| 00:09.13 | kaushal | paulc: i will update you in sometime |
| 00:09.22 | kaushal | paulc: can i pvt message you ? |
| 00:12.18 | paulc | kaushal: Sure. Not always at the PC but I'm usually logged in and not too far away.. |
| 00:12.28 | kaushal | paulc: sure |
| 00:18.28 | p3nguin | rotten777: What happens when the call comes "in" now? |
| 00:19.31 | rotten777 | it rings my tele and rings like a standard phone on the caller |
| 00:22.26 | p3nguin | And you want a specific sound clip, or would music be okay? |
| 00:22.51 | rotten777 | yeah i'm looking around right now for cc audio to use |
| 00:23.07 | rotten777 | i have a yeti mic i'm going to use to make a clip and save in the correct format |
| 00:23.31 | rotten777 | but just in general i wanted to know how to have the answer, play audio for 30s, then voicemail if no answer. |
| 00:24.29 | p3nguin | I would probably configure a musiconhold class with only that single sound file in it, then use that class to play musiconhold while the phone is ringing, and use a 30 second timeout on the Dial(). |
| 00:25.54 | p3nguin | You follow? |
| 00:26.12 | rotten777 | yeah i'm not sure how to create classes |
| 00:26.19 | p3nguin | musiconhold.conf |
| 00:26.29 | rotten777 | gotcha |
| 00:26.34 | p3nguin | Give it an arbitrary name for the new class. |
| 00:26.54 | p3nguin | Specify a directory for it, place your sound file in that directory by itself. |
| 00:27.16 | p3nguin | Then in the dial, you'll use option m(your-new-class) |
| 00:27.33 | p3nguin | The caller will hear the sound file, and the called phone will ring normally. |
| 00:27.59 | p3nguin | If you are going to make it a wave file, it needs to be mono 8 kHz. |
| 00:28.12 | rotten777 | mono 8khz ok i can do that with audacity |
| 00:28.44 | p3nguin | 16 bit |
| 00:28.49 | rotten777 | ok so i do NoOp(), m(class), dial(extension), voicemail(ext), Hangup() |
| 00:28.50 | rotten777 | ? |
| 00:29.04 | p3nguin | no, m(class) is a dial option |
| 00:29.16 | p3nguin | Dial(SIP/phone,30,m(class)) |
| 00:29.28 | rotten777 | ahh ok gotcha |
| 00:30.26 | p3nguin | Try it will music just for testing. Just omit the '(class)' part, using only the m. |
| 00:30.41 | p3nguin | I assume you have a default moh class. |
| 00:30.52 | p3nguin | err, try it with music |
| 00:31.17 | p3nguin | That's messed up. |
| 00:31.17 | rotten777 | ok trying now |
| 00:31.22 | p3nguin | I'm watching Hell's Kitchen... |
| 00:31.31 | p3nguin | And they were yelling at the dude named Will. |
| 00:31.38 | p3nguin | and I typed will instead of with. |
| 00:32.00 | rotten777 | exten => 18636584192,n,Dial(SIP/byrdits,30,m()); |
| 00:32.03 | rotten777 | ? |
| 00:32.11 | p3nguin | Just skip the () after the m. |
| 00:32.11 | rotten777 | do i need the m() or just m? |
| 00:32.13 | rotten777 | k |
| 00:32.34 | p3nguin | If you have a default moh, it should play music when you call it. |
| 00:32.59 | p3nguin | You can check if you have it: moh show classes |
| 00:33.03 | p3nguin | also: moh show files |
| 00:33.16 | rotten777 | nothing after moh show files |
| 00:33.21 | rotten777 | and no music but also no ringing lol |
| 00:33.29 | rotten777 | silence on caller |
| 00:33.36 | p3nguin | So there are no music files. Do you have a default class listed? |
| 00:33.46 | rotten777 | [Sep 12 20:33:37] WARNING[5788]: res_musiconhold.c:989 moh_scan_files: Cannot open dir /usr/share/asterisk/moh or dir does not exist |
| 00:33.51 | rotten777 | thank you ubuntu package maintainers.... |
| 00:33.56 | p3nguin | yeah, that's jacked. |
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| 00:34.11 | p3nguin | The normal path is /var/lib/asterisk/moh, I think. |
| 00:34.40 | rotten777 | the standard dir is empty |
| 00:34.48 | rotten777 | i created the one it's looking for as a link to the standard |
| 00:34.54 | p3nguin | Just one moment. |
| 00:35.32 | p3nguin | Do you have the file manolo_camp-morning_coffee.wav on the system? |
| 00:35.50 | rotten777 | nope |
| 00:36.17 | p3nguin | Is there a moh package that you didn't install? |
| 00:37.26 | rotten777 | nope installing them now |
| 00:39.57 | rotten777 | hmm same result |
| 00:40.11 | p3nguin | Now you have some moh files? |
| 00:40.36 | rotten777 | i don't see a moh class |
| 00:40.51 | p3nguin | Do you have some moh files installed now? |
| 00:41.00 | rotten777 | /var/lib/asterisk/moh/manolo_camp-morning_coffee.wav |
| 00:41.02 | rotten777 | yes |
| 00:41.22 | p3nguin | In musiconhold.conf, do you have a default class defined? |
| 00:41.46 | rotten777 | default mode=files directory=moh |
| 00:42.07 | p3nguin | Specify the full path to the files if they are non-standard for your build. |
| 00:42.44 | p3nguin | or just specify the path anyway. |
| 00:42.58 | rotten777 | same result |
| 00:43.02 | p3nguin | directory=/var/whatever |
| 00:43.25 | p3nguin | After you do that, save the file and run moh reload in the ast cli. |
| 00:43.31 | rotten777 | i did |
| 00:43.47 | p3nguin | moh show classes says you have the default class? |
| 00:44.02 | rotten777 | Class: default |
| 00:44.02 | rotten777 | Mode: files |
| 00:44.02 | rotten777 | Directory: /var/lib/asterisk/moh |
| 00:44.21 | p3nguin | And moh show files says nothing? |
| 00:44.32 | rotten777 | 5 files |
| 00:44.36 | p3nguin | Good. |
| 00:44.39 | rotten777 | under class default |
| 00:44.52 | *** join/#asterisk lanning (~lanning@173-8-187-197-SFBA.hfc.comcastbusiness.net) |
| 00:45.05 | p3nguin | Now if you call that extension, what happens? |
| 00:45.42 | rotten777 | silence when calling in and ringing on the ip phone |
| 00:45.52 | p3nguin | No errors? |
| 00:46.06 | rotten777 | nope |
| 00:46.10 | rotten777 | not in the asterisk console |
| 00:46.19 | p3nguin | You might not get music because the line is not up. Add an Answer() before the Dial(). |
| 00:46.32 | p3nguin | I hate doing that, but that could be the cause of no music. |
| 00:47.33 | rotten777 | sweeeeeeeeeeeeeeeeeeeet |
| 00:47.38 | p3nguin | When I play music instead of ringing, I playback a file that says to please wait while the call is connected. |
| 00:47.47 | rotten777 | well thats the goal |
| 00:47.51 | rotten777 | i will mod the wav file |
| 00:47.59 | p3nguin | Hmm? |
| 00:48.06 | p3nguin | What is the exact goal? |
| 00:48.24 | p3nguin | You can play a sound clip then play music instead of ringing. |
| 00:48.29 | p3nguin | No mod necessary. |
| 00:48.46 | rotten777 | the wav will have a voiceover on the music saying "please hold as we find someone to help you" or similar |
| 00:49.21 | p3nguin | Okay, change the Answer() to Playback(silence/1&vm-dialout&silence/1) |
| 00:49.46 | p3nguin | That'll suffice while you figure out what sound clip you want to record. |
| 00:50.13 | p3nguin | You don't need the Answer if you have Playback, so just change it.l |
| 00:51.19 | rotten777 | ok whats that function calling? |
| 00:51.22 | p3nguin | If you're doing it for a more professional type of thing as opposed to a home number, there are other sound files that will fit as well. |
| 00:51.31 | p3nguin | That's not a function, it's an application. |
| 00:51.54 | rotten777 | there sound files built into the asterisk platform? |
| 00:51.54 | p3nguin | And Playback is playing two files: silence/1 and vm-dialout |
| 00:52.00 | rotten777 | ahhhhh gotcha |
| 00:52.05 | p3nguin | They are included. |
| 00:52.10 | rotten777 | nice |
| 00:52.54 | p3nguin | Give it a try and let me know if it's satisfactory for the time being. |
| 00:52.55 | rotten777 | are they all in moh? |
| 00:52.58 | p3nguin | no |
| 00:52.59 | rotten777 | yeah it is good |
| 00:53.20 | p3nguin | The sound files are typically under /var/lib/asterisk/sounds(/en) |
| 00:54.31 | p3nguin | Is this for a home number? |
| 00:54.47 | rotten777 | nope |
| 00:54.50 | rotten777 | bidness number |
| 00:55.00 | p3nguin | Will you have several phones? |
| 00:55.16 | rotten777 | so far 1 did and 1 extension but will have 3 extensions soon |
| 00:55.33 | p3nguin | You mean 3 phones, probably. |
| 00:55.48 | p3nguin | You'll have LOTs of extensions. |
| 00:55.51 | rotten777 | yeah sorry i gotta learn the lingo |
| 00:56.12 | p3nguin | With a dozen phones, I have 399 extensions currently. |
| 00:56.21 | rotten777 | lol jeebus |
| 00:56.30 | p3nguin | -= 399 extensions (1412 priorities) in 82 contexts. =- |
| 00:56.38 | rotten777 | what kind of company is that? |
| 00:56.42 | rotten777 | voip services i'm assuming |
| 00:56.47 | p3nguin | general IT |
| 00:56.54 | rotten777 | ah where are you located? |
| 00:57.02 | p3nguin | I also hang my home phones off that box. |
| 00:57.05 | rotten777 | ah |
| 00:57.14 | p3nguin | I'm in IL. |
| 00:57.28 | rotten777 | ah I'm doing the same in FL |
| 00:57.34 | rotten777 | managed IT services for government and small business |
| 00:57.44 | rotten777 | not sure why I just now started to focus on voip |
| 00:58.38 | p3nguin | You're probably going to want to build an attendant pretty soon. |
| 00:58.57 | p3nguin | And you'll probably want to use a queue when you get more phones online with people to answer them. |
| 00:59.41 | rotten777 | piece at a time :) voip.ms doesn't have any did's here and i want to use them instead of flowroute long term |
| 00:59.59 | rotten777 | i'm still buying up ip phones now though |
| 01:00.44 | p3nguin | Did voipms finally activate you? |
| 01:01.03 | rotten777 | yeah they activated me and replied to my e-mail and said no DID's an no idea when they get here |
| 01:01.11 | p3nguin | :/ |
| 01:01.28 | p3nguin | You could always do toll-free for now. |
| 01:01.48 | p3nguin | You should be able to get one for $0.99/mo |
| 01:02.05 | p3nguin | and $0.025/minute |
| 01:02.07 | rotten777 | yeah that's the goal. i'm meeting with my cpa soon to start another s-corp and my partner will be taking over operations of that.. i think it'll be a toll-free based op |
| 01:02.28 | rotten777 | i don't want a toll free for my IT managed services company.. |
| 01:02.36 | p3nguin | oh |
| 01:04.08 | rotten777 | i'm trying to keep it local because i'm basically maxed out until i can find more help to hire |
| 01:04.59 | f2knight | rotten777, what part of FL? |
| 01:05.04 | f2knight | hi p3nguin |
| 01:05.09 | rotten777 | about an hour south of orlando |
| 01:05.12 | p3nguin | You could go ahead and build your attendant and set up a front, and the callers won't know if you are alone or have hired more people. |
| 01:05.13 | rotten777 | sebring |
| 01:05.25 | rotten777 | p3nguin i plan on doing that as i learn more about asterisk |
| 01:05.30 | f2knight | rotten777, just moved from Fort Lauderdale to Portland OR. Got tired of the heat :) |
| 01:05.33 | p3nguin | I'm here to help you. |
| 01:06.02 | f2knight | rotten777, what area code are you looking for? |
| 01:06.07 | rotten777 | f2knight i plan on moving also when i can sell my rental property and my house... probably wyoming or montana though to try to start a WISP |
| 01:06.13 | rotten777 | f2knight 863 |
| 01:06.34 | rotten777 | p3nguin i appreciate the help man you've already got me understanding the functioning of asterisk and the conf files |
| 01:06.42 | f2knight | rotten777, I still manage my WISP/ITSP in boca from all the way over hear :) |
| 01:06.51 | rotten777 | nice! |
| 01:07.18 | f2knight | I know I am coming in late, but i take it rotten777 you are just getting started? |
| 01:07.22 | rotten777 | i wish we could get some itsp's in our LEC |
| 01:07.27 | rotten777 | yeah i'm on day #3 |
| 01:07.30 | rotten777 | and ITSP #2 |
| 01:07.58 | p3nguin | rotten777: If you still have my example dialplan, it has a basic attendant configuration in it. |
| 01:08.16 | rotten777 | yeah i've bookmarked it lol |
| 01:08.42 | rotten777 | i've got to read piece at a time. i'm learning RouterOS and asterisk at the same time. |
| 01:09.16 | f2knight | We started with the routerboards moved to Ubuquity. |
| 01:09.50 | rotten777 | really? what are the advantages? |
| 01:10.11 | f2knight | performance! |
| 01:10.37 | rotten777 | oh oh oh sorry i didn't realize you said routerboards yeah for the high performance core routers i'm doing x86 builds |
| 01:10.58 | f2knight | I am able to actually get 100+Mbs over the air with them .. |
| 01:11.26 | rotten777 | woooow |
| 01:11.27 | rotten777 | mimo? |
| 01:11.49 | f2knight | are you using routeros for routing only? |
| 01:11.56 | rotten777 | yes i don't do wireless yet |
| 01:11.58 | f2knight | if so I would suggest looking at vyatta, |
| 01:12.06 | rotten777 | i'm setting up a 2km ptp shortly though |
| 01:12.30 | f2knight | Most of our clients are setup in point to multipoint |
| 01:12.51 | rotten777 | the hardware appliances? |
| 01:12.56 | f2knight | but our network is setup in more of a sonet mesh style. |
| 01:13.10 | f2knight | rotten777, actually if your doing x86 they have the os for download |
| 01:13.26 | rotten777 | yeah i've got a lot to learn when it comes to WISP stuff. |
| 01:13.34 | f2knight | you will get cisco quality out of the os. its actually been called the cisco killer. |
| 01:13.49 | rotten777 | the last ISP i was at was 8 years ago and it was simply dial-up/adsl |
| 01:13.49 | rotten777 | yeah |
| 01:13.53 | rotten777 | is it free? |
| 01:13.55 | f2knight | We started our wisp very small 2 clients and radios. |
| 01:14.03 | f2knight | yes Vyatta is free. |
| 01:14.08 | rotten777 | wow sweet thanks man |
| 01:14.18 | f2knight | np. |
| 01:14.43 | f2knight | I run a atom based box behind a few radios. |
| 01:14.52 | f2knight | and I do a neat trick for failover. |
| 01:15.14 | rotten777 | "Test Drive Vyatta Network OS" |
| 01:15.24 | rotten777 | you sure its free? |
| 01:15.34 | f2knight | if our main provider goes out, or the radio is down, I reroute everything over a ssh tunnel to another location that has a cable modem hooked up to it. |
| 01:15.52 | rotten777 | oh nice |
| 01:17.09 | f2knight | http://www.vyatta.com/downloads/vc6.3/vyatta-livecd_VC6.3-2011.07.21_i386.iso |
| 01:17.23 | f2knight | http://www.vyatta.com/downloads/vc6.3/vyatta-livecd_VC6.3-2011.07.21_amd64.iso |
| 01:17.55 | rotten777 | oh wow did I miss that? |
| 01:19.14 | f2knight | lol because I took the links from distrowatch and not the site :) |
| 01:19.25 | rotten777 | haha cool |
| 01:20.03 | Kobaz | yeah vyatta is free unless you want the certified hardware or support |
| 01:20.18 | hardwire | support |
| 01:20.44 | Kobaz | f2knight: ssh tunnel? why not openvpn |
| 01:20.54 | hardwire | no tap interface access? |
| 01:21.05 | Kobaz | could use tun |
| 01:23.32 | f2knight | Kobaz, I 'COULD' use openvpn but its a lot easier to just do SSH -L 80:localhost:80 remotebox |
| 01:24.09 | hardwire | -C |
| 01:24.33 | f2knight | Kobaz, I suppose there is a few ways to have done it, this just worked quickly with no software to install of configure. |
| 01:25.04 | Kobaz | i've done the ssh tunnel stuff before, it doesn't handle reconnections as gracefully as openvpn |
| 01:27.58 | f2knight | Kobaz, Luckily I have not had to use it much, but it is awesome when you don't need to install a software or configure it just to route some stuff over. 30 min later issues resolved and your back to working normally customer has little lag sometimes but other wise. Good to go. |
| 01:28.15 | f2knight | Kobaz, I might reconsider the VPN setup again though. |
| 01:30.56 | Kobaz | yeah |
| 01:30.59 | Kobaz | it is simple |
| 01:31.18 | Kobaz | could easily write a little perl script to routinely check the connectivity and restart the tunnel if needed |
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| 02:58.09 | dlisenby | Quick Question about Phantom extensions. I want to set up a voice mail box for a ring group. I've setup an extension but don't want to allow SIP or IAX authentication since the VM will be emailed. Any idea? |
| 02:59.18 | carrar | Just create a vmbox then |
| 03:00.38 | carrar | What extension is the "Phantom extensions"? |
| 03:00.47 | carrar | maybe use the same number for the VM |
| 03:01.30 | dlisenby | That's the issue. The GUI won't allow me to create an extension if I uncheck SIP and IAX. It requires an analog extension. I don't want any of them. |
| 03:01.45 | carrar | no gui in asterisk |
| 03:01.58 | dlisenby | I'm using Digium's GUI |
| 03:02.14 | carrar | I'm sorry |
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| 03:03.40 | carrar | Might try #asterisk-gui |
| 03:04.12 | carrar | Or dump the gui for the real power |
| 03:04.37 | dlisenby | I'm ok with the conf files. Just not sure where to code that. Point me to an example? |
| 03:04.46 | carrar | voicemail.conf |
| 03:04.57 | carrar | add a vmbox there |
| 03:05.48 | carrar | then add Voicemail to the end of your phantom exten w ith whatever vmbox you creat |
| 03:07.58 | carrar | Or dump the gui for the real power |
| 03:08.49 | dlisenby | ok.. thanks |
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| 03:20.23 | cstachris | hello |
| 03:24.47 | ChannelZ | hi |
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| 03:50.42 | Dovid | after upgrading to 1.8 when my phone gets a call I get numbe@IP. I know it can be disabled in the phone however I wanted to know if there is any way of "fixing this" in asterisk |
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| 03:58.17 | cstachris | Dovid, you can Set(CALLERID(num)=NUMBER) in the dialplan |
| 04:03.04 | Dovid | cstachris: is there any general setting for sip.conf that can do it? |
| 04:03.52 | cstachris | Dovid, sorry I'm a little bit rusty with my asterisk config files - especially with the new stuff in 1.8 |
| 04:04.40 | cstachris | i don't konw |
| 04:04.42 | cstachris | know |
| 04:05.30 | WIMPy | callerid= |
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| 04:57.07 | X-Rob | OK, so SIP Fax detection in 1.8 doesn't seem to actually work 8-\ |
| 04:57.21 | X-Rob | time to RTFM some more |
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| 05:14.08 | Dovid | WIMPy: where can I put that? Or are you saying to set it per peer ? (e.g. callerid=123456) |
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| 05:21.54 | fattsammy | Hello all, rookie here with a quick question for anyone who has a moment. |
| 05:23.32 | fattsammy | On the Definity G3 I used to admin, we had a thing called "cover paths." What is the asterisk terminology for this? |
| 05:27.22 | X-Rob | what does it do? |
| 05:28.43 | fattsammy | An extension doesn't pick up in x seconds, rolls call to another extension, or to a message that can then roll to another ring group, etc. |
| 05:33.01 | fattsammy | The end result desired is: call hits a ring group, if no one answers, a message is played, at end of message, another ring group is hit, if no answer, then to a voice mailbox. |
| 05:40.12 | kaldemar | fattsammy: timeout option in Dial will determine when to move to the next priority in the exten. |
| 05:42.26 | fattsammy | I thought about sending the call to a virtual extension's voicemail, but I don't know how to make the call go to the ring group after playing the message. |
| 05:43.48 | kaldemar | just put another priority after the message |
| 05:50.12 | fattsammy | ok, thanks for the tip. I will read up on that subject. |
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| 06:13.51 | fattsammy | So, I think what I will do is to set up multiple ring groups that contain the same extensions, let them ring through their timeouts, hit the virtual extension that initiates the playback of the gsm file, pass to the secondary ring group which finally (after its timeout) hands off to a preselected voice mail box. |
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| 06:22.31 | fattsammy | Thanks all, good morning. |
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| 06:29.36 | schmidts | good morning |
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| 07:19.54 | din3sh | is it possible to use asterisk as an E1 gateway for a polycom videoconferencing system? |
| 07:21.43 | irroot | din3sh could be tricky i suspect they data calls ?? ISDN dial up |
| 07:22.08 | din3sh | video calls |
| 07:22.27 | din3sh | data calls would be straight forward dialing a single channel |
| 07:22.52 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 07:22.55 | din3sh | is there any way to make asterisk use multiple available isdn channels for 1 particular call? |
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| 08:27.43 | *** join/#asterisk itguru (~itguru@unaffiliated/itguru) |
| 08:27.49 | itguru | greets the room |
| 08:28.24 | itguru | I had a rooted voip box dropped on my lap, and now I have to build my first asterisk box from scratch ... ouch |
| 08:31.16 | *** join/#asterisk Dovid (Dovid@office.mypbxmanager.net) |
| 08:34.24 | *** join/#asterisk jacc0 (~jacc0@D522448D.static.ziggozakelijk.nl) |
| 08:34.29 | jacc0 | good morning all |
| 08:45.05 | schmidts | morning jacc0 |
| 08:51.38 | ChannelZ | can't sleep. grrph |
| 08:52.03 | *** join/#asterisk RZero (~Mick_Love@gateway-hq-uk.oxygen8.com) |
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| 09:04.14 | *** join/#asterisk Rico29 (~rico@sar-b123.olm.fr) |
| 09:04.16 | Rico29 | hi ! |
| 09:05.24 | Rico29 | I've got a problem with an asterisk behind NAT. |
| 09:06.15 | *** join/#asterisk hetii (~Grzegorz@194.181.154.25) |
| 09:06.18 | hetii | Hello :) |
| 09:06.33 | Rico29 | when a do a call, the INVITE which is going from my asterisk to my voisp contains the private IP of my asterisk in the SDP owner field |
| 09:06.40 | Rico29 | hello hetii |
| 09:07.13 | Rico29 | i'm using asterisk 1.8 |
| 09:07.41 | Rico29 | can somebody tell me how I can modify this address ? I've put externip=<public ip address> in my sip.conf |
| 09:08.23 | Rico29 | should I add a "fromdomain" or something else ? |
| 09:09.11 | kaldemar | Rico29: you must have nat=yes and localnet [under general]. |
| 09:10.45 | Rico29 | ok |
| 09:10.59 | Rico29 | i already have nat=yes |
| 09:10.59 | hetii | I have strange issue with last asterisk, sometime the existing call is hangup here is log: http://pastebin.com/qgP8YVwk |
| 09:11.54 | hetii | its look like when 100142 answer call from 100125 the hangupcall is executed for 100118 |
| 09:12.16 | hetii | line 11 and 12 on log |
| 09:14.17 | kaldemar | hetii: can you reproduce that? that paste doesn't really give any reason for the hangup. try to get a hangup with sip debug enabled. |
| 09:15.10 | Rico29 | kaldemar > syntax for localnet is x.x.x.x/24 or x.x.x.x/255.255.255.0 ? |
| 09:15.18 | hetii | i can try |
| 09:15.23 | madduck | hello, what causes Asterisk to send a SIP re-invite? |
| 09:15.24 | madduck | X-asterisk-Info: SIP re-invite (Session-Timers) |
| 09:15.25 | ChannelZ | /24 is fine |
| 09:15.41 | madduck | after 15 mins, but the session expiry is set to 1800 |
| 09:15.42 | madduck | Session-Expires: 1800;refresher=uas |
| 09:16.05 | madduck | is it like DHCP where SIP tries to renegotiate as it approaches half-time? |
| 09:16.30 | *** join/#asterisk _omer (~omer@182.185.168.224) |
| 09:18.28 | madduck | I have this problem where on calls I receive (sipgate → asterisk → handset), after 14:45 Minutes, my asterisk sends a reinvite to sipgate |
| 09:18.38 | madduck | INVITE sip:017XCALLERX@217.116.117.7 SIP/2.0 |
| 09:18.49 | madduck | to which sipgate answers |
| 09:18.50 | madduck | SIP/2.0 100 Giving a try |
| 09:18.54 | madduck | SIP/2.0 420 Option Disabled |
| 09:19.08 | madduck | asterisk acknowledges that: |
| 09:19.09 | madduck | ACK sip:017XCALLERX@217.116.117.7 SIP/2.0 |
| 09:19.24 | madduck | and sipgate terminates the call |
| 09:19.24 | madduck | BYE sip:incoming@77.109.139.86 SIP/2.0 |
| 09:19.33 | madduck | this does not seem to happen on outgoing calls |
| 09:20.00 | madduck | oh, my asterisk responds to the BYE with |
| 09:20.00 | madduck | SIP/2.0 481 Call leg/transaction does not exist |
| 09:21.45 | hetii | can i set sip set debug on few peers on the same time ? |
| 09:21.53 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 09:22.08 | hetii | or it will switch to last one that i put ? |
| 09:22.14 | madduck | i think it switches |
| 09:22.23 | hetii | :( |
| 09:23.09 | madduck | turn it on, and use a text editor. ;) |
| 09:32.47 | *** join/#asterisk e7e5 (~rudenko@188.134.2.33) |
| 09:32.53 | kaldemar | or a little script to parse the wanted messages based on an ip address. |
| 09:39.05 | *** join/#asterisk pdtpatrick1 (~pdtpdt@mainstwan.farheap.com) |
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| 09:39.05 | *** mode/#asterisk [+o pabelanger] by niven.freenode.net |
| 09:44.57 | wdoekes2 | madduck: see the session-timers option in sip.conf, you can disable them for that particular peer |
| 09:46.26 | hetii | now i got that: http://pastebin.com/e6eT4VS2 so its break again the connection |
| 09:46.49 | madduck | wdoekes2: yeah, I found that and I will try it, but also inform sipgate. |
| 09:47.05 | madduck | I would love to find the real cause, not just fight symptoms ;) |
| 09:48.43 | hetii | so is its look like the trunk send bye sip message |
| 09:51.07 | *** join/#asterisk ean (~ean@unaffiliated/ean) |
| 09:53.03 | wdoekes2 | <PROTECTED> |
| 09:53.06 | wdoekes2 | <PROTECTED> |
| 09:53.31 | madduck | wdoekes2: so it does send at half-time, eh? |
| 09:53.52 | *** join/#asterisk chazzam (~chazz@173-24-236-90.client.mchsi.com) |
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| 09:55.18 | *** join/#asterisk Darksyre (~Darksyre@70.103.156.179) |
| 09:55.49 | *** join/#asterisk ocx (c27e0e65@gateway/web/freenode/ip.194.126.14.101) |
| 09:56.19 | ocx | hello, can someone point me to some documentation on integrating asteriskwith some analog pbx connected to analogue phones? |
| 09:56.30 | ocx | purpose is to allow analog phones to use voip network provided by asterisk |
| 09:56.35 | ocx | is it a tedious work? |
| 09:56.54 | ocx | does the analogue pbx need to be compatible or it does work with any plain pbx? |
| 09:56.55 | ocx | thanks |
| 10:00.36 | ChannelZ | depends on how you want to connect the two |
| 10:00.41 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
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| 10:01.45 | *** part/#asterisk wesphillips (~wphill04@192.160.117.137) |
| 10:01.46 | ChannelZ | If you want to basically use the old PBX as the analog interface for all the phones, does it have a T1/E1 interface as well (which you could use to bridge it to Asterisk)? |
| 10:02.04 | kaldemar | ocx: it doesn't matter what the pbx is, as long as it has an interface that can be used to connect it to asterisk. |
| 10:03.36 | madduck | wdoeskes2: I set session-timers=accept now and we'll see |
| 10:03.43 | madduck | i am also telling sipgate with debug info though |
| 10:11.05 | *** join/#asterisk enoch (~enoch@unaffiliated/enoch) |
| 10:11.07 | enoch | hi all |
| 10:12.19 | *** join/#asterisk godmachine-x6 (~godmachin@unaffiliated/godmachine-x6) |
| 10:28.37 | *** join/#asterisk joobie (~joobie@unaffiliated/moo0o0ooo00o0o0o) |
| 10:29.08 | joobie | hey guys.. is it possible to register a softphone and share the extension at the same time as having a hardphone on the same extension? |
| 10:29.15 | joobie | so if you ring the extenion both ring |
| 10:29.27 | joobie | or if the softphone is on, the hardfone doesnt ring? |
| 10:29.31 | joobie | what's the standard way to do this? |
| 10:29.47 | lanning | both have their separate channels |
| 10:30.08 | lanning | then in the dial plan, the Dial() statement can take multiple channels |
| 10:30.11 | kaldemar | two devices and some logic in dialplan. |
| 10:30.38 | lanning | first to answer, gets the call |
| 10:31.28 | joobie | i have an extension appear on the phone |
| 10:31.33 | joobie | like 4000 for example |
| 10:31.40 | joobie | the user knows this extension as their own |
| 10:31.56 | joobie | im hoping i can make 4000 appear on the softphone so they think it's their own same extension |
| 10:31.58 | joobie | can this be done? |
| 10:32.29 | lanning | the "appearance" is just a label on the phone. |
| 10:32.42 | Darksyre | How many channels can you set up for one extension, i.e. can I have a softphone, hardphone and cell phone all on the same extension? |
| 10:32.57 | lanning | the hard phone will be a channel |
| 10:33.12 | lanning | the SIP softphone will be a different channel |
| 10:33.21 | kaldemar | depends on the phone you use. with snom phones, it used to be possible (don't know the current status) to send a certain message that defined what the phone had in its display. |
| 10:33.47 | lanning | in the dial plan, when they dial 4000, you exec Dial(<channel1>&<channel2>) |
| 10:34.10 | kaldemar | Darksyre: no limit in extension usage, but if you mean a peer, a single peer per single phone. |
| 10:35.19 | kaldemar | Darksyre: let me correct that, there is a length limit in a dialplan command, around 250 characters IIRC. |
| 10:36.21 | Darksyre | kaldemar: I have a client who would like to have his cell and his hardphone both ring at the same time, similar to the scenario joobie was mentioning, is that possible as well? |
| 10:37.12 | Darksyre | And a different client mentioned wanting all 3 |
| 10:37.14 | kaldemar | Darksyre: yes. the downfall is that when one answers, the other sees a missed call. |
| 10:37.38 | kaldemar | Darksyre: you can have as many as you want to. |
| 10:37.50 | lanning | and then there are the competing voicemail systems |
| 10:38.16 | Darksyre | kewl, thank you... still getting my feet wet with Asterisk |
| 10:38.45 | kaldemar | lanning: they can all be configured with the same mailbox. |
| 10:39.03 | lanning | not your cell phone and the hard phone... |
| 10:39.19 | lanning | unless you are the cell carrier |
| 10:40.04 | kaldemar | cell phone voicemail is the problem. better deactivate it and use asterisk for voicemail. |
| 10:40.05 | lanning | internal end points are easy. it's when you forward outside of your system that you have to deal with stuff like that. |
| 10:41.04 | lanning | and if it is active and the cell phone is off or out of area, the call will go to the cell's voicemail before the user at the hard phone has time to pick up the call. :) |
| 10:41.45 | Darksyre | Ahh, thank you very much... that would be a bad thing |
| 10:42.49 | lanning | in most of the scenarios, the voicemail on the cell can't be turned off, because this new number being forwarded is an additional number (ie. a business number being forwarded to a personal cell...) |
| 10:43.32 | lanning | personal calls would have no voicemail to go to... |
| 10:44.07 | Darksyre | I don't see it being a problem with this particular client, but down the road that can be a definate issue, so just tell them the cell phone is now a work phone... or get them to get a Cisco 7921 |
| 10:46.21 | Darksyre | One other question, as I know Asterisk handles both and I keep hearing things from both sides of the fence... would you recommend SIP or SCCP? |
| 10:46.32 | kaldemar | SIP |
| 10:47.00 | *** join/#asterisk godmachine-x6 (~godmachin@unaffiliated/godmachine-x6) |
| 10:47.09 | Darksyre | kaldemar: why? |
| 10:47.38 | kaldemar | more features and support. |
| 10:47.55 | Darksyre | Thank you |
| 10:49.24 | Darksyre | I am studying up on Asterisk as my company has been with a hosting client who has been pushing SCCP and when I started realizing when somethng went down, and clients called me upset that they had no phones, I couldn't fix it... so I'm trying to understand and build a system I can give my clients full support on |
| 10:50.25 | *** join/#asterisk BuenGenio (~Gene@90.172.132.157) |
| 10:51.11 | Darksyre | So I'm reading the Second and Third Edition Asterisk books and trying to glean as much as I can hear as well |
| 10:52.57 | *** join/#asterisk FainaUkraina (~Gene@90.172.132.157) |
| 11:00.03 | ocx | how can asterisk know about the extensions configured on an anlog pbx, how can it route an incoming call and know about the dialplan of the analog pbx? |
| 11:00.22 | ocx | Phone line > Asterisk > analog pbx |
| 11:00.48 | ocx | consider this scenario where a call enters on the phone line and needs to be routed to extenion 500 defined on the analog pbx |
| 11:00.57 | kaldemar | ocx: you configure it in asterisk's dialplan. |
| 11:01.26 | ocx | i only have 1 FXO/FXS connection from asterisk to analogue pbx |
| 11:01.29 | ocx | can this be accomplished? |
| 11:03.56 | kaldemar | if the analog pbx side has the FXO. |
| 11:04.50 | kaldemar | dialing out will be a pain in the ass tough, since you can't get a dialed number through to the FXS side. |
| 11:06.29 | *** join/#asterisk singler (~singler@81-7-123-162.static.zebra.lt) |
| 11:08.44 | ocx | what if i connect the CO port of the analog directly into the FXS of the asterisk? |
| 11:08.54 | ocx | so ppl would should CO before dialing :/ ?> |
| 11:09.01 | ocx | i mean users* |
| 11:10.43 | kaldemar | the CO port being what? |
| 11:13.27 | ocx | FXO of the analog |
| 11:13.29 | ocx | pbx |
| 11:15.59 | kaldemar | what's your point? |
| 11:16.27 | ocx | can i pm you? |
| 11:16.51 | kaldemar | no. |
| 11:35.53 | *** join/#asterisk zewelor (~x@237-mo5-9.acn.waw.pl) |
| 11:38.46 | zewelor | hi i got some strange problem, there are connections sometimes that was done by noone, like yesterday at 5 am got connection in logs but noone did it and after anwsering it its only silence, any ideas what can be wrong or how to debug it ? |
| 11:41.13 | *** join/#asterisk rgsteele (~rgsteele@apo.aweber.com) |
| 11:45.10 | *** join/#asterisk cstachris (~chrismylo@202.182.147.24) |
| 11:51.01 | Gugge | zewelor: its magic |
| 11:51.07 | Gugge | or someone actually did it |
| 11:51.32 | Gugge | find out what IP did it, and what hardware is at that IP :) |
| 11:51.46 | *** join/#asterisk billmania (~bill@38.98.130.98) |
| 11:52.12 | *** part/#asterisk billmania (~bill@38.98.130.98) |
| 11:57.49 | zewelor | in cdr logs i only got account |
| 11:58.38 | zewelor | but that account is used by my voip gateway and its connected to my phone and i was sleeping or at least i hope i was sleeping :\ |
| 12:00.23 | Darksyre | gremlins? |
| 12:01.30 | *** join/#asterisk DelphiWorld (~VoIpGuy@openvpn/user/DelphiWorld) |
| 12:01.38 | DelphiWorld | Hey |
| 12:01.42 | DelphiWorld | svn co http://svn.digium.com/svn/dahdi/linux-complete/trunk dahdi |
| 12:01.45 | DelphiWorld | is not working for me |
| 12:01.48 | DelphiWorld | is frosen |
| 12:01.56 | zewelor | as i read at google i found some ppl wrote problems like that but without any solution |
| 12:02.12 | zewelor | i read some about queue manager or some lost connections ? |
| 12:02.27 | DelphiWorld | how do i checkout dahdi ? |
| 12:04.48 | kaldemar | DelphiWorld: works here. you can also get a specific release from http://downloads.asterisk.org/pub/telephony/dahdi-linux-complete/ |
| 12:05.16 | DelphiWorld | Thank you kaldemar |
| 12:08.34 | DelphiWorld | hahahaha kaldemar IPV6! |
| 12:09.38 | DelphiWorld | kamdownload/svn was not working due to ipv6. |
| 12:10.04 | *** join/#asterisk timahvo1 (~rogue@41.215.1.35) |
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| 12:40.10 | RZero | Hi Guys I am after some Asterisk realtime help |
| 12:41.29 | leifmadsen | ~ask |
| 12:41.29 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 12:41.44 | RZero | I have static sip.conf working great, inbound calls work, using switch in the extensions.conf, Im just having problems with Dialling |
| 12:43.12 | zamba | can someone recommend a mini-pci gsm card for asterisk? |
| 12:43.17 | RZero | SIP/MyIP-00000004", "SIP/Ox8-Sip-Sw2/0121" does not work it I get the error No such host: Ox8-Sip-Sw2 |
| 12:43.50 | p3nguin | Did you create a peer for it? |
| 12:43.57 | RZero | yes |
| 12:44.04 | RZero | its in the db table |
| 12:44.16 | p3nguin | Does "sip show peers" show it? |
| 12:45.48 | RZero | hmm no it does not gah brb :| |
| 12:46.56 | leifmadsen | RZero: rtcache=yes likely |
| 12:47.27 | leifmadsen | sorry... rtcachefriends=yes |
| 12:47.35 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-apqwxbwprpkoqwmj) |
| 12:48.15 | leifmadsen | although honestly it should try and do a lookup in the database when you do the Dial(). Are you using ODBC? You should enable SQL statement logging on the DB and look at that log and see what is being passed over for the lookup (it should be a SELECT statement) |
| 12:50.23 | *** join/#asterisk Naikrovek (~mbluth@unaffiliated/naikrovek) |
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| 12:58.27 | RZero | Im using mysql, working now. few things were incorrect in the db, is there a better way of using peers details rather than static sip.conf ? I can not get sippeers work at all |
| 12:58.46 | RZero | realtime static * |
| 13:04.09 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 13:04.09 | *** mode/#asterisk [+o malcolmd] by ChanServ |
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| 13:04.55 | *** mode/#asterisk [+o malcolmd] by ChanServ |
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| 13:18.17 | *** part/#asterisk zewelor (~x@237-mo5-9.acn.waw.pl) |
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| 13:20.23 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 13:21.04 | adnc | hello, since yesterday I can't get calls from some countries on my sipgate account. there has been no changes on my asterisk and I do get this error when someone calles in or out |
| 13:21.10 | adnc | Failed to authenticate on INVITE to handle response invite sipgate |
| 13:21.28 | adnc | this only happens to foreign country calls |
| 13:21.59 | wdoekes2 | adnc: sip set debug on |
| 13:22.17 | wdoekes2 | and watch the differences between the INVITE packets (and the peer IP:port) |
| 13:24.57 | adnc | wdoekes2, thank you very much. but what exactly do I look for? |
| 13:25.59 | adnc | [Sep 13 15:22:46] NOTICE[30962]: chan_sip.c:17863 handle_response_invite: Failed to authenticate on INVITE to '"Heybeli" <sip:4316088@sipgate.de>;tag=as3569b6ef' |
| 13:26.27 | adnc | X-Asterisk-HangupCause: Call Rejected |
| 13:26.27 | adnc | X-Asterisk-HangupCauseCode: 21 |
| 13:26.27 | adnc | Content-Length: 0 |
| 13:26.30 | adnc | and this |
| 13:27.24 | wdoekes2 | do they both get a 401/200? no.. one gets a 401 and then a 403, right? or? |
| 13:27.52 | ocx | i need some documentation to implement my traditional pbx with asterisk box, purpose is to keep extensions connected to traditional pbx and use asterisk for voip routing (incoming calls and outgoing calls to the internet) |
| 13:27.57 | ocx | please advise |
| 13:28.02 | RZero | how do I use sippeers in realtime, I can only get sip.conf working from the db but I have to reload asterisk if I made changes. When I enable sippeers in extconfig.conf and point it to the db. I make a test calls and nothing shows on the cli and phone just says calling |
| 13:28.03 | ocx | i cant find any resource on the internet |
| 13:28.17 | adnc | wdoekes2, I need to find out how to look for this to answer your question |
| 13:30.23 | *** join/#asterisk cerberus_za (~coert@8ta-151-99-103.telkomadsl.co.za) |
| 13:30.47 | ocx | any ebook |
| 13:30.49 | ocx | anything |
| 13:30.51 | ocx | i need resources... |
| 13:33.15 | *** join/#asterisk JustinCampbell (~justinCam@74-94-59-225-Philadelphia.hfc.comcastbusiness.net) |
| 13:34.18 | jaytee | ocx, what brand of traditional pbx do you have? |
| 13:34.23 | *** join/#asterisk GreatSUN (~greatsun@188.20.12.162) |
| 13:34.25 | ocx | Panasonic |
| 13:34.25 | RZero | I have followed this guide http://www.voip-info.org/wiki/view/Asterisk+RealTime+Sip yet I can not get it to work. |
| 13:34.34 | *** join/#asterisk McBoingBo (~McBoingBo@mail.hrsg.ca) |
| 13:34.57 | McBoingBo | Xlite has been dissapointing these days, what softphone clients do you guys use? free or not |
| 13:35.02 | McBoingBo | BTW, good morning! |
| 13:35.20 | adnc | wdoekes2, yes one gets 401 and than 403 |
| 13:36.29 | ocx | jaytee: |
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| 13:36.42 | *** join/#asterisk azv4 (~azv4@host-66-202-7-218.pit.choiceone.net) |
| 13:38.58 | JustinCampbell | McBoingBo: i use Telephone on OSX |
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| 13:41.06 | wdoekes2 | adnc: which direction? I assume your provider will send a 403 when you're not allowed to dial to foreignland (or when you're using the wrong international calling prefix) |
| 13:41.41 | adnc | wdoekes2, yes, but the caller has the same problem. and both numbers do belong to sipgate |
| 13:41.49 | adnc | this way we call for free to each other |
| 13:41.50 | McBoingBo | JustinCampbell: yeah I forgot to mention its for Winders |
| 13:42.35 | wdoekes2 | you're not calling for free if your call is not coming through ;) |
| 13:42.50 | adnc | wdoekes2, untill yesterday it was |
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| 13:45.11 | wdoekes2 | you probably want to talk to sipgate, tell them you get a 403, and supply the request URI (INVITE <this>) that you're sending |
| 13:50.23 | RZero | when I type realtime load sippeers name "Ox8-Sip-gw1" it shows nothing also I get no errors |
| 13:53.19 | adnc | wdoekes2, how can I dial a sip-id which is given to me by the provider? |
| 13:53.53 | adnc | for example my is 435959, how can someone call this sip id with his phone? the url would be 435959@sipgate.de |
| 13:54.28 | WIMPy | Sipgate stopped accepting external calls many years ago. |
| 13:54.51 | adnc | no not external calls |
| 13:54.54 | adnc | both are on sipgate |
| 13:56.14 | WIMPy | You can dial account numbers unless you have activated automatic area code, |
| 13:56.16 | WIMPy | . |
| 13:56.35 | adnc | WIMPy, ok thank you |
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| 14:15.33 | lucifurr | I have an architecture/design question. My dialplan is written in lua and I've built an extension (in C++ using OCCI) for calling oracle stored procs and returning result in lua table to the dialplan. It works very well. The problem I have is that I need to create a connection pool (perhaps a resource module) and somehow share the connection pool with all of the call threads. Is this the proper way to create a connection pool or is there a better way? H |
| 14:15.33 | lucifurr | I share the global pool with the threads? Thread storage? |
| 14:16.42 | adnc | how can i show the current calls in the command line? |
| 14:18.11 | p3nguin | core show channels |
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| 14:19.44 | p3nguin | <WIMPy> Sipgate stopped accepting external calls many years ago. <---- What is the meaning of this? |
| 14:20.08 | WIMPy | Guest calls. |
| 14:20.36 | p3nguin | I still don't understand what you're saying? You mean anonymous SIP? |
| 14:20.44 | WIMPy | yes |
| 14:20.45 | p3nguin | s/?/./ |
| 14:20.52 | p3nguin | Oh, okay. Got it. |
| 14:21.25 | p3nguin | I thought you were talking about not accepting calls, period. And I was really confused because I have a DID with sipgate and it's still taking calls. |
| 14:22.36 | p3nguin | I've never tried dialing any sipgate URIs, so I don't know anything about the acceptance or denial of that. |
| 14:24.51 | adnc | I now dialed the account number at sipgate and it goes directly to the voicemailbox of sipgate |
| 14:25.51 | adnc | i've someone whith an sipgate account in england and since two days i can't reach him anymore from his british number which i route in asterisk through my sipgate account |
| 14:26.13 | adnc | the same happens to the caler aswell |
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| 14:26.53 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
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| 14:32.24 | RZero | is it possible to look up sip peers details with out using realtime static sip.conf ? sippeers does not seem to work. |
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| 14:33.11 | RZero | the peers do not register as we trunk the calls directly to them |
| 14:37.35 | *** join/#asterisk kannan (kann@14.99.120.62) |
| 14:38.03 | Rico29 | hi again |
| 14:38.15 | Rico29 | I have a problem with sip channels not closed properly |
| 14:38.23 | Rico29 | here's what I can see in my logs : http://pastebin.com/0GZZ9E06 |
| 14:38.37 | Rico29 | has anyone ever met this problem ? |
| 14:39.16 | kannan | hello, when dialling a call thru a SIp service provider, i am using a simultaneous Dial option to call 4 PSTN number. How can find the number of the party that is the called party after the call is answered? |
| 14:39.46 | p3nguin | sip show channels |
| 14:40.38 | p3nguin | Look for any that have a codec listed under Format. |
| 14:40.52 | kannan | p3nguin , inside the priorities , so as to be able to pass it an AGI . The CDR(dstchannel) (and all SIP channels are having a format that does not show the number.. |
| 14:41.17 | kannan | ok one sec i will try this |
| 14:41.34 | Rico29 | p3nguin > http://pastebin.com/UgADqNVs |
| 14:41.37 | p3nguin | Maybe the info in core show channels could also be useful. Look for the one that says State is Up. |
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| 14:42.07 | kannan | ok there is one that has User/ANi , the number shows up there.. |
| 14:42.28 | kannan | user ?ANR sorry |
| 14:42.36 | kannan | how to get that as a Chanel variable? |
| 14:42.56 | Rico29 | p3nguin > core show channels : http://pastebin.com/fZDxrkrQ |
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| 14:43.24 | Katty | morning |
| 14:44.09 | p3nguin | hi |
| 14:44.26 | Rico29 | hi |
| 14:45.54 | Rico29 | p3nguin > do you need any other informations ? |
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| 14:50.23 | kannan | p3nguin, thanks is shows up under 'sip show peers' ; do i have to write into a file with System command and then retrieve ; or is the User / ANR for the called party available as a Channel System Var? |
| 14:51.56 | *** join/#asterisk cerberus_za (~coert@8ta-151-84-194.telkomadsl.co.za) |
| 14:55.17 | Rico29 | I'gve put many informations there : https://issues.asterisk.org/jira/browse/ASTERISK-18533 |
| 14:55.28 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 15:01.08 | *** join/#asterisk scubes13 (~scubes13@216.218.99.249) |
| 15:02.00 | kannan | p3nguin, soory for the wrong statement prevoiusly, the USER / ANR shows up under the sip show channels. It truncates to 10 chars (but i need more , including the dialprefixes) . IS this value available as a system variable inside the diaplan ? |
| 15:04.10 | *** join/#asterisk tyrrexrrg (~roger@190.147.143.151) |
| 15:04.30 | *** join/#asterisk cardy (~cardy@host65-16-97-138.birch.net) |
| 15:04.57 | kannan | ok ,i get the dial pefixes from EXTEN itself, but the SIP show channels output truncates the 11th digit of the called party's phone number, how can i get the 10 digit into a channel variable when dialling thru a SIP provider |
| 15:06.22 | *** join/#asterisk navaismo (~navaismo@fixed-203-96-202.iusacell.net) |
| 15:07.09 | navaismo | good morning |
| 15:08.04 | *** part/#asterisk sekil (~sekil@78.24.104.73) |
| 15:11.00 | kannan | are there any sip setting that will ask the Sip service to use the called number in the SIP channel ? |
| 15:11.42 | *** join/#asterisk mindCrime_ (~chatzilla@rrcs-70-62-112-146.midsouth.biz.rr.com) |
| 15:11.54 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 15:11.56 | kannan | actually the question is : how to specifically get Asterisk to include the Called Number inside the SIP channel |
| 15:15.08 | navaismo | dont understand |
| 15:15.27 | treborsux | What would cause this situation? A user calls an outside number with a dahdi trunk. SHe hears 2 rings the party on the other end answers but one of the rings continue. |
| 15:25.20 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 15:27.02 | *** join/#asterisk irroot (~irroot@197.173.85.208) |
| 15:36.28 | irroot | evening folks |
| 15:36.53 | anonymouz666 | irroot: I am following the app_queue review board :-) |
| 15:37.16 | irroot | thx ill dbl check it now been in car for 6hrs |
| 15:37.37 | irroot | found a problem last night with it |
| 15:38.10 | anonymouz666 | lock stuff? |
| 15:38.37 | irroot | yeah there was one that was needed still ive added it back |
| 15:40.54 | anonymouz666 | heh, that is a very sensible work. if I understand correctly, you need to take care to fix one part without broke another |
| 15:41.31 | irroot | that is the theory do no evil |
| 15:45.56 | kannan | is the system variable BRIDGEPEER only for PRIs? it show up blank inside my sip calls (i am calling pstn number thru a voip service) |
| 15:56.28 | *** join/#asterisk senator (lebbeous@nox.esilibrary.com) |
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| 16:03.51 | *** join/#asterisk cusco (~tralala@a79-168-174-232.cpe.netcabo.pt) |
| 16:03.56 | cusco | hello folks |
| 16:06.32 | *** part/#asterisk DelphiWorld (~VoIpGuy@openvpn/user/DelphiWorld) |
| 16:06.47 | senator | hello all. using call files for the pbx_spool module, and an AEX400 series card with 1 usable FXO trunk. any time i place more than one call file at a time into the spool directory beginning with the line "Channel: DAHDI/1/NNNNNNN", asterisk tries to place calls _at the same time_ for each file |
| 16:06.57 | senator | surely that's not the expected behavior, is it? |
| 16:08.09 | navaismo | if you dont set the time in the future, the spool process inmediatly |
| 16:09.19 | Freeaqingme | ~freepbx |
| 16:09.19 | 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 |
| 16:10.56 | senator | navaismo: ok, so supposing you have fifty odd calls you want to make, and you have no idea how long each will take to complete, and you want to put them all into the spool at once, pbx_spool can't try them one after another? |
| 16:11.36 | navaismo | you can set the retry time and maximuns retries |
| 16:12.23 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:acac:95:144d:f798) |
| 16:15.07 | senator | and you'd have to set them pretty high? if fifty calls might take eight hours, each call file would have to be prepared with retry time and maximum retries that multiply to eight hours? |
| 16:17.34 | *** join/#asterisk azv4 (~azv4@host-66-202-7-218.pit.choiceone.net) |
| 16:18.00 | navaismo | i think is better if you use an script for that |
| 16:19.18 | senator | pbx_spool is a rotten misnomer then. spooling implies serialization. ok thanks. |
| 16:19.25 | *** join/#asterisk bbryant (~brett@c-174-56-132-225.hsd1.sc.comcast.net) |
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| 16:20.07 | cusco | what is the url for that sip bandwidth calculator' |
| 16:20.26 | *** join/#asterisk SuperNull (~SuperNull@24-148-101-238.ip.mhcable.com) |
| 16:21.40 | SuperNull | i need help :-( with DTMF tones not working properly.. specifically i have inbound calls coming from a sip provider.. that we then transfer to other servers.. my ultimate question is.. how to make sure its all compatible all the way through. |
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| 16:26.08 | *** join/#asterisk Eitan (~Eitan@99.22.192.148) |
| 16:27.21 | Eitan | <PROTECTED> |
| 16:27.56 | Qwell | disk image? dd. |
| 16:28.22 | Eitan | figured something like that would work... any expereince using it? |
| 16:28.32 | Qwell | it's not difficult |
| 16:28.47 | Eitan | any specific one you like? |
| 16:28.54 | Qwell | any specific what? |
| 16:29.00 | *** join/#asterisk anonymouz666 (~anonymouz@189.25.65.185) |
| 16:29.07 | navaismo | clonezilla |
| 16:29.27 | paulc | SuperNull: RFC2833 all the way! ;-) |
| 16:29.48 | Eitan | thanks navaismo |
| 16:32.22 | *** part/#asterisk RZero (~Mick_Love@gateway-hq-uk.oxygen8.com) |
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| 16:46.37 | cusco | what viable GSM interface would you recommend to use with asterisk? We are looking for something that can handle about 70 concurrent calls (70 SIM slots) |
| 16:47.16 | *** join/#asterisk voxter (~hardcore@macpro.daytonhome.voxter.net) |
| 16:47.39 | irroot | used a orrion E1 channel bank and it was usefull mapping channel to sim |
| 16:48.14 | Freeaqingme | <PROTECTED> |
| 16:48.14 | 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 |
| 16:48.18 | irroot | some CB's dont they route the channels internally so depends on needs |
| 16:48.37 | irroot | Freeaqingme do the same with trixbox |
| 16:49.50 | Freeaqingme | irroot, nah, my employer is asking me to fix the telephony |
| 16:49.59 | Freeaqingme | but some moron decided to go for freepbx a long time ago |
| 16:50.01 | Freeaqingme | so I'm completely lost |
| 16:50.18 | irroot | ~trixbox <- |
| 16:50.21 | *** join/#asterisk kraptv (ryan@unaffiliated/kraptv) |
| 16:50.25 | *** join/#asterisk coppice (~chatzilla@116.92.31.241) |
| 16:50.29 | irroot | ~trixbox |
| 16:50.30 | infobot | [trixbox] SH1TB0X. Basically a CRAPPY, closed-source distro. STAY AWAY! |
| 16:50.37 | Freeaqingme | hehe |
| 16:50.49 | irroot | Freeaqingme dude format reinstall :P |
| 16:50.59 | Freeaqingme | yeah, lets do it overnight |
| 16:51.05 | Freeaqingme | :P |
| 16:51.24 | irroot | i got it down to 20min from CSV |
| 16:52.08 | Freeaqingme | neat |
| 16:52.17 | *** part/#asterisk senator (lebbeous@nox.esilibrary.com) |
| 16:52.44 | *** join/#asterisk d_preston215 (~chatzilla@173-12-4-137-panjde.hfc.comcastbusiness.net) |
| 16:52.46 | irroot | its a flash disk on a decent box fdisk / format /install < 10m then ready to go |
| 16:55.08 | kraptv | I'm a little troubled - upgraded from 1.6.2.10 (using macports spec file as a basis) to 1.8.3.3-1ubuntu1 ... pretty much everything works but am getting much less activity in rasterisk even when I set the verbosity high. (I used to see the applications being executed, SIP calls being received and dropped, etc..) - is there something I need to enable to bring that activity monitor back? |
| 16:55.40 | *** join/#asterisk Devon_ (~chatzilla@63.214.236.169) |
| 16:57.08 | kraptv | can't seem to find anything obvious to indicate this - Master.csv in the cdr-csv shows the active application at the time the call was dropped, and writes out accordingly... |
| 16:58.50 | kraptv | I would probably care less if it weren't for one of my SIP providers working flawlessly and another only working in half-duplex. (sounds like a classic SIP behind NAT problem, ehh?) |
| 17:06.59 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 17:12.44 | *** join/#asterisk floh79 (~quassel@62.53.224.37) |
| 17:13.09 | floh79 | Hi. |
| 17:13.17 | floh79 | I have a problem with extension. |
| 17:13.40 | floh79 | I have entries like exten => _XXXX.,1,Playback(conf-placeintoconf) |
| 17:14.06 | floh79 | But, if I call from a SIP-Client with 2000@IP-Address I always get such error: |
| 17:14.23 | floh79 | [Sep 13 19:08:47] NOTICE[25443] chan_sip.c: Call from '' to extension '2000' rejected because extension not found. |
| 17:14.28 | floh79 | Whats wrong here? |
| 17:14.55 | WIMPy | _XXXX. requires at leas 5 digits. |
| 17:15.09 | WIMPy | Remove the . or replace it byt a !. |
| 17:15.47 | floh79 | I even tried with 5 digits. |
| 17:15.51 | cusco | what viable GSM interface would you recommend to use with asterisk? We are looking for something that can handle about 70 concurrent calls (70 SIM slots) |
| 17:16.22 | floh79 | chan_sip.c: Call from '' to extension '20000' rejected because extension not found. |
| 17:17.12 | WIMPy | floh79: Are you hitting the correct context? |
| 17:17.26 | floh79 | WIMPy: How can I know that? |
| 17:17.28 | WIMPy | cusco: 2N Stargate or Teles i.pbx |
| 17:17.47 | *** join/#asterisk moy (~moy@173.239.155.74) |
| 17:17.52 | floh79 | WIMPy: I heard I should use bbb-voip or bigbluebutton |
| 17:18.33 | WIMPy | floh79: Check your configs or upgrade to a version that tells you in the error message. |
| 17:18.45 | WIMPy | floh79: Err, what? |
| 17:19.13 | floh79 | WIMPy: I'm using bigbluebutton on server this is a conference system. |
| 17:19.34 | WIMPy | Never heard of that. |
| 17:19.38 | floh79 | WIMPy: Its possible to call a conference by VoIP. But How can I tell Linphone which Context to use? |
| 17:19.57 | WIMPy | sip.conf context= |
| 17:20.30 | floh79 | argh... |
| 17:21.34 | floh79 | WIMPy: Great... now I hear somewhat. :) |
| 17:21.39 | floh79 | WIMPy: Thank you very much! |
| 17:21.46 | WIMPy | something |
| 17:22.12 | floh79 | WIMPy: So asterisk-server only use one context, right? |
| 17:22.14 | *** join/#asterisk JustinCampbell (~justinCam@74-94-59-225-Philadelphia.hfc.comcastbusiness.net) |
| 17:22.22 | WIMPy | hopefully not. |
| 17:22.45 | WIMPy | contexts are probably the most important security feature. |
| 17:22.58 | floh79 | WIMPy: I see. |
| 17:23.50 | WIMPy | Unless you don't do anything else on that server, except for hsting conferences, you should have different contexts for different (types of) peers. |
| 17:25.17 | floh79 | WIMPy: Ok. Just one remaining question before going further in documentations. |
| 17:25.49 | floh79 | WIMPy: Who decided which context is used? Asterisk server or VoIP-Clients? |
| 17:26.31 | WIMPy | Asterisk with the exception of IAX clients, they can be able to provide a context. |
| 17:27.02 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 17:27.14 | floh79 | WIMPy: Ok, thank you very much. |
| 17:27.14 | WIMPy | Security features that are controlled by the user would be pretty pointless. |
| 17:27.23 | floh79 | WIMPy: Sure. :) |
| 17:29.19 | floh79 | Well... have a nice day! :) |
| 17:29.20 | floh79 | cu |
| 17:52.24 | p3nguin | What the heck... CentOS doesn't have mmv nor a package for it? |
| 17:53.14 | *** join/#asterisk moy (~moy@CPE002719f00364-CM0026f3a1701d.cpe.net.cable.rogers.com) |
| 17:53.17 | navaismo | what is mmv? |
| 17:53.39 | p3nguin | useful |
| 17:53.56 | p3nguin | mmv - move/copy/append/link multiple files by wildcard patterns |
| 17:54.00 | cusco | WIMPy: what are the prices on those? they're not displayed on the website unless I register |
| 17:54.13 | navaismo | oh thx |
| 17:55.05 | *** join/#asterisk m_tadeu (~quassel@89.180.229.155) |
| 17:57.22 | p3nguin | I guess it is in EPEL, but for some reason, that repo is not in CentOS by default. |
| 18:00.19 | *** join/#asterisk scubes13 (~scubes13@rrcs-70-60-211-241.midsouth.biz.rr.com) |
| 18:00.50 | leifmadsen | that's because it's not a CentOS repo |
| 18:00.57 | leifmadsen | just like DagWieers isn't |
| 18:01.26 | p3nguin | But it is an RHEL-related repo, so I would have expected it would be included, just like extras repo is. |
| 18:01.57 | p3nguin | No problem, though... I'll just install the repo and be on my way. |
| 18:02.46 | anonymouz666 | leifmadsen: you will talk about inbound call centers at astricon? |
| 18:02.56 | leifmadsen | yes |
| 18:03.17 | leifmadsen | p3nguin: don't think EPEL is included in Fedora by default either |
| 18:03.18 | anonymouz666 | and what was your presentation in 2010 astricon? |
| 18:03.51 | p3nguin | As part of the fedora project, that actually surprises me. |
| 18:04.45 | anonymouz666 | the most interesting stuff I saw from astricon so far was the Qwell presentation and yours about CC |
| 18:09.24 | leifmadsen | anonymouz666: http://www.astricon.net/2010/confDescriptions.aspx?t=PS#PS-07 |
| 18:09.37 | leifmadsen | anonymouz666: although I ended up filling in for Jim |
| 18:09.52 | leifmadsen | anonymouz666: see here -- http://leifmadsen.com/node/5 |
| 18:10.39 | anonymouz666 | oh nice webpage |
| 18:12.25 | anonymouz666 | let me ask you a question, we hit some issues with ringinuse=no, that queues called a member more than once |
| 18:12.43 | anonymouz666 | then, we had two options, let the channel driver controls it or through group group_count |
| 18:13.15 | anonymouz666 | as local members, i didn't was able to put group_count to work, but the call-limit filled perfectly for our case. |
| 18:13.43 | anonymouz666 | how does it work the group_count in this case, when you put a value into a category, that's visible only like any other channel variable? |
| 18:13.45 | p3nguin | Is there any problem with using a Gosub() and never Return()ing, from a RAM usage standpoint? |
| 18:15.13 | leifmadsen | p3nguin: nope, it's bascially the same thing as a Goto() |
| 18:15.25 | anonymouz666 | very nice presentation called Distributed Call Center 2010 - we use everything that is there, except the xmpp that really sucks (using openais instead). |
| 18:15.45 | leifmadsen | p3nguin: I mean, it stores data like a channel variable, so I guess it is possible, but you'd have to call and not return from probably thousands of GoSub()s |
| 18:16.25 | leifmadsen | anonymouz666: I don't use call-limit (I use callcounter=yes) and I use real SIP device names for the interface_state data for Local channels and have zero problems |
| 18:16.39 | leifmadsen | you have to tell the Local channel where to get its device state from |
| 18:17.00 | anonymouz666 | oh yeah, that we already do. |
| 18:17.03 | anonymouz666 | works fine |
| 18:17.11 | anonymouz666 | callcounter we use it also |
| 18:17.25 | anonymouz666 | and also call-limit. |
| 18:17.53 | leifmadsen | call-limit is deprecated so I don't use it |
| 18:18.22 | beek | leifmadsen: But so is Macro and you use it! |
| 18:18.24 | beek | :D |
| 18:18.29 | leifmadsen | beek: barely :) |
| 18:18.35 | leifmadsen | only when necessary |
| 18:18.44 | leifmadsen | actually Macro() isn't really deprecated |
| 18:18.45 | anonymouz666 | that's the point. how do you limit without call-limit when you don't truste ringinuse=no ? using group and group_count... |
| 18:19.00 | leifmadsen | anonymouz666: I do trust ringinuse=no, that is the point. I don't have problems with it. |
| 18:19.57 | anonymouz666 | I just realized that you are problems-free :-) |
| 18:19.58 | anonymouz666 | hehe |
| 18:20.19 | anonymouz666 | j/k |
| 18:20.21 | leifmadsen | otherwise, ya, you need to use GROUP() and GROUP_COUNT() to limit |
| 18:20.40 | beek | This one is driving me NUTS: |
| 18:21.11 | beek | PRI -> * -> PRI -> LegacyPBX -> CHANNEL_BANK -> * (for voicemail) |
| 18:21.33 | beek | On * 1.6.0 and DAHDI 2.4, caller hangs up and voicemail terminates. |
| 18:21.48 | beek | On * 1.8.5 and DAHDI 2.5.0 caller hangs up and voicemail app times out. |
| 18:22.31 | beek | I can see no reason why the hangup isn't passed along so the call (and vm message recording) doesn't terminate properly. |
| 18:23.13 | beek | Any ideas where I should next look? I've compared the configurations of DAHDI on both systems and they're identical. |
| 18:23.52 | beek | I can't help think that there is some behaviour on DAHDI 2.5 or * 1.8 that has changed for which I need to use a switch to revert to the older behavior. |
| 18:24.05 | beek | Anyone? |
| 18:24.54 | p3nguin | If I have multiple peers all coming from the same host address, does sip set debug peer <peer> actually filter by peer name or by IP address? It says SIP Debugging Enabled for IP: <the IP address>. |
| 18:27.15 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
| 18:27.27 | p3nguin | I'm just wondering if this could be rewritten to say "SIP Debugging Enabled for Peer: <peer>" if it doesn't really mean everything from the IP address. |
| 18:34.29 | *** join/#asterisk talntid (~erict@li93-153.members.linode.com) |
| 18:34.40 | talntid | Anyone know of a viop provider, that can accept collect calls? |
| 18:34.44 | leifmadsen | p3nguin: peers are always matched via IP address |
| 18:35.06 | p3nguin | I guess you missed my point. |
| 18:35.44 | *** join/#asterisk DanFromUK (DanFromUK@2.27.37.78) |
| 18:36.00 | *** join/#asterisk eduzimrs (~eduzimrs@201.22.86.124.static.gvt.net.br) |
| 18:36.21 | eduzimrs | anyone knows "[Sep 13 15:27:38] WARNING[12806] config.c: Realtime mapping for 'sippeers' found to engine 'mysql', but the engine is not available" ??? |
| 18:36.49 | p3nguin | With a couple dozen phones coming from a single natted network, all phones come from the same IP address. If I can debug by peer, and it says it is enabled for IP address, it seems like the wrong filter is applied. |
| 18:37.09 | p3nguin | (even if it isn't wrong) |
| 18:38.06 | p3nguin | If I debug by IP address, I wouldn't expect it to say enabled for peer. |
| 18:38.23 | beek | What are the switches to use when Dialing a local channel? I'm remembering seeing something about that but I'll be damned if I can find it. |
| 18:38.40 | DanFromUK | hi, has anyone managed to get cisco 7945g phones to connect to asterisk? my phone just says "Registering", but doesnt do anything else. asterisk debug doesnt show anything. can anyone suggest how i can sort this out? |
| 18:38.48 | p3nguin | You don't necessarily need any. Dial(Local/123@context) |
| 18:38.58 | DanFromUK | i dont have a managed switch, so i can't monitor the packets coming out of the phone. |
| 18:39.02 | p3nguin | But there is /n which is pretty common. |
| 18:39.29 | ChannelZ | DanFromUK: does it know the hostname or IP of your Asterisk box? |
| 18:39.36 | beek | p3nguin: What does the '/n' do? Or better yet, where are these documented so that I can read about them. "Googling" asterisk 1.8 dial local isn't cutting it. |
| 18:40.03 | beek | Or are these parameters generic to Dial? |
| 18:40.08 | *** join/#asterisk BuenGenio (~Gene@80.30.212.226) |
| 18:40.30 | DanFromUK | ChannelZ: yes, i've set it in the provisioning files, and the settings are showing up on the device. |
| 18:41.18 | p3nguin | doc/localchannel.txt I think. |
| 18:41.37 | ChannelZ | Do you have verbose set up a little (3 or so) and/or have turned on SIP debug to see if it's even saying anything? (not "regular" debug which contains way too much crap for what you're trying to fix for now) |
| 18:41.59 | beek | p3nguin: Thanks. |
| 18:42.26 | p3nguin | I don't see localchannel.txt in my recent 1.8.6.0 source tree... but it is in all my 1.4s. |
| 18:42.38 | DanFromUK | ChannelZ: verbose was on 30, and ive got sip debug ip on. |
| 18:42.58 | DanFromUK | i can see the other phones from that network, just not the cisco phone. |
| 18:43.35 | ChannelZ | well.. it's either braindead or speaking on a different network or something |
| 18:44.18 | leifmadsen | p3nguin: it's on the Asterisk wiki now |
| 18:44.23 | leifmadsen | p3nguin: or in the AST.pdf file |
| 18:44.25 | ChannelZ | is it being configged by DHCP, does it have a legit IP/can you ping it from the server at all? |
| 18:45.10 | talntid | Anyone know of a viop provider, that can accept collect calls? |
| 18:45.54 | beek | p3nguin: Thanks... found it. |
| 18:47.55 | *** part/#asterisk m_tadeu (~quassel@89.180.229.155) |
| 18:48.44 | *** join/#asterisk shadowapex (~William@adsl-99-36-142-6.dsl.irvnca.sbcglobal.net) |
| 18:49.08 | shadowapex | Hey, anyone have any experience with using "shell_exec" with PHPAGI? |
| 18:49.28 | shadowapex | And Asterisk AGI in general. |
| 18:49.46 | shadowapex | or Asterisk AGI in general* |
| 18:51.57 | shadowapex | For some reason, whenever I run a particular command using shell_exec, the PHP script stops completely. This doesn't happen at all when executing directly from the command line, only when it is executed through Asterisk. |
| 18:53.35 | eduzimrs | anyone knows "[Sep 13 15:27:38] WARNING[12806] config.c: Realtime mapping for 'sippeers' found to engine 'mysql', but the engine is not available" ??? |
| 18:53.56 | *** part/#asterisk otwieracz (~gonet9@v6.gen2.org) |
| 18:55.26 | leifmadsen | eduzimrs: means the mysql engine isn't available (i.e. you don't have something configured correctly) |
| 18:55.47 | leifmadsen | eduzimrs: you're trying to map to mysql in extconfig, but haven't configured how to connect to 'mysql' |
| 18:56.42 | DanFromUK | ChannelZ: its got a legit IP via DHCP, and it successfully downloaded config files from a local TFTP server. |
| 19:02.17 | *** join/#asterisk MarKsaitis (~MarKsaiti@85-189-201-83.tmbsystems.managedbroadband.co.uk) |
| 19:04.32 | *** join/#asterisk DanFromUK (DanFromUK@2.27.37.239) |
| 19:09.56 | *** join/#asterisk Daejeo (~chatzilla@117.207.156.55) |
| 19:12.29 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v010-021.mobile.uci.edu) |
| 19:13.18 | x86 | has anyone configured DHCP with Active Directory to support WDS (windows deployment services), but still pass FTP server info to Polycom phones? |
| 19:13.31 | Naikrovek | ... yea |
| 19:13.45 | Naikrovek | in fact i don't recall any special configuration at all to make that happen |
| 19:15.15 | Daejeo | is there any commercial SMS gateway that can be integrated with asterisk for two way SMS communication? |
| 19:15.46 | x86 | I guess, is it possible to setup a "vendor class" or something to where the Polycom MAC address range gets IPs from its own scope? |
| 19:15.53 | Naikrovek | x86: how are you passing the ftp server to the phones? option 66 is the way i'm doing it. |
| 19:16.22 | x86 | Naikrovek: right, that's the way you should do it... but with WDS, option 66 and 150 (and others) are already used |
| 19:16.41 | Naikrovek | x86: i don't know about that... my default scope has option 66 configured, and WDS did whatever it needed to do when I installed it. I didn't have to change anything. |
| 19:16.59 | x86 | interesting... |
| 19:18.16 | *** join/#asterisk r33dtard (~r33dtard@gateway/tor-sasl/r33dtard) |
| 19:19.33 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v010-021.mobile.uci.edu) |
| 19:20.22 | *** join/#asterisk DanFromUK (DanFromUK@2.27.40.15) |
| 19:20.33 | r33dtard | can anyone point me as to where I could find documentation on setting up an IAX listener for warvox |
| 19:24.27 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-qisipfvrnyylqtpr) |
| 19:29.07 | *** join/#asterisk cyford (~cyford@c-24-99-17-185.hsd1.ga.comcast.net) |
| 19:34.43 | *** join/#asterisk tyrrexrrg (~roger@190.147.143.151) |
| 19:35.30 | *** join/#asterisk blitzrage (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 19:35.30 | *** mode/#asterisk [+o blitzrage] by ChanServ |
| 19:39.21 | *** join/#asterisk f2knight (~ben@c-76-115-44-207.hsd1.or.comcast.net) |
| 19:52.22 | *** join/#asterisk saisoma (~saisoma@client72.jdcc.edu) |
| 19:54.16 | *** join/#asterisk andresmujica (~Andres@ubuntu/member/andresmujica) |
| 19:54.16 | r33dtard | how would I setup an outgoing default route to a sip connection? |
| 19:57.12 | chazzam | r33dtard: what are you setting this up in? |
| 19:57.22 | chazzam | ~thebook |
| 19:57.23 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://ofps.oreilly.com/titles/9780596517342/ or ~buybook |
| 19:57.32 | *** join/#asterisk andresm (~Andres@ubuntu/member/andresmujica) |
| 19:58.08 | r33dtard | chazzam: just plain asterisk |
| 19:58.13 | r33dtard | not using freepbx |
| 19:59.51 | andresm | Hello asterisk fellows, I'm having a hard time with a perl AGI script... When I invoke the script the get_data or wait_for_digit functions don't work at all... any AGI expert who can give some tips? |
| 20:00.48 | chazzam | r33dtard: then yeah, check ~thebook's chapter on setting up SIP and basic dialplan examples for placing calls using it |
| 20:04.02 | r33dtard | thanks |
| 20:05.19 | saisoma | hey guys, question regarding a specific situation involving call files, meetme rooms and ringing outside lines: http://pastebin.com/g6PRxb0t |
| 20:11.07 | chazzam | r33dtard: it looks like chapters 5-7 will be related to your question, but 7 will probably have the most direct examples. 5 and 6 will be more background information and such |
| 20:11.31 | r33dtard | all right thanks |
| 20:11.50 | chazzam | and remember, you can view it for free online |
| 20:12.21 | p3nguin | If a call dials some phones and the call dies as soon as someone answers one of them, should extension h run or not? |
| 20:12.34 | treborsux | a call came in and it was transfered to extesion by ivr. She could see caller id. When she picked it up it was just ringing like she was making a call????? |
| 20:12.43 | r33dtard | cool thanks chazzam |
| 20:13.23 | *** join/#asterisk Faustov (~fst@gentoo/user/faustov) |
| 20:15.18 | treborsux | what directory is master.csv in |
| 20:15.23 | treborsux | where are the logs |
| 20:15.29 | *** join/#asterisk BuenGenio (~Gene@34.Red-83-39-174.dynamicIP.rima-tde.net) |
| 20:17.08 | pabelanger | treborsux: /var/log/asterisk/cdr-csv |
| 20:22.27 | f2knight | treborsux, has a similar problem a while ago. Look at your cus, cas timing. |
| 20:23.24 | f2knight | treborsux, flowroute actually debuged it for me on a client, they would be in a call and all of a sudden the call directions would switch, turns out its a timing issue with asterisk and the timing server. |
| 20:23.26 | *** join/#asterisk zerohalo (~zerohalo@173-13-92-27-NewEngland.hfc.comcastbusiness.net) |
| 20:24.02 | f2knight | treborsux, you can do a few things, one is make sure you have ntpd running to keep time syncs up. In our case it was and the problem was an upstream vendor had a bad timing |
| 20:24.17 | *** join/#asterisk zerohalo (~zerohalo@173-13-92-27-NewEngland.hfc.comcastbusiness.net) |
| 20:24.19 | *** join/#asterisk KNERD (~KNERD@adsl-99-35-23-16.dsl.hrlntx.sbcglobal.net) |
| 20:24.43 | f2knight | treborsux, the solution was to make a compromise. and set our time outs a little higher then we would other wise. forcing asterisk to wait a little longer . |
| 20:25.00 | KNERD | Why does GV and chan_jabber keep screwing up? ahhhhgggggg! |
| 20:25.38 | f2knight | treborsux, the result is that when the timing comes back from the client asterisk waits a little longer before assuming a new role. |
| 20:27.00 | f2knight | treborsux, this COULD have the bad result of if a hangup was not received from the caller that asterisk might keep the channel open until this timeout happens. we set ours for 3 min as 3 min was an acceptable time we were willing to pay for if this happened. It has not happened yet, so no big deal |
| 20:27.54 | f2knight | KNERD, What is your issue? |
| 20:28.10 | KNERD | fake ringing |
| 20:28.23 | KNERD | when dialing out |
| 20:28.43 | f2knight | KNERD, what exactly do you mean by 'fake ringing?' |
| 20:28.58 | chazzam | KNERD: you are aware of google changing the api for google voice constantly making it so chan_gvoice doesn't work properly anymore right? |
| 20:29.09 | KNERD | well, you know...when you pick up phone....dial a number...and it gives a ringing sound |
| 20:29.19 | chazzam | so that you pretty much have to always run SVN, and even then its often broken? |
| 20:29.25 | KNERD | it was working yesterday |
| 20:29.36 | f2knight | KNERD, what is your dial string? |
| 20:29.38 | KNERD | but on and off functioniality |
| 20:30.16 | KNERD | one day it works great for a while, then back to not functioning again |
| 20:30.27 | KNERD | this is a known bug, but |
| 20:30.33 | KNERD | it keeps appearing |
| 20:30.58 | chazzam | https://wiki.asterisk.org/wiki/display/AST/Help+Maintain+Google+Talk+and+Voice ? |
| 20:32.13 | KNERD | chazzam: yeah I saw that. I don't want Astricon tickets..maybe a soda fountain |
| 20:33.11 | f2knight | I run Gvoice on a box, and have had good luck so far. I do keep it uptodate with svn every week, and have about 20 GV numbers on it. (do some locking and checking on it to make sure I only use one channel at a time) |
| 20:33.42 | f2knight | KNERD, do you have a 'r' in your dial string? |
| 20:35.33 | *** join/#asterisk mateu (~mateu@missoula.org) |
| 20:38.43 | f2knight | guess it might be off topic but did Google Publish an API for Google Voice? |
| 20:39.03 | KNERD | f2knight: looking for s |
| 20:39.28 | KNERD | i mean r |
| 20:39.59 | KNERD | f2knight: actually they did for HTTP, but I do not know for other protocols |
| 20:42.09 | f2knight | KNERD, if you have an r or R in your dial string it would create a 'ringing' no matter what the other side is doing. |
| 20:42.50 | KNERD | i see..but that false ringing is a known documented bug, but I will look |
| 20:42.52 | f2knight | KNERD, so DIAL(gtalk/account/+15555551212@voice.google.com,45,rRTw) |
| 20:43.48 | f2knight | KNERD, would cause the ringing. I only suggest looking there because I have seen lots of people using rR on there dial strings, with out reallizing that they prob. shouldnt use it. |
| 20:44.27 | f2knight | KNERD, when things are 'working' you wouldn't really notice but when they are not working, well a different ball of wax is born |
| 20:49.29 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 20:51.07 | KNERD | actually I do..when you call and noboy anwers |
| 20:52.49 | KNERD | still looking..i never screw around with those conf file, so I forget where everything is |
| 20:57.16 | p3nguin | If a call dials some phones and the call dies as soon as someone answers one of them, should extension h run or not? |
| 21:00.11 | p3nguin | Also, are there any Asterisk decals for purchase, or can I have my own made for personal usage (not for resale)? |
| 21:01.13 | Qwell | p3nguin: decals, like case badges, or like stickers? |
| 21:01.23 | Qwell | there are stickers on the digium store |
| 21:02.08 | p3nguin | I wanted some stickers to put on my Asterisk appliances that I assemble. |
| 21:02.27 | p3nguin | I'll check the store now. |
| 21:02.39 | Qwell | http://store.digium.com/products.php?category_id=22 |
| 21:02.49 | Qwell | buy some coffee while you're there |
| 21:02.55 | f2knight | p3nguin, I would think the h should run |
| 21:03.29 | f2knight | p3nguin, set a noon(${DIALSTATUS}) and see what the status is you could at least catch it then. |
| 21:04.01 | p3nguin | These stickers are too large for my application. |
| 21:04.20 | f2knight | p3nguin, as for decals, I am not sure, but my Father is a sign maker and does custom vinyl letters logos and stuff |
| 21:04.22 | Qwell | p3nguin: yeah I figured. I know I've seen smaller ones in the past, but no clue where they came from. Might be a marketing custom order. |
| 21:04.23 | p3nguin | I need something like 1.5 x 2 inches or 1 x 1.5 inches. |
| 21:04.33 | *** join/#asterisk jkroon (~jkroon@197.169.211.9) |
| 21:04.44 | f2knight | p3nguin, how many are you needing? |
| 21:05.17 | f2knight | p3nguin, do you have / want your own logo on them? |
| 21:05.25 | p3nguin | If I had a Cricut, I'd do it myself. |
| 21:06.52 | p3nguin | I could probably use like five or so Asterisk stickers. |
| 21:08.36 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 21:08.36 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 21:10.34 | p3nguin | It needs to fit in a spot that is 1.125 inch wide, so it would probably be less than 1 inch tall. |
| 21:11.13 | Qwell | p3nguin: you'd have to be careful about trademark stuff |
| 21:12.00 | Qwell | I am a little surprised there isn't more demand for stickers that size though. |
| 21:12.30 | *** join/#asterisk Naikrovek (~mbluth@unaffiliated/naikrovek) |
| 21:13.09 | p3nguin | I think as long as I don't sell the stickers nor sell the equipment with the stickers affixed, I'm probably not going to get into too much of a problem. |
| 21:13.19 | p3nguin | I could be wrong, though, since I'm not a trademark attorney. |
| 21:13.55 | Qwell | I don't know, and I'm not willing to offer advice. :) |
| 21:14.00 | p3nguin | And if I would ever sell either, I'll be sure to send in my royalties check. :) |
| 21:14.15 | p3nguin | I know, I couldn't ask you to advise me on something that specific. |
| 21:15.21 | p3nguin | It's always good when someone just happens to know that kind of information, though. |
| 21:19.25 | f2knight | p3nguin, not sure if you would be 'legally' able to use the digium or asterisk logo, but you could use your own logo with out issues. |
| 21:19.56 | p3nguin | Maybe I could use the Asterisk logo if I pay the royalties. |
| 21:20.14 | f2knight | p3nguin, maybe you could use it if you just ask for permission. |
| 21:20.45 | f2knight | p3nguin, might have to have some refine ments like Built with asterisk(tm) |
| 21:21.07 | f2knight | of course I have an 'asterisk inside' logo :) |
| 21:21.46 | p3nguin | It's not really important enough for me to ask a qualified legal advisor, so maybe asking the right person within Digium could be adequate. |
| 21:22.56 | f2knight | http://www.asterisk.org/terms-of-use |
| 21:25.37 | f2knight | p3nguin, but if you have your own company logo, you could just use that ;) or put a simple "IP PBX" and a model number "X14r" |
| 21:26.39 | leifmadsen | p3nguin: ya just ask malcolmd |
| 21:26.49 | leifmadsen | p3nguin: he can direct you to the right person if he is not it |
| 21:27.00 | Qwell | pretty sure there's a trademarks@ email address |
| 21:27.17 | Qwell | yes, there is |
| 21:27.34 | p3nguin | I see that it states I may use the logo with consent. |
| 21:27.46 | ChannelZ | How does that go? It's better to ask forgiveness than permission? :P |
| 21:27.47 | anonymouz666 | kram still contribute with code to asterisk? |
| 21:27.52 | anonymouz666 | that's one curiosity |
| 21:28.08 | ChannelZ | s/easier/better/ |
| 21:28.23 | malcolmd | yop, if you've got a question about the Trademark Policy (http://www.digium.com/en/company/view-policy.php?id=Trademark-Policy), then trademarks@digium.com is what you want |
| 21:28.26 | leifmadsen | anonymouz666: not in quite some time |
| 21:28.28 | ChannelZ | or the other way around. sheesh my brain is jello today |
| 21:28.38 | ChannelZ | I've been battling dspam all day |
| 21:28.53 | leifmadsen | ChannelZ: that is how it goes, but when it comes to legal stuff, that is not the ideal solution :) |
| 21:29.00 | ChannelZ | heh I know |
| 21:29.16 | ChannelZ | Just ask Apple/Motorola/Google/Samsung/et al |
| 21:29.29 | anonymouz666 | leifmadsen: it's been a long time since I don't see him on IRC |
| 21:29.44 | leifmadsen | anonymouz666: he flys planes now |
| 21:30.03 | anonymouz666 | hehe I saw it on twitter |
| 21:30.10 | f2knight | p3nguin, exactly. Send off an email, you might be shocked. |
| 21:30.15 | anonymouz666 | is better than stay here, right? |
| 21:30.28 | anonymouz666 | :P |
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| 21:36.02 | p3nguin | malcolmd: Thank you. I may send off an email about it at some point. |
| 21:36.08 | f2knight | p3nguin, (d) A project is being sold or developed which incorporates a version of Asterisk which has not been modified from the form in which it was distributed by Digium and is described as being "Powered by Asterisk" or "Based on Asterisk". |
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| 21:39.04 | f2knight | p3nguin, Trademark in any manner that may be interpreted to be a logo by the public (such as with a stylized typeface for formatting), because the use of the Trademarks in logos of Digium is strictly limited to those licensed under the Digium Partner Program. |
| 21:39.30 | f2knight | p3nguin, Therefore, to ensure your use is in compliance with the Policy, you are encouraged to use only the "Word Form" (which means use of the words only, in a standard Arial font, without a design or stylization element) to avoid any use that may look like a logo. |
| 21:40.44 | f2knight | p3nguin, sounds like you can not use the 'LOGO' with out being a Digium Partner, but that you may use the "WORD" Asterisk. or "Powered by Asterisk" with no logo with out a problem. |
| 21:41.36 | p3nguin | Until I send that email, I'll just leave the box without any sticker on it. It's not like it can be seen, anyway. |
| 21:41.51 | p3nguin | I was just going to dress it up a little. |
| 21:43.16 | f2knight | p3nguin, thats a safe bet. But you could like I said always put YOUR logo or company name etc on it. Brand it yourself. |
| 21:43.31 | p3nguin | I could if I wanted to. |
| 21:44.59 | p3nguin | Huh. I think my monitor just gave up and quit. |
| 21:45.16 | p3nguin | Damn. That'll be two this year. |
| 21:45.39 | p3nguin | I was able to repair the last one by replacing capacitors in it. |
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| 21:51.51 | rotten777 | good afternoon |
| 21:57.22 | hardwire | ooh.. voip.ms is nice |
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| 22:03.29 | BladeMcCool | how do i turn on some general debugging info for applications? for example i am struggling with issues in getting Festival tts to work from within an AGI script and was wondering if there is any way to get some insight into where things are going wrong (Festival seems to work for me only from normal dialplan) |
| 22:03.34 | *** join/#asterisk DanFromUK (DanFromUK@2.27.40.15) |
| 22:03.47 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 22:03.58 | DanFromUK | hi, is there an option to get asterisk to add the date/time to the register header? |
| 22:04.43 | f2knight | BladeMcCool, agi debug |
| 22:04.48 | navaismo | BladeMcCool use gai set verbose on |
| 22:05.03 | BladeMcCool | tyvm for infos |
| 22:05.15 | f2knight | BladeMcCool, or agi set debug on if your on 1.8 |
| 22:05.32 | navaismo | agi* |
| 22:05.46 | f2knight | you wil basiclly see the send and recieve lines |
| 22:06.51 | f2knight | if you are using fastAGI you could also use netcat to listen on the 'server' side and manually send commands back. ... nc -l <port you want to listen on > |
| 22:07.30 | f2knight | DanFromUK, what do you mean by a register header? you mean a sip header? |
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| 22:08.13 | f2knight | navaismo, you mean agi set debug on :P |
| 22:09.39 | navaismo | yep |
| 22:09.56 | granola | do you guys know how npanxx's are resolved to location? or how accurate it can be? |
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| 22:15.16 | DanFromUK | f2knight: yes, i'm trying to configure a cisco phone to connect to asterisk. someone on voip-info.org says that the phone will set its date/time automatically using the date from the register header. (his words, not mine) |
| 22:15.42 | rotten777 | sntp server? |
| 22:15.52 | DanFromUK | "the phone will set its clock based on the Date header returned as part of the SIP proxy's registration response (200 OK)" |
| 22:16.15 | rotten777 | i don't know about ciscos but my polycom has a sntp server address |
| 22:16.28 | DanFromUK | i know. polycom are great! |
| 22:16.47 | f2knight | DanFromUK, can't speak for cisco phones I hate them personally, but most SIP phones get time from the DHCP server or an external NTPd service. |
| 22:17.08 | rotten777 | http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/5_0_1/ccmcfg/b02ntpsv.html |
| 22:17.09 | *** part/#asterisk granola (~david@stg.maculon.com) |
| 22:17.27 | DanFromUK | can the device date/time affect registration? the cisco phone keeps coming up as unauthorised, even though ive tested the same account on xlite and its fine. |
| 22:17.43 | f2knight | Okay do as rotten777 pointed out set up an NTP server |
| 22:17.43 | rotten777 | i've never seen it effect the auth |
| 22:17.55 | BladeMcCool | so yeah i think i'm gonna give up on trying to make Festival work within AGI on 1.8 .. not sure what is up bit it aint talking . i guess i should be happy that i even got espeak to work! haha .. it just does not sound awesome. i wish Allison was free LOL |
| 22:18.19 | f2knight | BladeMcCool, I tried getting festival up period it was a pain. |
| 22:18.39 | DanFromUK | is someone free to help me compare two SIP dialogs and see why one is unauthorized, and the other is OK? |
| 22:18.48 | f2knight | BladeMcCool, fiestival should not be used for any production work just for mock ups |
| 22:18.56 | p3nguin | It wasn't hard to make festival work, but the sound quality isn't very desirable. |
| 22:19.27 | f2knight | p3nguin, I kept getting dependency issues on my 10.04 LTs box |
| 22:19.37 | BladeMcCool | i got the festival software installed, and stuff seems like it should work .. festival logs the connect and disconnect, and there doesnt appear to be an error .. but it just doesnt speak. maybe i _have_ to enable the caching or something i'm not sure. there is a perl kludge to output the text to a file, generate the .wav and then have asterisk play that back but i'd rather not get into that if i dont have to lol |
| 22:20.13 | BladeMcCool | i'm on 10.04LTS as well. espeak is ok, i may just stick with that too. |
| 22:20.22 | f2knight | BladeMcCool, sounds like festival is not working at all if its not playing audio :) |
| 22:20.59 | p3nguin | My only problem making festival work the first time was that I didn't have an appropriate alsa.conf, so it could not play where it needed to play. |
| 22:21.13 | f2knight | for mock ups I will sometimes use SayALpha(what I want to say) (usually an 'error code' and keep a reference that way. |
| 22:21.18 | BladeMcCool | the funny thing is i did get it to output a sound ONCE .. via regular dialplan only but it actually worked. ... i mean i can tell the festival server process is running and watch it log communications at any rate :/ text2wave works as well (wait is that part of festival??) |
| 22:21.29 | f2knight | p3nguin, umm maybe thats my issue. |
| 22:22.08 | BladeMcCool | f2knight: i tryied the sayalpha stuff. espeak is reasonably intelligible TTS so i'll stick with that for now b/c it works. was hoping to get something that sounds a little nicer. and maybe one day if/when there is a budget I can hire Allison to talk into a mic for me haha |
| 22:23.23 | f2knight | BladeMcCool, you could also use http://www2.research.att.com/~ttsweb/tts/demo.php |
| 22:23.42 | f2knight | I used it a few times type it out , download the file put it on your box. |
| 22:24.11 | f2knight | if you ware being funky you could do an quick agi script to take your text do an html post and download or stream the file back :=) |
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| 22:31.54 | BladeMcCool | f2knight hehe i like the way you think |
| 22:32.19 | BladeMcCool | i may end up diong this when it comes time for some real polish |
| 22:34.45 | BladeMcCool | ouch pain at&t tells me that my text violated their policy b/c it sounds too commercially. i lol'd ah well i guess beggars like me cannot be chusers. |
| 22:35.50 | f2knight | BladeMcCool, and its for personal/non-commerical use :) |
| 22:36.49 | f2knight | BladeMcCool, if your developing something that your selling you charge the client for the prompts :-) if its for your own business, you grab a microphone and record them yourself, or write it all out and take your friend to lunch in exchange for reading them off for you. |
| 22:37.30 | BladeMcCool | i suppose a bitcoin e-wallet ivr interface thing would probably fall under 'commercial', even though I do plan to give it away! hehe |
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| 22:38.25 | BladeMcCool | its all good. i have more features to implement before i get my panties too much in a bunch over how the voice prompts sound. haha |
| 22:41.06 | p3nguin | finds it interesting that blademccool wears panties. |
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| 23:18.32 | p3nguin | f2knight: Did you say you use Vyatta? |
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