |
| 17:08.22 | robl^ | for a *chron? |
| 17:08.23 | Strom_M | los angeles? |
| 17:08.29 | Givemelove | Hey guys |
| 17:08.44 | Strom_M | yes, i'm in los angeles, and yes, I have a POTS circuit from AT&T California |
| 17:08.55 | Givemelove | anybody experiencing issues with Cisco 7900 series IP phones with Asterisk 1.4 and ODBCStorage? |
| 17:08.56 | *** join/#asterisk guillote_GNU (n=guillote@190.7.27.17) |
| 17:09.03 | Kurin- | adhearsion is so buggy |
| 17:09.13 | Givemelove | On all my phones, the light is not blinking on new voice messages |
| 17:09.27 | Strom_M | but I've got measured rate service, so calling the time number a million billion times is going to cost a fortune :) |
| 17:09.52 | robl^ | Strom: I mean do you have a number I can dial for an AT&T Audichron or Weahterchron? |
| 17:10.13 | Strom_M | robl^: in southern california: 213-853-1212 |
| 17:10.30 | Strom_M | robl^: in northern california: 415-767-8900 |
| 17:10.52 | Strom_M | but i dont know how successful you'll be calling that from outside |
| 17:11.54 | chemikk | i have gsm gate "alph line gsm" and i dont have idea how this function |
| 17:12.08 | Strom_M | chemikk: so.....read the manual? |
| 17:12.19 | robl^ | Strom_M: she hung up on me!!!!! |
| 17:12.36 | robl^ | after about 60 secs |
| 17:12.44 | Strom_M | robl^: uh, that's what I said |
| 17:12.48 | *** join/#asterisk kFuQ (n=somedude@c-67-185-112-20.hsd1.wa.comcast.net) |
| 17:12.52 | Strom_M | it only lets you listen for ten cycles |
| 17:12.55 | Strom_M | er, six cycles |
| 17:13.19 | Qwell[] | Strom_M: 767-2616 |
| 17:13.21 | robl^ | Strom_M: I mean.. I was sucessfull for getting through from Texas to that number |
| 17:13.27 | Strom_M | ah |
| 17:13.34 | Qwell[] | pop-corn FTW |
| 17:13.47 | Strom_M | Qwell[]: it's listed in the directory as 767-8900, which I always found easier to dial and remember |
| 17:13.50 | Qwell[] | erm |
| 17:13.54 | Qwell[] | 2676 |
| 17:14.12 | Qwell[] | it's 767-XXXX |
| 17:14.15 | Strom_M | yes, I know |
| 17:14.19 | Qwell[] | ooo, the 415 one works from here |
| 17:14.25 | Qwell[] | awesome |
| 17:14.29 | Strom_M | ah cool |
| 17:14.34 | Strom_M | the 415 one sounds better anyway |
| 17:14.35 | Strom_M | so record that one |
| 17:14.37 | Qwell[] | much |
| 17:14.37 | *** join/#asterisk supjigator (n=sysgod@152.53.16.10) |
| 17:14.47 | Strom_M | san / fran / cisco |
| 17:15.01 | Qwell[] | actually, the 213 one sounds great from here |
| 17:15.07 | Strom_M | uh, no |
| 17:15.11 | robl^ | ok.. lunch over.. time to setup video conference |
| 17:15.14 | Strom_M | the 213 one is overdriven |
| 17:15.16 | Strom_M | use 415 |
| 17:15.20 | *** join/#asterisk generalhan (n=asd@ip67-90-64-2.z64-90-67.customer.algx.net) |
| 17:15.22 | Qwell[] | it isn't loud at all |
| 17:15.27 | Strom_M | maybe you got lucky |
| 17:15.41 | Strom_M | but the 213 one tends to sound like the recording I sent you |
| 17:15.42 | generalhan | OMG Cisco = The Devil (in case anyone wanted to know) ! lol |
| 17:15.54 | blitzrage | that is a well known fact :) |
| 17:16.01 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 17:16.05 | Qwell[] | heh, I just synced my clock with ntp, and I'm watching my clock as it beeps... |
| 17:16.06 | Qwell[] | it's dead on |
| 17:16.15 | Strom_M | always has been |
| 17:16.17 | supjigator | I have an asterisk box with a PRI and another with IAX between the two. Basicall the PRI asterisk box is just a voice gateway. The other remote box is having dropped calls and some call out failures. Transfers don't seem to be reliable either. Any ideas? |
| 17:16.22 | Netgeeks | cisco is the devil? why? |
| 17:16.38 | Qwell[] | Strom_M: I'm amazed by the whole thing, really...1970s tech rocks :p |
| 17:16.39 | *** join/#asterisk [hC] (n=hardcore@66.119.167.162) |
| 17:17.02 | kolian123 | supjigator yes the same problems |
| 17:17.09 | kolian123 | very unreliable |
| 17:17.30 | supjigator | Any solution? |
| 17:17.31 | generalhan | ive been out of the country for the last 6 months working at our call center. our support contract ran out on the 7960. i need the newest firmware casue i am WAY behind, so i call them to get a new one setup, and they say that May was the last date for the 7960 contracts, and they no longer offer them anymore. so now i am screwed. they tell me there is NO WAY to get the new firmware versions |
| 17:17.49 | supjigator | kolian123: Have you tried using SIP instead? |
| 17:17.55 | generalhan | so eve nthough they are still releasing firmware versions i cant get a support contract to get them |
| 17:18.03 | Netgeeks | generalhan: bizarre |
| 17:18.39 | generalhan | Netgeeks: SERIOUSLY bizzare |
| 17:18.49 | Netgeeks | call up cisco and escalate to the office of the president |
| 17:19.24 | Netgeeks | be very annoying and you'll get taken care of. |
| 17:19.28 | generalhan | i talked with 3 people there and then a manager and they all assured me that the only option was to get a new 7960G or 7961. so then i asked them if i was to get one of those and get a support contract if i could then DL the frimware for the 7960, and they said no\ |
| 17:20.00 | Netgeeks | keep escalating, someone with an IQ greate than 50 will get on the phone eventually |
| 17:20.14 | Strom_M | and the moral of the story is "pay attention to EOL announcements and pay your bills on time" |
| 17:20.32 | Qwell[] | Netgeeks: at cisco? |
| 17:20.33 | kolian123 | supjigator, yes both will drop |
| 17:20.39 | Qwell[] | you're joking, right? |
| 17:20.40 | generalhan | Strom_M: thats not fair ... i was not in the country |
| 17:20.56 | generalhan | and no one here is smart enough to tell me that i have mail |
| 17:21.17 | Strom_M | EOL announcements are also online |
| 17:21.32 | Strom_M | so in theory you could call back home and say "renew this" |
| 17:21.35 | supjigator | kolian123: So connecting two asterisk boxes really isn't doable in the real world? |
| 17:21.36 | kolian123 | supjigator are you on 1.4? |
| 17:21.46 | supjigator | One box is. Other isn't |
| 17:21.47 | Netgeeks | Yeah, cisco is being a big company with employees who are not allowed to violate procedure. You need to get above that pay grade and get someone who has a vested interest in making you happy |
| 17:22.02 | supjigator | Both are running current stable verisons. |
| 17:22.15 | *** join/#asterisk prizark_ (i=prizark_@bb121-7-62-196.singnet.com.sg) |
| 17:22.17 | Strom_M | supjigator: define "current stable versions" |
| 17:22.50 | kolian123 | supjigator, i've filed a couple of bugs but there is no resolution |
| 17:22.56 | *** part/#asterisk prizark_ (i=prizark_@bb121-7-62-196.singnet.com.sg) |
| 17:23.11 | kolian123 | supjigator, are you running g729? |
| 17:23.29 | Qwell[] | Strom_M: any idea how many of those machines there are in production in CA? |
| 17:23.37 | Qwell[] | is it just 1 in southern, and 1 in northern? |
| 17:23.37 | supjigator | ULAW |
| 17:24.14 | *** join/#asterisk radicall (n=Zynth@189.168.164.235) |
| 17:24.17 | kolian123 | supjigator, just check where it's dropping though IAX debug |
| 17:24.23 | radicall | hello all |
| 17:24.35 | radicall | someone speak spanish? |
| 17:24.37 | kolian123 | Supjigator, which side is sending hangup |
| 17:24.57 | Strom_M | Qwell[]: IIRC there are machines in los angeles, santa ana, san diego, san francisco, oakland, san jose, and possibly sacramento |
| 17:25.02 | Qwell[] | wow |
| 17:25.08 | Strom_M | but I could be wrong |
| 17:25.22 | Strom_M | i havent looked at that part of the prefix routing table for a long long time |
| 17:26.57 | Qwell[] | I'm so going to reproduce this |
| 17:27.17 | Qwell[] | at least, as much of it as I can |
| 17:27.26 | Qwell[] | then I'll get the phone number 800-omg-time |
| 17:28.14 | Strom_M | haha |
| 17:28.29 | Strom_M | Qwell[]: check out the USNO clock on 719-567-6742 |
| 17:28.30 | ATXFormFactor | hmm, I just reinstalled from scratch AsteriskNow B6 and tried the stuff again without messing with rpath, and I got the same problem in Firefox. I tried Safari instead and it didn't do that |
| 17:28.33 | Qwell[] | which, oddly enough, would also be moh-time |
| 17:28.35 | Qwell[] | O.o |
| 17:29.19 | Qwell[] | neat |
| 17:29.29 | Qwell[] | ooo, 5 second resolution |
| 17:29.35 | Strom_M | well, one second |
| 17:29.37 | Strom_M | since it keeps ticking |
| 17:29.38 | Qwell[] | oh? |
| 17:29.41 | supjigator | kolian123: I haven't gotten that far in debugging. |
| 17:29.56 | Strom_M | technically it's on a fifteen second cycle |
| 17:30.04 | Strom_M | but there's a constant ticking in the background |
| 17:30.08 | Qwell[] | does mountain, then UTC is 5 seconds later |
| 17:30.19 | Qwell[] | then restarts 15 after |
| 17:30.29 | Strom_M | there's one in washington DC as well |
| 17:30.34 | Strom_M | but i dont remember that one |
| 17:30.51 | Strom_M | that's the voice of Fred Covington, who died in 1993 |
| 17:31.17 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 17:31.23 | Strom_M | also a Weatherchron brand machine |
| 17:32.36 | *** join/#asterisk lbow (n=lbow@41-195-77-82.access.uunet.co.za) |
| 17:33.50 | supjigator | kolian123: Do your have any of your bug IDs? |
| 17:34.02 | *** join/#asterisk Dovid (n=Dovid@bzq-82-81-160-215.red.bezeqint.net) |
| 17:34.05 | Dovid | hi guys |
| 17:34.42 | Dovid | i have set up call parking. it works fine. except I dont hear what digits the system was parked on. in the cli I see it playing the digits but the person parking the call never hears it. |
| 17:35.07 | Strom_M | Dovid: you're supposed to do an attended transfer |
| 17:35.10 | Strom_M | not a blind transfer |
| 17:35.36 | *** part/#asterisk generalhan (n=asd@ip67-90-64-2.z64-90-67.customer.algx.net) |
| 17:35.59 | *** join/#asterisk karleeto (i=karl@gentoo.karlhaines.com) |
| 17:40.53 | *** join/#asterisk McJerry (n=bryanj02@207-144-4-148.cstel.net) |
| 17:41.12 | Dovid | Strom_C: thanks |
| 17:41.34 | *** join/#asterisk r3n4n (n=renan@200.210.51.130) |
| 17:43.10 | McJerry | <PROTECTED> |
| 17:44.00 | Corydon76-dig | Yeah, that might work until someone's voice is detected as a busy signal... |
| 17:44.25 | Qwell[] | but with busycount=4, it would have to happen 4 times (in time) before it detected it, right? |
| 17:44.30 | Corydon76-dig | busydetect isn't very reliable that way... false positives... |
| 17:45.10 | Corydon76-dig | Qwell[]: if someone's voice has that tambre, very likely |
| 17:48.05 | codefreeze | MindTheGap: still around? Which ver of asterisk are you using, that you have probs with NoCDR? |
| 17:51.34 | bkruse | jsmith: thanks! ATXFormFactor: whats up? join #asterisk-gui, or file a bug report, or both, and we can talk about it hear also |
| 17:52.40 | *** join/#asterisk snook3r (n=snook3r@bzq-219-46-202.isdn.bezeqint.net) |
| 17:53.14 | MindTheGap | codefreeze, how are you? im using an old res_ldap trunk version based on 1.4.x... you were working on cdr yes? i had that problem on cdr recording "s" instead of the number inside macros, remember? |
| 17:53.32 | *** join/#asterisk drwelby (n=mpfister@mail.enplan.com) |
| 17:56.25 | codefreeze | MindTheGap: They say the memory goes first... mine started going when I was like 14. |
| 17:57.43 | codefreeze | MindTheGap: I just tried that scenario; and I don't get any output either way, if I invoke NoCDR.... Maybe you could try a more up-to-date 1.4? |
| 17:57.51 | MindTheGap | codefreeze, i see... thats why i log everything :) |
| 17:58.21 | codefreeze | MindTheGap: I tried that. Couldn't remember where in the logs I wrote the stuff! |
| 17:58.45 | MindTheGap | codefreeze, well... last time I checked that trunk was abandoned... got to go see if its up to date... |
| 18:00.32 | *** join/#asterisk darkfires (n=lwhite@d38-37-41.commercial1.cgocable.net) |
| 18:00.39 | *** join/#asterisk blitzrage (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 18:00.39 | *** mode/#asterisk [+o blitzrage] by ChanServ |
| 18:00.41 | MindTheGap | i remember you offered a cdr.c but it wouldnt compile, than i fixed the dialplan with a exten => s,1,Goto(${ARG1},1) and never got the chance to try the new cdr code... |
| 18:00.51 | darkfires | Does transfering work with SLA? |
| 18:01.01 | *** join/#asterisk mocker (n=mocker@198.247.173.227) |
| 18:01.51 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@c-98-195-23-75.hsd1.tx.comcast.net) |
| 18:02.35 | *** join/#asterisk CVirus (n=GoD@82.201.222.191) |
| 18:05.44 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) |
| 18:06.13 | *** join/#asterisk generalhan (n=asd@ip67-90-64-2.z64-90-67.customer.algx.net) |
| 18:06.21 | *** join/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net) |
| 18:07.26 | *** join/#asterisk CVirus (n=GoD@82.201.222.191) |
| 18:09.37 | generalhan | hey all, i need some troubleshooting advice on what to look for to get to the bottom of a issue that has just started recently. every one of our users are complaining that people on the other end are saying they are cutting in and out and there is a lot of echo on the line. I just dont know where to look, because nothing in my setup has changed, and its been working perfectly for a long time now. |
| 18:09.54 | MindTheGap | codefreeze, it seems somethings got done over there at res_ldap... will try out this weekend and ill let you know... |
| 18:10.22 | *** join/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net) |
| 18:10.39 | codefreeze | MindTheGap: OK; I just tried it on 1.4 and trunk, and both are obeying nocdr, with or without a gotoif. |
| 18:11.18 | MindTheGap | codefreeze, nice to know that, thanks... |
| 18:11.25 | *** join/#asterisk famicon (i=pastry@c51447ddc.cable.wanadoo.nl) |
| 18:13.16 | defsdoor | generalhan: whats the load on the server like ? or the network ? |
| 18:13.21 | *** part/#asterisk famicon (i=pastry@c51447ddc.cable.wanadoo.nl) |
| 18:13.43 | generalhan | defsdoor: according to the network graphs, low |
| 18:14.01 | defsdoor | generalhan: any problems on internal calls ? |
| 18:14.15 | *** join/#asterisk Lucky7 (n=Adam@207.200.28.175) |
| 18:14.17 | generalhan | defsdoor: internal calls are flawless |
| 18:14.20 | gerphimum | morning all - does anyone know if there is a hardware device that can broadcast to cb radio frequencies (ie: walkie talkies) ? specifically, is it possible for asterisk to broadcast on a cb freq instead of, say, an intercom, assuming such hardware exists |
| 18:14.32 | defsdoor | generalhan: what external lines ? |
| 18:14.37 | generalhan | PRI |
| 18:15.06 | jsmith | gerphimum: There are cards for doing amateur radio, but I don't know of any that do CB frequencies |
| 18:15.21 | generalhan | a long time ago i had to set the rx and txgain to fix an issue and i wonder if that would have anything to do with it ? |
| 18:15.24 | jcims | can you give us a basic application generalhan ? |
| 18:15.43 | jcims | is it two way or just one way 'broadcast' |
| 18:15.49 | generalhan | SIP > * > PRI |
| 18:15.49 | cpm | gerphimum, not legal w/CB |
| 18:15.57 | cpm | can do it with some ham stuff |
| 18:16.10 | cpm | yer looking for a 'phone patch' |
| 18:16.17 | Givemelove | anybody has an ODBC to voicemail script? I'd like to revert back to regular voicemail |
| 18:16.32 | generalhan | i havent had anyone tell me that they are hearing the echo, just the people on the other end. |
| 18:16.49 | *** join/#asterisk CVirus (n=GoD@82.201.222.191) |
| 18:17.16 | generalhan | i have been running zttest every day and i do get the occasional 99.93% but the average is always 99.989% or 99.99% |
| 18:17.38 | generalhan | but then i heard that, that is not always the most accurate test to run |
| 18:18.05 | gerphimum | jcims >> basic application: restaruant. cashier needs help with something but no manager is around. all the managers carry walkie talkies though, so the cashier dials an extension that broadcasts the message "manager needed at register" |
| 18:18.19 | generalhan | http://generalhan.pastebin.ca/676011 <<--- this is what im working with for zaptel and zapata |
| 18:20.50 | jcims | ah, ok |
| 18:21.45 | jcims | cona fide cb freq? |
| 18:21.51 | jcims | er bona |
| 18:21.55 | gerphimum | researching it now, no. |
| 18:22.16 | gerphimum | GMRS would be more suitable |
| 18:22.30 | [TK]D-Fender | generalhan: Buy a REAL card with HWEC, and stop relying on Zaptel SWEC |
| 18:23.09 | Kurin- | How can I have an AGI or dialplan working in the background while the caller is on hold? Currently it just hangs while playing hold music |
| 18:23.09 | generalhan | HWEC ? |
| 18:23.35 | generalhan | Hardware Echo cancel ? |
| 18:23.41 | Qwell[] | yes |
| 18:23.47 | generalhan | what kind of card ? |
| 18:23.59 | generalhan | i am using a digium TE210p |
| 18:24.02 | many | cpm: i wouldnt say that too loud |
| 18:24.06 | Qwell[] | There is a hwec module for that |
| 18:24.20 | many | cpm: on ham frequencies youre only allowed to interconnect when youre ham, too |
| 18:24.22 | Qwell[] | it just plugs right in to the card |
| 18:24.32 | cpm | many, yup. |
| 18:24.38 | many | cpm: however, there are a lot teamspeak<>CB Gateways, and they *are* lawful. |
| 18:24.52 | jcims | tear apart a walkie talkie, put a transistor across the talk switch..wire it to your serial port, raise DTR or something to talk, route audio to mic input on walkie talkie, say whatever, lower DTR |
| 18:24.54 | jcims | lol |
| 18:25.04 | Qwell[] | jcims: hack |
| 18:25.05 | Qwell[] | :p |
| 18:25.06 | cpm | pretty sure last I checked, you ain't sitting in front of the rig, with the locally connected mic, you ain't supposed to broadcast w/CB |
| 18:25.07 | jcims | haha |
| 18:25.17 | Qwell[] | There are pci radio cards that work with zaptel |
| 18:25.17 | many | cpm: thought so, too. |
| 18:25.20 | cpm | but who cares about CB anymore |
| 18:25.31 | Qwell[] | app_rpt I think is the app you'd use in asterisk |
| 18:25.33 | many | cpm: but that TS<>CB community is large and they are allowed to do so |
| 18:25.34 | jcims | i think that's part of the problme, didn't generalhan say it wasn't CB? |
| 18:25.38 | cpm | you *may* use a phone patch w/CB, but you still have to have a local operator |
| 18:25.38 | defsdoor | I doubt the managers carry CB radios |
| 18:25.50 | jcims | it's just walkie talkes, probably 900mhz or whatever |
| 18:26.00 | gerphimum | yeah, it would be GMRS, not CB.. |
| 18:26.02 | many | 900mhz? thats GSM. |
| 18:26.04 | gerphimum | that was my bad :| |
| 18:26.04 | cpm | FRS radios otoh, is very different |
| 18:26.09 | generalhan | Qwell[]: do you have a link for that part? |
| 18:26.15 | gerphimum | cpm >> or FRS |
| 18:26.33 | cpm | has to be used as it comes out of the box/package, The End. |
| 18:26.44 | cpm | you can't hook up nothing but a headset |
| 18:26.54 | jcims | http://www.qsl.net/n9zia/linux_frs/index.html |
| 18:27.04 | cpm | and you have to operate it by hand,. and only for Family stuff as the FRS implies. |
| 18:27.06 | many | walkie talkies are either CB at 27MHz or something between 400-500MHz |
| 18:27.08 | jcims | told you ya had to hack the radio...haha |
| 18:27.15 | cpm | GMRS is a bit more liberal, but you need to get licensed |
| 18:27.51 | cpm | and you still ain't allowed to fiddle with the actual radio itself. |
| 18:27.57 | cpm | Still can't remotely operate, that kinda thing |
| 18:28.05 | gerphimum | cpm >> what if the radio is a pci card.. |
| 18:28.10 | gerphimum | ah thats lame |
| 18:28.33 | jcims | google gmrs asterisk pbx, there are a few good looking threads |
| 18:28.38 | cpm | note, that when I say can/can't, I mean may/may-not |
| 18:28.41 | cpm | there is a difference |
| 18:28.42 | jcims | probably amount to squat :) |
| 18:30.07 | gerphimum | que es "amatuer radio repeaters" |
| 18:30.43 | many | oh, gmrs even uses a shift. what for? |
| 18:32.24 | generalhan | so i see the DSP-based echo cancellation module on the TE212 but i cant find the module to buy to fit onto the TE210P i already have ? |
| 18:35.29 | datachomper | Ok, so I had a sip phone registering fine and dandy in a remote location. This person moves to another location and all the sudden it's no longer registering. |
| 18:35.48 | datachomper | What are my avenues of troubleshooting? |
| 18:36.02 | jsmith | generalhan: Call Digium, and they'll sell one to you. |
| 18:36.38 | generalhan | jsmith: well i assumed that i could just call them ... but i was hoping to see it online, kinda price it out before i have to budget it in |
| 18:37.00 | [TK]D-Fender | generalhan: Phoning them up gets you the same answer... |
| 18:37.18 | [TK]D-Fender | datachomper: Gee, I don't know... check the networking maybe? |
| 18:38.49 | *** join/#asterisk Winkie (i=sd@87-194-8-125.bethere.co.uk) |
| 18:38.52 | *** join/#asterisk [T]ank (n=ckwall@206.71.78.172) |
| 18:39.40 | [T]ank | I am getting a complaint of "When he goes to transfer to a different internal ext the person he transfered to can't hear anything he says, however, the transfer works fine and both the new ext and the customer can hear each other" This just started randomly... what could be the issue? |
| 18:40.37 | datachomper | It's strange, the PC he's got daisy chained from the back of his polycom 501 has internet connectivity |
| 18:40.47 | defsdoor | [T]ank: user error |
| 18:40.47 | datachomper | Said PC can ping my asterisk box |
| 18:41.47 | Strom_M | [T]ank: perhaps not knowing the difference between blind and attended transfers? |
| 18:41.49 | [TK]D-Fender | datachomper: taht has nothing to do with making UDP work. |
| 18:42.11 | generalhan | crazy ... the EC module is going to cost as much as the card iteslf, this is gonna be rough to get passed the boss !lol |
| 18:42.39 | [TK]D-Fender | generalhan: Should have just bought one with HWEC in the first place. |
| 18:42.41 | datachomper | I'm watching the ftp logs for the configuration and I'm not getting any hits |
| 18:43.09 | generalhan | [TK]D-Fender: i swear that is new ... when i bought the TE 3 years ago i dont think that there was the TE212 |
| 18:43.33 | generalhan | im usually wrong ... but i didnt notice it |
| 18:43.35 | [TK]D-Fender | generalhan: the TE2XX line isn't even **TWO** years old! |
| 18:44.53 | generalhan | forgot that we started with * on VoIP |
| 18:44.59 | generalhan | man that was a long time ago |
| 18:45.57 | *** join/#asterisk lbow (n=lbow@41-195-77-82.access.uunet.co.za) |
| 18:46.07 | *** join/#asterisk ptiggerdine_ (n=ptiggerd@123-243-144-208.tpgi.com.au) |
| 18:46.51 | [TK]D-Fender | generalhan: And you're hopping from one source of pain to another systematically ;) |
| 18:47.10 | hmmhesays | just got back from the velvet revolver / alice in chains show |
| 18:47.14 | hmmhesays | one word... awesome |
| 18:47.40 | generalhan | [TK]D-Fender: i dont understand why this has been working soo well for so long, and then just decides to be a proble, |
| 18:48.18 | generalhan | problem even |
| 18:49.02 | generalhan | where would you have gone after the horrible VoIP failure ? |
| 18:49.50 | [TK]D-Fender | generalhan: You already know that answer :) |
| 18:50.05 | generalhan | you mean directly to a HWEC board |
| 18:57.49 | [TK]D-Fender | generalhan: Damn right. If you you're going to do a job, do it RIGHT. The FIRST time. |
| 18:58.58 | Strom_M | or else have [TK]D-Fender scream at you (much to the perpetual irritation of his doctor and the perpetual delight of the pharmaceutical companies that make blood pressure medication) :) |
| 19:01.34 | *** part/#asterisk r3n4n (n=renan@200.210.51.130) |
| 19:05.17 | *** join/#asterisk MaliutaWrk (n=nikolai@fw.hitwise.com) |
| 19:06.02 | *** join/#asterisk |dennis| (n=dennis@200.32.233.84) |
| 19:06.58 | Nix | is there a public SVN tree for asterisk somewhere?, |
| 19:07.30 | Dr-Linux | how many ms in one minute? |
| 19:07.31 | Strom_M | Nix: www.asterisk.org |
| 19:07.42 | Strom_M | Dr-Linux: 1000 ms per second |
| 19:07.49 | Strom_M | so therefore..... |
| 19:08.00 | Nix | I thought it needs a ssl cert... |
| 19:08.20 | Nix | ahh... not for annon |
| 19:08.21 | *** join/#asterisk PioneerVM2 (n=IceChat7@ool-45779466.dyn.optonline.net) |
| 19:08.22 | Nix | ignore me |
| 19:08.39 | Strom_M | Dr-Linux: come on, do the math :) |
| 19:08.51 | PioneerVM2 | hey anyone here know anything about CID Blocking + Asterisk + Linksys PAP2T (or similar) |
| 19:08.52 | Dr-Linux | Strom_M: thanks :) |
| 19:08.54 | Nix | i was reading the developers page.. |
| 19:09.10 | Strom_M | PioneerVM2: what are you attempting to do exactly |
| 19:09.12 | Dr-Linux | Strom_M: i'm very week in math |
| 19:09.23 | Dr-Linux | Strom_M: my asterisk shows time "2238 |
| 19:09.27 | Dr-Linux | "2238 |
| 19:09.30 | Strom_M | Dr-Linux: 60 x 1000 is not exactly a difficult calculation |
| 19:09.54 | Dr-Linux | Strom_M: so it it seconds are ms :S |
| 19:10.14 | PioneerVM2 | I have PAP2T -> Asterisk box (i control) -> Voip provider -- PAP2T has options built in for *67 and its turned on -- when i hit *67 i get stutter dialtone confirming, then dial # -- asterisk box sees only the # dialed, am wondering how i get it to detect (either on its own or thru a dialplan setup) that *67 was typed |
| 19:10.46 | PioneerVM2 | i would like to do it proper way and have PAP2T accept *67 there and only pass phone # to box so i can hear stutter tone confirm |
| 19:10.46 | Strom_M | Dr-Linux: that made no sense |
| 19:11.04 | Dr-Linux | Strom_M: i understand that time is in seconds :) |
| 19:11.06 | Dr-Linux | i got it :P |
| 19:11.07 | Dr-Linux | Thanks |
| 19:11.19 | Strom_M | yes, and mili- means "one thousandth of" |
| 19:11.32 | PioneerVM2 | i can find no documentation anywhere on properly setting up CID blocking :( |
| 19:12.00 | Strom_M | PioneerVM2: how about...SetCallerPres() |
| 19:12.10 | PioneerVM2 | Not familiar with that. |
| 19:12.14 | *** join/#asterisk Telamon (n=telamon@blk-222-160-100.eastlink.ca) |
| 19:13.15 | PioneerVM2 | hmm that seems more for incoming rather than outgoing? |
| 19:13.27 | Strom_M | no |
| 19:13.34 | PioneerVM2 | well lets jump back a second -- lets say i have 2 lines and 2 PAP2T boxes and an asterisk box |
| 19:13.47 | PioneerVM2 | so line with phone #111 and line with phone #222 (shortened for typing) |
| 19:14.01 | PioneerVM2 | and box A is to use 111 outgoing and box B is to use 222 |
| 19:14.42 | PioneerVM2 | where is the proper place to put the caller ID info -- in the sip handset config for A/B as "callerid=" option -- or setting manually with Set when handling an outgoing call, or in the PAP2T box itself |
| 19:16.07 | Strom_M | what do you mean "line" in this context? |
| 19:16.19 | *** join/#asterisk jetlagmk2 (n=jetlag@pool-70-17-41-64.pskn.east.verizon.net) |
| 19:16.20 | PioneerVM2 | hmm |
| 19:16.25 | Telamon | PioneerVM2: In the callerid field of the user record on Asterisk. Sounds like you are using a GUI though, so I'm not sure what screen that owuld be in. |
| 19:16.33 | PioneerVM2 | no im using text |
| 19:16.38 | PioneerVM2 | not sure how to answer your question strom |
| 19:16.41 | PioneerVM2 | let me re-explain |
| 19:16.49 | NOT_guru | ~Grandstream |
| 19:16.49 | jbot | extra, extra, read all about it, grandstream is the Yugo of VoIP hardware. Run. Run away now. |
| 19:17.05 | PioneerVM2 | Asterisk box, 2 Voip lines through Teliax provider |
| 19:17.07 | PioneerVM2 | DIDs i suppose |
| 19:17.08 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
| 19:17.14 | PioneerVM2 | 111-1111 and 222-2222 for example |
| 19:17.24 | PioneerVM2 | i can accept and dial out calls using those lines |
| 19:17.58 | PioneerVM2 | now, i have 2 PAP2T boxes in my home (make believe they only have one analog port each for simplicity) -- each one has a different login to the asterisk box in config "handset_a" and "handset_b" |
| 19:18.06 | [hC] | anyone ever see a polycom return 'not acceptable here' and return circuit busy when you try to call it? |
| 19:18.19 | PioneerVM2 | handset_a gets and makes calls with CID 111-1111 |
| 19:18.27 | PioneerVM2 | handset_b gets and makes calls with CID 222-2222 |
| 19:18.38 | PioneerVM2 | (think of 111... as home line and 222.. as biz line) |
| 19:18.45 | Telamon | In asterisk, set the callerid field for users handset_a and handset_b. If you aren't using a DB backend, that would be in the /etc/asterisk/sip.conf file |
| 19:18.45 | *** join/#asterisk |dennis| (n=dennis@200.32.233.84) |
| 19:19.00 | PioneerVM2 | ok i figured that out yesterday and tried it and it works |
| 19:19.16 | PioneerVM2 | i set callerid="Joe Smith" <111-1111> for example |
| 19:19.22 | PioneerVM2 | in the handset config, that does work. |
| 19:19.37 | PioneerVM2 | so that is the "correct" way to do it im guessing |
| 19:20.07 | PioneerVM2 | i use to use a different outgoing dialplan for each handset and set the callerid manually using SET(CALLERID...) depending on which handset was being used |
| 19:20.21 | PioneerVM2 | then figured out that callerid= setting for handset was more efficient |
| 19:20.40 | wunderkin | [hC], with a bad config, yeah, i don't remember what specifically, start from a fresh config and go step by step |
| 19:20.54 | PioneerVM2 | so the question comes back to, how to i get *67 to properly work from PAP2T -- when i type *67, PAP2T box stutters to indicate accepted code |
| 19:20.55 | Telamon | Does anyone know where I can find "official" (or at least, with all required fields) db schemas for Asterisk 1.4? Specifically, I'm looking for sip_users, extensions, and CDR. |
| 19:21.14 | PioneerVM2 | then i type 555-1212 for example and Asterisk only sees 555-1212 and does not block the callerid= setting for the handset |
| 19:21.52 | PioneerVM2 | is the PAP2T box suppose to pass some kind of code or signal to asterisk to indicate CID should be blocked? |
| 19:22.05 | *** join/#asterisk |dennis| (n=dennis@200.32.236.10) |
| 19:22.20 | PioneerVM2 | (FYI, CallerID block option in config for PAP2T says "*67" and is turned "on") |
| 19:23.25 | *** join/#asterisk Blackthorn (i=blacktho@76.77.160.10) |
| 19:24.24 | Blackthorn | Hi, I'm having a little trouble with my asterisk 1.4.4 server in that in the console I will get a set of repeating messagse over and over and never stop untill i sotp the server and restart it. those messages start with |
| 19:24.35 | Blackthorn | WARNING[18733]: app_dial.c:674 wait_for_answer: Unable to forward voice frame |
| 19:24.50 | *** join/#asterisk btsteve (n=btsteve@204.10.20.30) |
| 19:25.23 | [hC] | oh problem #2, anyone heard of a polycom phone that when you call it, it doesnt ring and the line key indicators dont blink?? |
| 19:25.33 | Blackthorn | Have any idea what is causing this? I also noticed that most of the time when the messages start repeating it stops answering calls. |
| 19:26.10 | wunderkin | [hC], yeah, i think that was a nat problem.. what version of firmware are you using? |
| 19:26.15 | *** join/#asterisk msetim (n=marcos@200.195.161.164) |
| 19:26.34 | msetim | hi guys |
| 19:29.06 | msetim | How can I add a personalized column extra to cdr? ( database cdr ) Thanks :-D |
| 19:31.45 | [hC] | wunderkin: phones not behind nat, and 2.1.2 |
| 19:32.45 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-79-180-14-100.red.bezeqint.net) |
| 19:35.19 | *** join/#asterisk saftsack (n=saftsack@p57A76966.dip.t-dialin.net) |
| 19:35.22 | [TK]D-Fender | PioneerVM2: this will only work if you do not use "callerid=" for your sip entry |
| 19:35.37 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 19:35.37 | *** mode/#asterisk [+o anthm] by ChanServ |
| 19:38.02 | simond | anthm: shidan mentioned that you got fs running on an arm processor, did you use buildroot to do that or something? |
| 19:38.35 | anthm | arm11 or armel as they call it with scratchbox |
| 19:38.37 | anthm | for n800 |
| 19:39.15 | *** part/#asterisk [T]ank (n=ckwall@206.71.78.172) |
| 19:39.21 | [hC] | wunderkin: are you running 2.2.0 yet? |
| 19:40.20 | wunderkin | [hC], never had that problem on 2.1.2, but i think some of our wackyness was from registering with the local and extern ip (for failover) so i just changed to the extern ip only, not sure what i was thinking there... 2.1.2 has been great for us but we don't use anything fancy like sla, haven't tried or had the need to try 2.2.0 yet |
| 19:40.48 | *** join/#asterisk powerkill (n=powerkil@84.205.154.247) |
| 19:41.20 | [TK]D-Fender | 2.2.0 adss some wonderful stuff... |
| 19:41.21 | [hC] | wunderkin: i just downloaded it and im gonna upgrade to it, but theve changed a bunch of file naming again.. i hope its not totally messed up again with some major config format change. |
| 19:41.37 | [hC] | bunch of files in the new bootrom such as |
| 19:41.37 | [hC] | BootRom_4_0_0_release_sig/2345-11500-020.bootrom.ld |
| 19:41.45 | [hC] | [TK]D-Fender: whats new in 2.2.0 thats so cool? |
| 19:41.59 | wunderkin | yeah, the firmware and bootrom archives are f'n huge now.. |
| 19:42.03 | PioneerVM2 | D-F.. so how do i set the CID number on outgoing calls? |
| 19:42.06 | jsmith | Imagine that.. |
| 19:42.14 | jsmith | Polycom, with big firmware files? |
| 19:42.14 | PioneerVM2 | If i don't put it in callerid= then i don't have a caller id # set |
| 19:42.21 | [TK]D-Fender | wunderkin: huge & small at the same time if you look at the actual breakdown. |
| 19:42.21 | PioneerVM2 | it just comes out blank all the time |
| 19:42.28 | [hC] | big is a little bit of an understatement |
| 19:42.41 | [hC] | not to mention fighting with the extranet site for 5 minutes to get it to actually give me the file. |
| 19:42.56 | [TK]D-Fender | PioneerVM2: All CID must come from the ATA. And as for what you send to the PSTN, you that answer isn't any different than before. |
| 19:42.57 | wunderkin | at least you have access |
| 19:43.11 | [hC] | i went thru the pain of registering as a certified reseller |
| 19:43.16 | PioneerVM2 | ATA do you mean the Linksys PAP2t? |
| 19:43.20 | [hC] | it didnt exactly make anything much easier, |
| 19:43.22 | [TK]D-Fender | PioneerVM2: Yes. |
| 19:43.27 | [hC] | it would likely be easier to convince someone else to give me the zip. |
| 19:43.28 | [hC] | :) |
| 19:43.33 | PioneerVM2 | do you know how to set the DID in the PAP2t? |
| 19:43.40 | PioneerVM2 | i couldnt find an option for that |
| 19:43.46 | [TK]D-Fender | PioneerVM2: go look at the dan UI! |
| 19:43.48 | PioneerVM2 | but i figured that was a possible solution |
| 19:43.48 | [TK]D-Fender | damn* |
| 19:43.54 | PioneerVM2 | i did, it's huge and i couldnt find it anywhere |
| 19:44.09 | [TK]D-Fender | PioneerVM2: how about a blatant thing like "display name" ? |
| 19:44.37 | PioneerVM2 | AHh is that what DIsplay name is? |
| 19:44.42 | PioneerVM2 | does that do phone # as well? |
| 19:45.04 | [TK]D-Fender | PioneerVM2: Go try stuff. Then read the MANUAL. Then check out Voxilla's forums. |
| 19:45.05 | PioneerVM2 | you would think it would be called something logical like "Phone Number" or "DID" or something |
| 19:45.15 | PioneerVM2 | yea i've searched all over on this, and tried stuff |
| 19:45.35 | PioneerVM2 | thats how i even got this far, there is so little documentation on caller ID settings |
| 19:45.55 | PioneerVM2 | whenever i search on a term in a linksys pap2t box, i wind up with some idiots config page that is insecure with no password |
| 19:45.57 | [TK]D-Fender | PioneerVM2: You keep thikning that this box is exclusively for ITSP usage. DID doesn't mean SHIT to this box. its a friggen SIP UA. Nothing more. Youa re attaching YOUR INTENDED USE to its "fuction". |
| 19:46.17 | PioneerVM2 | "DIsplay name" does not indicate "number" in any fashion |
| 19:46.29 | PioneerVM2 | they refer to "CID" all over the place in the config |
| 19:46.40 | PioneerVM2 | you would figure the label would say "CID Display" or something |
| 19:46.44 | [TK]D-Fender | PioneerVM2: get over it. |
| 19:46.53 | [hC] | Its all in the same like.. 3 lines in the Web ui |
| 19:46.54 | PioneerVM2 | i'm over it, your yelling at me for some unknown reason |
| 19:46.56 | [hC] | its not hidden |
| 19:47.00 | darkfires | PioneerVM2 then change it yourself and quite your bitching |
| 19:47.02 | PioneerVM2 | all you had to say was "It's called Display Name" |
| 19:47.13 | *** join/#asterisk [T]ank (n=ckwall@206.71.78.172) |
| 19:47.43 | PioneerVM2 | yes, i'll 'quite' my bitching |
| 19:48.13 | darkfires | this isn't english class |
| 19:48.44 | PioneerVM2 | I'm not sure why every question in this room is answered obnoxiously. |
| 19:48.57 | PioneerVM2 | Or at least a vast majority of them |
| 19:49.12 | [hC] | PioneerVM2: cause youre asking fender :) |
| 19:49.13 | darkfires | because every question you have asked is blatantly obvious |
| 19:49.20 | PioneerVM2 | Not quite |
| 19:49.36 | [hC] | well regarding the CID question that is pretty damn obvious man. |
| 19:49.48 | PioneerVM2 | Apparently not to me. |
| 19:49.49 | [hC] | its the line above display name |
| 19:49.57 | [T]ank | someone flame me... i am feeling left out |
| 19:49.58 | [hC] | or below it or whatever it is |
| 19:49.59 | PioneerVM2 | THere is no line above display name |
| 19:50.07 | PioneerVM2 | above is nothing, below is "password" |
| 19:50.07 | [hC] | its within 2cm of display name. |
| 19:50.15 | [hC] | whats below that? |
| 19:50.16 | *** join/#asterisk Ebola (n=Ebola@host86-138-6-48.range86-138.btcentralplus.com) |
| 19:50.20 | PioneerVM2 | user id, use auth id, auth id, password, displya name |
| 19:50.25 | [TK]D-Fender | PioneerVM2: What do YOU think "display name" implies? What made you associate "DID" with "callerid"? But I only use my ATA's for.. EXTENSIONS. Maybe it should be called "extension"! No wait.... |
| 19:50.27 | [hC] | User Id = CID |
| 19:50.41 | PioneerVM2 | Actually, i inferred display name referred to a name to appear on a phone text label |
| 19:50.54 | [hC] | Heres what it is, so you're clear |
| 19:51.00 | [hC] | User ID = CID Number |
| 19:51.08 | PioneerVM2 | no, User ID = asterisk login name |
| 19:51.12 | [hC] | Auth ID = authentication username you authenticate to asterisk with |
| 19:51.14 | Blackthorn | WARNING[18733]: app_dial.c:674 wait_for_answer: Unable to forward voice frame --- Anyone know what causes this message to be repeated ver quickly until server restart? |
| 19:51.17 | PioneerVM2 | thats the sip login name |
| 19:51.20 | [hC] | Display Name = Caller ID NAME |
| 19:51.31 | PioneerVM2 | User ID in my case is the sip authentication name |
| 19:51.35 | [TK]D-Fender | [hC]: That makes a fair amount of sense actually.... |
| 19:51.42 | [hC] | User Id and Auth ID kind of go hand in hand for CID or not, depending on if you use 'Use Auth ID' |
| 19:51.58 | [TK]D-Fender | [hC]: I hadn't thought about the number aspect, but that explains the seperate "auth name" perfectly. |
| 19:52.00 | [hC] | PioneerVM2: and your sip authentication name would then be sent as your CID number. |
| 19:52.02 | PioneerVM2 | hC -- thank you for making some sense and not being obnoxious |
| 19:52.16 | [hC] | [TK]D-Fender: yeah, its incase you authenticate as 'joe' but joe is ext 102. |
| 19:52.20 | [TK]D-Fender | [hC]: largely because I would never let a UA detimine its own CID :) * enforcement all the way! |
| 19:52.20 | *** join/#asterisk ccesario (n=ccesario@189-19-9-100.dsl.telesp.net.br) |
| 19:52.22 | PioneerVM2 | I was about to say "Display NAME" would infer "NAME" not number |
| 19:52.49 | [hC] | [TK]D-Fender: I let the UA handle its own cid for internal calls. * enforcement for anything that leaves * |
| 19:52.53 | PioneerVM2 | hC -- so select "use auth id" and change my user ID to my auth id field, then put phone # in user id? |
| 19:52.55 | [TK]D-Fender | PioneerVM2: And in so doing be closer than your blind question earlier left you. You really should "just try stuuff" more... |
| 19:52.55 | VJFROMGT | how do i end a call from shell? |
| 19:53.07 | PioneerVM2 | DF, have you seen the size of this config menu"? |
| 19:53.09 | [TK]D-Fender | VJFROMGT: "soft hangup [channel]" |
| 19:53.10 | [hC] | PioneerVM2: let me ask you... what is your sip peer in asterisk's sip.coinf set to |
| 19:53.15 | [T]ank | if you mean from the cli> type hangup |
| 19:53.16 | [TK]D-Fender | PioneerVM2: Yes. |
| 19:53.16 | PioneerVM2 | i have tried thousands of options, i didnt know if it was something in asterisk or the menu |
| 19:53.18 | [hC] | PioneerVM2: is it your extension number/DID? |
| 19:53.36 | *** join/#asterisk rvhi (n=chatzill@66.135.230.96) |
| 19:53.38 | PioneerVM2 | not sure what you mean hc |
| 19:53.48 | [T]ank | or if you are killing a specific call like [TK]D-Fender said, soft hangup [channel] |
| 19:53.52 | PioneerVM2 | i have a [home_phone] section for registering my pap2t box |
| 19:53.55 | PioneerVM2 | and it has a password, etc. |
| 19:53.56 | [TK]D-Fender | PioneerVM2: "yes" to SPA-2000, 1001, 3000, 3102. I figure the PAP is only SMALLER than the rest. |
| 19:54.01 | PioneerVM2 | i was setting "callerid=" to set # |
| 19:54.11 | [hC] | PioneerVM2: okay see you're getting a ton of terms confused here. |
| 19:54.19 | [hC] | PioneerVM2: so your SIP username is "home_phone" ? |
| 19:54.29 | rvhi | hi, is there any limit of number of arguments passed to macro? |
| 19:54.31 | [TK]D-Fender | PioneerVM2: and I told you to REMOVE "callerid=" |
| 19:54.36 | PioneerVM2 | ok, so what should i set to "home_phone" -- the auth name and set use auth name to yes? |
| 19:54.44 | PioneerVM2 | DF yes i understand that, i was stating how i use to do it |
| 19:54.46 | [T]ank | rvhi: i dont think there is |
| 19:54.51 | [hC] | PioneerVM2: we're going back to basics here cause this is where you're getting confused. |
| 19:54.52 | [T]ank | i have used quite a few in the past |
| 19:54.55 | PioneerVM2 | yes sip username = home_)phone |
| 19:54.59 | [T]ank | how many are you thinkig of using? |
| 19:55.04 | [hC] | Normal people dont register as 'home_phone' they would register as the DID or the Extension number |
| 19:55.27 | rvhi | [T]ank: so just use arg1... all the way to arg20, arg100 and etc? |
| 19:55.28 | darkfires | is vm-duration.gsm recorded for english yet? |
| 19:55.33 | [T]ank | yep |
| 19:55.42 | PioneerVM2 | hC -- i want to use the same # on two different devices |
| 19:55.43 | [TK]D-Fender | PioneerVM2: ok. leave your * setup as is but remove "callerid=". on your PAP2 add the display name, userid, and authuser=[your * userhere] |
| 19:56.05 | [hC] | PioneerVM2: so two devices both show up as CID like.. 1-###-###-#### |
| 19:56.09 | PioneerVM2 | so authuser = home_phone, userid = 123-456-7890 |
| 19:56.13 | [hC] | in that case you can register as whatever you want |
| 19:56.13 | PioneerVM2 | hc yes |
| 19:56.18 | [hC] | forget auth user |
| 19:56.23 | *** join/#asterisk jtexter3 (n=jtexter3@66.0.66.130) |
| 19:56.23 | [hC] | and leave use auth id empty |
| 19:56.27 | PioneerVM2 | ok |
| 19:56.29 | PioneerVM2 | thats what im doing now |
| 19:56.34 | [hC] | register with User ID as home_phone |
| 19:56.34 | PioneerVM2 | so where do i put the phone # in the pap2t box |
| 19:56.40 | PioneerVM2 | ok im doing that now |
| 19:56.42 | [hC] | you dont, in this case. |
| 19:56.42 | PioneerVM2 | azlready that way |
| 19:56.57 | [TK]D-Fender | [hC]: About 2.2.0 : |
| 19:57.04 | [hC] | In this case you must set callerid= in sip.conf for home_phone, or do it in a dial plan macro for when you dial, to Set ${CALLERID(num)} |
| 19:57.06 | PioneerVM2 | right now on pap2t: displayname = blank, username=home_phone auth_name=blank |
| 19:57.14 | PioneerVM2 | lol hc |
| 19:57.18 | PioneerVM2 | thats where the problem lies :) |
| 19:57.21 | [hC] | its safe enough to just put callerid=Some guy <cidnumber> |
| 19:57.24 | PioneerVM2 | im trying to get *67 to work |
| 19:57.29 | [hC] | dude |
| 19:57.33 | [hC] | one problem at at time here man. |
| 19:57.35 | PioneerVM2 | i already do that but am trying to get *67 to work thats how we got here |
| 19:57.37 | [hC] | what do you want.. |
| 19:57.39 | PioneerVM2 | its all the same problem |
| 19:57.49 | PioneerVM2 | ok lets back up u missed the whole first part of conversation |
| 19:57.55 | PioneerVM2 | i already have PAP2t working with caller id displaying |
| 19:58.00 | [hC] | Then you need to implement caller id setting in y our extensions.conf, not sip.conf |
| 19:58.05 | PioneerVM2 | i want to get *67 working, which seems to mean i have to put the phone # into pap2t box |
| 19:58.24 | PioneerVM2 | i need to know how to get asterisk to detect that *67 was typed |
| 19:58.26 | [hC] | dont set callerid= in sip.conf, but instead in your extensions.conf, in the macro you use to dial external numbers, put in (before the dial!!) |
| 19:58.29 | PioneerVM2 | so asterisk can set to blank |
| 19:58.38 | PioneerVM2 | yes i did that way too that was the original way i did it |
| 19:58.43 | *** join/#asterisk MitchM (n=MitchM@unaffiliated/MitchM) |
| 19:58.43 | PioneerVM2 | but how do i detect user typed *67 |
| 19:58.47 | [hC] | exten => whatever,1,Set(CALLERID(num)=somenumber) |
| 19:58.49 | PioneerVM2 | on pap2t, as pap2t intercepts it |
| 19:58.59 | PioneerVM2 | hc -- already done that, that was original way i did it |
| 19:59.09 | PioneerVM2 | i need to have asterisk set empty caller id when *67 typed |
| 19:59.16 | [hC] | Theres a page in the pap2 config menu with a huge list of * codes |
| 19:59.19 | [hC] | for local configs |
| 19:59.21 | PioneerVM2 | yes |
| 19:59.25 | [hC] | blank those all out |
| 19:59.30 | [hC] | and reboot it |
| 19:59.33 | [hC] | then the pap2 wont catch it |
| 19:59.38 | PioneerVM2 | so then my # comes thru as *67 1-555-1212 for example |
| 19:59.43 | [hC] | and you can implement everything in your asterisk dial plan |
| 19:59.45 | PioneerVM2 | i wanted to avoid that if possible |
| 19:59.53 | [hC] | exten => *671NXXNXXXXXX,1,Blah |
| 20:00.08 | PioneerVM2 | i was hoping to do it on pap2t side which would seem to be the "correct" way unless there is no other way |
| 20:00.11 | [hC] | exten => *671NXXNXXXXXX,1,Dial(Zap/g1/${EXTEN:3}) |
| 20:00.19 | [hC] | ok so |
| 20:00.24 | [hC] | you just told me you dont want the pap2 to catch it |
| 20:00.27 | PioneerVM2 | i figured that way out already i just thought there was a "correct" way |
| 20:00.33 | [T]ank | PioneerVM2: may not apply, but does the phone you use already have a dialplan for *67 that is messing you up? |
| 20:00.35 | PioneerVM2 | nono, i want to use pap2t to catch it, if possible |
| 20:00.36 | [hC] | the correct way is whatever way you do it. |
| 20:00.42 | [TK]D-Fender | [hC]: ring while busy (not jsut beep), timeouts from menus, seperate model specific firmwares, big MB improvements, call list reformatting, more EC options, improved no-answer/busy forwarding, support for backlit expansion module, and of course tons of fixes |
| 20:00.46 | PioneerVM2 | ank -- asterisk never sees *67 |
| 20:00.50 | PioneerVM2 | lol |
| 20:00.55 | jtexter3 | I heard once that Cisco uses Asterisk for some stuff. A google reveals lots about them using OpenSER, but not Asterisk. Anyone know if they do? |
| 20:01.02 | PioneerVM2 | hc -- if i were to have the pap2t box set the phone # how would it be setup |
| 20:01.04 | [T]ank | the linksys phones for example already have a dialplan for *67 so it would not work. |
| 20:01.33 | [TK]D-Fender | [hC]: You don't need to tell * about your CID block. You do it on the ATA and it'll give you another tone and you proceed from there |
| 20:01.46 | *** join/#asterisk NightMonkey (n=NightMon@pdpc/supporter/sustaining/NightMonkey) |
| 20:02.31 | Blackthorn | WARNING[18733]: app_dial.c:674 wait_for_answer: Unable to forward voice frame --- Anyone know what causes this message to be repeated ver quickly until server restart? |
| 20:03.06 | [hC] | [TK]D-Fender: yeah i mean you can do it any number of ways |
| 20:03.29 | b11d|bbl | . |
| 20:03.36 | [hC] | [TK]D-Fender: oooooo nice change list |
| 20:03.40 | [hC] | [TK]D-Fender: im gonn ahave to look into this more |
| 20:03.50 | vader-- | what would be the job title of someone who installs, admins, designs phone systems for a company? |
| 20:03.57 | vader-- | Telecommunications Technician? |
| 20:04.01 | PioneerVM2 | DF -- yes im trying to use the *67 detect on pap2t |
| 20:04.05 | PioneerVM2 | it gives stutter tone |
| 20:04.15 | PioneerVM2 | so its like a regular land line in that respect |
| 20:04.35 | PioneerVM2 | but doing CID the ways described (which i have already done) doesn't block CID |
| 20:04.44 | PioneerVM2 | its like pap2t accepts *67 and tosses it aside and does not inform * |
| 20:05.11 | [hC] | Ive always just done *67 in asterisk... I think all it does is send a blocked CID to asterisk, so if you override it in asterisk, its going to re-set your caller id |
| 20:05.26 | [hC] | im not sure how you would deal with that, so that you get caller id normally, but have it NOT set it if *67 was dialed |
| 20:05.59 | *** join/#asterisk collink (n=speedrac@206.166.236.197) |
| 20:06.06 | [hC] | cause i think all the ATA is doing is dialing like a regular call, but suppressing or mangling the CID to asterisk, or not, depending on if you dialed *67. Im not sure how you could differentiate that |
| 20:06.09 | *** part/#asterisk collink (n=speedrac@206.166.236.197) |
| 20:06.35 | datachomper | So I've got some sip phones that moved to another network, I turned on sip debug, and now they are all giving me "SIP/2.0 401 Unauthorized" ? No configuration has changed whatsoever. |
| 20:07.07 | *** join/#asterisk rodent|S (i=nobody@wsip-24-234-241-112.lv.lv.cox.net) |
| 20:07.11 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 20:07.16 | [TK]D-Fender | datachomper: show us your configs and the SIP debug info. |
| 20:07.24 | datachomper | Sure |
| 20:07.37 | [TK]D-Fender | PioneerVM2: hgell... do the same! |
| 20:08.31 | *** join/#asterisk lbow (n=lbow@41-195-77-82.access.uunet.co.za) |
| 20:08.39 | datachomper | Sip Debug : http://pastebin.com/d5afa2b34 |
| 20:09.55 | datachomper | Sample sip.conf : http://pastebin.com/m7349a675 |
| 20:10.20 | *** join/#asterisk jmesquita (n=jmesquit@c915230f.virtua.com.br) |
| 20:13.38 | *** join/#asterisk brianmohr (n=brianmoh@68.96.17.50) |
| 20:15.19 | *** join/#asterisk dr0ck (i=dr0ck@nat/digium/x-fd9c5c872a6c7e19) |
| 20:15.24 | PioneerVM2 | DF -- you were partially correct -- "Display Name" is the caller ID NAME field |
| 20:15.30 | PioneerVM2 | but does not seem to be number field |
| 20:15.37 | PioneerVM2 | and if you do *67 it seems to block it |
| 20:16.01 | PioneerVM2 | the reason i know this is that VoicePulse is screwed up and uses the NAME field for the CID number incorrectly |
| 20:16.18 | PioneerVM2 | when i used Teliax for outgoing it didnt work -- so i have to figure out how to set the CID number in the PAP2T |
| 20:17.51 | *** join/#asterisk |dennis| (n=dennis@200.32.233.84) |
| 20:18.28 | PioneerVM2 | hey i have a solutioin |
| 20:18.38 | PioneerVM2 | i could set CID NUMBER in the dialplan of asterisk |
| 20:18.45 | PioneerVM2 | and put a display name in the PAP2T |
| 20:18.54 | PioneerVM2 | if the display name comes through emty, i set the CID NUmber to empty |
| 20:24.09 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 20:24.52 | *** join/#asterisk jmacz (n=jmacz@200.71.41.92) |
| 20:26.48 | datachomper | Anybody? Non registering phones giving SIP/2.0 401 Unauthorized ? |
| 20:29.34 | *** part/#asterisk brianmohr (n=brianmoh@68.96.17.50) |
| 20:29.39 | *** join/#asterisk konqi_ (n=konqi@217.193.163.2) |
| 20:32.00 | *** join/#asterisk lbow (n=lbow@41-195-77-82.access.uunet.co.za) |
| 20:32.15 | konqi_ | anybody with experiene in getting a Digium TE420 Card to work with asterisk ... or maybe someone who could tell me where to get a good manual for what to do |
| 20:33.37 | Qwell[] | konqi_: There should be instructions somewhere on digium.com, but, we also offer free installation support... |
| 20:37.05 | *** join/#asterisk maxyz (n=maxy@190.55.86.61) |
| 20:37.54 | *** join/#asterisk krondorl (n=chatzill@acid.auricnet.ca) |
| 20:38.03 | krondorl | Hi All |
| 20:38.41 | krondorl | did the IF statement change from 1.2.11 to 1.4.4? |
| 20:39.02 | Qwell[] | krondorl: If it did, it would be mentioned in UPGRADE.txt |
| 20:39.09 | maxyz | Hi, is there a trivial way to have a poll of fxos? So if somebody wants to make a call through the pstn would get "any of the available ones". |
| 20:39.12 | Qwell[] | but, no, it probably did not |
| 20:39.20 | Qwell[] | maxyz: setup zap groups |
| 20:39.45 | maxyz | Qwell[]: even for sip fxos ? |
| 20:40.14 | krondorl | We upgraded our server from 1.2.11 to 1.4.4 but for some reason the if code stopped working. I'll see if it's something else maybe.. |
| 20:41.46 | brad_mssw | krondorl: are you using extensions.conf or extensions.ael ? |
| 20:41.56 | *** part/#asterisk [T]ank (n=ckwall@206.71.78.172) |
| 20:42.00 | krondorl | bugger. I just figured it out.. |
| 20:42.24 | Qwell[] | brad_mssw: coming from 1.2, I assume not AEL.. |
| 20:42.31 | krondorl | Another person updated the system to a new machine... but he didn't create the base db entries... |
| 20:42.35 | brad_mssw | Qwell[]: why? I used ael on 1.2 |
| 20:42.43 | Qwell[] | brad_mssw: Then you of all people would know :) |
| 20:42.54 | brad_mssw | Qwell[]: had to make a few changes to get it to work on 1.4 |
| 20:43.03 | Qwell[] | AEL2 is far superior to what was in 1.2 |
| 20:43.12 | brad_mssw | yep, I'd agree there |
| 20:43.19 | konqi_ | Qwell[]: thing is, if i edit the config-files mentioned in the manual, the config file says "Autogenerated by /usr/local/sbin/gnzaptelconf -- do not hand edit Zaptel Configuration File |
| 20:43.38 | brad_mssw | it's pretty much compatible ... the main thing that bit me was missing functions going from 1.2->1.4 |
| 20:44.36 | codefreeze | brad_mssw: you mean deprecated funcs? |
| 20:45.19 | brad_mssw | yep ... with the 1.2 AEL, you had to use some deprecated functions because it couldn't handle some things (like the DB stuff didn't work) |
| 20:49.06 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 20:51.29 | *** join/#asterisk jsmith (n=jsmith@000-143-916.area3.spcsdns.net) |
| 20:51.29 | *** mode/#asterisk [+o jsmith] by ChanServ |
| 20:53.01 | *** join/#asterisk MACscr (n=MACscr@adsl-75-23-94-138.dsl.peoril.sbcglobal.net) |
| 20:53.21 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 20:53.35 | MACscr | Is there a particular format the name should be in for directory use in voicemail.conf? |
| 20:53.55 | *** join/#asterisk smultron (n=lukas@cpe-72-179-47-78.austin.res.rr.com) |
| 20:54.05 | Qwell[] | MACscr: the same as the samples |
| 20:55.17 | [TK]D-Fender | MACscr, first last |
| 20:55.25 | PioneerVM2 | DF & Hc I have the solution |
| 20:55.27 | smultron | if i have an ISDN PRI line which only has 1 line on it, but still have 6 other lines handled by the phone company, do I need the individual phones to have 6-line capability? |
| 20:55.34 | MACscr | Hmm, ok, thanks |
| 20:55.42 | PioneerVM2 | "Display Name" only sets the CID Name from the PAP2T |
| 20:55.50 | smultron | need to have* |
| 20:56.06 | PioneerVM2 | as HC said, username is the caller ID # but if you cant make it the number for whatever reason, i have a workaround |
| 20:56.34 | PioneerVM2 | in PAP2T set "DIsplay Name" to something like "Joe Smith", in Asterisk handset SIP config, set "callerid=<###>" but don't set a name |
| 20:56.53 | PioneerVM2 | PAP2T will pass the display name as CID NAME and the callerid= line will set the CID NUMBER |
| 20:57.04 | jsmith | konqi_: I'd be happy to try to help you with your TE420 problem |
| 20:57.13 | PioneerVM2 | now when you do caller ID blocking from PAP2T, it will set Display Name (CID NAME) to "Anonymous" |
| 20:57.27 | konqi_ | jsmith: that would be most welcome ;-) |
| 20:57.32 | PioneerVM2 | so just put something in dialplan to check if "Anonymous" set as CID NAME to set CID NUM=<empty> |
| 20:58.35 | *** join/#asterisk deeperror (n=deeperro@69-215-202-195.ded.ameritech.net) |
| 20:59.28 | deeperror | need some assistance with sip packet handling |
| 21:01.01 | *** join/#asterisk guillote_GNU (n=bancaria@host185.190-136-203.telecom.net.ar) |
| 21:01.33 | deeperror | when i receive a BYE packet from my provider asterisk hangsup all parties including my zap channel this causes the handset to hear fast busy until it is hung up...any way to drop BYE packets or handle them differently? |
| 21:04.09 | [TK]D-Fender | PioneerVM2, what is this "for whatever reason" with regards to the username? What vaule is "not working" for you? |
| 21:05.14 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 21:06.42 | *** join/#asterisk imagitronics (n=tofarley@YRMfb-01p2-113.ppp11.odn.ad.jp) |
| 21:07.18 | elixer | ~101 |
| 21:07.19 | jbot | [101] Telephony 101, which is a good read if you're unfamiliar with traditional TDM telephony. You can download it at http://www.stromcarlson.com/docs/basics/NTtelephony101.pdf |
| 21:07.19 | Kurin- | Is there a way to keep asterisk listening for a given digit no matter what the context? I have an AGI script which spends most of its time in Dial(), but I need to be listening for '0' during that time |
| 21:07.29 | elixer | (sorry, needed that URL :)) |
| 21:07.48 | *** join/#asterisk kombi (n=kombi@213.160.14.18) |
| 21:08.16 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 21:08.16 | *** mode/#asterisk [+o anthm] by ChanServ |
| 21:09.09 | kombi | when connecting a thomson st2020 I get these weired capi errors on outgoing calls. In what way does a sip phone interact with the capi driver, I wonder? |
| 21:09.09 | *** part/#asterisk MitchM (n=MitchM@unaffiliated/MitchM) |
| 21:09.53 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 21:09.56 | MACscr | Is it possible for me to have extensions like 100, but still allow extensions to be dialed directly at the IVR and still use options like 1, 2, etc, during the menu? Do i just need to setup some type of timeout? |
| 21:10.08 | MACscr | Or is my idea just bad |
| 21:10.18 | *** join/#asterisk CVirus (n=GoD@82.201.222.191) |
| 21:10.42 | kombi | your idea might be good, I just don't get it.. |
| 21:10.58 | Kurin- | I have that set up |
| 21:11.34 | MACscr | Kurin- : how did you handle it? |
| 21:11.39 | MACscr | Er, set it up |
| 21:11.40 | Kurin- | I have exten => _[1-3]XX,1,goto(extensions); exten => _[1-9],1,goto(single option place) |
| 21:11.50 | Kurin- | where ; is a newline |
| 21:12.09 | [TK]D-Fender | MACscr, No, there is nothing special to do |
| 21:12.37 | [TK]D-Fender | MACscr, It will wait for your digit timeout if you enter a single digit that may be part of a longer matching exten |
| 21:14.19 | MACscr | Ah, my bad, i screwed up when testing that. Thanks for the confirmation |
| 21:15.01 | *** part/#asterisk lirakis (n=etamme@65.200.191.253) |
| 21:15.51 | *** join/#asterisk whywontitwork (n=d@196.211.34.2) |
| 21:15.51 | *** part/#asterisk Nix (n=Nix@opensuse/developer/nix) |
| 21:16.13 | PioneerVM2 | DF -- sorry was working in another window... regarding "for whatever reason" |
| 21:16.30 | PioneerVM2 | my handsets registration names are not "[number]" they are names |
| 21:16.44 | PioneerVM2 | i have two locations i want to have the same number but maybe different settings, passwords etc. |
| 21:16.58 | *** join/#asterisk dunder (n=grndslm@24-116-87-97.cpe.cableone.net) |
| 21:17.02 | PioneerVM2 | so i can't have the registration tag be the phone number because it would have to be different for two places |
| 21:17.18 | PioneerVM2 | so i call my handsets for example "[house1]" "[house2]" -- |
| 21:17.40 | PioneerVM2 | it seems asterisk uses the registration tag as a phone # but in this case its not valid so i have to set it manually in the callerid= setting |
| 21:17.45 | *** join/#asterisk blitzrage (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 21:17.45 | *** mode/#asterisk [+o blitzrage] by ChanServ |
| 21:17.50 | PioneerVM2 | i got it all to work the way described before , just got done with it |
| 21:18.27 | whywontitwork | hi there need some help with caller id using a patton isdn gateway i can receive incoming calls and if a route it to a xlite softphone(100) i can see the call originated from external number 2002 witch is the phone number i am phoning from(used on a pbx) how can i display this 2002 in asterisk using the noop application, any help welcome |
| 21:18.38 | PioneerVM2 | so the only way to set a CID Number in a PAP2T is through the "user id" -- display name is only the name field |
| 21:18.51 | [TK]D-Fender | PioneerVM2, you CAN have them the same, its jsut used for CID. thats what AUTH NAME is for. |
| 21:18.59 | [TK]D-Fender | PioneerVM2, AUTH NAME is what has to be unique. |
| 21:19.20 | Dovid | whywontitwork: if you want to see the CID in the cli simply do |
| 21:19.25 | [TK]D-Fender | PioneerVM2, and of course you need to say "yes" to "use auth id" |
| 21:19.27 | Dovid | noop(${EXTEN}) |
| 21:19.29 | PioneerVM2 | DF -- i tried to get that to work but i couldnt register |
| 21:19.44 | PioneerVM2 | i did try that but if i didnt set user ID and auth ID the same, i couldnt login |
| 21:19.51 | PioneerVM2 | unless there was an asterisk setting i was missing |
| 21:20.05 | PioneerVM2 | i tried user ID = phone #, auth id = home1 and "use auth id=yes" |
| 21:20.12 | PioneerVM2 | but it wouldnt login unless user id was home1 too |
| 21:20.23 | [TK]D-Fender | PioneerVM2, tell you what, I've got an SPA-2000 here I'll test with either late tonight, or tomorrow. |
| 21:20.34 | PioneerVM2 | ok -- i tried what you described |
| 21:20.40 | PioneerVM2 | but could not get login to work -- i had thought of that already |
| 21:20.52 | PioneerVM2 | unless there is an asterisk setting i have to change when i want to use auth_id to login |
| 21:21.15 | PioneerVM2 | so it seems id like to set "user id: ###" and "auth id: home1" and "use auth id: yes" but dont know how to get it to work |
| 21:21.21 | PioneerVM2 | but i have a temporary work around |
| 21:21.36 | PioneerVM2 | btw, in my local area, when we dial *67 on a land line it blocks caller id for one call only |
| 21:21.48 | PioneerVM2 | but in my pap2t the default is to turn off caller id all together until turned back on -- |
| 21:22.05 | PioneerVM2 | how does it work from what you are use to? |
| 21:22.40 | *** join/#asterisk blitzrage (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 21:22.40 | *** mode/#asterisk [+o blitzrage] by ChanServ |
| 21:22.45 | PioneerVM2 | DF yes i did that |
| 21:23.01 | whywontitwork | hi there need some help with caller id using a patton isdn gateway i can receive incoming calls and if a route it to a xlite softphone(100) i can see the call originated from external number 2002 witch is the phone number i am phoning from(used on a pbx) how can i display this 2002 in asterisk using the noop application, any help welcome |
| 21:23.11 | PioneerVM2 | when you say "unique" do you mean auth name must be the registration [xxx] name in the sip file |
| 21:23.12 | PioneerVM2 | or the phone # |
| 21:23.43 | PioneerVM2 | nevermind last comment was responding to scrollback msg by accident |
| 21:24.57 | kombi | has anyone had issues with dtmf and authenticate() such as some phone's dialtones not being read? |
| 21:25.34 | whywontitwork | yes kombi |
| 21:25.50 | kombi | how did you cope? |
| 21:25.59 | *** part/#asterisk deeperror (n=deeperro@69-215-202-195.ded.ameritech.net) |
| 21:26.05 | kombi | nice name, btw..;) |
| 21:26.19 | whywontitwork | had to change the settings in indications.conf took forever to get it right |
| 21:27.01 | kombi | never even looked into that file yet.. what sort of params are there to be tweaked? |
| 21:27.37 | Dovid | whywontitwork: Did you see my response to u ? |
| 21:28.17 | whywontitwork | record info stutter ect |
| 21:28.35 | whywontitwork | no dovid let me scroll back |
| 21:28.46 | Dovid | seems u needed help with something else |
| 21:28.47 | Dovid | nm |
| 21:29.28 | *** join/#asterisk brianmohr (n=brianmoh@68.96.17.50) |
| 21:30.03 | whywontitwork | Dovid: i've tried that i dont get any info, but what boggles me is that i do see the cli in the xlite softphone |
| 21:30.47 | *** part/#asterisk brianmohr (n=brianmoh@68.96.17.50) |
| 21:31.39 | kombi | whywontitwork: indications.conf seems for the tones * plays, not so much for the ones in reads at first glance.. |
| 21:32.55 | whywontitwork | what interface cards do you use? |
| 21:33.50 | kombi | none at all.. |
| 21:33.58 | kombi | box is all sip |
| 21:34.18 | *** join/#asterisk darkskiez (n=mhb@bb-87-81-62-203.ukonline.co.uk) |
| 21:34.27 | kombi | oh, sorry, not me right? |
| 21:34.57 | whywontitwork | no wyou lost me so you saying that you can not read dtmf digits from some softphones or hardphones? |
| 21:35.22 | kombi | from a hardphone called AEG Fame |
| 21:35.58 | whywontitwork | dtmfmode=rfc2833 |
| 21:36.20 | whywontitwork | add this to all users in sip.conf |
| 21:36.43 | kombi | thanks! |
| 21:36.59 | whywontitwork | let me know if it works |
| 21:37.12 | kombi | the said phone calls from outside though, it's not local |
| 21:40.52 | whywontitwork | explain some more i don't get the question |
| 21:42.57 | kombi | I have this extension with authenticate() that let's callers do stuff after they punched in the right code. There is this one caller that does not seem to be able to authenticate, even though she uses the right code and has a phone that has working dtmf otherwise |
| 21:47.04 | datachomper | <PROTECTED> |
| 21:47.10 | datachomper | wtf is that? |
| 21:47.32 | whywontitwork | is she dialing out or in? |
| 21:48.38 | tzafrir_laptop | datachomper, /bin/sh is not bash? |
| 21:48.44 | *** join/#asterisk ToyMan (n=Stuart@c-24-63-160-77.hsd1.ma.comcast.net) |
| 21:49.04 | datachomper | Sorry? |
| 21:49.33 | tzafrir_laptop | You use Ubuntu(?) and you /bin/sh is not bash? |
| 21:49.55 | datachomper | <PROTECTED> |
| 21:49.57 | tzafrir_laptop | And the author of safe_asterisk uses some bash-specific syntax in it (a.k.a bashism) |
| 21:50.03 | Nugget | ewwwww |
| 21:50.07 | Nugget | that's evil |
| 21:50.28 | *** join/#asterisk wunderkin (n=wunderki@ip72-223-86-126.ph.ph.cox.net) |
| 21:50.35 | Nugget | that's like when console apps emit ansi color codes even when your TERM=vt100 |
| 21:50.41 | Wonka | so he should use #!/bin/bash |
| 21:50.46 | tzafrir_laptop | edit safe_asterisk and replace /bin/sh in the header with /bin/bash |
| 21:51.08 | Wonka | would someone apply a LART please? |
| 21:52.16 | datachomper | Ah, thanks fellas |
| 21:52.19 | datachomper | You're lifesavers |
| 21:52.42 | MACscr | Can i setup IVR's so they just ignore invalid options? |
| 21:53.01 | Nugget | MACscr: the answer to every question that begins with "Can i setup..." is pretty much a "yes" |
| 21:53.21 | MACscr | Sry, the question should be. How can i do.... |
| 21:53.22 | MACscr | =P |
| 21:53.54 | Nugget | well, let's break it down. What happens now when a caller enters an invalid option? |
| 21:54.08 | MACscr | Says the user has entered an invalid option |
| 21:54.37 | Nugget | and what is making asterisk say that? |
| 21:55.14 | Nugget | hint: look for "exten => i," in your dialplan |
| 21:55.19 | MACscr | Guess i need to dig a little deeper. I didnt set that part up. Someone else did. |
| 21:55.31 | MACscr | Ok, what would i change it to then? |
| 21:55.38 | MACscr | Or would i just remove it? |
| 21:55.39 | Nugget | asterisk jumps to extension "i" when an invalid extension is entered |
| 21:55.41 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 21:55.49 | Nugget | change it to whatever you'd prefer asterisk do |
| 21:55.57 | MACscr | I dont want it to do anything =P |
| 21:56.04 | Nugget | technically that's not accurate. |
| 21:56.04 | MACscr | I want it to ignore it |
| 21:56.38 | MACscr | So the IVR would just keep playing if someone entered an incorrect option |
| 21:56.42 | Nugget | what does "ignore" mean in that context? Probably "give the user another chance to enter an extension" |
| 21:57.01 | MACscr | correct |
| 21:57.11 | [TK]D-Fender | MACscr, exten => i,1,goto(s,whateverthatlastprioritywas) |
| 21:57.24 | Nugget | so you'd want it to Goto() the prompt again. You could jump to the very end of that prompt so it doesn't play the sound files again if you wanted. |
| 21:58.01 | *** join/#asterisk kiscokid (n=Ron_Laut@208.106.35.66) |
| 21:58.41 | [TK]D-Fender | MACscr, may the last priority in your ivr exten a NoOp |
| 22:00.02 | Nugget | May the NoOp be with you. |
| 22:00.25 | Wonka | NoOp(${EXTEN}); |
| 22:00.43 | [TK]D-Fender | Wonka, Nah... why was cycles evaluating a variable? :) |
| 22:00.47 | [TK]D-Fender | NoOp() <- |
| 22:00.54 | [TK]D-Fender | waste* |
| 22:00.55 | Nugget | NoOp(telnet is eeeeeeevil) |
| 22:00.57 | MACscr | Hmm, if its a single recording for the entire menu listings (IVR), then having any type of action (going to the end) or saying invalid option, would completely negate what im trying to do |
| 22:01.20 | [TK]D-Fender | MACscr, who said anything about it SAYING anything!? |
| 22:01.22 | Nugget | MACscr: You're not fully understanding the advice |
| 22:01.32 | Wonka | noop - three sweet cycles of nothingness... |
| 22:01.32 | MACscr | Nugget : you are correct =P |
| 22:01.45 | [TK]D-Fender | MACscr, it jsut runs a NoOp tht the user can't tell the difference on, and then runs on to keep listing for a VALID option! |
| 22:01.47 | whywontitwork | hi there need some help with caller id using a patton isdn gateway i can receive incoming calls and if a route it to a xlite softphone(100) i can see the call originated from external number 2002 witch is the phone number i am phoning from(used on a pbx) how can i display this 2002 in asterisk using the noop application, any help welcome |
| 22:01.51 | Nugget | exten => ivr,1,Play("Hey! I'm an IVR") |
| 22:01.56 | whywontitwork | TK please help |
| 22:01.58 | Wonka | [TK]D-Fender: NoOp is very nice to output something to the console with verbosity high enough... |
| 22:01.58 | Nugget | exten => ivr,2,NoOp(I don't do anything) |
| 22:02.04 | Nugget | exten => i,1,Goto(ivr,2) |
| 22:02.28 | [TK]D-Fender | Nugget, you REALLY shouldn't run IVR's off anything but "s"..... |
| 22:02.28 | Wonka | ivr => { NoOp(hello world); HangUp(); } |
| 22:02.37 | Nugget | fair point. |
| 22:02.47 | [TK]D-Fender | Wonka, Not very... umm... "interactive" there.... |
| 22:02.54 | Wonka | *g* |
| 22:02.57 | [TK]D-Fender | ~stdextens |
| 22:02.57 | jbot | extra, extra, read all about it, stdextens is "s" Standard Extension : Where a call goes to when * does not know the destination of the call. Ex : Calls coming in on FXO ports (no DID), a call coming in from an ITSP that doesn't specify or where it was not set in the REGISTER line, or FXS port goes off-hook and "immediate=yes" in zapata.conf. "s" is also used to ... |
| 22:03.00 | *** join/#asterisk explidous (n=explidou@rrcs-24-173-134-222.se.biz.rr.com) |
| 22:03.14 | [TK]D-Fender | WTF! |
| 22:03.17 | [TK]D-Fender | truncated! |
| 22:03.30 | [TK]D-Fender | some moron wnet and modded it! |
| 22:03.34 | [TK]D-Fender | went* |
| 22:03.36 | [TK]D-Fender | aksdfhlasdjhaslfd |
| 22:04.04 | Nugget | ~stdtexans |
| 22:04.04 | jbot | extra, extra, read all about it, stdtexans is extra, extra, read all about it, stdtexans are bigger. Everything is bigger in Texas! |
| 22:04.14 | Strom_M | ~cohujibuggle |
| 22:04.14 | jbot | methinks cohujibuggle is gublgubbglggugglbuglgbugblgbgbgbgbglbglgbulgblugbgubgublgbglulllbgbb |
| 22:04.20 | *** join/#asterisk fujin (n=aj@unaffiliated/fujin) |
| 22:04.27 | J4k3 | everything is bigger in texas... except my * box. |
| 22:04.29 | [TK]D-Fender | oh well,time to head out... later all |
| 22:04.48 | J4k3 | speaking of, I just had to replace the CPU fan in my * box... the old one was only maybe 8 years old. |
| 22:04.50 | Wonka | ~lart jbot |
| 22:04.50 | jbot | chops wonka in half with a free AOL CD |
| 22:05.13 | [TK]D-Fender | Wonka, pwned |
| 22:06.24 | *** join/#asterisk |dennis| (n=dennis@200.32.236.18) |
| 22:06.57 | fujin | ~lart |
| 22:07.15 | J4k3 | ~gs |
| 22:07.16 | jbot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
| 22:07.33 | fujin | ~lart fujin |
| 22:07.34 | jbot | puts on a hockey mask and jumps out at fujin |
| 22:07.38 | blitzrage | FYI: The 2nd Edition of Asterisk: The Future of Telephony is now shipping from Amazon --> http://tinyurl.com/382dz2 |
| 22:07.50 | fujin | ebook avail? |
| 22:07.57 | fujin | probly don't need it these days, personally |
| 22:08.00 | fujin | did they cover AEL ;] |
| 22:08.09 | J4k3 | yeah, but can you buy it from anywhere credible? |
| 22:08.25 | J4k3 | amazon is the "you mighta paid for it but you prolly aren't going to get it" online retailer of 2007 |
| 22:08.29 | fujin | ;\ |
| 22:08.38 | fujin | can buy it from safari |
| 22:09.00 | blitzrage | PDF will be available at some point (i.e. as soon as O'Reilly generates it for me). AEL is not covered in this edition (it was scheduled, but I decided to write a chapter on Relational Database Integration and func_odbc) |
| 22:09.51 | fujin | i see |
| 22:12.20 | *** join/#asterisk dioedu (n=dioedu@c915230f.virtua.com.br) |
| 22:13.37 | dioedu | hello all, i have a problem here in my asterisk where i have agents in a queue and they don't receive any calls because asterisk tell that call limit is 1, but this agent is not in a call... |
| 22:15.21 | dioedu | this sip channel is not in a call and in the command "sip show inuse" i have 1 (one) in the column peer inuse |
| 22:15.51 | jmesquita | But 'show channels' command listing does not show any channels related to that SIP peer at all |
| 22:15.51 | dioedu | someone knows possible causes of this problem ? |
| 22:15.56 | codefreeze | fujin: don't dispair! you can eke by (hopefully) with the stuff on the voip-info wiki about AEL2 |
| 22:16.22 | fujin | lol |
| 22:16.28 | fujin | not running ael2 here, yet |
| 22:16.29 | fujin | still ael |
| 22:16.37 | fujin | wrote my entire callcentre in it. it's beautiful |
| 22:17.01 | jmesquita | fujin: How many agents are you running? |
| 22:17.23 | jmesquita | fujin: And what version of asterisk are you running if you don't mind me asking |
| 22:18.16 | fujin | we have 20 agents, dynamic & hotdesking |
| 22:18.28 | fujin | Asterisk 1.4.10.1 built by root @ asterisk01 on a i686 running Linux on 2007-08-20 04:11:38 UTC |
| 22:18.57 | fujin | not really agents though, I'm using aqm/rqm for queue member management |
| 22:19.02 | jmesquita | fujin: Not using app_queue tho |
| 22:19.14 | fujin | yeah, using app-queue |
| 22:19.18 | fujin | just not chan_agent :) |
| 22:19.25 | fujin | chan_local / chan_sip |
| 22:19.32 | jmesquita | fujin: aqm/rqm? |
| 22:19.41 | fujin | yeah |
| 22:19.46 | fujin | addqueuemember/removequeuemember |
| 22:19.51 | codefreeze | fujin: if you are running 1.4, you are using AEL2 (it's just called AEL in asterisk) |
| 22:19.55 | dioedu | fujin, in our case too |
| 22:19.59 | fujin | codefreeze: oh, awesome :) |
| 22:20.05 | fujin | I wondered why the new AEl2 syntax was working. |
| 22:20.05 | fujin | :] |
| 22:20.11 | fujin | I love ael2 ;] |
| 22:20.15 | dioedu | fujin, AQM/RQM... |
| 22:20.49 | fujin | yeah |
| 22:20.49 | fujin | aqm/rqm are awesome |
| 22:21.00 | dioedu | and my problem is... i need to RQM every time that i receive CHANUNAVAIL in a Dial |
| 22:21.17 | *** join/#asterisk sergey (n=sergey@gw4-130.iks.ru) |
| 22:21.28 | dioedu | because when I receive this status, the softphone is not open... right ? |
| 22:22.03 | blitzrage | AEL will make it into the 3rd edition! :) |
| 22:22.04 | dioedu | i do that... i don't know if this is the better way to this... |
| 22:22.08 | blitzrage | I'm gonna make sure of it :) |
| 22:22.27 | codefreeze | blitzrage: Yay for AEL! |
| 22:22.37 | blitzrage | YAY! :) |
| 22:22.43 | fujin | dioedu: use a LOCAL channel |
| 22:22.49 | blitzrage | one of these days I'll get ambitious again and try it |
| 22:22.51 | fujin | that way you can dial the local channel and handle what happens when it is dial |
| 22:23.05 | dioedu | fujin, but i'm using local channel... |
| 22:23.27 | fujin | yeah, so handle your ${DIALSTATUS} when you dial your agent |
| 22:23.32 | dioedu | yeah |
| 22:23.33 | fujin | that'sa ll you need to do.. pretty easy really |
| 22:23.38 | dioedu | exactly |
| 22:23.51 | dioedu | but... |
| 22:24.05 | fujin | here's my local channel |
| 22:24.07 | fujin | and call delivery |
| 22:24.07 | fujin | http://rafb.net/p/ajnVgM13.html |
| 22:25.23 | dioedu | thanks... but don't you treat the CHANUNAVAIL ? |
| 22:25.31 | fujin | nope |
| 22:25.59 | dioedu | when i receive chanunavail, i RQM my agent... because the sip client is not avaible |
| 22:26.06 | dioedu | available |
| 22:26.11 | fujin | yeah, so add another case: |
| 22:26.18 | fujin | case CHANUNAVAIL: |
| 22:26.34 | fujin | &queue-remove(${MACRO_EXTEN}); |
| 22:26.35 | fujin | or whatever. |
| 22:26.36 | fujin | ; |
| 22:26.36 | fujin | ] |
| 22:27.14 | *** join/#asterisk Aeudian (n=Aeudian@204.52.131.22) |
| 22:29.06 | Aeudian | evening everyone how goes it. |
| 22:32.01 | *** join/#asterisk `Sean (i=Un1x@CPE000c256d416d-CM000a73a1559d.cpe.net.cable.rogers.com) |
| 22:34.26 | jmesquita | fujin: If you dont handle CHANUNAVAIL how do you know if your agent did not crash earlier? |
| 22:34.55 | jmesquita | fujin: I mean, you still rely on queue timeout b4 you pass it to another agent? |
| 22:36.31 | *** join/#asterisk Tond (n=t@CPE0014bf30c190-CM00194747ae5e.cpe.net.cable.rogers.com) |
| 22:36.51 | fujin | jmesquita: I check state of the agent before delivering |
| 22:36.55 | Tond | Hi is there a way to set a differnt MOH for different sip users? |
| 22:36.57 | fujin | and have a really low tiemout/retry |
| 22:37.04 | fujin | Tond: music=blah |
| 22:37.11 | Tond | tnx |
| 22:37.35 | dioedu | fujin, yeah, your agents are softphone ? |
| 22:37.43 | fujin | na hardphone |
| 22:37.45 | Tond | fujin> it's not working |
| 22:38.03 | dioedu | ok... |
| 22:38.14 | Tond | fujin> music=mymusic and my music is set in the musiconhold.conf file |
| 22:38.33 | fujin | uh, I'm sure you can put music in sip.conf |
| 22:38.43 | *** join/#asterisk Benny_132 (n=benmarti@144.131.132.107) |
| 22:38.54 | whywontitwork | i am routing a call from isdn gateway to sip phone, i can see the cli on the xlite softphone but can not see the information in asterisk console please help |
| 22:39.07 | fujin | Noop(${CALLERID(all)}; |
| 22:39.30 | whywontitwork | i need to see the information to route thr call to the right extension |
| 22:39.40 | Tond | fujin> you can but it will effect all peers |
| 22:40.30 | Tond | i need to set a different moh for a sip user. One way is to do it through setmoh in extensions.conf but I want to avoid typing that in for all the extensions this user will be dialing |
| 22:40.55 | fujin | just make a macro |
| 22:41.08 | *** join/#asterisk CVirus (n=GoD@82.201.222.191) |
| 22:41.49 | Benny_132 | hi all hi guys when i run'sip show chanels' it comes up with something like 188 active SIP channels i have about 16 phone extensions is this something to be worried about |
| 22:42.05 | Tond | fujin> several ways come to my mind... anything in particular u recommend? |
| 22:43.28 | *** join/#asterisk cirgal (n=robert@pool-71-102-137-33.snloca.dsl-w.verizon.net) |
| 22:43.40 | *** join/#asterisk anthm (n=anthm@adsl-76-229-233-50.dsl.milwwi.sbcglobal.net) |
| 22:43.40 | *** mode/#asterisk [+o anthm] by ChanServ |
| 22:44.49 | whywontitwork | yes benny |
| 22:44.55 | fujin | create a macro |
| 22:44.58 | fujin | set musiconhold |
| 22:45.00 | fujin | call person |
| 22:49.19 | whywontitwork | there must be a way to see the cli if asterisk passes it to the xlite softphone above sample does not work:fujin |
| 22:52.28 | *** join/#asterisk kFuQ (n=somedude@c-67-185-112-20.hsd1.wa.comcast.net) |
| 22:52.57 | *** join/#asterisk purpoze (n=purposeZ@adsl-66-126-122-10.dsl.lsan03.pacbell.net) |
| 22:53.24 | purpoze | hey guys, i couldn't find a VOIP channel, I am having some difficulties getting CounterPath eyeBeam software working with my BroadVoice account |
| 22:53.40 | purpoze | I get the following error message: 15:47:46.5 Call (l:'2138050567' r:'sip:18183579709@147.135.0.6') - Call being terminated. Reasons: "Forbidden", (code: 403) |
| 22:54.30 | *** join/#asterisk jm|laptop (n=jm|home@zen.jamiem.com) |
| 23:01.00 | *** join/#asterisk mindCrime (n=chatzill@cpe-065-190-188-124.nc.res.rr.com) |
| 23:01.34 | Benny_132 | whywontitwork: is there any way to colse all those active sip channels in the cli other than a restart |
| 23:03.47 | JunK-Y | Benny_132: soft hangup chan |
| 23:05.14 | Benny_132 | 10.19.21.118 00085D1813 4ae9c59728c 00102/00000 unkn No Tx: ACK |
| 23:05.14 | Benny_132 | 10.19.21.200 00085D1814 547fa6fb4bd 00102/00000 unkn No Tx: ACK |
| 23:05.14 | Benny_132 | 10.19.21.118 00085D1813 1bd4100c122 00102/00000 unkn No Tx: ACK |
| 23:05.15 | Benny_132 | 10.19.21.158 00085D03AC 35144519598 00102/00000 unkn No Tx: ACK |
| 23:05.15 | Benny_132 | 10.19.21.118 00085D1813 1bab469f06b 00102/00000 unkn No Tx: ACK |
| 23:05.15 | Benny_132 | 10.19.21.200 00085D1814 795cd807788 00102/00000 unkn No Tx: ACK |
| 23:05.17 | Benny_132 | 10.19.21.118 00085D1813 63f8896d2ad 00102/00000 unkn No Tx: ACK |
| 23:05.19 | Benny_132 | 10.19.21.200 00085D1814 7a833a6b098 00102/00000 unkn No Tx: ACK |
| 23:05.47 | Benny_132 | thats an example of what happens when i go sip show channels how would i end one of them with soft hangup "channel" |
| 23:06.37 | fujin | sip show channels |
| 23:06.39 | fujin | show channesl |
| 23:06.43 | fujin | soft hangup <channel> |
| 23:07.38 | Benny_132 | yeah but where is the channel number |
| 23:07.45 | bkruse | module remove chan_sip.so |
| 23:07.48 | bkruse | lol |
| 23:08.00 | bkruse | actually, it should maintain state for the calls up, i think, could be wrong. |
| 23:08.07 | bkruse | Benny_132: please dont post > 3 lines without |
| 23:08.08 | bkruse | ~pb |
| 23:08.09 | jbot | A Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 23:10.39 | *** join/#asterisk Peaceful (n=peaceful@70.98.162.62) |
| 23:11.16 | Peaceful | Can anyone recommend some good non-cisco POE switches? |
| 23:12.42 | *** join/#asterisk asdx (n=diego@adsl-145-105.click.com.py) |
| 23:14.00 | *** join/#asterisk famicon (i=pastry@c51447ddc.cable.wanadoo.nl) |
| 23:16.38 | Peaceful | ...to use with Polycom phones? |
| 23:17.35 | wothinn | I've always liked the ProCurve switches, myself, but depends how big-iron you're talking here. |
| 23:17.45 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 23:18.05 | Trevor_b | Peaceful: I have been looking into the Linksys and NetGears, dont have one in the shop to run through the ringer yet though. |
| 23:19.21 | fujin | Linksys POE are alright |
| 23:19.27 | fujin | I roll cisco 3560G's here, though |
| 23:19.39 | fujin | they're nice, let me do all my TOS/DiffServ stuff, CDP, blabla |
| 23:20.15 | *** part/#asterisk jm|laptop (n=jm|home@zen.jamiem.com) |
| 23:22.04 | Peaceful | I've got 58 employees and growing, so I don't need super-huge and expensive, but 48-port stackables would still be nice. |
| 23:23.35 | JT | and in reality linksys and netgears are toys |
| 23:24.10 | Peaceful | So ProCurves are HP... |
| 23:24.31 | Peaceful | what series should I be looking at for my size? |
| 23:29.25 | Peaceful | Ok, should I be looking at some HP reseller for these ProCurves? |
| 23:34.44 | *** join/#asterisk mvardanega (n=root@189.26.18.38.adsl.gvt.net.br) |
| 23:35.20 | fujin | Peaceful: you can generally only buy hp gear through resellers |
| 23:37.02 | Peaceful | fujin, k, cool. Looks like the regulars apply.. Thanks guys |
| 23:37.15 | *** join/#asterisk ToyMan (n=Stuart@c-24-63-160-77.hsd1.ma.comcast.net) |
| 23:41.55 | whywontitwork | here is a strange one i am using asterisk purely for voicemail on a philips is3000 pbx, i do a divert on no answer from a users analog handset to asterisk giving the caller an option to leave a message for the user i need to get the callerid of the user not the caller to link him to the right mailbox how does one go about it? |
| 23:41.57 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 23:42.27 | whywontitwork | anyone,anyone? |
| 23:42.41 | whywontitwork | does that make sence? |
| 23:43.02 | fujin | have to pass it to asterisk |
| 23:44.40 | JT | whywontitwork: you mean "sense"? |
| 23:47.02 | whywontitwork | well does it? |
| 23:49.07 | Strom_M | it's one incredibly long run-on sentence |
| 23:49.23 | JT | and impatience usually makes others ignore the problem |
| 23:55.26 | *** join/#asterisk asdx (n=diego@adsl-145-105.click.com.py) |
| 23:58.26 | *** join/#asterisk MCelo (n=mcelo@201.50.81.83) |
| 23:59.59 | *** join/#asterisk Krurst (n=me@eth244.wa.adsl.internode.on.net) |