| 00:05.24 | *** join/#asterisk ariel_ (~chatzilla@99-1-236-49.lightspeed.miamfl.sbcglobal.net) |
| 00:06.15 | MikeH | Is there a 'get started' guide for AsteriskGUI? |
| 00:09.49 | *** join/#asterisk TimeRider (steve@5ac7b321.bb.sky.com) |
| 00:16.06 | *** join/#asterisk fbc_ (~fbc@200.92.91.102) |
| 00:16.25 | fbc_ | ChannelZ, I like the electrodes on a chair idea |
| 00:17.05 | fbc_ | MikeH, dunno I use elastix. |
| 00:36.45 | *** join/#asterisk MikeH (~mike@86.63.17.141) |
| 00:36.48 | MikeH | Well, this is fun |
| 00:37.02 | MikeH | The GSM card I bought requires that I use their own supplied version of asterisk and dahdi |
| 00:40.53 | WIMPy | You shoot yourself in the foot hardware. |
| 00:41.09 | MikeH | I found no other GSM interface cards when looking :/ |
| 00:41.36 | MikeH | Are there source packages available for either AsteriskGUI or FreePBX? |
| 00:42.21 | ariel_ | there is a full open source for Freepbx |
| 00:42.45 | pabelanger | MikeH: which card? |
| 00:42.49 | WIMPy | But the GUIs don't have anything to do with hardware. |
| 00:43.16 | MikeH | pabelanger, atcom ax4g |
| 00:43.24 | *** join/#asterisk smps (~maher@78.104.151.89) |
| 00:43.51 | WIMPy | Ugh |
| 00:44.14 | MikeH | WIMPy, Not saying they do, but once I've had to manually compile asterisk, I'm assuming that the packages for FreePBX/AsteriskGUI will fail on their dependency for asterisk |
| 00:45.08 | WIMPy | Ok, things to think of when buying hardware for use with Asterisk #1: Don't fall for a "works with Asterisk" or "Designed for Asterisk". Always ask if it works with the current version. |
| 00:45.49 | WIMPy | Not sure, they are that closely related, but I don't know much about the GUIs. |
| 00:46.58 | WIMPy | But that card has too many antennas and too little sim readers at first sight. |
| 00:48.35 | MikeH | huh? |
| 00:48.40 | MikeH | Its modular |
| 00:48.50 | MikeH | you have one antenna and one sim card slot per module..... |
| 00:49.10 | WIMPy | Yes, indeed. |
| 00:50.34 | *** join/#asterisk moy_ (~moy@70.31.14.144) |
| 01:13.40 | *** join/#asterisk beebuu (3afc49a5@gateway/web/freenode/ip.58.252.73.165) |
| 01:13.50 | beebuu | hello,all |
| 01:14.38 | beebuu | how to know which called part pickup when dial(sip/1001&sip/1002) run? |
| 01:14.55 | DND | guys. im having problems installing nvfax |
| 01:15.02 | DND | i mean lading nvfax in asterisk |
| 01:15.07 | DND | *loading |
| 01:15.25 | DND | i tried module load app_nv_faxdetect.so |
| 01:15.36 | DND | but its not showing any confirmation that its loaded |
| 01:15.44 | DND | its not also in core show application |
| 01:16.12 | WIMPy | turn up debug and verbose and try again. |
| 01:16.13 | *** join/#asterisk rrb3942 (~rbullock@cpe-67-242-215-62.rochester.res.rr.com) |
| 01:16.31 | WIMPy | It should tell you why it doesn't load. |
| 01:16.59 | DND | my verbose is asterisk -vvvvvvvvvvr |
| 01:17.34 | WIMPy | 'core set debug 9' |
| 01:19.12 | DND | nope |
| 01:19.14 | DND | nothing |
| 01:19.51 | *** join/#asterisk root52 (~root52@ip70-191-116-76.cl.ri.cox.net) |
| 01:21.09 | DND | hmm |
| 01:21.15 | root52 | Hey All. Asterisk 1.8.1 func_odbc readsql works just fine. however writesql seems to just be ignored. I never see the query in the mysql log. Any thoughts? |
| 01:23.50 | *** join/#asterisk csnook (~chris@c-76-19-64-161.hsd1.ma.comcast.net) |
| 01:25.53 | WIMPy | o/~ |
| 01:25.58 | WIMPy | oops |
| 01:26.02 | WIMPy | not here :-) |
| 01:26.14 | *** join/#asterisk delroy_ (~dion@S0106001310246ce7.ss.shawcable.net) |
| 01:27.22 | delroy_ | anyone using/used IMAP storage? |
| 01:29.49 | delroy_ | anyone using/used IMAP storage? I want to know if one can use normal storage and IMAP storage for specific extensions only. |
| 01:37.50 | root52 | Wait. I'm sorry forgot what server I was on. It is asterisk 1.8.0 not 1.8.1 |
| 02:04.47 | *** join/#asterisk TimeRider (steve@5ac7b321.bb.sky.com) |
| 02:07.58 | drmessano | delroy_, you should be able to mix them |
| 02:08.05 | drmessano | Last time I checked you could |
| 02:32.58 | MikeH | ANyone here use a Cisco phone with Asterisk? |
| 02:33.03 | MikeH | I'm trying to get it to connect |
| 02:33.32 | MikeH | but I'm unsure of some settings - I've filled it in with 'best guess' and the phone is giving me 'Failed(404)' |
| 02:51.26 | *** join/#asterisk Defraz (~Defraz@96.18.85.158) |
| 02:51.34 | MikeH | Anyone? |
| 03:05.11 | *** join/#asterisk werdan7 (~w7@freenode/staff/wikimedia.werdan7) |
| 03:09.53 | beebuu | how to know which called part pickup when dial(sip/1001&sip/1002) run? |
| 03:14.53 | *** part/#asterisk root52 (~root52@ip70-191-116-76.cl.ri.cox.net) |
| 03:23.59 | MikeH | is there a way to test channels. |
| 03:24.16 | MikeH | I've setup incoming and outgoing calling rules, dial plans and users |
| 03:24.43 | MikeH | but when I try calling out, I get this: http://pastebin.com/JGPa74EK |
| 03:34.43 | MikeH | ah |
| 03:34.45 | MikeH | solved that one |
| 03:52.38 | *** join/#asterisk CaptWho (~farina@76.201.156.215) |
| 03:53.57 | CaptWho | Is "asterisk now" the best build to set up a DID network for voip? |
| 03:54.34 | WIMPy | A what? |
| 03:56.20 | CaptWho | we're going to be issued some DIDs from a company that provides access to the telephone switched network and i'm wondering if there is a specific build that would be good for handling that? I've been exposed to "asterisk now", but are there different builds available? |
| 03:57.44 | MikeH | hrm, not I really am stuck. I've devised a voice menu in asterisk, which goes - Answer, Background, digit timeout |
| 03:57.58 | MikeH | however when calling that associated extension, it calls and immediately hangs up |
| 03:58.47 | WIMPy | Sure, but that shouldn't make any difference. There are differen GUIs however, but all of them limit your possibilities, if you plan to do a little more fancu stuff. |
| 03:59.49 | CaptWho | is there a breakdown of the different GUIs somewhere? |
| 04:00.15 | WIMPy | Not AFAIK. |
| 04:01.20 | CaptWho | thanks... i wasn't aware that asterisk now was just a GUI for asterisk |
| 04:02.11 | WIMPy | It isn't. It's centos+Asterisk[+freepbx|asteriskgui] |
| 04:02.14 | MikeH | Can anyone help me with my voice menu issue? Is there something I need to do before Answer()? or perhaps another step I'm missing? |
| 04:05.43 | CaptWho | so there could be different GUIs installed over Asterisk NOW? |
| 04:06.13 | MikeH | this is the menu: http://pastebin.com/mTDx9TYS |
| 04:06.20 | WIMPy | AFAIK it comes with those two. But you can install without. |
| 04:07.03 | WIMPy | MikeH: Maybe you want to add a WaitExten? |
| 04:07.53 | MikeH | hrm, I'm not sure I do? DOes that not just wait for the user to enter an extension number first? |
| 04:08.13 | WIMPy | Or thereafer. |
| 04:08.35 | WIMPy | When your welcome has finished, it's over. |
| 04:08.59 | WIMPy | I personally prefer to do it with read() instead. |
| 04:09.08 | MikeH | My welcome never seems to start |
| 04:09.18 | MikeH | the call is answered then dropped instantly |
| 04:09.30 | WIMPy | Oh, right. |
| 04:10.03 | WIMPy | Always strip the extension from the file name. |
| 04:10.09 | MikeH | ah |
| 04:10.30 | WIMPy | Asterisk will decide itself what's the best available file. |
| 04:11.01 | MikeH | [Jan 30 12:09:46] WARNING[7151]: file.c:650 ast_openstream_full: File welcome does not exist in any format |
| 04:11.01 | MikeH | [Jan 30 12:09:46] WARNING[7151]: file.c:953 ast_streamfile: Unable to open welcome (format 0x4 (ulaw)): No such file or directory |
| 04:11.07 | MikeH | could have something to do with it I guess :P |
| 04:11.18 | WIMPy | indeed |
| 04:11.38 | MikeH | it exists somewhere, as asteriskgui can play it back to me |
| 04:11.46 | MikeH | I wonder actually |
| 04:12.08 | MikeH | aah, I'm an idiot. |
| 04:12.23 | justdave | anyone using asterisk w/ dahdi on RHEL5? having trouble trying to get dahdi-linux to compile against the kernel in rhel 5.6. here's the compile error I'm getting: http://it.pastebin.mozilla.org/1001944 |
| 04:12.37 | WIMPy | MikeH: That's usually a "was" :-) |
| 04:12.53 | MikeH | 'for custom recordings, please prepend the file with record/' |
| 04:12.59 | MikeH | :D |
| 04:13.47 | MikeH | wow |
| 04:13.49 | MikeH | this is brilliant. |
| 04:14.24 | MikeH | I've got my voice menu, sip phone and incoming calling rules mostly sorted |
| 04:14.31 | MikeH | outgoing works as well |
| 04:18.32 | MikeH | hrm |
| 04:18.37 | MikeH | I do have one more question |
| 04:19.19 | MikeH | Is it possible to add a user for an FXO line? (ie. to dial out a number when that user is called) |
| 04:19.24 | MikeH | I guess its just call forwarding really? |
| 04:20.04 | WIMPy | You can put whatever you like in to Dial(). |
| 04:20.35 | MikeH | where does that Dial() go? |
| 04:20.52 | MikeH | I basically need to create a call group |
| 04:20.54 | WIMPy | The extension. |
| 04:21.01 | MikeH | ok |
| 04:21.16 | MikeH | can you refer me to documentation for this, so I can read up later? |
| 04:21.30 | WIMPy | And you can put multiple channels in to one Dial(). |
| 04:21.58 | MikeH | Basically (I think its a call group I need) when calling the group, both one extension should ring, and another 'extension' should ring by dialing a number over one of the trunks. |
| 04:21.59 | WIMPy | It's just an extension like any other. |
| 04:22.59 | WIMPy | Just put both destionations in there. |
| 04:23.35 | WIMPy | There is no concept of something being local. There are just destinations. |
| 04:27.51 | MikeH | thanks |
| 04:28.00 | MikeH | plagued with teething issues this evening :/ |
| 04:28.17 | MikeH | when dialing out via the gsm line, it works fine |
| 04:28.28 | justdave | looks like something it doesn't like in the new kernel source |
| 04:28.31 | MikeH | but when dialing out via the analogue card there isn't any audio :/ |
| 04:28.36 | justdave | if I back down two versions of kernel it works |
| 04:29.36 | WIMPy | Analog is evil. |
| 04:29.43 | MikeH | but when dialing in via the analogue card, everything works fine. |
| 04:30.32 | WIMPy | justdave: The only thin i know of it that DAHDI relies on the BKL being enables which is no longer the default, but your issue looks like something I haven't heard about. |
| 04:30.38 | p3nguin | ~newbook |
| 04:30.39 | infobot | Help review the 3rd edition of the Asterisk book published by O'Reilly at http://ofps.oreilly.com/titles/9780596517342/ Released under a Creative Commons license: http://creativecommons.org/licenses/by-nc-nd/3.0/us/ |
| 04:30.47 | p3nguin | mikeh: ^^^^^^ |
| 04:32.40 | MikeH | right, just this dial out extension and fax detection to go |
| 04:32.58 | MikeH | I'll take a look at that book, thanks p3nguin |
| 04:33.34 | beebuu | how to know which called part pickup when dial(sip/1001&sip/1002) run? |
| 04:36.20 | justdave | WIMPy: anything I can provide to help debug? |
| 04:36.23 | p3nguin | beebuu: You want to know which device answers the call? |
| 04:37.09 | WIMPy | justdave: I guess you have to dig the source. |
| 04:39.39 | *** join/#asterisk neurosys (~neurosys@c-65-34-190-58.hsd1.fl.comcast.net) |
| 04:40.23 | beebuu | p3nguin: yes,like dahdi/1 etc |
| 04:40.57 | p3nguin | beebuu: If you Dial SIP/1001 and SIP/1002, it would be pretty hard for Dahdi/1 to answer it. |
| 04:41.25 | WIMPy | Forwarding? :-) |
| 04:41.51 | beebuu | p3nguin: just for example. i want to know which one is answer the call by dial mulit number? |
| 04:42.39 | beebuu | anyone know this? |
| 04:43.13 | WIMPy | beebuu: I think you need to listen on AMI for that. |
| 04:43.50 | beebuu | WIMPy: can it be known in dialplan? |
| 04:44.21 | WIMPy | I don't think so. But I'm not sure. |
| 04:44.22 | p3nguin | I don't think it can be done in dialplan until after the call ends. |
| 04:45.01 | WIMPy | Yes, the CDR variables should show it. |
| 04:45.32 | beebuu | Oh, any suggestion else? please |
| 04:46.04 | WIMPy | Use AMI. |
| 04:46.21 | beebuu | not easy way? |
| 04:46.43 | *** join/#asterisk sourcode (~code@ppp-115-87-241-106.revip4.asianet.co.th) |
| 04:46.49 | WIMPy | Whe exactely do you want to know? |
| 04:46.54 | WIMPy | Or in order to do what? |
| 04:48.02 | beebuu | i want to dial mulit numbers and when one is answer, i want to redirect it to some one |
| 04:49.03 | beebuu | dial mulit number can save time than call one by one |
| 04:49.13 | WIMPy | So you want to end the connection and setup the next? |
| 04:50.02 | beebuu | no,let it talk to a agent if someone pickup |
| 04:50.37 | beebuu | as you know,many number maybe no one answer |
| 04:50.50 | WIMPy | That's what happens when someone picks up. |
| 04:51.04 | WIMPy | I don't get what you try to do. |
| 04:51.16 | beebuu | redirect to a agent |
| 04:51.27 | carrar | How about let the Agent answer? |
| 04:51.38 | carrar | using queues and agents |
| 04:51.52 | WIMPy | Who do you dial then? |
| 04:51.59 | p3nguin | Someone doesn't understand call setup and dialplan. |
| 04:52.11 | carrar | heh |
| 04:52.17 | carrar | now who could that be |
| 04:52.57 | WIMPy | A local saying goes: Always the one who asks. |
| 04:53.52 | beebuu | i want to do this: make asterik to call 3 or more number,when one of them pickup, call a agent to answer. |
| 04:54.04 | p3nguin | Let me make sure I understand this. He's asking to call an arbitrary SIP device and have that device forward to some agent channel? |
| 04:54.10 | beebuu | i think this will save some time |
| 04:54.11 | WIMPy | And who is the one that answers? |
| 04:54.28 | beebuu | WIMPy: it doesn't matter |
| 04:54.45 | p3nguin | Why not just call the agent channels to begin with? |
| 04:54.52 | WIMPy | If it doesn't matter, what is the question? |
| 04:55.10 | beebuu | WIMPy: sorry,i make a mistake. |
| 04:55.11 | WIMPy | The whple story doesn't make any sense to me so far. |
| 04:55.30 | WIMPy | Maybe you should describe exactely what you want to do/happen. |
| 04:55.39 | p3nguin | Stop worrying about the SIP devices and call the Agent channel you want to take the call. |
| 04:55.56 | beebuu | WIMPy: yes,let me try it following |
| 04:56.33 | WIMPy | Start at the beginning. |
| 04:57.30 | beebuu | i got many numbers, now my agents call them one bye one,i think that's waste time, why can i make mulit number at one time? |
| 04:57.50 | carrar | haha |
| 04:58.35 | WIMPy | Ah, so you want an extremely dumb version of an "predictive" dialler? |
| 04:58.42 | beebuu | and if someone is answered,call back my agents,it does't need my agents to wait so many time |
| 04:58.47 | WIMPy | To piss even more customers off? |
| 04:59.24 | beebuu | yes |
| 04:59.26 | WIMPy | That "off" would be better in the front, wouln't it? |
| 04:59.53 | beebuu | predictive--- this is good name for me |
| 05:00.09 | WIMPy | Clearly a misnomer. |
| 05:00.41 | beebuu | WIMPy: so what's that name in fact? |
| 05:01.04 | WIMPy | That is the name. |
| 05:01.16 | WIMPy | But there's nothing predictive about it. |
| 05:01.18 | beebuu | and now,how can i do that? |
| 05:01.47 | WIMPy | It's just a way to maximise customers that complain to the relevant bodies. |
| 05:02.14 | WIMPy | You can buy or rent them. |
| 05:02.20 | carrar | http://tinyurl.com/4nj2b6f |
| 05:02.36 | WIMPy | And in some countries you risk getting shut down. |
| 05:03.08 | beebuu | WIMPy: in fact,my agents need to talk to them |
| 05:03.33 | beebuu | carrar: thanks |
| 05:03.47 | WIMPy | Then make up a good excuse why you hung up on them several times before. |
| 05:04.20 | p3nguin | heh |
| 05:04.29 | p3nguin | Yeah. That's illegal here. |
| 05:04.42 | WIMPy | Here as well. |
| 05:05.14 | WIMPy | Well, not exactely, but there have been cases of call centers being cut off the net for doing so. |
| 05:06.16 | WIMPy | That was ruled as being harassment |
| 05:06.43 | WIMPy | Which I think is fair enough. |
| 05:06.44 | beebuu | WIMPy: i want to know the way, could you tell me please? |
| 05:07.11 | WIMPy | You got that link to google, didn't you? |
| 05:07.52 | beebuu | Oh....thanks,at lease i got it's realname |
| 05:08.44 | beebuu | least |
| 05:08.47 | WIMPy | And better ask a lawyer. |
| 05:09.11 | beebuu | WIMPy: thanks for your help |
| 05:09.23 | beebuu | and carrar,p3nguiin |
| 05:09.24 | p3nguin | Too bad you aren't in the US. It is written out very clearly by the FCC and FTC. |
| 05:09.45 | p3nguin | Not much reason to ask a lawyer here. Just read the documents by yourself. |
| 05:10.07 | beebuu | p3nguin: Oh, can i read that in any URL about FCC and FTC? |
| 05:10.43 | WIMPy | If it's not illeagal, the customers will take care of that, I'm sure. |
| 05:11.09 | p3nguin | http://www.fcc.gov/cgb/consumerfacts/tcpa.html |
| 05:11.17 | beebuu | WIMPy: yes it's not |
| 05:11.31 | beebuu | in my place |
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| 07:08.55 | DND | hi guys i have digium fax free license installed. i wanted to know if this can only receive fax or can also send? |
| 07:14.15 | p3nguin | both |
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| 07:18.18 | DND | p3nguin you know hoe can i send faxes? |
| 07:18.29 | DND | can i do it like tel-number@domain.com? |
| 07:20.18 | DND | or maybe i setup faxing in windows ? |
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| 07:32.30 | Dovid | hi all. |
| 07:33.04 | Dovid | if i want to build sound files for all codecs. is there any simple way of doing it with out rebuilding the entire asterisk? |
| 07:38.47 | sranil | <PROTECTED> |
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| 07:55.03 | baldrailers | hello world |
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| 08:04.29 | p3nguin | dnd: Yeah, use the fax apps. |
| 08:05.02 | p3nguin | dnd: You can use the free stuff or the Digium stuff (free or non-free). |
| 08:05.36 | p3nguin | dnd: I've done it both ways and if I remember right, I had better luck with the free Digium stuff. |
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| 10:26.11 | a001101011 | hello |
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| 12:47.30 | MikeH | hrm |
| 12:47.54 | MikeH | I'm trying to create an extension that automatically forwards to an external number via a specific trunk |
| 12:48.00 | MikeH | can anyone point me in the right direction? |
| 12:48.53 | Dovid | MiekH: Are you using some sort of gui or "vanilla Asterisk" ? |
| 12:51.35 | MikeH | Dovid, I'm using AsteriskGUI but have no qualms in doing it 'by hand' as it were. |
| 13:08.51 | MikeH | I can do it via a voice menu, but I want to do it as a user/extension ideally so I can use it as part of a call group |
| 13:09.01 | *** join/#asterisk af_ (~getsmart@78.134.21.247) |
| 13:12.41 | MikeH | anyone? :/ |
| 13:13.25 | Tozz_ | not sure if I understand right |
| 13:13.35 | Tozz_ | but wouldn't Dial(SIP/number@trunk) do the trick? |
| 13:13.50 | MikeH | where? |
| 13:14.10 | Tozz_ | in your extensions.conf |
| 13:14.29 | Tozz_ | exten => 1234,1,Dial(SIP/number@trunk) |
| 13:14.32 | Tozz_ | where 1234 is the incoming number |
| 13:15.32 | MikeH | under any particular heading? |
| 13:16.38 | *** join/#asterisk csnook (~chris@c-76-19-64-161.hsd1.ma.comcast.net) |
| 13:17.23 | Tozz_ | perhaps you need to read some documentation to get the basics |
| 13:18.40 | MikeH | well, I've managed everything else but this. |
| 13:33.01 | MikeH | hrm, so I fugured that out |
| 13:36.41 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 13:36.43 | MikeH | hrm, it would seem you cannot use a ringgroup in Goto() |
| 13:36.47 | MikeH | is there another way to do it? |
| 13:39.08 | Tozz_ | what is it you want to accomplish? |
| 13:40.45 | MikeH | to have both a SIP extension and an 'extension' by dialing out ring at the same time |
| 13:40.47 | MikeH | I'm close |
| 13:41.01 | MikeH | I've set up 6001 which dials an external number. |
| 13:41.18 | *** join/#asterisk guilhermebr (~Guilherme@189.63.57.56) |
| 13:41.19 | MikeH | In my voice menu, I've done exten = 1,1,Dial(SIP/6000&Local/6001,40,${DIALOPTIONS}i) |
| 13:41.29 | MikeH | DOesn't quite work as desired though |
| 13:41.44 | MikeH | 6001 only rings for one ring. |
| 13:43.14 | MikeH | so I think it is calling 6000 for one ring, giving up, then you get blank while it tryies to connect my DIalout on 6001, then you hear the dialout ringing. |
| 13:44.36 | MikeH | *6000 only rings for one ring, rather |
| 13:52.14 | leifmadsen | MikeH: well that would mean 6000 is either answering, rejecting the call, or something else |
| 13:52.32 | MikeH | nope |
| 13:52.47 | MikeH | I think my issue is that I'm using failover macro for 6001 |
| 13:52.58 | leifmadsen | well all we can do is guess since you haven't provided any debugging information |
| 13:53.25 | MikeH | I'm not sure how to select an analogue trunk when using DIAL() |
| 13:55.54 | *** join/#asterisk slackytude (~slacky@drms-590ddd31.pool.mediaWays.net) |
| 13:59.10 | MikeH | nope, that isn't the issue :/ |
| 13:59.50 | MikeH | I figured out how to do it directly to the trunk |
| 14:00.03 | MikeH | the only thing in console other than echo cancellation errors is ' == Spawn extension (voicemenu-custom-1, 1, 1) exited non-zero on 'DAHDI/5-1'' |
| 14:00.28 | MikeH | this is what I have in my voice menu: exten = 1,1,Dial(SIP/6000&DAHDI/g2/075154xxxx,40,${DIALOPTIONS}i) |
| 14:04.27 | leifmadsen | glad you figured it out |
| 14:05.03 | *** join/#asterisk hehol (~Adium@ip-78-94-0-76.unitymediagroup.de) |
| 14:05.03 | MikeH | I get the same issue though, 6001 rings once, then silence while the second dial connects, then the mobile rings. |
| 14:05.17 | MikeH | It seems I can do this with 'followme' in AsteriskGUI though, so I'm going to try this. |
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| 14:16.42 | MikeH | why is nothing ever simple |
| 14:16.45 | MikeH | followme doesn't seem to work |
| 14:19.26 | *** join/#asterisk linuxplatform (~fedora@188.148.96.149) |
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| 14:43.27 | MikeH | Hrm, the above issue is present when using queues |
| 14:43.36 | MikeH | I set up a queue with two agents |
| 14:43.55 | MikeH | a SIP extension and DAHDI/g2/07xxxxxx |
| 14:44.05 | MikeH | call the queue, sip extension rings once |
| 14:44.12 | *** join/#asterisk ariel_ (~chatzilla@63.214.236.169) |
| 14:44.16 | MikeH | then it moves on to ringing the mobile |
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| 14:54.19 | *** join/#asterisk mechbangirc (~mechbangi@mbl-65-157-32.dsl.net.pk) |
| 14:55.16 | mechbangirc | hi i am calling a person in a meetme conference through Local channel. Why dont i hear his ring tone? |
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| 15:25.50 | cusco | how do you call someone trough a meetme?» |
| 15:25.58 | cusco | once you're in a meet me you'r not calling anyone |
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| 15:59.20 | benlangfeld | Hey guys, I'm trying to get Dial() NOT to cause the calling party to hear a ringing tone, but I can't find anything to this effect. I can only find the inverse. Does anyone know if this is possible? |
| 15:59.26 | *** join/#asterisk iulhk (iulhk@117.102.38.37) |
| 15:59.28 | iulhk | hi all |
| 15:59.44 | cusco | benlangfeld: there is a flag that makes them ear moh instead |
| 16:00.04 | benlangfeld | cusco: So just give them silence for MOH? Good idea. Cheers |
| 16:00.42 | iulhk | i hv installed asterisk now testing video call, at my server i hv firewall configured, i hv allowed ports for sip 5060, for rtp 10000:20000, do i need to enable ports for video rtp too? |
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| 16:02.56 | cusco | check /etc/asterisk/h323.conf there is a port arg |
| 16:02.59 | cusco | other than that, no |
| 16:03.50 | *** join/#asterisk SparFux1 (~raoul@g224208016.adsl.alicedsl.de) |
| 16:05.09 | SparFux1 | Hello all. With the DAHDI driver, is the xpp part able to drive HFC PCI 2 channel bri cards by today? |
| 16:05.33 | iulhk | <cusco> : for video i m using sip not h323 ,,, do i still need to check h323? |
| 16:05.47 | WIMPy | SparFux1: xpp? |
| 16:07.05 | youngproguru | Anyone used Android 2.3 SIP with Asterisk. It is Awesome on Verizon Data. The only catch is that your SIP Realm needs to match your public Domain or SIP will not authenticate. |
| 16:07.47 | chiwawa_42 | WIMPy: in case you're still interested in our little experiment, yate + IAXconfig + mgetty let us get a serial shell on a linux box with a v.21 modulation (300bps). That all we could get yet with existing software, because most hardware modems don't support any other modulation |
| 16:07.59 | WIMPy | SparFux1: What kind of hardware are you on about? |
| 16:08.03 | chiwawa_42 | having v32 or v34 would mean a great deal |
| 16:08.57 | *** join/#asterisk angryuser_laptop (~angryuser@90-156-167-83.reverse.alphalink.fr) |
| 16:09.34 | MikeH | Any suggestions why simultaneous dials and queues would cause a sip client in the list to jsut ring once then stop, and then move onto the second number (external) |
| 16:10.13 | MikeH | http://forums.digium.com/viewtopic.php?f=1&t=76948&sid=2c9b0487d6df40b0aec24f2c0c415cb2 |
| 16:10.17 | WIMPy | chiwawa_42: V.21 sonds VERY different from V.34. I can test up to V.90 (bot directions). |
| 16:10.18 | *** join/#asterisk weta (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com) |
| 16:11.04 | chiwawa_42 | WIMPy: well, we have no software implementation of v.90 or v92 to glue to spandsp :s |
| 16:11.39 | WIMPy | chiwawa_42: If you have V.34, V.90 should be easy, I guess. |
| 16:12.11 | chiwawa_42 | we don't have v.34, spandsp author said its cluttered by too many pattents to be distributed in an open-source package |
| 16:12.28 | coppice | V.90 takes a *lot* more code than V.34, and V.34 is a big job |
| 16:12.33 | SparFux1 | WIMPy: it's a cologne HFC-S PCI bri card. |
| 16:12.54 | SparFux1 | WIMPy: I find stuff in drivers/dahdi/xpp about HFC and bri. That's why I am asking myself. |
| 16:13.10 | chiwawa_42 | coppice: v32 or v32bis would be sufficient |
| 16:13.25 | WIMPy | SparFux1: Then you want wcb4xxp. |
| 16:13.37 | WIMPy | Try http://voice.yeti.dk/Asterisk_vs_ISDN :-) |
| 16:13.59 | coppice | well, V.32bis is like 2 V.17 modems back to back + echo cancellation + an elaborate new training sequence |
| 16:14.12 | SparFux1 | WIMPy: means, I do not even need the zaphfc stuff indeed? |
| 16:14.28 | WIMPy | coppice: Why should V.90 be such a big deal once you've got V.34 working? But I agree that V.34 prabably is a big deal. |
| 16:14.44 | chiwawa_42 | coppice: so, having v17 already supported by spandsp, v32bis could actually be possible to implement ? |
| 16:15.21 | coppice | WIMPy: V.90 is V.34 in one direction, and something horribly complex to doin the other |
| 16:16.14 | WIMPy | Complex? As it uses the full bandwidth there can't be any modulation? |
| 16:16.27 | chiwawa_42 | coppice: all we want is a full duplex modulation, fast enough to use text-mode IP applications |
| 16:16.33 | SparFux1 | WIMPy: I think wcb4xpp is for quad bri. I have single bri I am afraid. |
| 16:17.08 | WIMPy | SparFux1: Didn't you say dual above? |
| 16:17.24 | SparFux1 | WIMPy: sry, dual channel I ment two channels only. |
| 16:17.31 | chiwawa_42 | coppice: the goal of the project is to spawn as many dial up services as possible for the egyptian people, using SIP trunks to transport the channels from local phone numbers |
| 16:17.33 | coppice | WIMPy: what do you mean by that? |
| 16:17.34 | WIMPy | Ah |
| 16:17.45 | SparFux1 | WIMPy: couldn't in theory the quad bri be tweaked to support single bri? |
| 16:17.46 | MikeH | WIMPy, When you mentioned about simply using dial() when wanting to divert to an outside line last night, are there any caveats to it working like this? |
| 16:17.56 | MikeH | I can't get simultaneous calls working properly like this |
| 16:18.03 | coppice | chiwaw42: are you the person who e-mailed me yesterday about this? |
| 16:18.04 | WIMPy | SparFux1: Ok, then you find the answer(s) on that page. |
| 16:18.21 | chiwawa_42 | coppice: if you're the author of spandsp, yes, that's me |
| 16:18.37 | chiwawa_42 | Steve ? is that you ? :p |
| 16:18.38 | SparFux1 | ok thx. just stepped over it again because of the bri stuff in xpp dir. I once saw this page, but 'll check it out again more thoroughly :-) |
| 16:18.53 | WIMPy | coppice: It transmits 64kbps on a 64kbps channel. So that must be bit transparent. |
| 16:19.12 | chiwawa_42 | WIMPy: yes but it has to be supported by hardware PSTN modems too |
| 16:19.29 | WIMPy | MikeH: There's nothing special, no. |
| 16:19.40 | MikeH | WIMPy, any ideas on this - http://forums.digium.com/viewtopic.php?f=1&t=76948&sid=2c9b0487d6df40b0aec24f2c0c415cb2 ? |
| 16:19.48 | coppice | WIMPy: not at all. it has to get through an analogue link of poorly known characteristics to get to the 64k digital channel, and you need to jump through hoops to make that work |
| 16:20.14 | MikeH | essentially, the sip extenion rings once then stops, then it dials out to the mobile |
| 16:20.23 | MikeH | rather than them both rining simultaneously |
| 16:20.26 | MikeH | *ringing |
| 16:20.38 | coppice | WIMPy: the client and server V.90 modems are very different, but both very complex |
| 16:20.41 | WIMPy | coppice: But there is nothing you CAN do. You only have 64kbit. |
| 16:21.00 | SparFux1 | WIMPy: have you ever tried to use the quad bri with single bri hfc card? |
| 16:21.40 | WIMPy | MikeH: Maybe that doesn;t work with analog lines? No idea. |
| 16:21.41 | *** join/#asterisk moy (~moy@70.25.53.39) |
| 16:22.04 | WIMPy | SparFux1: That doesn't work. |
| 16:22.14 | SparFux1 | hm... ok, thx for the info :-) |
| 16:22.26 | *** join/#asterisk Tim_Toady (~moi@188.4.101.65.dsl.dyn.forthnet.gr) |
| 16:22.58 | WIMPy | coppice: I know. The "server" side needs a bit transparent connection. |
| 16:23.16 | *** join/#asterisk |Physis| (~physishec@187.79.206.208) |
| 16:23.26 | WIMPy | SparFux1: Maybe it could be patched. I don't know. |
| 16:24.03 | SparFux1 | WIMPy: yes, would be interesting if this would be possible. Maybe it's more elegant than having zaphfc for single bri. |
| 16:24.32 | WIMPy | I don't know how much differences there are between single and multi port chips. |
| 16:27.21 | WIMPy | But I didn't notice the xpp stuff was about HFC. I should take a closer look at that, I guess. |
| 16:27.43 | SparFux1 | WIMPy: I found it by accident due to latest svn update :-P |
| 16:28.26 | SparFux1 | drivers/dahdi/xpp/card_bri.c is all about bri stuff. |
| 16:28.55 | MikeH | WIMPy, Are there any other options? |
| 16:29.32 | WIMPy | MikeH: Turn up debug and verbose and see if there's any hint, what's going on. |
| 16:30.18 | WIMPy | MikeH: But I suspect interface configuration/limitations. |
| 16:30.43 | coppice | A V.90 modem is an amazing piece of work, it uses practically every modulation trick so far devised to get the data through a single analogue link |
| 16:31.20 | MikeH | WIMPy, is there any such mechanism such as a virtual line, or even a sip client that could handle the call forwarding? |
| 16:31.43 | WIMPy | MikeH: Not sure what you mean by that. |
| 16:32.01 | chiwawa_42 | coppice: but is it robust ? I mean, can it still work on a degraded voice channel ? I guess older protocols will be more resilient, aint them ? |
| 16:32.02 | MikeH | WIMPy, Well you seem to suggest that perhaps its a limitation of the interface |
| 16:32.15 | WIMPy | coppice: I'd still like to know how you could possibly modulate 64kbps into 64kbps. |
| 16:32.29 | MikeH | WIMPy, Perhaps a solution is to have something third party like a sip client take the call and call forward back via asterisk? |
| 16:33.01 | WIMPy | MikeH: Yes, it probably isn;t ablte to or not configured to detect answer. |
| 16:33.20 | WIMPy | MikeH: What difference could that make? |
| 16:33.57 | MikeH | hrm fair point |
| 16:34.00 | coppice | chiwawa42: if you want a modem that's robust enough to work over a variety of paths, you want V.32bis or V.34 |
| 16:34.14 | MikeH | so you're suggesting that it is probably a case that what I want to do just isn't possible? |
| 16:35.08 | chiwawa_42 | coppice: so, let's focus on the simpliest to implement ;) |
| 16:35.31 | WIMPy | MikeH: Possibly. I don't know if thare's any polarity reversal on answer possible that might just need to be configured. |
| 16:35.42 | MikeH | I've enabled console debug and verbose in logging.conf, but I don't seem to get anything more through the console than I did before :/ |
| 16:35.48 | WIMPy | MikeH: But analog does have some limitations. |
| 16:38.52 | MikeH | WIMPy, I think you were right. If I turn Answer on polarity switch to on for the GSM card, it works |
| 16:39.03 | MikeH | except when I answer it doesn't detect. :/ |
| 16:39.56 | WIMPy | On the GSM??? That is digital and shouldn't cause any problems. |
| 16:40.33 | MikeH | GSM is listed as analog |
| 16:41.12 | WIMPy | That doesn't make sense. |
| 16:45.11 | MikeH | :/ |
| 16:45.12 | MikeH | pass |
| 16:45.36 | MikeH | call progress doesn't work either |
| 16:46.49 | MikeH | is there no way to detect answer like followme does? |
| 16:47.06 | MikeH | Ie. play a message and wait for a specified dtmf tone? |
| 16:47.10 | WIMPy | MikeH: Maybe the driver is just incomplete? |
| 16:47.59 | WIMPy | I think there is an option to Dial(), yes. |
| 16:48.02 | Kyosh | besides x-lite, whats a decent SIP phone thats not crippled (like x-lite)? |
| 16:48.30 | Kobaz | twinkle |
| 16:48.43 | *** join/#asterisk hehol (~Adium@ip-78-94-0-76.unitymediagroup.de) |
| 16:48.59 | Kobaz | it would be cool if there was a twinkle for windows though |
| 16:49.50 | MikeH | WIMPy, Perhaps I'll speak to Atcom and see if it can be implemented |
| 16:50.54 | WIMPy | MikeH: I think that kind of functionality should be expected from a product with that price. |
| 16:51.13 | WIMPy | But maybe it's jsut some fancy configuration option. |
| 16:51.18 | MikeH | are there any alternative gsm cards that actually work? |
| 16:51.45 | WIMPy | I haven't tried any. |
| 16:51.57 | WIMPy | Junghanns.net springs to mind. |
| 16:53.03 | WIMPy | As those guys also did the bristuff, I think it's safe to assume that kind of thing works with their cards. |
| 16:53.11 | MikeH | lol |
| 16:53.13 | MikeH | £772 |
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| 16:54.40 | WIMPy | The cheapest option would be an USB stick with chan_datacard. But I can't comment on that, either. The sticks I've got have voice disabled :-( |
| 16:57.16 | *** join/#asterisk manji (~manjiki@adsl-141-49.adsl.ntua.gr) |
| 17:05.42 | WIMPy | Hmm. xpp "without BRISTUFF support". Looks like that's getting ugly once again. |
| 17:10.19 | WIMPy | But certainly nothig from dahdi_hardware or dahdi_scan. |
| 17:11.39 | WIMPy | halt |
| 17:11.49 | WIMPy | oops |
| 17:13.11 | iulhk | getting problem when trying video call, call disconnecting after 15-20 seconds, pls check the console logs at "http://paste.ubuntu.com/560304/"? |
| 17:30.53 | youngproguru | Your disconnect might be the SIP Timer setting. I had the same issue with Aastra phones, and Asterisk 1.6 |
| 17:40.17 | manji | iulhk, you are probably behind NAT |
| 17:40.33 | manji | and the ACK packet from the other end |
| 17:40.41 | manji | never made it to your asterisk bo |
| 17:40.42 | manji | x |
| 17:40.56 | manji | so you need a port forwarding |
| 17:41.37 | iulhk | <manji>: one other thing, for audio we have to enable 10000:20000 ports for rtp, for video do i need to enable any other ports? |
| 17:41.54 | manji | iulhk, I don't think so |
| 17:43.03 | iulhk | <manji>: another things .. as i m already behind the nat and where i m , there is sip port block so i hv already forwareded some other non-standard ports redirect to 5060, do u think it can b disconnecting ? |
| 17:43.42 | manji | could be the reason, yeap |
| 17:44.04 | *** join/#asterisk ruied (~ruied@po-217-129-154-202.netvisao.pt) |
| 17:46.57 | p3nguin | RTP is "media." I think that encompasses all media, like audio and video. |
| 17:47.56 | drmessano | Yep, which is why asterisk is able to do video to the limited extent that it does |
| 17:48.08 | WIMPy | That's what I think as well. And the message moans about SIP, not RTP. |
| 17:48.10 | drmessano | It just directs the RTP stream |
| 17:56.49 | ruied | Hi! I'm using asterisk 1.8.3. When I',m compiling lcr with chan_lcr, it reports an error ralated to asterisk channel.h:1099. I'm using lcr 1.7 and 2.6.37 kernel. Could this be something related to asterisk or to lcr? |
| 17:57.51 | p3nguin | 1.8.3 has already been released? |
| 17:58.00 | tzafrir_laptop | ruied, please pastebin more details |
| 17:58.08 | tzafrir_laptop | it has been tagged |
| 17:58.25 | tzafrir_laptop | but maybe he means 1.8.2.3 |
| 17:59.34 | ruied | tzafrir_laptop, yes, it's 1.8.2.3 http://pastebin.com/kx9rbQAW |
| 17:59.45 | *** join/#asterisk fskrotzki_ (~fskrotzki@cpe-74-74-245-250.rochester.res.rr.com) |
| 17:59.50 | WIMPy | ruied: For Asterisk 1.8 you need the lcr asterisk_1_8 branch from git. |
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| 18:01.01 | WIMPy | Or the patch from http://voice.yeti.dk/patches/ |
| 18:02.22 | ruied | WIMPy, ok, going to take a try |
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| 20:42.57 | neurosys | I have a cisco 871 router. From wireshark info, it appears that somehow, asterisk picks up the routers externip for the record-route. it is at .177... but the asterisk system is natted to .178. It happens randomly and of course the call disconnects when the SIP ACK doesnt return to .178, it returns to .177 |
| 20:43.00 | neurosys | Any thoughts? |
| 20:47.01 | StaRetji | check your router, it must be some mistake with NAT |
| 20:47.20 | neurosys | StaRetji: I would believe you're right. Im just not a proficent with IOS and NAt :( |
| 20:47.37 | neurosys | StaRetji: I just find it strange that it's seemingly random |
| 20:47.46 | StaRetji | IOS? |
| 20:47.52 | StaRetji | like iphone os? |
| 20:48.00 | neurosys | No. Cisco IOS |
| 20:48.03 | StaRetji | lol |
| 20:48.40 | StaRetji | well, don't know about cisco, but I'm familiar with Mikrotik and networking |
| 20:48.46 | StaRetji | it goes like this |
| 20:49.05 | StaRetji | local ip (i.e 10.0.0.2) is address of your asterisk server |
| 20:49.47 | StaRetji | public ip (ie 192.168.0.178) is public address which you have to assign to Asterisk |
| 20:51.07 | StaRetji | so, src-nat or source nat should be: src-address 10.0.0.2 src-nat 192.168.1.178 |
| 20:51.12 | neurosys | Oh i understand the concept. Like I said.. it works... but every so often, * will get the routers externIP and place it in the record-route. The address for the router (.177) not the staticed nat IP (.178) |
| 20:51.16 | StaRetji | now, you have to do vice versa for dst-nar |
| 20:51.24 | manji | neurosys, your router is sip aware |
| 20:51.37 | manji | and it is messing with your sip messages |
| 20:51.50 | manji | search on google how to disable SIP on IOS nat |
| 20:51.50 | neurosys | manji: Not that Im aware of. also... I do not have any type of inspect sip or global policy for classes |
| 20:52.05 | manji | neurosys, might be by default |
| 20:52.29 | manji | but if your router is changing things inrecord route |
| 20:52.38 | StaRetji | well, I'll be damned, who would put something like that in router, by default :/ |
| 20:52.46 | neurosys | hmm. If that were the case, I should be able to remove the externip variable and it should "just work" |
| 20:52.54 | manji | then it is doing more than forwarding packets |
| 20:52.56 | neurosys | StaRetji: CISCO! :P |
| 20:53.15 | manji | NAT is rewritting the TCP/UDP headers |
| 20:53.30 | manji | and only |
| 20:53.55 | neurosys | manji: The packet header has the source correctly (.178) , its inside the sip header that has the new record-route added |
| 20:54.08 | manji | neurosys, that's what I am saying |
| 20:54.39 | manji | no one but a sip proxy can do that |
| 20:54.41 | manji | write? |
| 20:54.46 | manji | right? |
| 20:55.19 | manji | neurosys, I may be wrong, but you can just chase it up a bt |
| 20:55.19 | neurosys | manji: I would say that's right. |
| 20:55.20 | manji | bit |
| 20:55.32 | StaRetji | well, cisco sux |
| 20:55.41 | neurosys | StaRetji: hehe |
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| 20:55.58 | manji | StaRetji, it doesnt suck |
| 20:56.16 | StaRetji | I just checked Mikrotik, if you want to set something like that, you use mangle, blah blah |
| 20:57.22 | manji | neurosys, mmm, you are not behind nat, I just realized |
| 20:57.31 | neurosys | manji: I am |
| 20:57.48 | manji | ok |
| 20:57.53 | manji | I misread |
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| 21:11.57 | zgor | hi :) |
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| 23:59.42 | CoffeeIV | I am debugging why 1.6.2 (on Ubuntu) doesn't play a entry and exit beep on conference calls. It's using ConfBridge, and the q option is not set. But in logs I don't see any failure to read a wav file or anything. Any suggestions ? |