| 00:46.02 | p3nguin | hlieberman: Why in the world would you want all those options? |
| 00:46.31 | hlieberman | p3nguin, No clue! I'm just following the example. Or, was, theoretically. Haven't actually implemented any of it yet. |
| 00:46.40 | p3nguin | You want both the callER and the callEE to bne able to transfer the call and toggle recording? |
| 00:47.06 | p3nguin | Usually that's not the case. |
| 00:47.58 | hlieberman | No, that's definitely not the case. |
| 00:48.41 | hlieberman | I think I really just want M(screen) |
| 00:50.25 | p3nguin | I've never used it, but I'm about to take a look at Dial()'s options to see if we can get away from FollowMe(). |
| 00:51.46 | hlieberman | Thanks! |
| 00:58.21 | p3nguin | Well... |
| 00:58.43 | p3nguin | Privacy mode works similar to FollowMe(), but has more options. |
| 00:59.43 | p3nguin | So instead of FollowMe(${medical}), Dial(SIP/flowroute/${medical},,p) |
| 01:00.04 | p3nguin | actually... |
| 01:00.08 | p3nguin | Dial(SIP/flowroute/1${medical},,p) |
| 01:00.11 | p3nguin | Forgot the 1 |
| 01:00.28 | p3nguin | We're doing 10-digit numbers everywhere, so need the 1. |
| 01:03.15 | WIMPy | What does p do? |
| 01:03.35 | p3nguin | It enables privacy mode. |
| 01:03.56 | WIMPy | Yes, I see that. But what does that do? |
| 01:04.00 | WIMPy | Or mean? |
| 01:05.50 | p3nguin | I call you, you have privacy mode enabled... it asks me to record my name, then dials your phone. It tells you that you have a call from someone who identified himself as "Mr. X." To accept the call, press 1. To reject and send to voicemail, press 2. To send the caller to the torture menu, press 3. To play a polite message telling the caller not to call anymore, press 4. |
| 01:05.55 | p3nguin | I'm not sure what else. |
| 01:06.12 | p3nguin | or even how accurately I repeated the options. |
| 01:07.00 | p3nguin | I pressed 3 just to see what happened, and it set the DIALSTATUS to TORTURE. |
| 01:07.02 | WIMPy | I see. A bit like getting a reverse charged call. |
| 01:07.45 | p3nguin | It's a little heavy for hlieberman's application. FollowMe() is better than privacy mode, but a macro could still create the same effect. |
| 01:09.12 | WIMPy | That reminds me that I need to get a recording of that "The person you have called does not whish to receive calls at the moment.". |
| 01:23.39 | *** join/#asterisk boogerfly (~rainkid@cpe-66-65-54-130.nyc.res.rr.com) |
| 01:23.57 | p3nguin | The only thing I don't like about using the macro like that is that the caller gets dead air while the callee is listening to the option to accept/reject the call. |
| 01:24.07 | p3nguin | I'd rather have ringing or moh. |
| 01:24.51 | hlieberman | I would prefer that as well. |
| 01:25.09 | hlieberman | Perhaps there is an extension to Dial(,,p)? |
| 01:25.55 | p3nguin | Dial's p option doesn't seem to have any configuration for it. You get all or nothing. |
| 01:26.29 | p3nguin | With the macro, once the macro starts, there is no way to play ringing or music to the caller that I can see. |
| 01:26.41 | p3nguin | I'm going to play with that idea for a second. |
| 01:28.33 | p3nguin | Solved. |
| 01:28.55 | p3nguin | Use Dial() option r for ringing or m for MusicOnHold. |
| 01:29.18 | p3nguin | Dial(SIP/flowroute/1${medical},,rM(screen)) |
| 01:29.22 | p3nguin | Dial(SIP/flowroute/1${medical},,mM(screen)) |
| 01:30.03 | hlieberman | Let me give that a shot. Hang on. |
| 01:30.10 | p3nguin | If you want moh, you'll want to configure it first, and you'll want to play some type of a message to the caller to let them know why they are hearing music. |
| 01:30.22 | p3nguin | Otherwise, it just goes right to music as it calls the on-call. |
| 01:30.23 | hlieberman | I think ringing is better. |
| 01:30.43 | hlieberman | moh is too... corporate. |
| 01:31.26 | WIMPy | F***. Why doesn't MixMonitor announcements? |
| 01:33.09 | WIMPy | For "incoming" calls it starts recording befor the call is bridged. But it doesn't seem to work in the other direction. |
| 01:35.42 | *** join/#asterisk adolfomaltez (~taro@190.62.220.209) |
| 01:37.50 | p3nguin | hmm? |
| 01:38.25 | WIMPy | And when calling out the ITSP I don't even hear anything I could record. |
| 01:39.50 | p3nguin | I'd have to see the extensions to understand. |
| 01:40.30 | *** part/#asterisk adolfomaltez (~taro@190.62.220.209) |
| 01:41.18 | WIMPy | Calling to PSTN. |
| 01:42.09 | hlieberman | p3nguin, OK, here's one twist. I want to play a soundfile to the original caller in the event that the call is rejected or not answered. How would you swing that? Just append it after the Dial()? But then, won't it get played if the call is disconnected after the callee hangs up and the caller is still on the line? And if I put it in the macro, it'll end up getting played to the callee, not the caller. |
| 01:42.15 | p3nguin | It doesn't matter where you're calling, I'd have to see the extensions to understand what you're confused about. |
| 01:43.10 | WIMPy | exten => _010*5.,1,Set(file=/var/spool/asterisk/monitor/${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)}-${CALLERID(num)}-${EXTEN:5}.wav) |
| 01:43.12 | WIMPy | exten => _010*5.,2,MixMonitor(${file},,chmod 740 ${file}) |
| 01:43.15 | WIMPy | exten => _010*50.,3,Dial(sip/1und1/${EXTEN:5},${dto}) |
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| 01:44.17 | WIMPy | Well, that was the 2nd try using SIP. |
| 01:45.08 | *** join/#asterisk adolfomaltez (~taro@190.62.220.209) |
| 01:45.18 | *** join/#asterisk rainkid (~rainkid@cpe-66-65-54-130.nyc.res.rr.com) |
| 01:46.11 | rainkid | I migrated to a new asterisk box. Some users voicemail greeting is working. It isn't for others, even though /var/spool/asterisk/voicemail/default/$EXTEN/unavail.wav exists. |
| 01:46.24 | rainkid | Instead, vm-intro.gsm is played. Any idea how to troubleshoot? |
| 01:46.37 | WIMPy | Did you change formats? |
| 01:46.42 | rainkid | Nope |
| 01:46.51 | p3nguin | If you're dialing a number that starts with 010*50, then the MixMonitor should begin before the Dial(). You'll even record the ringing sound as the callee is contacted. |
| 01:47.14 | WIMPy | That's what I thought, but that's not happening. |
| 01:48.21 | p3nguin | hlieberman: If the callee hangs up, it should immediately go to extension h. Put the Playback after the Dial. If the callee does not accept, dialplan should continue to the Playback and the caller would hear that message. |
| 01:52.24 | hlieberman | Well, that's odd. If the callee hangs up, it's just leaving the caller on the line. |
| 01:52.57 | WIMPy | This is rather strange. I tried using an IAX trunk, but no recording, either. |
| 01:53.37 | WIMPy | OTOH, calls from PSTN to local phones start recording before the call is answered. |
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| 01:54.02 | hlieberman | But the rejection message is working. :) |
| 02:00.42 | hlieberman | Also, the r function to Dial() doesn't seem to be working, oddly enough. I wonder if the macro is suppressing it somehow. |
| 02:04.42 | p3nguin | So if the callee presses 1, does the caller get connected to the callee? |
| 02:04.52 | hlieberman | Yes. And if they press 2, they get rejected. |
| 02:04.54 | *** join/#asterisk HSaus (3ce2264d@gateway/web/freenode/ip.60.226.38.77) |
| 02:04.58 | HSaus | hello |
| 02:05.03 | hlieberman | But if they press 1, talk, then hang up, the caller just kind of stays there. |
| 02:05.08 | hlieberman | Connected to no one. |
| 02:05.17 | p3nguin | I see. |
| 02:06.09 | HSaus | small problem that is driving me crazy. I have an asterisk server running behind a NIC, I have forwarded port 5060 on the router to the server's IP but I still can't register a client outside the lan. The apache webserver is working from the same server with port 80 forwarded, so port forwarding should be working. |
| 02:06.15 | *** join/#asterisk darkbasic (~quassel@host37-245-static.119-2-b.business.telecomitalia.it) |
| 02:06.47 | WIMPy | Did you forward UDP? |
| 02:06.54 | p3nguin | SIP is UDP. |
| 02:07.11 | WIMPy | Does anyone have an Idea how I could record an announcement? |
| 02:07.26 | p3nguin | Use Record(). |
| 02:08.06 | WIMPy | How do I get it to the record application? The call is never answered. |
| 02:09.35 | p3nguin | Normally, you'd Answer(), then Record(). I'm not seeing what the problem is. |
| 02:10.04 | WIMPy | The other end will never answer. |
| 02:10.10 | p3nguin | Other end of what? |
| 02:10.22 | WIMPy | The announcement I want to steal. |
| 02:10.29 | p3nguin | OH! |
| 02:10.33 | p3nguin | That's completely different. |
| 02:10.47 | p3nguin | For that, I'd use MixMonitor(). |
| 02:11.01 | p3nguin | But they are going to have to play the message for you to be able to record it. |
| 02:11.07 | WIMPy | Yes, but that won't record anything, either. |
| 02:11.31 | p3nguin | If they play a sound while mixmonitor is running, it will record the sound. |
| 02:11.37 | p3nguin | If it doesn't, something is broken. |
| 02:11.54 | WIMPy | I even tried to use the conference function of the phone or to transfer it, but the phone won't allow me to do that on inactive calls :-( |
| 02:12.03 | *** join/#asterisk adolfomaltez (~taro@190.62.220.209) |
| 02:12.07 | WIMPy | Well, then something is broken. |
| 02:12.21 | WIMPy | It won't even create a file. |
| 02:12.24 | p3nguin | Why would the call be "inactive" if you're making a call? |
| 02:12.24 | *** join/#asterisk darkbasic (~quassel@host37-245-static.119-2-b.business.telecomitalia.it) |
| 02:12.48 | WIMPy | Because it is only an announcement. The call is never answered. |
| 02:13.33 | p3nguin | You're saying they are playing early media, without ever changing to state Up. |
| 02:13.42 | WIMPy | yes |
| 02:13.56 | p3nguin | What happens if you put an Answer() before your Dial()? |
| 02:14.04 | p3nguin | MixMonitor(file.wav,a) |
| 02:14.06 | p3nguin | Answer() |
| 02:14.14 | p3nguin | Dial(SIP/peer/exten) |
| 02:14.25 | WIMPy | Hmm. Answer my side? Let me try. |
| 02:14.28 | p3nguin | Then it's Up before it reaches the dial. |
| 02:14.43 | p3nguin | As soon as the channel goes Up, MixMonitor() begins. |
| 02:15.27 | WIMPy | Well, as I stated earlier, usually it starts before. That's why I'm a little confused about what's going on here. |
| 02:16.18 | *** join/#asterisk darkbasic (~quassel@host37-245-static.119-2-b.business.telecomitalia.it) |
| 02:17.36 | WIMPy | Ha, that did the trick. |
| 02:17.59 | p3nguin | A trick it was. |
| 02:18.48 | WIMPy | But why do I ned to answer there while I don't have to on incomming calls? |
| 02:19.03 | WIMPy | That seems rather strange. |
| 02:19.08 | p3nguin | When you Dial(), the state is Ringing. |
| 02:19.24 | p3nguin | When you pick up the phone, the channel state goes to Up. |
| 02:19.43 | WIMPy | Yes, but that's exactely what happens when someone calls me. But in that case MixMonitor will even record a fake ringing. |
| 02:19.56 | p3nguin | If you use Playback() before a Dial(), there is an implicit Answer() in the Playback app. |
| 02:20.08 | WIMPy | I don't answer. |
| 02:20.36 | WIMPy | Neither explicitly nor implicitly. |
| 02:20.42 | p3nguin | Using the r option probably provides an implicit Answer, which changes the state to Up. r provides a ringing sound file. |
| 02:21.20 | WIMPy | I don't use r, either. |
| 02:21.21 | p3nguin | Call yourself using an r in the Dial, and check core show channels. |
| 02:21.36 | p3nguin | You said fake ringing... how do you get fake ringing? |
| 02:22.21 | WIMPy | I only hear it in what MixMonitor saves. |
| 02:22.47 | p3nguin | You'd have to check the channel state. If it's Up, that explains it. |
| 02:23.48 | *** join/#asterisk coppice (~chatzilla@m121-203-236-16.smartone-vodafone.com) |
| 02:23.58 | WIMPy | Nope, none is up. |
| 02:24.06 | hlieberman | Hm. |
| 02:24.15 | HSaus | WIMPy, sorry for the delay... had to log into the server and the router and check the port forwarded. I forwarded both TCP and UDP. |
| 02:24.37 | WIMPy | Hmm. but then I didnt get fake ringing now, either. |
| 02:25.45 | WIMPy | I have had fake ringing, however. But it might only have happend on IAX. Need to watch out for that. |
| 02:26.27 | p3nguin | How is this fake ringing generated? |
| 02:26.38 | WIMPy | I have NFI. |
| 02:26.46 | p3nguin | Who is producing it? |
| 02:27.06 | WIMPy | NFI, eiter. |
| 02:27.12 | p3nguin | What type of device is providing it? |
| 02:27.17 | p3nguin | Where is it coming from? |
| 02:27.21 | p3nguin | Give me SOMETHING. |
| 02:27.46 | WIMPy | I wouldn't even know if it is coming form a device, a channel, or Dial(). |
| 02:27.51 | p3nguin | Ringing doesn't just happen, unless you've got something like tinnitus or whatever. |
| 02:28.01 | WIMPy | The only thin I know is that I heard it. |
| 02:28.25 | WIMPy | Well, actually I do, but not in that frequency range :-) |
| 02:35.18 | WIMPy | Ok, it must have been Dial(). I used to have option r at some time, even if that didn't make a difference to the caller. |
| 02:35.40 | WIMPy | Except when the calld phone was unreacable. |
| 02:35.42 | WIMPy | +h |
| 02:35.51 | p3nguin | I would think the r option brings the channel Up so it can play sound. |
| 02:36.14 | p3nguin | And once it's up, MixMonitor records. |
| 02:36.18 | WIMPy | It certainly doesn't answer. |
| 02:36.42 | p3nguin | When the ringing sound is present from the r option, the state is not up? |
| 02:37.09 | WIMPy | The callers have not been connected in that state. |
| 02:37.24 | p3nguin | Up does not imply bridged. |
| 02:37.45 | hlieberman | Any ideas why Dial(,,r) isn't causing it to ring? |
| 02:38.02 | p3nguin | No. It works for me. |
| 02:38.18 | p3nguin | I tried Dial(M(screen)) |
| 02:38.21 | WIMPy | The only thing I know for certain is that callers got the telcos ring back sample, not mine. |
| 02:38.24 | p3nguin | then Dial(rM(screen)) |
| 02:38.28 | p3nguin | then Dial(mM(screen)) |
| 02:38.43 | p3nguin | Without r or m, there was silence while I waited for a response. |
| 02:42.35 | SeRi | p3nguin, you ever used gsm gateway? |
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| 02:44.44 | hlieberman | Interesting. |
| 02:44.50 | hlieberman | p3nguin, When I remove the timeout, it rings. |
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| 03:20.53 | HSaus | I can register and call from my laptop connected to the server on the same lan. I have TCP/UDP port 5060 and 10000 to 20000 forwarded on the router to the server, so I should be able to connect from outside my lan. |
| 03:21.42 | HSaus | are there any other ports that asterisk uses for voip clients like bria to register? or just 5060? |
| 03:22.42 | WIMPy | Not unless you're using TLS. |
| 03:22.54 | WIMPy | Are you allowed to reply to the internet? |
| 03:22.57 | p3nguin | SIP is UDP 5060, and RTP is whatever UDP range specified in rtp.conf. |
| 03:23.03 | WIMPy | Is the client reachable? Is it behind NAT? |
| 03:23.10 | p3nguin | IAX2, if you're using that, is UDP 4569. |
| 03:23.20 | p3nguin | ~sipnat |
| 03:23.20 | infobot | [~sipnat] Quick guide on configuring Asterisk + SIP behind NAT: http://www.aocomputing.net/?p=3 otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions . Typically, you MUST configure these settings with appropriate values in the [general] section of sip.conf: nat, canreinvite, externhost or externip, and localnet. |
| 03:24.18 | HSaus | I had the client connect to a STUN just to check |
| 03:24.23 | HSaus | didn't help |
| 03:25.10 | WIMPy | Using STUN usually makes things worse when the other End is Asterisk. |
| 03:26.00 | HSaus | sip.conf has nat=yes and externip= and localnet=192.168.0.18 |
| 03:26.19 | p3nguin | localnet=192.168.0.18 isn't quite right. |
| 03:26.24 | p3nguin | You're almost there, though. |
| 03:26.40 | p3nguin | localnet=192.168.0.0/255.255.255.0 could work. |
| 03:27.11 | WIMPy | Far too much typing. |
| 03:27.47 | HSaus | tried 192.168.0.0/255.255.255.0 didn't work either |
| 03:40.21 | p3nguin | That part works. |
| 03:40.28 | p3nguin | It's obviously something else that isn't. |
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| 04:37.45 | psilikon | Anyone wanna help me find out why an AGI script (* 1.8.6, pyst python library) is giving unexpected results when processing DTMF input from the user? http://pastie.org/2669094 |
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| 06:14.16 | schmidts | good morning |
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| 06:17.39 | ChannelZ | well it's monday morning, so not so good |
| 06:18.13 | irroot | morning people |
| 06:24.50 | ollii | g'mornin |
| 06:25.39 | wdoekes2 | morning |
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| 07:00.11 | adeel | anyone happen to know if there's a ser/openser/kamailio freenode chan? |
| 07:01.12 | wdoekes2 | #kamailio and #opensips |
| 07:01.27 | wdoekes2 | where the latter is a lot more active |
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| 07:01.50 | adeel | ah, i must have misspelt it earlier |
| 07:01.57 | adeel | and thanks |
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| 07:48.07 | Freeaqingme | hi! |
| 07:50.50 | ollii | hi |
| 07:51.10 | Freeaqingme | I'm looking for some good hardphones that support ipv6, any ideas? |
| 07:52.01 | ollii | can someone suggest me a working cti solution for asterisk 1.4 / 1.8 on linux (Ubuntu 10.04+) ? It should look like this one: http://phonesuite.de/de/images/screenshot/PhoneSuite_snapshot.gif |
| 07:52.35 | ollii | or maybe just a thunderbird plugin |
| 08:03.28 | *** join/#asterisk oej (~olle@ns.webway.se) |
| 08:09.51 | SunTsu | Freeaqingme: that depends on what you call good and what's available to you - and what your budget is. Some people like cisco phones, some like snom, others Aastra or polycoms |
| 08:10.50 | Freeaqingme | SunTsu, we're currently running some Cisco SPA504G's (and SPA942's). That's the kind of phone I'm after, but then I want it to support ipv6 |
| 08:11.54 | SunTsu | Freeaqingme: That normally is a firmware issue, did you look at cisco's firmware releases? |
| 08:12.50 | Freeaqingme | it says that it doesn't support it, but might with future fw releases |
| 08:14.36 | *** join/#asterisk oej (~olle@2001:470:1f15:d79:789a:8f3:16c8:f700) |
| 08:14.45 | coppice | chances are they won't support IPv6, and they hope to use this at some point to force you to buy new phones |
| 08:14.48 | *** part/#asterisk didnot (~didnot@unaffiliated/didnot) |
| 08:15.22 | SunTsu | Freeaqingme: snom supports IPV6 AFAIK and has phones similar to the SPA504G |
| 08:16.12 | *** join/#asterisk Faustov (~fst@gentoo/user/faustov) |
| 08:16.12 | Freeaqingme | SunTsu, tnx. gonna check it out |
| 08:17.49 | *** join/#asterisk Azrael808 (~peter@212.161.9.162) |
| 08:23.46 | *** join/#asterisk jacc0 (~jacc0@D522448D.static.ziggozakelijk.nl) |
| 08:30.08 | coppice | its amazing how little consumer grade kit, even for new models, supports IPv6 |
| 08:35.04 | Freeaqingme | well, voip hardphones aren't really consumergrade imho |
| 08:36.13 | *** join/#asterisk nix8n82-phone (~AndChat@24.143.28.16) |
| 08:36.58 | coppice | most of them go into businesses, but huge numbers are made in consumer like ways - especially things like the Linksys phones |
| 08:37.00 | *** join/#asterisk cerberus_za (~coert@196.25.231.147) |
| 08:38.51 | Freeaqingme | coppice, true. But what then would you characterize as a t ypical business phone? |
| 08:39.37 | coppice | a cisco vs a linksys maybe |
| 08:39.49 | *** join/#asterisk af_ (~getsmart@78.134.22.161) |
| 08:40.00 | Freeaqingme | coppice, even linksys phones are being rebranded to cisco? |
| 08:40.26 | coppice | the cisco is a more business phone, where you get screwed for software licences |
| 08:40.28 | coppice | the linksys is a more consumer phone, where you get screwed by new things not being available |
| 08:40.47 | Freeaqingme | heh |
| 08:41.08 | coppice | rebranding Linksys as Cisco is soooooo last month. This month they seem to be trying to separate them again, so they can sell off Linksys |
| 08:41.25 | Freeaqingme | ya serious? |
| 08:43.05 | coppice | there has been a lot of talk of a selloff. a year ago most of the Linksys stuff said "Linksys by Cisco". now I see lots of stuff that doesn't mention CIsco. |
| 08:44.19 | Freeaqingme | hmmz |
| 08:44.23 | Freeaqingme | hadnt seen that yet |
| 08:44.27 | Freeaqingme | interesting move |
| 08:46.23 | Madkiss | Is there some easy way available to test whether a ConfBridge conference exists and how many people are in there? |
| 08:50.47 | Madkiss | basically I am looking for MeetMeCount for ConfBridge ;) |
| 09:03.03 | kaldemar | Madkiss: there is no dialplan application or function to do that. |
| 09:06.01 | Madkiss | kaldemar: ahum. okay. so when using ConfBrige, I do not have a way to test whether a conference room is empty or not? |
| 09:06.31 | kaldemar | there are cli commands for it. |
| 09:07.54 | kaldemar | unless you consider it to be too ugly, you could issue cli commands from the dialplan through the SHELL function. |
| 09:08.56 | kaldemar | might be that the commands are available in asterisk 10 only. |
| 09:09.39 | Madkiss | ?! |
| 09:09.40 | Madkiss | wtf, |
| 09:10.38 | Madkiss | so confbridge as such exists |
| 09:10.58 | kaldemar | which would leave the option of parsing core show channels... |
| 09:11.05 | Madkiss | but in asterisk 1.8 there is no way whatsoever to find out *anything* within asterisk about existing confrooms? |
| 09:12.10 | kaldemar | if you have an extension per conference, you could use EXTENSION_STATE. |
| 09:12.57 | Madkiss | that sounds like a plan |
| 09:13.18 | Madkiss | What I am doing right now is doing MeetMeCount($foo) on a randomly chosen extension |
| 09:13.30 | Madkiss | and if the result is <1, I assume that the conference doesn't exist |
| 09:18.36 | ollii | <PROTECTED> |
| 09:23.33 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
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| 10:02.28 | *** part/#asterisk Jckf (~jckf@2001:470:27:155::2) |
| 10:24.45 | zamba | i'm trying to set up dahdi against mx-one.. where can i find a simple to follow howto for that? |
| 10:24.57 | zamba | first of all, how do i know that the card is supported and discovered? |
| 10:28.53 | kaldemar | zamba: isn't mx-one SIP-based? |
| 10:29.03 | zamba | kaldemar: not this version, i'm afraid not |
| 10:30.28 | kaldemar | zamba: what interface are we talking about here then? |
| 10:31.35 | kaldemar | zamba: and what card are you trying to use? |
| 10:38.25 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:4c81:be6:d57a:5e73) |
| 10:39.17 | zamba | 03:02.0 Network controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface |
| 10:39.20 | zamba | could it be that card? |
| 10:40.23 | zamba | it's using the netjet kernel module |
| 10:42.55 | kaldemar | yes it could. what is the card model? |
| 10:43.14 | zamba | Subsystem: Device 795e:0001 |
| 10:43.23 | zamba | dunno if that gives you anything? |
| 10:43.28 | zamba | i'm configuring this server remotely |
| 10:44.25 | kaldemar | if that's a PCI ID, it doesn't show up in http://pciids.sourceforge.net/pci.ids |
| 10:44.48 | *** join/#asterisk Azrael808 (~peter@212.161.9.162) |
| 10:45.11 | zamba | but it's a card that i know works.. it was working in the previous installation of asterisk running the same machine |
| 10:45.19 | zamba | it was running 1.4 of asterisk, and i just upgraded to 1.8 |
| 10:45.25 | kaldemar | what card is it? |
| 10:45.37 | zamba | i don't know any more than lspci -v tells me |
| 10:45.52 | zamba | http://pastebin.com/vmurxZeH |
| 10:46.00 | kaldemar | how the hell are you going to use it for something if you have no idea what the card is? |
| 10:59.12 | *** join/#asterisk zamba (marius@flage.org) |
| 10:59.46 | *** join/#asterisk puzzled (~patrick@puzzled.xs4all.nl) |
| 11:01.11 | zamba | network hiccup.. |
| 11:02.08 | zamba | kaldemar: it's a digium tdm400 |
| 11:02.35 | *** join/#asterisk cerberus_za (~coert@196.25.231.147) |
| 11:13.11 | zamba | and please don't tell me it's too old and that i have to upgrade the card.. i know it's an old card, but it used to work, and this is just temp. while we're waiting for the sip trunk |
| 11:13.59 | kaldemar | interesting. it's not a digium card according to the PCI ID. |
| 11:14.37 | kaldemar | and especially not an analog card, which TDM400 is, because it has an ISDN circuit on it. |
| 11:14.58 | *** join/#asterisk joat (~joat@ip70-160-216-251.hr.hr.cox.net) |
| 11:15.09 | zamba | but what's the general procedure of setting up an isdn card like this? |
| 11:15.13 | zamba | do i need kernel modules? |
| 11:15.47 | zamba | linux has autoprobed it with the netjet module, but i guess that's wrong |
| 11:16.51 | kaldemar | i'll take that last comment back. don't use the hisax and netjet modules, use one from DAHDI. |
| 11:17.06 | zamba | yeah, so how do i "use one from DAHDI"? |
| 11:17.44 | kaldemar | a DAHDI module is a driver for the card and provides the interface that asterisk uses through chan_dahdi. |
| 11:17.52 | kaldemar | have you installed DAHDI? |
| 11:18.02 | zamba | i have installed asterisk-dahdi |
| 11:18.13 | kaldemar | which is? |
| 11:18.13 | zamba | the apt package |
| 11:18.22 | zamba | i'll try reinstalling it |
| 11:18.38 | kaldemar | what's in the package? |
| 11:18.40 | zamba | maybe it didn't do anything with the card since linux had already auto detected with the netjet modules |
| 11:18.41 | *** join/#asterisk kareena (~k@unaffiliated/kareena) |
| 11:19.13 | kaldemar | or maybe it doesn't even have a module for the card, but the asterisk module. |
| 11:19.48 | *** join/#asterisk atan (atan@unaffiliated/atan) |
| 11:20.32 | kaldemar | zamba: the asterisk-dahdi package only includes asterisk stuff. you need dahdi-linux and dahdi-tools. |
| 11:21.13 | zamba | dahdi-linux and dahdi-tools are automatically installed with asterisk-dahdi |
| 11:21.38 | zamba | oh, not dahdi-tools, though, but dahdi-dkms and dahdi-linux |
| 11:22.17 | Madkiss | zamba: is that Debian? |
| 11:22.31 | zamba | Madkiss: ubuntu/debian, yeah |
| 11:22.38 | Madkiss | ah |
| 11:23.10 | kaldemar | zamba: stop the hisax and netjet modules from loading, then configure DAHDI in /etc/dahdi/system.conf and /etc/dahdi/modules and load the DAHDI modules. |
| 11:23.33 | kaldemar | then you need to configure asterisk in chan_dahdi.conf to use the card. |
| 11:25.05 | zamba | kaldemar: aight |
| 11:25.34 | zamba | no /etc/dahdi/system.conf though |
| 11:29.22 | zamba | ah, there it appeared.. had to purge the package and install it again |
| 11:30.52 | kaldemar | if you have no knowledge of system.conf, try if dahdi_genconf knows how to configure the card. |
| 11:34.10 | *** join/#asterisk endemic (~endemic@2001:49f0:400e::5) |
| 11:36.15 | zamba | # dahdi_genconf |
| 11:36.15 | zamba | Empty configuration -- no spans |
| 11:36.23 | zamba | i have the previous /etc/dahdi/system.conf file here |
| 11:36.30 | zamba | and the previous /etc/dahdi/modules |
| 11:36.41 | zamba | so i've copied over the relevant settings.. how do i then try to reload the dahdi modules? |
| 11:37.30 | zamba | it should be wct1xxp |
| 11:37.37 | zamba | E100P i guess |
| 11:40.12 | kaldemar | the module for TDM400P is wctdm. |
| 11:40.51 | zamba | yeah, but i guess it's not TDM400P |
| 11:41.02 | zamba | because this is an ISDN card |
| 11:41.15 | kaldemar | E100P is an E1 card. |
| 11:41.30 | zamba | isn't that ISDN? |
| 11:41.42 | zamba | (i'm new to the telephony world ;)) |
| 11:42.05 | kaldemar | ISDN can run on an E1. they are not synonyms. |
| 11:42.30 | zamba | ok |
| 11:42.44 | kaldemar | you should have told what the card is used for an hour ago. |
| 11:42.45 | zamba | but i have now edited /etc/dahdi/system.conf.. how do i apply these settings? |
| 11:42.59 | zamba | as i said, i new to this.. :) |
| 11:43.05 | coppice | if it appears to be a netjet its probably a TDM400 |
| 11:43.06 | kaldemar | probably with /etc/init.d/dahdi |
| 11:43.29 | kaldemar | or "dahdi_cfg -vv" if you want to do it manually. |
| 11:43.31 | zamba | coppice: TDM400 is an analog card? |
| 11:43.37 | kaldemar | zamba: yes |
| 11:43.40 | zamba | FATAL: Module dahdi not found. dahdi: error dahdi_dummy: error dahdi_transcode: error wct1xxp: error |
| 11:43.43 | zamba | Error: missing /dev/dahdi! |
| 11:43.45 | coppice | so? |
| 11:43.47 | zamba | coppice: then it's not TDM400 |
| 11:43.58 | coppice | wrong |
| 11:44.01 | zamba | just can't be |
| 11:44.30 | coppice | oh, yes it can |
| 11:44.52 | zamba | coppice: ok? |
| 11:45.16 | zamba | i feel this is a bit counter-productive |
| 11:45.32 | zamba | if i have 32 channels to the mx-one.. how can that be an analog card? |
| 11:46.01 | *** join/#asterisk mraky_1 (~mraky@ip-94-112-113-65.net.upcbroadband.cz) |
| 11:46.07 | zamba | with "have", i of course mean "had", since that's the problem right now |
| 11:47.13 | coppice | counter intuitive maybe, but quite productive |
| 11:47.28 | zamba | coppice: is it possible to see 32 channels with an analog card? |
| 11:48.13 | coppice | the biggest actual analogue card I know of with that chip has 24 channels, but I guess 32 is possible. its on a 2Mbps backplane |
| 11:50.53 | kaldemar | it seems the card is a TE110P. the module for it is wcte11xp. |
| 11:51.12 | kaldemar | that information is based on the PCI ID, 795e:0001. |
| 11:51.24 | zamba | http://pastebin.com/Yz7rMYHp |
| 11:51.52 | zamba | one step further |
| 11:51.57 | zamba | got the dahdi modules installed |
| 11:52.04 | zamba | <PROTECTED> |
| 11:53.28 | zamba | oh.. and there wct1xxp suddenly came as well |
| 11:53.29 | zamba | strange |
| 11:59.12 | zamba | hm |
| 11:59.20 | zamba | chan_dahdi.so fails now |
| 11:59.50 | zamba | http://pastebin.com/QpZ2bLfB |
| 11:59.55 | zamba | probably wrong driver then.. ? |
| 12:00.47 | *** join/#asterisk Naikrovek (~mbluth@unaffiliated/naikrovek) |
| 12:00.57 | kaldemar | zamba: read what i said last. |
| 12:02.17 | zamba | oooh |
| 12:02.23 | zamba | looks like it loaded successfully |
| 12:02.40 | zamba | http://pastebin.com/GpM0zX7d |
| 12:02.49 | zamba | where do i go from here? |
| 12:02.54 | zamba | tone zone 0 is wrong, though |
| 12:03.12 | zamba | aight |
| 12:03.18 | zamba | changed in /etc/dahdi/system.conf |
| 12:04.18 | zamba | where do i go from here? |
| 12:04.23 | kaldemar | pastebin your system.conf and chan_dahdi.conf if they don't work. |
| 12:05.02 | zamba | http://pastebin.com/XKPSFtvU |
| 12:05.04 | zamba | that's system.conf |
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| 12:05.36 | kaldemar | zamba: all of it? |
| 12:09.17 | zamba | kaldemar: yup |
| 12:09.56 | kaldemar | fill it up with configuration... |
| 12:10.30 | zamba | like what? :) |
| 12:11.16 | kaldemar | what kind of an interface is the trunk on the mx-one? |
| 12:14.03 | kaldemar | http://ofps.oreilly.com/titles/9780596517342/asterisk-OutsideConn.html#OutsideConnectivity_id291062 |
| 12:14.39 | zamba | added this: http://pastebin.com/QDBW8EE7 |
| 12:14.44 | zamba | that was from the old configuration |
| 12:17.16 | kaldemar | span=1,1,0,ccs,hdb3,crc4 should probably be span=1,0,0,ccs,hdb3,crc4. |
| 12:18.21 | zamba | tried that |
| 12:19.35 | kaldemar | what does not work? |
| 12:20.03 | zamba | same problem as earlier.. the chan_dahdi.so won't load |
| 12:20.28 | zamba | http://pastebin.com/3wQ9yzBi |
| 12:20.43 | zamba | i've even copied the /etc/asterisk/chan_dahdi.conf file from the link you pasted |
| 12:21.23 | kaldemar | it's probably wrong. |
| 12:21.26 | zamba | DAHDI_SPANCONFIG failed on span 1: Invalid argument (22) |
| 12:21.41 | zamba | get that when i run dahdi_cfg -vvv |
| 12:21.52 | kaldemar | are the dahdi modules loaded? |
| 12:21.55 | zamba | yup |
| 12:23.23 | zamba | aight |
| 12:23.30 | zamba | i removed dahdi_dummy and then i got a different output |
| 12:24.06 | zamba | aaah |
| 12:24.10 | zamba | let's try now |
| 12:24.31 | zamba | the dahdi module loads! |
| 12:24.32 | *** join/#asterisk cerberus_za (~coert@196.25.231.147) |
| 12:25.50 | zamba | oh yeah |
| 12:26.11 | zamba | got it working! :) |
| 12:26.19 | zamba | removing the dahdi_dummy was the trick |
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| 12:30.03 | stix_ | Guys. Do you know of any decent hard-phones which support pptp-vpn? |
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| 12:38.16 | defswork | hi - have a system on 1.6.2.20 and occasionally we get a call in a queue that no longer exists. it locks the queue up. I cannot find a way to clear the call - channel request hangup says channel unknown |
| 12:38.46 | defswork | I can hangup calls stuck behind it |
| 12:51.17 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 12:54.44 | zamba | now i got this: [Oct 10 14:53:36] WARNING[12909]: chan_dahdi.c:4847 dahdi_enable_ec: Unable to enable echo cancellation on channel 1 (No such device) |
| 12:55.34 | cusco | hi |
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| 13:00.54 | *** mode/#asterisk [+o russellb] by ChanServ |
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| 13:15.43 | Katty | stretches |
| 13:15.56 | Katty | naps on energy drink |
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| 13:34.53 | chuckf | Katty: did you enjoy the celebratory shenanigans? |
| 13:39.19 | Katty | yessss, except for having to sit in the back seat 2 hours to, and 2 hours back, from stl |
| 13:39.25 | Katty | there were childrens. |
| 13:39.38 | Katty | and they were hyper on the way up, and tired and whiny on the way back |
| 13:40.11 | chuckf | ugh, I dislike those types of trips |
| 13:43.42 | chuckf | but as long as the rest of the time was good, what's 4 hours of your day? |
| 13:45.17 | ollii | anyone knows a good cti client for linux which is compatible to asterisk 1.4 ? |
| 13:52.08 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
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| 14:00.45 | Katty | chuckf: ehnn |
| 14:00.59 | Katty | chuckf: well the two people with kids ended up making me feel like the third wheel |
| 14:01.08 | Katty | chuckf: day could have been worse tho |
| 14:01.17 | chuckf | sorry about that |
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| 14:03.03 | Katty | ah it's fine. |
| 14:03.47 | chuckf | so are you working today or taking the holiday? |
| 14:04.04 | Katty | working. |
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| 14:08.24 | defswork | can anyone suggest anything about my problem - * 1.6.2.20 - running queues. Got a call stuck at top of queue that doesn't exist anymore and it locks the queue up. Cannot hang it up - says channel unknown |
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| 14:09.50 | *** join/#asterisk raden (~J@66-168-15-100.dhcp.stpt.wi.charter.com) |
| 14:09.55 | raden | hugs Katty |
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| 14:18.40 | Katty | hi raden :> |
| 14:18.42 | Katty | hugs raden |
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| 14:21.44 | schmidts | hi Katty |
| 14:22.34 | raden | Katty, how u been ? |
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| 14:23.39 | puzzled | WIMPy: you around? |
| 14:23.56 | Katty | hai schmidts |
| 14:23.58 | puzzled | hi Katty |
| 14:23.59 | Katty | schmidts: do we hug? |
| 14:24.10 | Katty | raden: good, good, just had my 27th birthday on friday ^_^ |
| 14:24.16 | Katty | hugs puzzled |
| 14:24.21 | puzzled | belated congrats :) |
| 14:24.22 | Katty | puzzled: how're you dear? |
| 14:24.25 | Katty | ty |
| 14:24.29 | puzzled | getting old and grey |
| 14:24.48 | zamba | oh my.. one girl and the whole channel goes OT :) |
| 14:25.04 | dym | (safe bets its a dude) |
| 14:25.05 | _Corey_ | Katty: Congrats on 27... :) |
| 14:25.26 | n3hxs | dym, not a chance. |
| 14:25.37 | dym | pretty certain :) |
| 14:26.05 | puzzled | like it matters... |
| 14:26.10 | zamba | somehow i don't think the guys in here'd care.. as long as they believe it's a girl themselves, they're fine :) |
| 14:26.26 | FlashDeluxe | hi! i got a question, i am using * 1.8 with dahdi and after every call i want to execute an agi-script and that works so far. The script should write something into a mysql database and sometimes it doesn`t do it and i get the error "utils.c:1164 ast_carefulwrite: write() returned error: Broken pipe" The load is quite high on the server, is it possible that the server is to slow? Or could it be another cause? |
| 14:26.45 | zamba | FlashDeluxe: can't you see we're busy chatting up girls? :) |
| 14:26.56 | puzzled | FlashDeluxe: is your db on the same box? |
| 14:27.01 | Katty | zamba is totally right. |
| 14:27.08 | Katty | zamba i'm just a male in disguise |
| 14:27.09 | Katty | please carry on |
| 14:27.17 | zamba | that wasn't me, that was dym :) |
| 14:27.18 | Katty | _Corey_: aww ty |
| 14:27.33 | Katty | oh so it was |
| 14:28.29 | *** join/#asterisk Praise- (~Fat@unaffiliated/praise) |
| 14:28.39 | puzzled | FlashDeluxe: if the mysql db is on the same box and the box is quite loaded it might very well be too slow to respond in time |
| 14:28.42 | Katty | puzzled: you're not old dear. |
| 14:30.31 | zamba | when using my dahdi channels i get the following warning in the asterisk console: [Oct 10 16:30:10] WARNING[12909]: chan_dahdi.c:4847 dahdi_enable_ec: Unable to enable echo cancellation on channel 1 (No such device) |
| 14:30.39 | zamba | something i've missed in /etc/dahdi/system.conf? |
| 14:31.34 | *** join/#asterisk cerberus_za (~coert@196.25.231.147) |
| 14:32.02 | kaldemar | zamba: are the modules loaded and the channels configured? |
| 14:32.06 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 14:32.06 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:32.22 | zamba | kaldemar: that's a very loaded question.. and to that i only have the answer, "i don't know" :) |
| 14:32.25 | zamba | but i believe they are |
| 14:32.30 | zamba | i'm able to dial on them |
| 14:32.43 | kaldemar | zamba: what does cat /proc/dahdi/1 say? |
| 14:33.05 | zamba | http://pastebin.com/JZ6wnH2X |
| 14:33.06 | puzzled | FlashDeluxe: see more info about the error here: http://www.mail-archive.com/asterisk-users@lists.digium.com/msg231722.html |
| 14:33.10 | kaldemar | i don't believe you're able to make calls over channels that don't exist. |
| 14:33.22 | zamba | kaldemar: dahdi_cfg -vv shows stuff like this: |
| 14:33.27 | zamba | Setting echocan for channel 28 to none |
| 14:33.35 | zamba | so i'm guessing it has to be related to this somehow? |
| 14:33.36 | FlashDeluxe | puzzled: ok thanks :) |
| 14:33.37 | kaldemar | zamba: you haven't configured chan_dahdi.conf at all, have you? |
| 14:33.41 | zamba | kaldemar: sure i have |
| 14:34.01 | kaldemar | zamba: show it. |
| 14:34.31 | zamba | hm.. let's se.. just have to do some grep magic to get rid of all the commented lines |
| 14:34.38 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 14:34.46 | zamba | grep -v ^\; chan_dahdi.conf |
| 14:34.54 | zamba | what if i want to get rid of all lines starting with whitespace? |
| 14:36.13 | zamba | oh, nevermind. |
| 14:36.18 | zamba | http://pastebin.com/b5NTx3zD |
| 14:36.21 | zamba | there you go |
| 14:39.37 | *** join/#asterisk mirelab (~mirelab@212.200.146.253) |
| 14:40.26 | kaldemar | zamba: have you configured echocancellers in system.conf? |
| 14:40.28 | mirelab | hello, does anyone have experience with SIPp testing tool? :) |
| 14:40.57 | zamba | kaldemar: nope, i don't know what that is :) |
| 14:41.13 | zamba | where is the man page for system.conf? i can't find any documentation for it |
| 14:42.33 | *** join/#asterisk chazzam (~chazz@173-24-236-90.client.mchsi.com) |
| 14:44.29 | kaldemar | zamba: http://svn.digium.com/svn/dahdi/tools/trunk/system.conf.sample |
| 14:44.55 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 14:48.22 | zamba | kaldemar: so this has to be defined in /etc/dahdi/system.conf? |
| 14:48.23 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-mwddxwouprjkqzzv) |
| 14:48.37 | zamba | kaldemar: not in chan_dahdi.conf? |
| 14:48.55 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-mwddxwouprjkqzzv) |
| 14:49.31 | zamba | and which echo canceller should i be using? i have absolutely no idea about this :) |
| 14:50.16 | puzzled | oslec or the hardware EC cards from digium or sangoma |
| 14:50.35 | kaldemar | the echocanceller parameter is a system.conf one, obviously. try mg2. |
| 14:51.23 | zamba | so two lines, one for channels 1 to 15 and one for 17 to 31? |
| 14:51.55 | zamba | yup, that worked |
| 14:51.58 | *** part/#asterisk mirelab (~mirelab@212.200.146.253) |
| 14:51.58 | zamba | let's check the asterisk side |
| 14:52.22 | *** join/#asterisk irroot (~irroot@197.109.154.120) |
| 14:52.36 | zamba | cool, cool.. though i'm not entirely sure what i just did :) |
| 14:52.41 | puzzled | hi irroot |
| 14:52.54 | irroot | hi there folks |
| 14:53.27 | irroot | puzzled hi there how goes |
| 14:54.07 | puzzled | irroot: good once I get this mISDN figured out. iirc I saw your nick in a few chan_misdn commits. do you know which version of mISDN is required for chan_misdn in 1.8 or 10? |
| 14:54.36 | irroot | puzzled what kernel you using ? |
| 14:54.44 | puzzled | zamba: if yu have no idea what you are doing, I recommend you read the book |
| 14:54.57 | kaldemar | zamba: you just enabled an echo canceller for asterisk on the DAHDI side. |
| 14:55.22 | irroot | chan_misdn is a lib based system and you need to use upto 1.1.9.2 |
| 14:55.31 | zamba | puzzled: i'm in the process of doing so :) |
| 14:55.43 | puzzled | irroot: it's CentOS 6 with 6.1 updates (the CR repo) running kernel-2.6.32-131.12.1.el6.x86_64 |
| 14:55.44 | irroot | but it only works on kernels < 2.6.2X [4 if i recall] |
| 14:56.02 | puzzled | zamba: good :) |
| 14:56.18 | puzzled | irroot: I see. that 2x does not sound very good with the kernel I'm running |
| 14:56.34 | *** part/#asterisk sekil (~sekil@78.24.104.73) |
| 14:56.41 | irroot | i have a patched version on my SVN server |
| 14:56.59 | puzzled | irroot: I tried compiling Digium-mISDN-1.1.3 (aka 1_1_9.1) but it blew up a second after starting |
| 14:57.04 | irroot | http://pbx.distrotech.co.za/svn/mISDN and then mISDNuser |
| 14:57.14 | puzzled | cool I'll give that one a try. thanks! |
| 14:57.27 | irroot | i use it with linux 3.0 |
| 14:57.47 | irroot | you will need to change the astrerisk configure.ac file |
| 14:57.59 | irroot | to use alternate header file |
| 14:58.32 | puzzled | ok, I'll give it a try. thanks |
| 15:00.08 | irroot | http://pbx.smartdns.co.za/viewsvn/asterisk/distrotech/1.8/1.8.9.0/configure.ac?view=patch&r1=548&r2=547&pathrev=548 <- run ./bootstrap.sh after patching to build configure to detect it |
| 15:00.10 | *** join/#asterisk cerberus_za (~coert@196.25.231.147) |
| 15:00.27 | defswork | got this call stuck in queue problem again - cannot hang it up - is not a known channel |
| 15:00.39 | defswork | anyone know if there is a workaround ? |
| 15:00.58 | puzzled | irroot: is you tree based on Digium's mISDN tree now at 1.1.3 which is based on 1_1_9.1 with a ton of COLP and other fixes? See http://svnview.digium.com/svn/thirdparty/ |
| 15:01.38 | irroot | defswork what version and was a pickup involved ? |
| 15:01.47 | defswork | 1.6.2.20 |
| 15:01.50 | defswork | no pickup |
| 15:01.57 | irroot | puzzled its based on the original mISDN |
| 15:01.58 | defswork | it's on a SIP trunk |
| 15:02.26 | irroot | defswork ok cant really help on 1.6 its been a while and i did not use it much |
| 15:02.32 | irroot | sorry |
| 15:02.45 | defswork | irroot, :| we migrated down from 1.8 as we had deadlocks |
| 15:03.34 | irroot | what ver 1.8 you use with 1.8.8 we have no more deadlocks i know i killed a few and the other developers killed plenty |
| 15:03.58 | defswork | it was 1.8.5 at the time |
| 15:04.29 | irroot | ok .6/7 have improved |
| 15:05.43 | defswork | do you know if specically in queues ? I could lock it repeatably |
| 15:07.08 | irroot | there was a nasty deadlock in queues thats been fixed |
| 15:07.19 | irroot | its in the latest RC |
| 15:08.09 | puzzled | irroot: I get the same error compiling your mISDN as described in https://issues.asterisk.org/jira/browse/ASTERISK-18676 |
| 15:09.15 | irroot | you need the fritz ? |
| 15:09.19 | puzzled | nope |
| 15:09.39 | irroot | how you building it ? |
| 15:09.46 | defswork | anyway to kick the queue to lose it's calls ? |
| 15:09.56 | puzzled | irroot: make in the root of the src tree |
| 15:09.58 | defswork | so the one thats not a call anymore goes ? |
| 15:10.34 | snocetti | Hello, I am using asterisk 1.8.6 and SIP stop working after 3-4 days |
| 15:10.41 | snocetti | somebody has same problem? |
| 15:10.42 | defswork | atm I'm having to kill * and restart - not a popular move |
| 15:11.05 | irroot | make KSRC=/usr/src/linux KVERS=<check output of uname> |
| 15:11.13 | puzzled | irroot: ok |
| 15:11.48 | irroot | the thing is i have it for version 3 and there changes i have put in with out making backward compat potentially |
| 15:12.15 | irroot | puzzled should be safe for 2.6.39 and up |
| 15:12.23 | puzzled | ok |
| 15:12.28 | irroot | bellow that i need to make compat some time |
| 15:12.36 | irroot | i cant garuntee |
| 15:12.39 | *** join/#asterisk davlefou (~david@41.225.9.81) |
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| 15:15.22 | raden | EVIL KDE CRASHES |
| 15:15.56 | puzzled | irroot: no joy. with fritz disabled I get the same "error: expected ‘)’ before ‘*’ token" as here: https://issues.asterisk.org/jira/secure/attachment/41326/Digium-mISDN-1.1.3_build_failure_on_CentOS6_x86_64.txt |
| 15:16.13 | irroot | puzzled it does fail |
| 15:16.27 | irroot | i get lots of warnings but not a error |
| 15:16.45 | puzzled | ok, let me do some more searching then |
| 15:16.58 | irroot | diff it to 1.1.9.2 |
| 15:17.05 | irroot | and see the changes |
| 15:17.12 | puzzled | ok |
| 15:17.29 | puzzled | ah wait, it's caused by error: mISDN/mISDNif.h: No such file or directory |
| 15:17.34 | irroot | need to watch out for the BKL changes you will need the BKL |
| 15:17.50 | irroot | puzzled ok copy the include files there |
| 15:18.01 | irroot | in include in root |
| 15:18.10 | puzzled | sounds like a good idea :) |
| 15:18.20 | irroot | sorry my bad |
| 15:19.58 | ollii | hey..we have a pbx with 1.4.42 where a SIP trunk is used to call the world from sip phones...nothing special so far...but lately there occured multiple warnings about: chan_sip.c: Too many SIP headers added, max 50 |
| 15:20.09 | ollii | how can i interpret this? |
| 15:20.14 | *** join/#asterisk navaismo (~navaismo@fixed-203-96-202.iusacell.net) |
| 15:20.33 | ollii | there're ~90 extensions |
| 15:20.47 | ollii | <PROTECTED> |
| 15:21.32 | puzzled | irroot: getting somewhere. now I only need to fix the location of oslec.h & friends |
| 15:21.48 | irroot | puzzled you want oslec ? |
| 15:21.57 | irroot | if not you can revert that patch |
| 15:22.07 | irroot | i use the echo kernel driver |
| 15:22.11 | irroot | from staging |
| 15:22.24 | *** join/#asterisk d-_-b- (~d-_-b-@2607:f370:9999:dead:292e:ee70:8ff8:9ad6) |
| 15:22.55 | irroot | ../mISDN/drivers/isdn/hardware/mISDN/dsp.h <- comment / uncomment as you wish |
| 15:23.14 | puzzled | irroot: oslec is not in the CentOS 6 kernel but I add oslec.ko with the dahdi modules so I'll figure it out |
| 15:23.55 | *** join/#asterisk d-_-b- (~d-_-b-@204.238.177.5) |
| 15:24.48 | irroot | puzzled the oslec wrapper in my branch was redone from scratch to work with latest ala dahdi i use common oslec |
| 15:24.56 | irroot | so should be little problem |
| 15:25.00 | puzzled | ok |
| 15:29.10 | *** join/#asterisk brdude (~brdude@12.155.183.30) |
| 15:29.14 | Katty | this server is bananas, B A N A N A S |
| 15:29.16 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 15:30.38 | tuxx- | bananas \o/ |
| 15:31.08 | n3hxs | wants a banana split |
| 15:31.16 | irroot | Katty find monkeys |
| 15:34.08 | Katty | hugs n3hxs |
| 15:34.11 | Katty | hugs irroot |
| 15:34.29 | irroot | thx need it monday and all |
| 15:34.37 | irroot | {{{}}} back at you |
| 15:34.38 | n3hxs | Katty, gets a hug from n3hxs |
| 15:34.55 | *** join/#asterisk kareena (~k@unaffiliated/kareena) |
| 15:35.13 | Katty | hi fender bender |
| 15:35.28 | [TK]D-Fender | Katty: Mew. |
| 15:36.47 | jaytee | now I'm gonna have the theme song to "The Banana Splits" children's show going through my head all day |
| 15:37.14 | *** join/#asterisk kdmessano (~nonya@unaffiliated/kdmessano) |
| 15:37.36 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 15:39.11 | n3hxs | is fortunate that he has never seen such a show. |
| 15:39.33 | jaytee | it was a Sid and Marty Kroft production back in the late 60's early 70's |
| 15:40.17 | jaytee | and it's been the main culprit in several "earworm" episodes I've experienced. |
| 15:40.37 | n3hxs | I guess I was too busy working back then. |
| 15:41.12 | n3hxs | Maybe my kids were watching. |
| 15:41.21 | jaytee | I once had a Nickelback earworm once. After 2 hours I'd started contemplating various suicide scenarios. |
| 15:41.55 | Katty | lol |
| 15:42.47 | *** join/#asterisk r0m|u (~are@darkstar.rice.edu) |
| 15:45.40 | *** join/#asterisk rampage73 (~rampage73@vpn.dctechonline.com) |
| 15:46.09 | rampage73 | anyone have a simple script for call forwarding in asterisk ? |
| 15:47.46 | p3nguin | Typically you press the CFwd key on the phone and enter the number. No script needed. |
| 15:48.46 | WIMPy | Including a number of drawbacks. |
| 15:48.49 | puzzled | irroot: with the includes fixed and oslec disabled I get a zillion errors: http://pastebin.com/s2EqGndY |
| 15:49.11 | jaytee | "It was a cold January day in '99 when I went into the rollerskating rink with my K2 Ascent rollerblades desperate to get some skate time in out of the cold and wet. I could only handle 20 minutes in there after hearing both Backstreet Boys "Larger than Life" and "Summer Girls" by Lyte Funky Ones play twice in that small span of time. I barely escaped with my sexual orientation intact." |
| 15:50.25 | [TK]D-Fender | jaytee: http://www.youtube.com/watch?v=c9q30Ce2vwE |
| 15:50.57 | jaytee | is that the "full disclosure" video? |
| 15:51.07 | rampage73 | p3nguin, huh weird the phone says that it is forwarding calls but asterisk has not noticed and when I call the # it does not forward like it says |
| 15:51.37 | [TK]D-Fender | rampage73: Show us the call from * CLI |
| 15:51.38 | jaytee | yeah, I'd seen that. |
| 15:51.39 | irroot | puzzled the path makes no sense /trunk/drivers ?? |
| 15:51.43 | *** join/#asterisk vinhdizzo (~vinh@vqn-portege.ics.uci.edu) |
| 15:51.46 | [TK]D-Fender | ~pb |
| 15:51.46 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com , http://pastebin.us , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ ; or install pastebinit with yum or aptitude. |
| 15:51.52 | [TK]D-Fender | ^^ |
| 15:52.03 | puzzled | irroot: just did an svn co <your repo> |
| 15:52.16 | jaytee | I'm an anachronism.....I rollerblade but I'm hetero |
| 15:52.31 | puzzled | irroot: make that svn co <your repo>/mISDN |
| 15:52.33 | p3nguin | What's wrong with having a drivers subdirectory under the trunk directory? |
| 15:53.18 | WIMPy | puzzled: What was the start of the story? |
| 15:53.19 | irroot | the nothing im trying to make sure you working on right files |
| 15:53.31 | irroot | in the repo there is mISDN/drivers/.... |
| 15:53.47 | puzzled | WIMPy: can't compile mISDN 1.1.9.2 on up-to-date CentOS 6 x86_64 |
| 15:54.21 | WIMPy | Sure. And you are sure, you want cahn_misdn? |
| 15:54.42 | puzzled | WIMPy: errors here: http://pastebin.com/s2EqGndY |
| 15:54.53 | puzzled | WIMPy: yes I want to play with my USB TA units |
| 15:55.19 | puzzled | I want my 2x 10 euro's money worth :) |
| 15:55.20 | WIMPy | I'd recommend chan_lcr. |
| 15:55.23 | irroot | same problem i haz when i have time id like to try port the USB |
| 15:55.43 | irroot | chan_lcr does support the USB too |
| 15:56.18 | WIMPy | chan_lcr is only a bridge to LCR, but misdn(2) supports the USB thingies. |
| 15:56.30 | WIMPy | i.e. the standard Linux kernel. |
| 15:56.34 | puzzled | mISDN(1) has hfcsusb too |
| 15:56.46 | irroot | WIMPy yeah thx for clarifying |
| 15:57.02 | irroot | puzzled mISDN 1/2 both support USB |
| 15:57.23 | puzzled | good |
| 15:57.28 | WIMPy | I'm not sure the bugs is the USB handling have ben fixed for misdn(1) as well. |
| 15:57.37 | WIMPy | s/is/in/ |
| 15:58.09 | puzzled | WIMPy: I just want to use it for some playing around. nothing serious so a bug or two are ok |
| 15:58.22 | *** join/#asterisk kareena (~k@unaffiliated/kareena) |
| 15:59.06 | WIMPy | Sould be ok. |
| 15:59.24 | WIMPy | But misdn(2) should work out of the box. |
| 15:59.29 | puzzled | yes if it compiles which it does not |
| 15:59.55 | WIMPy | Yes, so just don't. |
| 16:00.08 | puzzled | I can't do that Dave |
| 16:00.21 | WIMPy | Can't do what? |
| 16:00.33 | puzzled | not (trying to) compile v1 |
| 16:00.56 | rampage73 | [TK]D-Fender, sorry someone called me, uh ok I cannot get call forward to engage? so i have not tried calling |
| 16:01.08 | puzzled | prolly just needs a few tweaks by someone who groks C enough |
| 16:01.16 | puzzled | I came up empty |
| 16:01.16 | WIMPy | Why not? The kernel you've got should already include what you need. |
| 16:01.52 | puzzled | WIMPy: they included rmudgett's COLP etc. fixes in the C6 kernel? |
| 16:02.00 | [TK]D-Fender | rampage73: You haven't told us what action you are doing that you are thinking is "call forwarding" for us to guess what is wrong? How are you trying to do this? Details are required here... |
| 16:02.30 | WIMPy | puzzled: The new stack works entirely different. It just has the same name, unfortuinately. |
| 16:02.57 | puzzled | WIMPy: new stack? is the one in the C6 kernel v2 or v1? |
| 16:03.44 | WIMPy | The new stack has replaced the old one in Linux 2.6.27. |
| 16:03.54 | WIMPy | So that was some time ago. |
| 16:04.05 | rampage73 | [TK]D-Fender, sorry I am not trying to be cryptic I have a basic asterisk setup running on ubuntu with no gui and nothing in extensions.conf for *72 to enable call forwarding , right now there is nothing there at all for *72 or * anything for that matter because i have not put it there yet |
| 16:04.10 | irroot | WIMPy not replaced but was merged upstream |
| 16:04.53 | WIMPy | It is not backwards compatible or anything. So I thing replaced is the right word. |
| 16:05.00 | WIMPy | think |
| 16:05.16 | [TK]D-Fender | rampage73: Ok that would be a dialplan based decision on how your call would be routed an you have no invented it yet. What are you using for "phone devices"? |
| 16:06.31 | puzzled | WIMPy: sorry still not clear to me. is the mISDN stuff in the CentOS 6 kernel based on mISDN version 2 or version 1? |
| 16:06.36 | irroot | WIMPy it v1 never made it to the kernel so it was never incompatible cause of 1/2 the problem v1 was meant to go in in 2.6.0+ but was blocked till v2 with sockets |
| 16:06.42 | irroot | v2 |
| 16:06.46 | puzzled | ah ok |
| 16:07.04 | WIMPy | puzzled: Depends on the version <2.6.27 v1, >= 2.6.27 v2. |
| 16:07.23 | rampage73 | [TK]D-Fender, our phones? are cisco spa5255 using sip (if i understand your question that is sorry to say it but i am a newb) |
| 16:07.27 | irroot | WIMPy he on 32 so dev 2 |
| 16:07.30 | WIMPy | irroot: v1 has been in the stabdard kernel before. |
| 16:07.30 | puzzled | WIMPy: got it. C6 kernel is 2.6.32 |
| 16:07.56 | WIMPy | That's where all the confusion came from. |
| 16:08.00 | irroot | ah ok |
| 16:08.20 | irroot | so not mainline but some distros included it |
| 16:08.32 | WIMPy | And I4L is still included :-) |
| 16:08.51 | [TK]D-Fender | rampage73: Those phones have their own forwarding right on them. |
| 16:08.55 | WIMPy | No, it was in the standard kernels. |
| 16:09.06 | [TK]D-Fender | rampage73: that has nothing to do with * configuration |
| 16:09.07 | irroot | yeah and some capi stuff that does not play nice |
| 16:09.10 | WIMPy | Nothing distro specific. |
| 16:09.22 | puzzled | so what do I need to use my USB TA's with the mISDN v2 version in the CentOS 6 2.6.32 kernel and asterisk 1.8? |
| 16:09.29 | irroot | can see the mess |
| 16:09.41 | WIMPy | puzzled: LCR |
| 16:09.54 | irroot | puzzled need to get LCR wimpy is there chan_lcr for 1.8 ?? |
| 16:09.57 | *** join/#asterisk Igneous (~igneous@208.86.215.38) |
| 16:10.02 | WIMPy | puzzled: http://voice.yeti.dk/Asterisk_vs_ISDN |
| 16:10.11 | puzzled | thanks guys |
| 16:10.17 | WIMPy | Yes, but there is no chan_lcr for 10, yet. |
| 16:10.41 | irroot | maybe i can make that a project |
| 16:11.42 | rampage73 | [TK]D-Fender, ok i see your point there problem is I am trying to set this up as an option for everyone who uses our system (sorry i was unclear about that) while we have cisco spa525g others have ata's and different models of phones , so I was trying to set this up so we could all have a standard way to call forward if possible |
| 16:11.44 | WIMPy | If I got that correctly, KK added facility support to the kernel. That should give room for improvements. :-) |
| 16:12.07 | WIMPy | No, that was a lib IIRC. |
| 16:12.58 | WIMPy | Anyway: Currently chan_dahdi is the only one supporting ECT, which is quite dull for the others. |
| 16:13.04 | puzzled | WIMPy: very informative website! |
| 16:13.10 | [TK]D-Fender | rampage73: Ok, then you are going to have to create some dialplan. First an extension to take in their choice of where to send calls to them instead (and deactivate); and another bit of code just before you would actually call the device to see if the call was "forwarded" somewhere and needs to jump somewhere eslse |
| 16:13.17 | WIMPy | Thanks |
| 16:14.36 | puzzled | WIMPy: do I need to install mISDNuser v2 too? |
| 16:15.26 | rampage73 | [TK]D-Fender, thanks will give it a try |
| 16:15.46 | WIMPy | puzzled: yes |
| 16:15.50 | puzzled | thanks |
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| 16:21.38 | *** join/#asterisk Janos (~Janos@186.32.185.5) |
| 16:22.11 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 16:24.49 | Janos | hi, still having troubles with that E1, i changed the cable as suggested by WIMPy, and now i got BLUE + RED alarms, if anyone can give me a hand |
| 16:27.18 | Igneous | I apologize, this isn't exactly asterisk related.. but does anyone know a way to define which line/extension is used (to dial out) on a Polycom idle-screen $TINVITE$ EFK button? |
| 16:27.30 | WIMPy | Janos: If I understand that right that beans you're receiving an 'all ones' signal. That means there's a upstream problem. |
| 16:27.53 | WIMPy | s/bea/mea/ |
| 16:28.42 | Janos | WIMPy, kk, thanks a lot, will double check the cable again just in case and will call the provider afterwards, thanks again |
| 16:29.23 | WIMPy | Janos: Doesn't the NT ave some status LEDs? |
| 16:33.57 | Janos | WIMPy, i'm don't really at the moment, i'm going to the place right now to check it out myself since i was working remotely on Friday, i know things are a bit more complex on this setup because they have both data and the E1 coming through the same fiber and there is some kind of multiplexer in the middle, will report back once i get there and check it out myself, thanks again |
| 16:34.11 | Janos | s/really at/ really know at/ |
| 16:34.56 | WIMPy | Well, that multiplexer may be th NT in that case. But ist shouldn't make things more complex. |
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| 17:30.00 | Katty | i haz chicken nuggets! |
| 17:30.58 | WIMPy | You must look funny then. |
| 17:32.09 | *** join/#asterisk SwK (~SwK@freeswitch/developer/swk) |
| 17:32.12 | r0m|u | hello all. |
| 17:32.56 | WIMPy | Hello one. |
| 17:33.04 | r0m|u | :) |
| 17:33.22 | r0m|u | felling like crap today.... I haz flu shot |
| 17:34.29 | WIMPy | Bird flu or swine flu? |
| 17:35.14 | WIMPy | BTW: Or what is it this year? |
| 17:36.59 | puzzled | WIMPy: wow those v2 modules from git compiled without an issue. very nice |
| 17:37.03 | r0m|u | shit I dont know... lol I was told you are getting the flu shot... I was like ok... |
| 17:37.38 | *** join/#asterisk nhaynes (~nhaynes@c053880-02-v1031-static.nvrninaa.cinergymetronet.net) |
| 17:37.47 | chuckf | who are 'they' that forced a needle into your arm? |
| 17:37.55 | nhaynes | has anyone made faxing work over a digium t1 card? |
| 17:38.25 | nhaynes | I have several fax machines connected to audiocodes mp-118 gateways and they can send/receive faxes from eachother internally |
| 17:38.39 | nhaynes | but if i try to send/receive via the t1 it fails every time |
| 17:38.45 | WIMPy | puzzled: Yu made the modules yurself anyway? |
| 17:39.09 | nhaynes | it has to be some dahdi setting that's not right |
| 17:39.11 | nhaynes | but i can't figure it out |
| 17:39.49 | puzzled | WIMPy: your webpage said to use latest git for everything so I complied :) |
| 17:40.05 | WIMPy | Ok :-) |
| 17:40.22 | *** join/#asterisk citywok (~citywok@67-134-194-33.dia.static.qwest.net) |
| 17:40.36 | WIMPy | Well, at least there's a recent official release of LCR again. |
| 17:40.50 | puzzled | you mean the 1.10 release? |
| 17:40.55 | WIMPy | Before that you had to use the git version with 1.8. |
| 17:41.00 | WIMPy | yes |
| 17:46.27 | nhaynes | nobody? |
| 17:47.14 | WIMPy | thinks it's time to tone that comment down a little. |
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| 18:18.16 | puzzled | WIMPy: do you perhaps know which var controls the install destination of the modules? I would like to install them to /lib/modules/$KVERSION/extra/mISDN instead of just extra/ |
| 18:21.11 | *** join/#asterisk didnot (~didnot@unaffiliated/didnot) |
| 18:22.00 | WIMPy | I haven't tried to put them elsewhere. In fact I haven't tried a stand-alone build for ages. |
| 18:22.36 | puzzled | ok |
| 18:37.58 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
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| 19:00.55 | r0m|u | p3nguin, you avail? |
| 19:03.44 | p3nguin | Yes. |
| 19:16.19 | *** join/#asterisk mcnon (cffb52e2@gateway/web/freenode/ip.207.251.82.226) |
| 19:16.46 | mcnon | hello i need to setup a asterisk box as a gateway that will handle all incoming/outgoing DIDs and pass to another asterisk box that will handle extensions |
| 19:17.08 | mcnon | can someone point me in the right direction? |
| 19:17.33 | WIMPy | You need at least to tell us from/to what you want to convert. |
| 19:17.50 | mcnon | Ah |
| 19:18.21 | mcnon | Asterisk/FreePBX/A2Billing >>> multi-tenant setup of somesort |
| 19:18.42 | mcnon | i googled multi-tenant and found some commercial software |
| 19:18.55 | mcnon | but they recommended i setup a box to handle the external calls |
| 19:19.23 | mcnon | i have played with asterisk but not in this nature |
| 19:21.11 | mcnon | ? |
| 19:24.21 | WIMPy | You still didn't mention where those call come from or go to. |
| 19:24.47 | WIMPy | And if you want to use FreePBX, try #freepbx. |
| 19:25.23 | mcnon | im debating about freepbx |
| 19:25.29 | mcnon | but the calls will be from voip providers |
| 19:25.38 | p3nguin | ~freepbx |
| 19:25.38 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there, or fpbx |
| 19:25.38 | mcnon | in a nutshell |
| 19:26.14 | WIMPy | That does not sound much like a gateway. Why two boxes? |
| 19:26.30 | mcnon | Because. the MT software heavily modifies the CDRs |
| 19:26.30 | *** join/#asterisk heffer (~felix@fedora/heffer) |
| 19:26.47 | mcnon | so a2billing wont bill properly |
| 19:26.48 | p3nguin | Set up extensions on the intermediate asterisk to accept calls from a peer and send them to another peer. |
| 19:27.04 | mcnon | p3nguin: englih |
| 19:27.08 | mcnon | english? |
| 19:27.13 | WIMPy | A simple procy might be more appropriate, I guess. |
| 19:27.24 | WIMPy | s/cy/xy/ |
| 19:27.24 | p3nguin | Yes, I am speaking English. |
| 19:27.52 | p3nguin | Plain English, to be more exact. |
| 19:28.12 | mcnon | can you put it in n00b terms |
| 19:29.35 | p3nguin | Do you know peers and extensions? |
| 19:29.48 | mcnon | yes |
| 19:29.57 | p3nguin | Then create a peer where calls will come FROM. |
| 19:30.04 | mcnon | im coming from elastix,freepbx,trixbox background |
| 19:30.12 | p3nguin | Create extensions to process those calls. |
| 19:30.22 | mcnon | right |
| 19:30.26 | p3nguin | Create extensions that will accommodate sending the calls to another peer. |
| 19:30.34 | p3nguin | Create another peer to send calls TO. |
| 19:30.35 | mcnon | but how do i get the call from Server A to Server B |
| 19:30.49 | p3nguin | I just told you. |
| 19:30.53 | mcnon | ok |
| 19:30.57 | p3nguin | extensions |
| 19:31.01 | mcnon | back up a page |
| 19:31.12 | p3nguin | reverses |
| 19:31.26 | mcnon | The Asterisk Box w/ a2b will have all the dids right? |
| 19:31.37 | p3nguin | DIDs are provided by your ITSP. |
| 19:31.42 | mcnon | right |
| 19:31.52 | mcnon | so i configure the DIDs on Server 1 right |
| 19:32.17 | p3nguin | They send calls which have arrived at your DID to your systems, where the call will match a peer entry and go into a context, where a extension processes the call. |
| 19:32.26 | *** join/#asterisk Azrael808 (~peter@cpc17-walt12-2-0-cust657.13-2.cable.virginmedia.com) |
| 19:32.40 | mcnon | can i share contexts beteween servers? |
| 19:33.05 | p3nguin | I don't know why you'd need to in this case. |
| 19:33.12 | mcnon | ok just a question |
| 19:33.20 | mcnon | ok so i make a extenstion can it be the DID? |
| 19:33.23 | p3nguin | It can be done, but not just as easily as you've written it. |
| 19:33.31 | mcnon | such as 2125550101 |
| 19:33.35 | p3nguin | Yes, the extension will most often be the DID number. |
| 19:33.52 | p3nguin | But you can also use a pattern to match many numbers. |
| 19:34.22 | mcnon | ok so someone calls 2125550101 the call comes from the provider >>>>> */A2B >>>>> through the 2125550101 extenstion >>> MT Serve |
| 19:34.23 | mcnon | right? |
| 19:34.54 | p3nguin | That seems like a fairly accurate representation of the call flow. |
| 19:35.13 | mcnon | now. how would i setup the 2125550101 on the MT Server? |
| 19:35.15 | mcnon | via a trunk? |
| 19:35.39 | *** join/#asterisk devmikey (~irc@unaffiliated/devmikey) |
| 19:35.54 | p3nguin | Configure a peer on one system for the other system. Configure a peer on the other system for the first system. |
| 19:36.15 | mcnon | p3nguin: ever hear of thirdlane? |
| 19:37.16 | mcnon | ok i see how its gonna work |
| 19:37.24 | mcnon | when i setup a trunk |
| 19:37.28 | mcnon | 2125550101 is user |
| 19:37.32 | mcnon | password is whatever i set |
| 19:37.41 | mcnon | caller id is whatever also |
| 19:38.18 | mcnon | right? |
| 19:38.57 | WIMPy | ...and in most cases your number will be non-DID... |
| 19:39.18 | mcnon | non-did? |
| 19:40.27 | WIMPy | It's just like sip trunks, ppl sitting at extensions and so on. |
| 19:40.35 | mcnon | what do you mean |
| 19:40.35 | mcnon | ? |
| 19:40.50 | WIMPy | misused terms |
| 19:41.22 | mcnon | isnt the DID a number like 212-555-1010>? |
| 19:42.04 | p3nguin | yes |
| 19:42.09 | WIMPy | It is a number, yes, but it probably isn't DID. |
| 19:42.23 | p3nguin | But the peer name is most commonly something representative of the device on the other side. |
| 19:42.38 | p3nguin | Like a peer name of "other-server" |
| 19:43.33 | mcnon | ok |
| 19:52.04 | irroot | WIMPy just got chan_lcr to compile with asterisk 10 |
| 19:52.43 | *** join/#asterisk tzafrir_laptop (~tzafrir@212.179.75.202) |
| 19:56.04 | WIMPy | Cool. |
| 19:56.56 | WIMPy | irroot: Was it only the new media description stuff or more? |
| 19:57.32 | irroot | ive posted the patch on i4l but it seems like its only codec formats |
| 19:57.36 | irroot | compiles clean no errors |
| 19:57.43 | irroot | but now to test it |
| 19:59.51 | WIMPy | Ok, so I will see that in a few moments. |
| 20:00.35 | WIMPy | I guess I will have to write a lot of updates for 10. |
| 20:13.33 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 20:18.03 | irroot | WIMPy found why i was not getting it working before |
| 20:18.42 | irroot | my firewall was to agresive it needs to allow protocol family 34 mISDN uses this protocol to comunicate |
| 20:20.13 | *** join/#asterisk devmikey (~irc@unaffiliated/devmikey) |
| 20:23.15 | WIMPy | Oh, wow. You filtered non-IP on lo? |
| 20:24.25 | irroot | WIMPy yeah lo could be opened |
| 20:24.54 | irroot | makes little sense to have it |
| 20:26.32 | WIMPy | I fell for that one as well some time. I only allowed 127/8, 0/32 and 255.255.255.255/32. |
| 20:27.15 | WIMPy | Later I learned that all local traffic goes via lo, even if it uses addresses of other interfaces. |
| 20:27.25 | WIMPy | A little strange thing on Linux. |
| 20:27.56 | beek | If I'm in the VoiceMail application and press the # key, what is supposed to happen? |
| 20:28.35 | beek | My problem is that the # key is being recorded along with the message and Asterisk is reacting to it. I need to somehow get it to be ignored while in that application. |
| 20:30.00 | WIMPy | If recording, it should terminate recording, otherwise it will hang up. |
| 20:30.25 | *** join/#asterisk sassyn (~sassyn@bzq-82-80-242-217.cablep.bezeqint.net) |
| 20:30.28 | sassyn | hi all |
| 20:30.34 | sassyn | I just installed asterisk 1.8.7 |
| 20:30.39 | WIMPy | If you want # to be ignored completely, you can't use VoiceMail, I think. |
| 20:30.41 | sassyn | with freepbx 2.9 and meetme 4.0 |
| 20:30.52 | Qwell | wtf is meetme 4.0? |
| 20:31.01 | Micc | is there a way to tell asterisk to stop accepting new calls? |
| 20:31.08 | Micc | but leave up existing calls? |
| 20:31.13 | Qwell | Micc: core stop gracefully |
| 20:31.13 | sassyn | web-meetme |
| 20:31.22 | Micc | thanks |
| 20:31.28 | Qwell | I think it's gracefully.. |
| 20:31.29 | irroot | Micc night mode / stop gracefully ?? |
| 20:31.36 | Qwell | there are 2. I forget which is which |
| 20:31.46 | WIMPy | yes |
| 20:32.02 | sassyn | Qwell, http://areski.net/Web-MeetMe/about.php |
| 20:32.04 | WIMPy | 'when convenient' will still accept new calls. |
| 20:32.31 | sassyn | It is working, but I don't get to which conf number I need to call |
| 20:32.39 | Micc | yeah I was using when convenient, that was still accepting calls. |
| 20:32.46 | irroot | WIMPy yeah convenient waits till all has stoped allowing new calls |
| 20:32.50 | Micc | trying gracefully now. |
| 20:32.57 | Qwell | WIMPy: right - thanks |
| 20:33.13 | sassyn | anyone have any idea? |
| 20:33.16 | irroot | im off folks bed calling |
| 20:33.25 | sassyn | irroot, good night |
| 20:33.38 | WIMPy | can_bed? |
| 20:33.42 | WIMPy | chan_bed? |
| 20:34.24 | irroot | WIMPy unfortunately chan_beer did not compile tonight |
| 20:34.37 | WIMPy | Oh, that's bad. |
| 20:37.28 | azv4 | can anyone find me Panasonic Td-500 Maintience Console program? |
| 20:39.12 | sassyn | anyone? |
| 20:39.13 | sassyn | please? |
| 20:39.20 | sassyn | I'm stuck :-( |
| 20:40.45 | Micc | is it still preferable to run fax for asterisk on 1.8 or is app_fax better? |
| 20:41.08 | Micc | If its all the same, I won't worry about moving the licenses and I'll just use app_fax. |
| 20:41.25 | p3nguin | gracefully should leave existing calls and deny any new calls. Eventually there will be no calls and it can stop or restart. |
| 20:41.26 | WIMPy | Or iaxmodem or t38modem? |
| 20:41.45 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 20:41.53 | Micc | I do use t.38, is that still only available with res_fax_digium? |
| 20:42.09 | Micc | or maybe it was always available in both. |
| 20:42.27 | Micc | I remember there was a reason why we got fax for asterisk licenses. |
| 20:42.27 | WIMPy | I think you might want to take a look at Asterisk 10. |
| 20:42.48 | Micc | I don't want to do t38 gateway. |
| 20:43.06 | Micc | at least not yet. Our customers like us taking the fax and sending it to them in email. |
| 20:44.31 | Micc | I just want to do the least amount of work to fix what we have already which is fax for asterisk on 1.6 which only gets about 50% success rate. |
| 20:44.50 | Micc | It seems like 1.8.8 has some fixes that might help. |
| 20:44.59 | WIMPy | That's bad. |
| 20:45.34 | Micc | I have a feeling it has something to do with reinvites. |
| 20:45.58 | Micc | I've found other incompatibilities with asterisk and sonus equipoment, which is what 360 uses. |
| 20:48.44 | beek | WIMPy: Thanks. That means the # transfer is happening elsewhere and I need to find that. |
| 20:55.25 | CrossWired | I have cdr setup in my context so I get all the details of connected calls using originate, but how do iget the status of those that go unanswered, or are busy? |
| 20:55.52 | *** join/#asterisk elb (~elb@99-52-196-7.lightspeed.crmlin.sbcglobal.net) |
| 20:56.27 | elb | ASTERISK-18221 was my bug; I submitted it months ago, along with a detailed description of the problem and a self-evident patch |
| 20:56.58 | elb | a week or two ago I was asked to provide additional information, including a backtrace, which would require rebuilding an old, known broken version of Asterisk just to reproduce |
| 20:57.25 | elb | I planned to do that, but hadn't gotten to it -- and frankly, it wasn't high on my priority list because the code in question is obviously broken to casual visual inspection |
| 20:57.37 | elb | the bug was just closed, with a note that it can be reopened here |
| 20:57.53 | elb | now, I'm not asking to have it reopened, but I did want to note that I'm aware of the process and I'll get to it if I can get to it |
| 20:59.09 | mjordan | elb: actually, #asterisk-bugs |
| 20:59.19 | mjordan | that's where the bug marshalls hang out |
| 20:59.32 | elb | oops, did I misread the email |
| 20:59.36 | mjordan | or at least, one of the many places |
| 20:59.49 | elb | yeah, I did, sorry |
| 20:59.54 | mjordan | no problem :-) |
| 21:04.27 | sassyn | anyone know about web-meeme? |
| 21:04.30 | sassyn | meetme? |
| 21:05.34 | *** join/#asterisk ajohnson (~aaron@65-122-4-130.dia.static.qwest.net) |
| 21:14.51 | ajohnson | I'm having some difficulty with group counting and I was hoping someone could double check that what I'm doing is sane |
| 21:15.06 | ajohnson | http://pastebin.com/qN1bFfrt |
| 21:15.22 | *** part/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
| 21:16.09 | ajohnson | I've added local channels to queues so that I can limit the number of queue calls that go to a user, but the group counting doesn't appear to be working. No matter how many calls the user is on, the count always returns 0 |
| 21:17.11 | WIMPy | Did you use the local channel with /n ? |
| 21:17.55 | p3nguin | If you're counting local channels, they'll go away once the call is bridged if you aren't using /n on it. |
| 21:18.05 | ajohnson | eureka! |
| 21:18.19 | ajohnson | Thanks, I did this once a while ago and figured I must be missing something |
| 21:18.30 | *** part/#asterisk elb (~elb@99-52-196-7.lightspeed.crmlin.sbcglobal.net) |
| 21:18.40 | ajohnson | This is what I had: member => Local/ajohnson@exit-queue,1,,SIP/ajohnson |
| 21:18.47 | p3nguin | By default chan_local optimizes itself out of the call. |
| 21:19.24 | ajohnson | thanks |
| 21:19.45 | Katty | peeks in |
| 21:21.44 | ajohnson | that worked, awesome. Thanks |
| 21:24.31 | beek | waves to Katty |
| 21:26.35 | sassyn | sent into invalid extension 's' in context 'custom-meetme3' ? what does it mean? |
| 21:27.03 | p3nguin | It means the call was sent to extension s in context custom-meetme3, and that extension does not exist. |
| 21:27.25 | sassyn | p3nguin, OK |
| 21:27.36 | sassyn | p3nguin, did u ever used meetme? |
| 21:27.43 | *** join/#asterisk navaismo (~navaismo@187.170.11.234) |
| 21:27.45 | p3nguin | I use MeetMe every day. |
| 21:28.01 | sassyn | p3nguin, maybe please can u help? |
| 21:28.08 | p3nguin | I can attempt. |
| 21:28.13 | sassyn | i want to used web-meetme |
| 21:28.17 | sassyn | ODBC is configure |
| 21:28.19 | p3nguin | Never heard of it. |
| 21:28.34 | sassyn | well it doesn't really matter |
| 21:28.37 | p3nguin | okay |
| 21:28.43 | sassyn | can u tell how u configure meetme |
| 21:29.04 | p3nguin | If you can use words I understand, it'll make this a lot easier. |
| 21:29.17 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 21:29.35 | sassyn | see here http://areski.net/Web-MeetMe/about.php |
| 21:29.46 | mjordan | sassyn: there is a big difference between MeetMe, which is an application in Asterisk (and what most people are going to be familiar with), and Web-MeetMe, which is a 3rd party application you are trying to use |
| 21:29.58 | mjordan | you may want to note that before attempting to get support out of others |
| 21:30.11 | sassyn | mjordan, OK thanks I didn't know that |
| 21:30.19 | sassyn | mjordan, Sorry |
| 21:30.24 | mjordan | no problem |
| 21:30.44 | p3nguin | Did you do steps 1-4 on that page? |
| 21:30.59 | sassyn | It just as far as I understand the Web-MeetMe is build on top of the MeetMe |
| 21:31.09 | sassyn | p3nguin, which 4 steps? |
| 21:31.24 | p3nguin | Did you read the page you linked me to? |
| 21:31.32 | sassyn | yep |
| 21:31.33 | sassyn | I did |
| 21:31.44 | p3nguin | There's a 4-step process. Steps 3 and 4 appear to be optional. |
| 21:31.55 | sassyn | the Readme file is mush more helpfull |
| 21:32.31 | sassyn | I did, the system seems to work, the site is working. and also when running asterisk -vvvvr I can see the ODBC binding to the DB |
| 21:32.44 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 21:32.57 | sassyn | However when dailing *66 - the system gives nothing |
| 21:33.13 | p3nguin | I'll need to see what extension *66 does. |
| 21:34.21 | sassyn | http://sourceforge.net/projects/web-meetme/files/ here is where u can find the files |
| 21:35.51 | ChannelZ | it calls satan |
| 21:35.59 | sassyn | :-) |
| 21:36.05 | p3nguin | I thought that would be extension 616. |
| 21:37.44 | sassyn | I'm lost . |
| 21:37.47 | sassyn | :-( |
| 21:38.34 | p3nguin | The number of the Beast, ya know? |
| 21:39.03 | sassyn | Yes |
| 21:39.05 | sassyn | 666 |
| 21:39.08 | p3nguin | 616 |
| 21:42.24 | *** join/#asterisk lanning (~lanning@173.8.187.197) |
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| 22:04.22 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-mwddxwouprjkqzzv) |
| 22:19.00 | Micc | So now I have a problem where t.38 negotiation times out when call comes directly into 1.8.8-rc1 server but it works fine if call comes into 1.6.2.17.3 first then to 1.8.8-rc1 |
| 22:19.58 | Micc | I'm gonna try putting back a wait in the dialplan. |
| 22:22.40 | *** join/#asterisk f2knight (~ben@c-98-246-74-16.hsd1.or.comcast.net) |
| 22:48.32 | Micc | I registered fax for asterisk licenses on a new machine but now after reloading it shows 0 licenses after it was working before. |
| 22:50.55 | Micc | oh crap, I think I accidentally deleted them |
| 22:51.50 | Micc | yeah it overwrote the licenses. |
| 22:53.49 | Micc | looks like its letting me do it again |
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| 23:02.30 | CrossWired_ | anyone familar with starpy? |
| 23:04.52 | p3nguin | Is that anything like a star crunch? |
| 23:05.56 | *** join/#asterisk Hierbenik (~Hier_ben_@a83-163-196-243.adsl.xs4all.nl) |
| 23:07.37 | Hierbenik | Anyone got overrideswitch to work with realtime extensions? |
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| 23:09.28 | *** join/#asterisk autofsckk (~autofsckk@unaffiliated/autofsckk) |
| 23:18.39 | Hierbenik | nobody uses realtime with different contexts for each user? |
| 23:23.33 | pdtpatrick | what is usually the cause for this ? |
| 23:23.34 | pdtpatrick | http://pastebin.com/jSDEnKez |
| 23:26.38 | pdtpatrick | nvm figured ito ut |
| 23:38.09 | WIMPy | What is overrideswitch? |
| 23:41.32 | *** join/#asterisk batfastad (~benbradle@87-194-37-131.bethere.co.uk) |
| 23:51.51 | Hierbenik | see here: https://issues.asterisk.org/jira/browse/ASTERISK-5860?page=com.atlassian.jira.plugin.system.issuetabpanels%3Aall-tabpanel |
| 23:53.51 | batfastad | Hi everyone. What's the best way of having a script (PHP) run when an extension starts (or is connected to) and ends a call? With access to various vars about the call. |
| 23:55.07 | *** join/#asterisk gxdssoft (~gxdssoft@201.240.107.202) |
| 23:55.50 | batfastad | I've been experimenting with AMI and I can read information out of Asterisk ok. But according to this page http://www.voip-info.org/wiki/view/Asterisk+manager+API there's Events which "inform clients about state changes within Asterisk". Could those events be used to trigger my script? Unfortunately there seems to be pretty much no documentation on the AMI Events that I can find, and any that I have found is certainly not complete or up |
