| 00:00.00 | sawgood | 5555 will always match, but it might not be online at the time |
| 00:00.13 | sawgood | so, this would be a failover to a cell phone if 5555 does not accept the call |
| 00:00.38 | beek | sawgood: You need to decide how long to ring the phone, then if there's no answer just dial your cell phone number. if it answers the call will be bridged. |
| 00:00.44 | beek | I do this at home. |
| 00:00.46 | WIMPy | sawgood: Check DIALSTATUS and/or HANGUPCAUSE and then do a 2nd Dial(). |
| 00:00.54 | sawgood | nice ... |
| 00:01.14 | citywok | what wimpy said. |
| 00:03.00 | WIMPy | And maybe some time we also get a HANGUPLOCATION if we're lucky. |
| 00:03.19 | sawgood | This example is for SIP trunking not a PRI call (sorry) |
| 00:03.43 | beek | sawgood: A dial is a dial... |
| 00:03.55 | sawgood | is a dial (got it) |
| 00:03.59 | beek | Don't think of the cell phone as anything special. |
| 00:04.05 | WIMPy | Or at least it should be. |
| 00:04.15 | beek | The only issue you may have is no outbound line available (or SIP provider, if you're using that) |
| 00:04.39 | sawgood | Is DIALSTATUS more or less like a NoOp statement |
| 00:04.48 | beek | sawgood: One other complication is if you're expecting to dump into voicemail box. |
| 00:05.06 | beek | DIALSTATUS is a channel variable that shows the results of the last DIAL call |
| 00:05.22 | sawgood | beek: just a fail-over so calls are not missed if the IP PBX is down |
| 00:05.43 | beek | sawgood: Again, not a biggie but something to keep in mind. |
| 00:07.19 | sawgood | So, if the first statement is the DIAL command to the SIP line, and the 2nd line is to print off the status of the dial, what would the 3rd line (2nd dial command look like)? |
| 00:07.40 | sawgood | why do I need to know the status ... why not just go straight to a 2nd dial command? |
| 00:08.05 | beek | You can get away without it. |
| 00:08.43 | beek | It depends on your use case. |
| 00:08.43 | beek | Let's say that the extension of interest is yours. |
| 00:08.58 | beek | If DIALSTATUS is "BUSY" then do you really want your cell phone to ring? |
| 00:09.01 | sawgood | So, I could have two identical DIAL lines (and if the 1st fails), the 2nd attempt will start a short time after that? |
| 00:09.11 | sawgood | no I don't! |
| 00:09.12 | beek | You're already on a call, so you might want to go straight to voice mail. |
| 00:09.15 | sawgood | very good point! |
| 00:09.28 | beek | has been there, done that. |
| 00:09.38 | beek | @#$%ing cell phone |
| 00:11.13 | sawgood | exten => 4155554330,1,Dial(SIP/5555/${EXTEN}) |
| 00:11.13 | sawgood | exten => 4155554330,1,Dial(SIP/2009/4155551212) |
| 00:11.16 | sawgood | would this work? |
| 00:11.24 | WIMPy | no |
| 00:11.33 | sawgood | that was a fast NO! |
| 00:11.42 | beek | Well, you'd need to change 0,1,Dial to 0,n,Dial on the second. |
| 00:11.44 | WIMPy | The priorities must be numbered correctly. |
| 00:12.05 | sawgood | exten => 4155554330,1,Dial(SIP/5555/${EXTEN}) |
| 00:12.05 | sawgood | exten => 4155554330,n,Dial(SIP/2009/4155551212) |
| 00:12.10 | beek | Yes |
| 00:12.28 | beek | But again, if ${EXTEN} is busy then it's gonna dial SIP/2009/... |
| 00:12.43 | sawgood | how will * Asterisk 'know' when to give up on the priority 1 and move to the 'n' |
| 00:12.57 | WIMPy | It always will if the forst dial wasn't answered. |
| 00:13.16 | WIMPy | You should give a timeout to Dial. |
| 00:13.17 | beek | There's a timeout |
| 00:13.27 | beek | Yeay, what WIMPy said. |
| 00:13.40 | WIMPy | In your exaple it will only move on if the first Dial failed for whatever reason; |
| 00:13.48 | beek | And don't forget, the entire time this process is going on your caller is wondering WTF is going on. |
| 00:15.32 | beek | Also sawgood, look up the FollowMe application |
| 00:15.42 | beek | That's essentially what you're implementing. |
| 00:16.07 | sawgood | ty |
| 00:17.02 | beek | If you're really lazy, just dial 'em both at the same time and pick up whichever you wish |
| 00:17.11 | beek | sawgood: ^^^^^^^^^^^ |
| 00:18.22 | beek | Dial(SIP/5555/${EXTEN}&SIP/2009/4155551212) |
| 00:18.41 | sawgood | that won't work but I thought about it |
| 00:19.29 | beek | That's what I use at home, actually. I did what you did before then set it up to dial a phone at home, my cell phone and a SIP phone on my office desk simultaneoulsy. I only give my home phone number out now. |
| 00:20.42 | sawgood | that could be very handy too |
| 00:25.27 | beek | That's the problem with this damned Asterisk project... so many toys, so little time. |
| 00:26.32 | carrar | Best to just say to hell with it and go hiking |
| 00:26.53 | beek | carrar: I'd just end up clutching my chest and dropping over in the middle of the woods. |
| 00:26.53 | *** join/#asterisk csnook (~chris@209-6-38-66.c3-0.smr-ubr1.sbo-smr.ma.cable.rcn.com) |
| 00:29.40 | *** join/#asterisk coppice (~chatzilla@210.17.255.115) |
| 00:32.20 | carrar | thats not a good sign |
| 00:42.46 | *** join/#asterisk SaiSoma|AtHome (~SaiSoma@adsl-074-167-136-030.sip.mob.bellsouth.net) |
| 00:44.08 | SaiSoma|AtHome | evening folks. anyone know of a decent survey system for asterisk? not necessarily free (but that always helps!) |
| 01:01.47 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 01:04.36 | pabelanger | SaiSoma|AtHome: roll your own ;) |
| 01:04.42 | *** join/#asterisk kn0x (~pinochle@74.63.68.42) |
| 01:04.44 | pabelanger | vicidial would also be an option |
| 01:05.23 | SaiSoma|AtHome | pabelanger: *nod* i certainly can roll my own . :) . .but thought I'd try to see if there were existing packages we could easily insert and save my client some $$ |
| 01:05.29 | SaiSoma|AtHome | i'll check out vicidial, thanks |
| 01:05.55 | justdave | if my asterisk server has more than one network interface, is there a way to specify a source IP that a traffic to a given sip peer should always go out via? |
| 01:06.17 | justdave | or will static-routing that IP at the OS level be sufficient? (and asterisk will figure out the correct source IP by the route?) |
| 01:06.23 | WIMPy | justdave: Only your routing table. |
| 01:07.32 | WIMPy | It's more like you need to take care of everything at OS level. Last time I tried a multi-homed Asterisk that didn't wotk out too well |
| 01:09.09 | *** join/#asterisk distributed (~root@66.241.104.83) |
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| 01:15.07 | *** part/#asterisk bougyman (bougyman@pdpc/supporter/gold/bougyman) |
| 01:15.27 | cj | can one of you recommend a 1U tdm to sip gateway with an rj-21 interface? |
| 01:16.10 | cj | s/sip/iax/ if you like |
| 01:18.40 | kn0x | russellb: where did TK go? |
| 01:19.22 | coppice | to a better place |
| 01:19.40 | kn0x | ;) |
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| 01:20.00 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 01:22.54 | cj | hurm. I'm thinking I'll just trunk the copper at the firehouse and patch them in directly to the 911 echange there. I don't want to have to do foreign exchange. wah. |
| 01:23.13 | cj | do any of you people run CLECs? |
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| 01:27.49 | *** mode/#asterisk [+o russellb_] by ChanServ |
| 01:29.02 | sawgood | WIMPy: anything special or different need to be done when using a Sagnoma PRI card vs a TE110 by Digium? |
| 01:29.15 | sawgood | Asterisk is not seeing the card in /etc/dahdi/system.conf |
| 01:31.27 | WIMPy | You need the sangoma drivers. |
| 01:32.19 | *** join/#asterisk root52 (~root52@ip70-191-116-76.cl.ri.cox.net) |
| 01:32.24 | sawgood | oh ... ok ... from the Sagnoma site I take it then |
| 01:32.36 | leifmadsen | heh ya :) |
| 01:32.50 | leifmadsen | Sangoma card, Sangoma drivers. Digium card, Digium drivers :) |
| 01:33.26 | sawgood | hello Mr. Madsen nice to meet you actually |
| 01:34.00 | leifmadsen | hides |
| 01:34.24 | leifmadsen | and now I have to run off and finish my Guinness and the hardware I'm installing for my wife |
| 01:34.35 | pabelanger | Is Gosub() > Macro() ? |
| 01:34.38 | beek | That last statement begs the question... |
| 01:34.42 | beek | leifmadsen: ^^^^^^ |
| 01:35.12 | sawgood | Sangoma it is |
| 01:35.30 | kn0x | cj: carrier, but not a clec |
| 01:35.39 | leifmadsen | pabelanger: that is a question with no answer |
| 01:35.50 | leifmadsen | pabelanger: they do similar things in a different manner |
| 01:36.02 | beek | I thought that Macro() was being phased out. |
| 01:36.06 | leifmadsen | fuck off |
| 01:36.11 | leifmadsen | that is not true |
| 01:36.12 | leifmadsen | :) |
| 01:36.21 | leifmadsen | people keep spreading that nasty rumour and it's absolutely not true. |
| 01:36.28 | leifmadsen | Macro() is not being removed |
| 01:36.55 | beek | I can't remember where I read that it was being deprecated, but I'm glad that it's not. I have a ton of Macro()s written and wasn't looking forward to converting them. |
| 01:36.58 | leifmadsen | I use both on a regular basis, but try to use GoSub() as my first line of defense |
| 01:37.16 | leifmadsen | deprecation does not mean removal |
| 01:37.33 | beek | It usually does infer it however... |
| 01:37.45 | leifmadsen | deprecation means, "hey, that thing over there is better than this, and will be supported better than this thing, but if this works for you, it won't go away" |
| 01:37.51 | leifmadsen | in the case of Asterisk, it does not |
| 01:37.58 | leifmadsen | we do not remove functionality (anymore) |
| 01:38.01 | *** join/#asterisk ariel_ (~chatzilla@c-98-254-138-7.hsd1.fl.comcast.net) |
| 01:38.18 | leifmadsen | it may get lower or nearly no support, but the code will not be removed |
| 01:38.19 | beek | That makes me happy as there are some reasons for using it still that are compelling. |
| 01:38.23 | leifmadsen | exactly |
| 01:38.28 | leifmadsen | like I said, I use both |
| 01:38.42 | leifmadsen | and with all of that said, please spread the rumour that Macro() isn't being removed |
| 01:38.48 | beek | Happily. |
| 01:38.52 | leifmadsen | and now I'm going back to being an awesome husband |
| 01:38.59 | beek | does the happy dance. |
| 01:39.19 | beek | GN leifmadsen |
| 01:41.37 | root52 | So, Trying to do a pickup. Using default feature code *8 I have pickup and call groups defined in sip.conf. (in fact I have almost the same setup working just fine on two other servers). When I try to do it I get. "chan_sip.c:21613 handle_request_invite: Nothing to pick up for..." and a sip debug show that it is "looking for *8 in my phones context. Is that supposed to happen? Any Thoughts? |
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| 02:10.37 | sawgood | I can't seem to install Sangoma Wanpipe driver suite because during the installation it keeps asking for a path to the Zaptel drivers/modules |
| 02:10.46 | sawgood | any idea on how to over-come this? |
| 02:11.26 | sawgood | Please select the working zaptel directory |
| 02:18.49 | sawgood | got it |
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| 02:42.09 | jaytee | ~book |
| 02:42.09 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook, or http://ofps.oreilly.com/ |
| 02:59.34 | carrar | ~library |
| 03:02.30 | jaytee | ~snark |
| 03:02.31 | infobot | rumour has it, snark is at http://www.sans.org/newlook/resources/IDFAQ/knark.htm |
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| 03:07.58 | pabelanger | heh, flag /n for Local/ FTW |
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| 03:18.36 | yagoo | I'm getting this issue. I have asterisk and two lan softphone SIP, this occurred yesterday as well -- with ekiga it's saying 'Transport error' under the account box (the first time it work perfectly).. i'm tring canreinvites=yes but it doesn't help.. |
| 03:18.39 | yagoo | anyone? |
| 03:20.05 | pabelanger | I would have though, same => n,Dial(${ARG1},${ARG2},M(UpdateRecord^${DIALSTATUS})) would pass the correct DIALSTATUS to UpdateRecord macro. Seems not to be set |
| 03:20.25 | pabelanger | I guess I can assume the channel is ANSWERED since the macro was fired |
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| 03:54.18 | pabelanger | WTB: *CLI> core show variable DIALSTATUS |
| 03:54.22 | pabelanger | that would be nice |
| 03:58.22 | *** join/#asterisk yagoo (~jb@modemcable197.51-201-24.mc.videotron.ca) |
| 03:58.25 | yagoo | hi |
| 03:58.36 | yagoo | anyone has video working with asterisk? |
| 03:59.01 | yagoo | a simple local ethernet test would do.. |
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| 04:12.06 | carrar | Who wants to really see who they are talking too? |
| 04:13.09 | carrar | Maybe if I did Video conferences to Paradise Island |
| 04:13.14 | carrar | might be worth it |
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| 04:39.15 | cj | kn0x: can you recommend some hardware to provide minimum regulatory requirements for a clec? |
| 04:39.41 | cj | kn0x: I don't want to provide local exchange service, just 911 and dsl on the copper |
| 04:40.10 | cj | once the dsl is paying for the copper, I will start offering voice as well |
| 04:40.44 | atan2 | Similar to exten =>a,1,VoiceMailMain(${mailbox}) is there a way I can allow the called party to "press # to accept fax" type deal? |
| 04:41.25 | atan2 | So an inbound fax to the asterisk box gets picked up by a sip peer, and that peer learns it's a fax, so they press ## or something which moves the call to another context, like, fax-in or something. |
| 04:54.43 | pabelanger | atan2: features.conf, applicationmap |
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| 05:05.01 | atan2 | pabelanger, great! |
| 05:05.27 | atan2 | I have Set(__DYNAMIC_FEATURES=testfeature) within my inbound dialplan and the tt-monkey testfeature uncommented, reloaded, and #9 doesn't trigger it. |
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| 05:12.31 | atan2 | Wrong order in my dialplan, my bad! Things are good now. |
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| 05:53.35 | UnixDev | how can I get the source ip of the inbound call inside a context? |
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| 06:02.54 | sranil | I have a requirement where I have to allow DTMF to be accepted when a call is going on between the aparty and bparty. Can anyone of you please tell me how to do this ? |
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| 06:43.13 | kaldemar | sranil: do you want something to happen on the server side upon dtmf sequences or just relay them? |
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| 07:06.37 | sranil | kaldemar, I would require to play a message depending on the DTMF |
| 07:14.02 | kaldemar | sranil: see [applicationmap] in http://svn.digium.com/svn/asterisk/tags/1.8.1.1/configs/features.conf.sample |
| 07:16.07 | *** join/#asterisk dimm (~guest@unaffiliated/dimm) |
| 07:16.57 | sranil | kaldemar, thanks very much for your help |
| 07:25.16 | asterisk-learner | hello |
| 07:25.36 | asterisk-learner | does anyone knows if "beep.gsm" has some silence at the end that might cause problems ? |
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| 07:29.45 | kaldemar | asterisk-learner: why are you assuming that silence might cause problems? |
| 07:30.32 | asterisk-learner | kaldemar: I am having the same problem as described here |
| 07:30.34 | asterisk-learner | http://www.opensubscriber.com/message/asterisk-dev@lists.digium.com/3895733.html |
| 07:30.45 | asterisk-learner | however i am just using the "beep.gsm" and not a custom file |
| 07:34.19 | kaldemar | there's about 0.01 s silence at the end of beep.gsm. |
| 07:35.42 | asterisk-learner | kaldemar: mmm do u think this might be a problem ? |
| 07:36.26 | asterisk-learner | i am using it in a custom application and sometimes it executes in the same second (playing file, waiting for stream to end and continue executing my code ) |
| 07:36.44 | asterisk-learner | while in some random cases it can take up to 2 or even 3 minutes to resume to my code |
| 07:41.48 | kaldemar | i really don't see a problem with the sound file here, since there is practically no silence at the end. |
| 07:45.25 | asterisk-learner | kaldemar: mmm do u know what is the problem of silence at the end of the file? i mean why it might cause an delay ? |
| 07:46.41 | kaldemar | there is no problem. if there is silence in the file, asterisk will play silence. |
| 07:47.10 | asterisk-learner | ah i thought there is a bug or so..... |
| 07:48.26 | kaldemar | doesn't look like there was a bug in the mailing list issue, just unexpected silence at the end of the file. that's not the base with beep.gsm. |
| 07:49.02 | asterisk-learner | kaldemar: ok |
| 07:50.26 | asterisk-learner | I am asking this beacuse the threadid of my application is indicating in the full file : <SIP/3822-17690a90> Playing 'beep' (language 'en') |
| 07:50.35 | asterisk-learner | then it resume to my code but after around 3 minutes |
| 07:51.10 | asterisk-learner | there is no deadlock or crash, and asterisk keeps accepting incoming calls in the meantime.... |
| 07:51.25 | asterisk-learner | and the wird thing is |
| 07:51.36 | kaldemar | probably something fishy with your code. |
| 07:51.45 | asterisk-learner | that in other cases, it executes normally, playing the beep and resuming to my code in the same second |
| 07:52.02 | asterisk-learner | mmm |
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| 07:53.59 | asterisk-learner | i forgot to mention that before resuming to my code |
| 07:54.05 | asterisk-learner | i get this in full: channel.c: No/unknown event '0' on timer for 'SIP/3822-17690a90'? |
| 07:54.20 | asterisk-learner | many times (around 6) |
| 07:54.35 | asterisk-learner | that's why i suspected a problem with the file playing..... |
| 07:55.06 | asterisk-learner | where as in the normal execution (same second) i don't get those messages |
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| 08:04.45 | schmidts | good morning |
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| 08:08.48 | yagoo | hey hey hey.. i need an asterisk gui end.. is there an official one? |
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| 08:31.04 | carrar | no you don't |
| 08:31.39 | sranil | In features.conf, [applicationmap], can we configure it such that it can play a dynamic music file (i.e. can the user select the music file at runtime) ? |
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| 09:03.56 | *** join/#asterisk chopp (~chopp@unaffiliated/chopp) |
| 09:08.54 | *** join/#asterisk mechbangirc (~mechbangi@mbl-65-157-37.dsl.net.pk) |
| 09:10.04 | mechbangirc | hi i have two asterisk boxes i want box-1 to be the SIP/IAX provider for box-2. what should i do? |
| 09:11.07 | mechbangirc | also what is the name of this technology in VoIP jargon? |
| 09:11.39 | tuxx- | i think you're searching for the term 'trunking' |
| 09:13.33 | mechbangirc | tuxx: i have just come across 'peering' sounds like what i am looking for. is it same? |
| 09:14.37 | kaldemar | mechbangirc: http://ofps.oreilly.com/titles/9780596517342/ch07.html#OutsideConnectivity_id291235 |
| 09:15.17 | kaldemar | mechbangirc: see "Configuring SIP Trunks Between Asterisk Systems" and "Configuring IAX Trunks Between Asterisk Systems" |
| 09:17.27 | *** join/#asterisk SeTTleR (~bernd@p5DDEE067.dip.t-dialin.net) |
| 09:19.25 | *** join/#asterisk TimeRider (steve@188-220-33-123.zone11.bethere.co.uk) |
| 09:24.22 | mechbangirc | kaldemar: great meterial. In another document i read about register statement. So i do not need a register statement in trunking two boxes togater right? |
| 09:24.34 | mechbangirc | *togather |
| 09:27.38 | kaldemar | mechbangirc: registering is a way to let the other end know where you are. if you define hosts with an address, you don't need registrations. |
| 09:28.37 | mechbangirc | kalmemar: absolutely sensible |
| 09:31.25 | *** part/#asterisk JAMMAN2110 (~JAMMAN211@unaffiliated/jamman2110) |
| 09:33.23 | mechbangirc | kaldemar: so i were to use dynamic ip, i would define host=dynamic, type=friend and use register statement for routing right? |
| 09:35.39 | kaldemar | for example, yes. |
| 09:36.22 | Iron_Chef | I've got a box that was registering for incomming calls, but now refuses to, i can use an ata on the same network with the same vsp, what could be going wrong? |
| 09:37.23 | Iron_Chef | sip debug throws up SIP/2.0 401 Unauthorized over and over, but I can't see anything more than that |
| 09:39.50 | Iron_Chef | perhaps i need to register to an outbound proxy? the ata has that set |
| 09:40.04 | kaldemar | what registers to where and when do you get 401? |
| 09:40.07 | mechbangirc | kaldemar: one last thing. secrets on box-1 and box-2 should be same right? |
| 09:41.24 | *** join/#asterisk SeTTleR (~bernd@p5DDEE067.dip.t-dialin.net) |
| 09:41.33 | kaldemar | mechbangirc: if done by the book (the oreilly site one), yes. |
| 09:43.29 | mechbangirc | kaldemar: i will follow this book for now. this technology is really complex. thanks. |
| 09:47.51 | *** join/#asterisk evharten (~evharten@vpn.evertje.net) |
| 09:57.56 | *** join/#asterisk bjornts (~BTS@it010226.klientdrift.uib.no) |
| 10:06.06 | Iron_Chef | kaldemar: I just can't seem to register with an externam sip provider |
| 10:06.12 | Iron_Chef | *external |
| 10:08.03 | Iron_Chef | I've got a couple of "register" directives in sip.conf, they both return 401 Unauthorized |
| 10:08.37 | Iron_Chef | I think it's a network problem - dns, nat something like that, but I can't tell |
| 10:09.25 | Iron_Chef | If we try the same accounts with a cisco sip phone or linksys box they register and dial/receive fine. |
| 10:19.40 | *** join/#asterisk CoderForLife (~Miranda@cpe-174-103-152-61.cinci.res.rr.com) |
| 10:21.20 | kaldemar | Iron_Chef: if you get an answer from the provider, it's not a network problem. you have an authentication problem. |
| 10:23.33 | Iron_Chef | kaldemar: I get the 401 Unauth. message when trying to connect to 2 different providers - my local one and sip.skype (just signed up to test this) |
| 10:30.54 | Iron_Chef | kaldemar: do you mean it's a problem with my config files? i wouldn't be surprised... |
| 10:31.22 | kaldemar | yours or theirs. |
| 10:31.51 | *** join/#asterisk bsaxon (~bsaxon@12.107.149.61) |
| 10:33.26 | *** join/#asterisk guilhermebr (~Guilherme@200.103.96.98) |
| 10:34.57 | Iron_Chef | kaldemar: heh, that's kinda what I'm looking for help with :-) |
| 10:35.48 | *** join/#asterisk creativx (~creadurex@226.62-97-205.bkkb.no) |
| 10:38.28 | kaldemar | Iron_Chef: can you make calls via the provider? do you get the 401 to registrations, call attempts or both? |
| 10:38.41 | Iron_Chef | kaldemar: I can make calls |
| 10:40.39 | kaldemar | Iron_Chef: compare the credentials on the registration statement to the ones used to make a call. |
| 10:41.25 | *** join/#asterisk iulhk (~iulhk@117.102.32.179) |
| 10:44.34 | Iron_Chef | kaldemar: they are the same here's my config for this VSP |
| 10:44.35 | Iron_Chef | http://pastebin.com/8QsM25TV |
| 10:48.18 | kaldemar | how do you dial? |
| 10:49.06 | Iron_Chef | dial out with a handset plugged into the phone, dialing in (to test) with skype or mobile |
| 10:49.16 | Iron_Chef | *plugged into the switch |
| 10:49.22 | kaldemar | is asterisk... |
| 10:49.30 | iulhk | do i need a Asterisk license key for testing purposes too? |
| 10:49.34 | kaldemar | in asterisk, what does your dial line look like? |
| 10:50.01 | kaldemar | iulhk: what is an asterisk license key? |
| 10:50.11 | Iron_Chef | iulhk: asterisk is free |
| 10:50.47 | Iron_Chef | i can sell you a linux license key if you want one |
| 10:51.22 | schmidts | Iron_Chef its free to get, but you have to pay with time, nerves and sweat for it :D |
| 10:52.29 | Iron_Chef | schmidts: tanstaafl |
| 10:52.56 | kaldemar | luckily, asterisk is not a lunch. |
| 10:52.58 | Iron_Chef | "there aint no such thing as a free lunch" - a classic for economics undergrads |
| 10:53.05 | Iron_Chef | kaldemar: :-) |
| 10:54.39 | iulhk | <Iron_Chef>: sorry with appology, i was asking about asterisk-skype license, do i need to purchase license fee for testing purposes too ? |
| 10:54.52 | Iron_Chef | ah yes i think so |
| 10:55.30 | Iron_Chef | it works extremely well though, i use it and it's simple enough for me to use (linux 9/10 asterisk 2/10) |
| 10:57.44 | Iron_Chef | my licence for skype/aster expires in 2030, i like that, pretty sure there will be a different standard by then |
| 10:58.13 | Iron_Chef | i hope that by 2030 i'm so far done with IT, on a yacht somewhere |
| 10:58.36 | Iron_Chef | in fact by 2030 i want to be sick of the yacht too!! |
| 10:58.38 | iulhk | going for the test drive for asterisk=skype version ... :) |
| 10:59.05 | Iron_Chef | iulhk: it's a hoot, i totally recommend it |
| 11:00.58 | iulhk | before this i hv tried karaka for skype chat and freeswitch skypopen,,, and it was bad experience .. let me c with it now |
| 11:01.25 | iulhk | Iron_chef: does it support chat too ? |
| 11:01.43 | kaldemar | http://www.digium.com/en/products/software/skypeforasterisk.php |
| 11:01.54 | Iron_Chef | iulhk: dunno, don't think so, but do you want to chat with your phone? |
| 11:01.58 | ironm | Good morning - can you recommend an asterisk support to get .rpm packages of asterisk 1.8.x ? - thank you in advance for any hints ;) |
| 11:02.21 | ironm | I don't have any idea what would it cost ... |
| 11:02.56 | Iron_Chef | ironm: i will sell it to you for $20 ;-) |
| 11:02.57 | ironm | it looks like preparing own .rpm packages for asterisk 1.8.x is a painful issue |
| 11:02.59 | *** join/#asterisk erez_ (~erez@carmel8.bb.netvision.net.il) |
| 11:03.13 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 11:03.18 | schmidts | Iron_Chef do you play salesman today? |
| 11:03.19 | erez_ | hello |
| 11:03.31 | ironm | Hi Iron_Chef - are they your own packages ? |
| 11:03.40 | Iron_Chef | ironm: just kidding |
| 11:03.52 | Iron_Chef | pretending to be some dude trying to sell open source |
| 11:04.07 | Iron_Chef | ironm: have you tried rpmfind etc etc? |
| 11:04.21 | schmidts | Iron_Chef do you want a Wifi Cable? it will cost only, ... 50 $ |
| 11:04.42 | erez_ | i have a problem with peers becoming unreachable over time |
| 11:04.50 | ironm | Iron_Chef, I have tried to find last .rpm packages for 1.8.0 but I was not successful |
| 11:04.53 | Iron_Chef | schmidts: actually I'm looking for a left handed screwdriver, willing to pay extra |
| 11:04.58 | wdoekes2 | erez_: nat keepalives on the phone? |
| 11:05.31 | kaldemar | ironm: http://packages.asterisk.org/ |
| 11:05.32 | erez_ | i have qualify=yes, wouldnt that suffice ? |
| 11:05.49 | ironm | thank you very much kaldemar :) |
| 11:06.05 | schmidts | Iron_Chef thats bad. i sold the last one yesterday but maybe you can buy a right handed from australia :D |
| 11:06.07 | wdoekes2 | it should, if they're non-realtime (static) peers |
| 11:06.25 | schmidts | erez_ how many peers do you have and get all of them unreachable the same time? |
| 11:06.31 | schmidts | or even close together |
| 11:06.59 | *** join/#asterisk AliRezaTaleghani (~AliRezaTa@unaffiliated/AliRezaTaleghani) |
| 11:07.03 | AliRezaTaleghani | hi dear all |
| 11:07.31 | AliRezaTaleghani | can someone tell me the command to check variable lenght? |
| 11:07.35 | AliRezaTaleghani | length? |
| 11:07.38 | schmidts | LEN |
| 11:07.45 | Iron_Chef | schmidts: yeah we need the left handed ones down here to work on things made up there ;-) the screws turn the other way in the southern hemisphere |
| 11:08.00 | erez_ | wdoekes2, thanks for unswering, i am trying to connect my n900 to asterisk. i opened udp ports 5060 on *, and my n900 is dynamic |
| 11:08.18 | schmidts | Iron_Chef i know, it must be hard :D |
| 11:08.25 | AliRezaTaleghani | schmidts: is this a function |
| 11:08.27 | AliRezaTaleghani | ? |
| 11:08.28 | AliRezaTaleghani | am i right? |
| 11:08.39 | AliRezaTaleghani | ye |
| 11:08.40 | Iron_Chef | schmidts: it's the standing on the head to read the manuals really bites |
| 11:08.42 | AliRezaTaleghani | i get it |
| 11:08.44 | AliRezaTaleghani | tnx |
| 11:08.45 | erez_ | i tried setting the keepalive option on the n900 both via register and via options, but neither helped |
| 11:08.46 | *** part/#asterisk AliRezaTaleghani (~AliRezaTa@unaffiliated/AliRezaTaleghani) |
| 11:09.00 | schmidts | AliRezaTaleghani yes for example exten => 123,1,Noop(length of fubar ${LEN(${fubar})}!) |
| 11:09.17 | schmidts | Iron_Chef lol ;) |
| 11:09.37 | *** join/#asterisk AliRezaTaleghani (~AliRezaTa@unaffiliated/AliRezaTaleghani) |
| 11:09.47 | kaldemar | erez_: try a smaller qualify interval with the qualifyfreq option. the default is 60 seconds. |
| 11:09.55 | *** part/#asterisk AliRezaTaleghani (~AliRezaTa@unaffiliated/AliRezaTaleghani) |
| 11:10.06 | *** join/#asterisk AliRezaTaleghani (~AliRezaTa@unaffiliated/AliRezaTaleghani) |
| 11:10.33 | erez_ | qualify=30 ? |
| 11:10.45 | kaldemar | erez_: "qualifyfreq" |
| 11:11.10 | ironm | are there any problems when installing http://packages.asterisk.org/centos/5/current/x86_64/RPMS/asterisk18-core-1.8.1.1-1_centos5.x86_64.rpm on CentOS 5.5 ? |
| 11:11.16 | kaldemar | erez_: search for "qualify" in http://svn.digium.com/svn/asterisk/tags/1.8.1.1/configs/sip.conf.sample |
| 11:11.51 | kaldemar | ironm: http://www.asterisk.org/downloads/yum |
| 11:13.23 | ironm | thanks a lot kaldemar :) |
| 11:13.51 | erez_ | kaldemar, even if i make a call from my n900 via asterisk (and the call works), at the same time it still unreachable |
| 11:16.10 | erez_ | kaldemar, changed to 30 sec, no change |
| 11:16.22 | kaldemar | erez_: the phone can be unreachable from asterisk's point of view even if you can make calls with it. |
| 11:16.53 | *** part/#asterisk sol326 (~studyofli@c-71-56-135-99.hsd1.wa.comcast.net) |
| 11:18.23 | kaldemar | erez_: see with sip debug if the n900 client answers to the qualify messages at all. |
| 11:21.43 | erez_ | kaldemar, i'll check |
| 11:31.26 | *** join/#asterisk sekil (~sekil@80.93.247.26) |
| 11:38.16 | Iron_Chef | kaldemar: is the register string the only setting for incoming sip? |
| 11:48.01 | *** join/#asterisk puzzled (~foobar@puzzled.xs4all.nl) |
| 11:56.48 | *** join/#asterisk Cain` (~Geek@unaffiliated/cain) |
| 11:59.46 | *** join/#asterisk timahvo1 (~rogue@41.72.215.94) |
| 12:00.38 | *** part/#asterisk sekil (~sekil@80.93.247.26) |
| 12:12.39 | erez_ | although i can see *'s qualify (sip debug shows option message and reply to it) - it renders my client unreachable |
| 12:15.54 | schmidts | erez_ do you have set qualify just to yes or used a time value? |
| 12:16.37 | erez_ | 30 second |
| 12:17.00 | erez_ | after i see a reply to qualify, my peer is still unreachable |
| 12:17.45 | schmidts | erez_ thats 30 milliseconds not seconds maybe the traversal time between your server to your client and back is higher than 30 ms, you should try the default value with just set it to yes |
| 12:18.47 | *** join/#asterisk eject_ck (~eject_ck@haitek.cosmonova.net.ua) |
| 12:19.25 | erez_ | i tried both |
| 12:19.32 | eject_ck | Hi guys, after upgrade to Asterisk 1.6.2.9-2 I'm getting error during start [2011-01-12 14:14:51] ERROR[21895]: pbx.c:3309 __ast_custom_function_register: Function MAILBOX_EXISTS already registered. |
| 12:20.15 | erez_ | * receives a reply to the 'option' message. so it should find the client reachable, however it doesn't |
| 12:20.34 | schmidts | erez_ could you pastebin the output of the sip debug |
| 12:20.35 | eject_ck | how to fix |
| 12:25.07 | eject_ck | I had to remove app_voicemail_imap |
| 12:35.32 | *** join/#asterisk csnook (~chris@209-6-38-66.c3-0.smr-ubr1.sbo-smr.ma.cable.rcn.com) |
| 12:40.38 | *** join/#asterisk wesphillips (~wphillips@64-19-240-7.caprock.net) |
| 12:43.00 | *** part/#asterisk erez_ (~erez@carmel8.bb.netvision.net.il) |
| 12:58.43 | *** join/#asterisk ruyo (~psantos@a83-132-152-91.cpe.netcabo.pt) |
| 13:11.07 | *** join/#asterisk ariel_ (~chatzilla@63.214.236.169) |
| 13:11.33 | *** part/#asterisk ariel_ (~chatzilla@63.214.236.169) |
| 13:24.38 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 13:26.55 | *** join/#asterisk WIMPy (~wimpy@e183095026.adsl.alicedsl.de) |
| 13:27.46 | *** part/#asterisk AliRezaTaleghani (~AliRezaTa@unaffiliated/AliRezaTaleghani) |
| 13:28.39 | eject_ck | I'm running * on xen DomU and got this warning |
| 13:28.39 | eject_ck | <PROTECTED> |
| 13:28.45 | eject_ck | how can I fix this issue ? |
| 13:32.55 | tuxx- | hey guys, is there a similair settings for iax as progressinband? (http://www.voip-info.org/wiki/view/Asterisk+sip+progressinband) |
| 13:33.19 | tuxx- | eject_ck: what version of asterisk? |
| 13:33.50 | tuxx- | 1.6.2.9-2? |
| 13:35.50 | *** join/#asterisk osiro (~osiro@189.111.254.251) |
| 13:37.06 | leifmadsen | eject_ck: install DAHDI |
| 13:37.13 | leifmadsen | eject_ck: then recompile asterisk |
| 13:38.52 | tuxx- | is that still necessary with 1.6? i thought the timing was incorporated in the asterisk code since 1.6.x |
| 13:40.27 | leifmadsen | it is if you have the right kernel modules |
| 13:40.36 | tuxx- | mkay |
| 13:41.00 | leifmadsen | but if it's not working, either the modules couldn't be compiled, or potentially they are unloaded, but I would say if you're getting that, then the res_timing modules did not compile |
| 13:41.39 | *** join/#asterisk timahvo1 (~rogue@41.72.215.94) |
| 13:44.43 | osiro | Can anybody help me? When astetisk 1.4.30 originate a call the call's disposition is "FAILED", but when I make the call manually I get "NO ANSWER". Any idea? |
| 13:46.09 | osiro | Why the originated call didn't get "NO ANSWER" either? |
| 13:46.57 | *** join/#asterisk rrb3942 (~rbullock@208.34.105.161) |
| 13:51.29 | ectospasm | osiro: please pastebin your Asterisk CLI output |
| 13:51.36 | ectospasm | ~pb | osiro |
| 13:51.42 | ectospasm | ~pb |
| 13:51.42 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude. |
| 14:00.10 | Katty | morning |
| 14:04.32 | *** join/#asterisk dimm (~guest@unaffiliated/dimm) |
| 14:05.08 | *** join/#asterisk coppice (~chatzilla@210.17.255.115) |
| 14:05.17 | Katty | infobot: seen jblack |
| 14:05.24 | infobot | jblack <~jblack@71.181.209.104> was last seen on IRC in channel #asterisk, 43d 19h 45m 1s ago, saying: 'Does asteriskgui walk over manual configuration changes?'. |
| 14:06.16 | *** join/#asterisk chuckf (~chuckf@ubuntu/member/chuckf) |
| 14:12.22 | *** join/#asterisk jkroon (~jkroon@dsl-241-233-157.telkomadsl.co.za) |
| 14:12.55 | jkroon | hi guys, i'm trying to compile dahdi-linux-2.4.0 against a 2.6.37 kernel and came accross two things, the one relates to the other. |
| 14:13.33 | jkroon | dahdi-base.c does a check for HAVE_UNLOCKED_IOCTL and then prefers to use that if it's available, and the only added functionality is to grab the BKL, which has been a no-op on most systems. |
| 14:14.11 | jkroon | can i rip out all that code or is there actually a good reason to grab the BKL ? |
| 14:15.16 | Chainsaw | Morning Katty :) |
| 14:16.49 | Katty | morning Chainsaw |
| 14:21.26 | *** join/#asterisk root52 (~root52@ip70-191-116-76.cl.ri.cox.net) |
| 14:28.01 | *** join/#asterisk cesar_CR (~cesar@201.192.86.30) |
| 14:32.54 | root52 | So I have callgroups and pickupgroups set up in SIP.conf. When I try to pickup a ringing phone it will say. "chan_sip.c:21613 handle_request_invite: Nothing to pick up for..." When I look at the SIP trace it says "Looking for *8 in mycontext" as if it is trying to dial the *8 exten. Am I missing something. Is that how it is supposed to work? |
| 14:34.59 | *** join/#asterisk X-Raimo (~asmpub@opensuse/member/raimoschmidt) |
| 14:35.15 | X-Raimo | hello. Can I use STRP with IAX? |
| 14:38.29 | jkroon | no |
| 14:38.46 | WIMPy | IAX does not use RTP. |
| 14:39.49 | X-Raimo | is anyway to encrypt media stream via IAX?] |
| 14:42.27 | *** join/#asterisk ihor (~Miranda@194.44.15.90) |
| 14:42.30 | hurdman | X-Raimo: with a vpn ^^ |
| 14:44.08 | X-Raimo | we need to avoid VPN somehow, but still make secure signalling and voice stream |
| 14:44.36 | X-Raimo | we decined SIP because of NAT problems |
| 14:49.25 | eject_ck | my sip peers stopped registry after upgrade from 1.6.0 to 1.6.2 |
| 14:49.33 | eject_ck | how can I figure why ? |
| 14:49.58 | eject_ck | can't find in messages :( |
| 14:50.23 | *** join/#asterisk gerhard7 (~gerhard7@212-123-146-122.ip.telfort.nl) |
| 14:50.29 | X-Raimo | eject_ck: set sip debug on |
| 14:50.46 | X-Raimo | and raise debug level |
| 14:52.36 | eject_ck | X-Raimo: what level should be ok ? |
| 14:52.54 | X-Raimo | eject_ck: 15 |
| 14:53.31 | X-Raimo | is IAX encrypt media stream? |
| 14:53.40 | X-Raimo | *does |
| 14:58.57 | eject_ck | X-Raimo: I see no events with my peers |
| 14:59.04 | eject_ck | what I need to register with |
| 15:01.58 | eject_ck | Configs are the same |
| 15:02.06 | eject_ck | it was working before |
| 15:02.29 | eject_ck | now my phones are registered, but peers (outgoing) not |
| 15:02.30 | *** part/#asterisk ihor (~Miranda@194.44.15.90) |
| 15:06.12 | *** join/#asterisk felimwhiteley (~quassel@109.255.104.145) |
| 15:07.54 | *** join/#asterisk twanny796 (~chatzilla@c141-181.i01-4.onvol.net) |
| 15:07.59 | *** join/#asterisk Carlos_PHX_ (~Carlos@ip68-99-199-10.ph.ph.cox.net) |
| 15:12.32 | twanny796 | what is the software for automating calling DISA from the mobile called? |
| 15:17.10 | eject_ck | How can I force peer to register from cli ? |
| 15:18.09 | eject_ck | I've just ran sip qualify peer internetcalls |
| 15:18.36 | eject_ck | where internetcalls my peer name and it was qualified |
| 15:19.14 | *** join/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 15:23.39 | kaldemar | eject_ck: you can't. |
| 15:23.45 | eject_ck | :( |
| 15:24.06 | eject_ck | why it stopped ? chan_sip.so is only module involved ? |
| 15:24.14 | eject_ck | I have no errors or even warnings |
| 15:24.30 | eject_ck | I can make outbound calls |
| 15:26.13 | eject_ck | how to fix ? |
| 15:26.50 | eject_ck | *CLI> sip show registry |
| 15:26.51 | eject_ck | sip.internetcalls.com:5060 N myusername 120 Unregistered |
| 15:32.49 | kaldemar | that show where asterisk has registered itself |
| 15:34.29 | *** join/#asterisk russellb (~russell@asterisk/digium-open-source-team-lead/russellb) |
| 15:34.29 | *** mode/#asterisk [+o russellb] by ChanServ |
| 15:36.09 | eject_ck | kaldemar: that show that it's Unregistered |
| 15:36.27 | eject_ck | I need to make Asterisk registered at sip.nternetcalls.com |
| 15:36.37 | eject_ck | it even not tries |
| 15:36.45 | eject_ck | that's my problem |
| 15:36.58 | eject_ck | sorry, I said "peer" which is wrong |
| 15:41.10 | *** join/#asterisk SuPrSluG (~SuPrSluG@8.22.96.106) |
| 15:42.25 | lordvadr | Can anybody give me a rundown of hardphone that support ipv6? |
| 15:43.41 | WIMPy | Snom since 7.3.17 |
| 15:45.20 | lordvadr | or any ideas when some of the bigger players might be? Aastra? Polycom? |
| 15:49.12 | kaldemar | eject_ck: show the register statement, masking username and secret. |
| 15:50.38 | kaldemar | eject_ck: what did you change when it stopped working? |
| 15:51.52 | kaldemar | X-Raimo: IAX can encrypt media. see the sample config for parameters. |
| 15:52.49 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 15:59.26 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 15:59.53 | *** join/#asterisk fskrotzki_ (~fskrotzki@cpe-74-74-245-250.rochester.res.rr.com) |
| 16:00.20 | eject_ck | kaldemar: I've upgraded to 1.6.2 from 1.6.0 |
| 16:00.59 | eject_ck | I've tried to sniff traffic and can say that there is no even packets from my server to sip.internetcalls.com |
| 16:01.02 | eject_ck | strange |
| 16:02.25 | eject_ck | kaldemar: my config is |
| 16:02.26 | eject_ck | [general] |
| 16:02.26 | eject_ck | context=default ; Default context for incoming calls |
| 16:02.26 | eject_ck | register => XXXXXXXXXX:YYYYYYYY@sip.internetcalls.com/inetcalls-in~3600 |
| 16:02.36 | *** join/#asterisk ClintGoudie-Nice (~clint@smtp.callware.com) |
| 16:05.13 | *** join/#asterisk Carlos_PHX1_ (~Carlos@ip68-99-199-10.ph.ph.cox.net) |
| 16:05.55 | kaldemar | eject_ck: change the ~3600 to something else or remove it and try again. |
| 16:06.20 | *** join/#asterisk russellb (~russell@asterisk/digium-open-source-team-lead/russellb) |
| 16:06.20 | *** mode/#asterisk [+o russellb] by ChanServ |
| 16:07.36 | citrus2 | can anyone recommend a cheap BRI card for asterisk? one that will work in the USA |
| 16:08.25 | WIMPy | Just any half way recent el-cheapo will do. |
| 16:08.36 | eject_ck | kaldemar: removed |
| 16:08.37 | russellb | AFAIK, Asterisk doesn't support BRI in the US |
| 16:09.01 | WIMPy | Why shouldn't it? |
| 16:09.09 | citrus2 | wuuuuh? |
| 16:09.26 | russellb | BRI in the US != BRI in Europe at the protocol level IIRC, and we don't support it |
| 16:09.32 | tzafrir_laptop | russellb, have you actually tried? |
| 16:09.41 | russellb | nope, there's a giant patch on mantis somewhere though |
| 16:09.42 | eject_ck | kaldemar: not helped :) |
| 16:09.51 | WIMPy | You certainly have to use dahdi in th US, but otherwise I'd expect it to work as usual. |
| 16:09.56 | russellb | BRI is incredibly rare in the US |
| 16:10.30 | citrus2 | verizon has it.. its 80$ a month compared to there PRI wich is 500$ |
| 16:10.32 | tzafrir_laptop | russellb, they use National switchtype an mulaw, right? |
| 16:10.44 | WIMPy | Ouch |
| 16:10.46 | *** join/#asterisk russ (foobar@ip70-176-251-1.ph.ph.cox.net) |
| 16:11.11 | russellb | maybe I'm wrong, I'm no BRI expert, just going off what I thought I remembered |
| 16:12.21 | eject_ck | why asterisk can not initiate registering itself on other sip server ? |
| 16:13.45 | *** join/#asterisk timahvo1 (~rogue@41.223.57.74) |
| 16:16.43 | kaldemar | eject_ck: make sure that is does not even send a single attempt when you reload sip. |
| 16:18.57 | eject_ck | did it |
| 16:19.13 | kaldemar | the unregistered state is the initial state when no attempts have been sent or in a case when the registrar responds with a 423 which means that the expiration time is too brief. |
| 16:19.24 | eject_ck | there is no even ip packet :( |
| 16:21.13 | kaldemar | quite odd. did anything else change besides your asterisk? |
| 16:21.25 | eject_ck | no :( |
| 16:21.40 | eject_ck | pings goes well and I see them with tcpdump |
| 16:22.01 | eject_ck | when I start/restart * and do sip reload no packets to that host :( |
| 16:22.10 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 16:23.12 | eject_ck | rebooting server |
| 16:23.54 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 16:25.57 | eject_ck | same :( |
| 16:26.08 | eject_ck | kaldemar: any ideas ? |
| 16:33.25 | *** join/#asterisk distrubuted (~root@66.241.104.83) |
| 16:38.03 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 16:56.19 | *** join/#asterisk timahvo1 (~rogue@41.223.57.75) |
| 16:57.09 | *** join/#asterisk FlashDeluxe (~benedict@static-87-79-94-28.netcologne.de) |
| 16:58.07 | distrubuted | hello, was wondering if there is a channel dedicated to asterisk help and diagnosing probelms... |
| 16:59.04 | FlashDeluxe | hi! I have question, i want incoming calls to be passed to my sip telephones with a 0 at the beginning of the incoming number, how can i realize that? if i take this string Set(CallerID=(0${CALLERID(num)})) the callerid is set, but in the cli i see Answer("DAHDI/i1/numberwithout"0" can anybody help me? |
| 17:00.05 | *** join/#asterisk w4llnu55 (~w4llnu55@dslb-094-217-088-061.pools.arcor-ip.net) |
| 17:01.38 | FlashDeluxe | i guess theres something wrong in the way i am thinking, btw i am using asterisk 1.8 with dahdi |
| 17:08.21 | leifmadsen | distrubuted: this would be that channel |
| 17:09.04 | leifmadsen | FlashDeluxe: Set(CALLERID(num)=0${CALLERID(num)}) |
| 17:09.44 | tzanger | leifmadsen: shhh if you tell everyone we'll be flooded with help requests |
| 17:10.48 | distrubuted | oh cool...just wanted to make sure...just wanted to make sure before i started asking questions.... I am a novice with Asterisk, but i am preferring it over any other softpbx...thank you |
| 17:11.03 | leifmadsen | ya asterisk is kind of a big deal :) |
| 17:11.17 | leifmadsen | distrubuted: might want to check this out too... |
| 17:11.19 | leifmadsen | ~newbook |
| 17:11.19 | 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/ |
| 17:11.35 | leifmadsen | tzanger: YOU!@ |
| 17:11.47 | FlashDeluxe | leifmadsen, i did that too, but it doesn`t work |
| 17:11.48 | distrubuted | :) thank you |
| 17:11.54 | tzanger | leifmadsen: now you sound like joshua |
| 17:12.09 | leifmadsen | tzanger: we are the same person |
| 17:12.25 | tzanger | now *that* is scary |
| 17:12.26 | leifmadsen | FlashDeluxe: what are you expecting? The output you show does not show the callerid -- it shows the number you are trying to call |
| 17:12.32 | leifmadsen | leif... file.... |
| 17:12.38 | file | I didn't do it. |
| 17:12.46 | leifmadsen | that's the problem! |
| 17:12.52 | file | tzanger, ewww you called me joshua |
| 17:12.59 | leifmadsen | mr joshua |
| 17:13.00 | tzanger | file: you should be proud, my son is named joshua |
| 17:13.08 | file | did you name him after me? thanks! |
| 17:13.13 | tzanger | my firstborn son in fact |
| 17:13.22 | leifmadsen | I've been trying to convince girls to name their babies 'Leif' |
| 17:13.33 | leifmadsen | none of which I'm aware to be mine |
| 17:13.49 | tzanger | my secondborn is named matthew and it wasn't until after I did so that I realized that I had so many friends with the name |
| 17:13.55 | FlashDeluxe | i mean, it does work the CALLERID(num)= is set rigt, but the number in the dialstring Dial("DAHDI/i4/number hasnt got the digit 0 :( |
| 17:13.57 | file | my brother is Matthew |
| 17:14.10 | paulc | My brother is Matthew too! |
| 17:14.12 | paulc | spooky! |
| 17:14.15 | leifmadsen | FlashDeluxe: and it wouldn't -- the outbound number dialed isn't the callerID |
| 17:14.15 | FlashDeluxe | i will pastebin it for better understanding |
| 17:14.23 | tzanger | although is middle name was chosen after my wife's brother... when she told him he said "that's a terrible name." so she said "ok then, we named him after our father then." to which he replied "oh ok, that's much better" |
| 17:14.25 | leifmadsen | your expectations are incorret |
| 17:14.27 | FlashDeluxe | oh |
| 17:14.33 | tzanger | file: that's too weird. |
| 17:14.53 | FlashDeluxe | ok so how can i set the outbound number? |
| 17:14.54 | leifmadsen | FlashDeluxe: your Dial() line needs to be modified |
| 17:14.55 | tzanger | two of my sons are named the same as you and your brother |
| 17:14.56 | paulc | file: actually, I didn't know you had a brother |
| 17:15.00 | tzanger | me either |
| 17:15.03 | file | paulc, I do. |
| 17:15.13 | tzanger | I always explained away file's idiosyncrasies due to being an only child |
| 17:15.13 | leifmadsen | exten => _NXXNXXXXXX,1,Dial(DAHDI/i1/0${EXTEN}) |
| 17:15.16 | *** join/#asterisk bubo (~bubo@178-191-211-33.adsl.highway.telekom.at) |
| 17:15.16 | tzanger | so much for that theory |
| 17:15.23 | distrubuted | awesome, thank you for the link to that book |
| 17:15.39 | tzanger | although now if I had to guess I'd guess file was the younger sibling |
| 17:15.58 | FlashDeluxe | leifmadsen: and how do it for incoming calls? |
| 17:16.14 | file | tzanger, I am. |
| 17:16.17 | FlashDeluxe | if i want to get a digit infront of incoming numbers? |
| 17:16.23 | file | my brother is older and in no way like me |
| 17:16.36 | tzanger | file: heh. |
| 17:16.51 | FlashDeluxe | because incoming calls are passed to SIP (exten => blaXX,n,Dial(SIP/${EXTEN:-2},,tT)) |
| 17:17.02 | tzanger | my "little" brother is 2.5 years younger and 5" taller than I am |
| 17:17.14 | leifmadsen | tzanger: holy crap, 5" and you're not short either |
| 17:17.22 | leifmadsen | be must be like 6'7" or something |
| 17:17.25 | tzanger | I'm 5'11", shortest male in my direct family |
| 17:17.35 | leifmadsen | really? I always thought you were over 6' |
| 17:17.37 | tzanger | my 15yo stepson is about an inch taller than me right now |
| 17:17.39 | leifmadsen | you must wear tall shoes |
| 17:17.44 | tzanger | leifmadsen: high heels. |
| 17:17.47 | leifmadsen | makes sense |
| 17:17.50 | paulc | but only at weekends, right? ;) |
| 17:17.55 | tzanger | exactly! |
| 17:18.11 | tzanger | how else do you think I got leifmadsen to owe me a pitcher of beer |
| 17:18.49 | tzanger | this body don't come cheap |
| 17:19.29 | FlashDeluxe | leifmadsen: perhaps i am explaining it too weird... I ll explain it this way: if i get a call, i want to see the number of the caller +a digit infront of that number in my display :) how do i do it? |
| 17:19.59 | tzanger | FlashDeluxe: oht hat's easy... Set(CALLERID(number)=9${CALLERID(number)}) |
| 17:20.06 | *** join/#asterisk ChrisH (~dg7pc@dslb-094-221-218-096.pools.arcor-ip.net) |
| 17:20.13 | tzanger | FlashDeluxe: I do that at home for the business calls that I route to the home phone |
| 17:21.03 | FlashDeluxe | tzanger: i tried that, but it doesn`t work |
| 17:21.16 | tzanger | FlashDeluxe: odd, that's exactly what I am doing |
| 17:21.42 | leifmadsen | FlashDeluxe: then you probably don't have something like sendrpid=yes and trustrpid=yes in sip.conf |
| 17:21.55 | leifmadsen | I'm assuming you're sending this to a SIP device |
| 17:22.24 | *** join/#asterisk Defraz (~tim@96.18.85.158) |
| 17:22.27 | FlashDeluxe | yyes, to a sip phone |
| 17:22.36 | leifmadsen | then try what I just suggested |
| 17:24.50 | FlashDeluxe | doenst work |
| 17:24.58 | FlashDeluxe | here i pasted the output |
| 17:24.59 | FlashDeluxe | http://pastebin.de/13771 |
| 17:25.07 | FlashDeluxe | maybe its easier to understand |
| 17:25.46 | leifmadsen | so you're calling SIP/55 and you're expecting to see 002282xx on your display right? |
| 17:26.19 | FlashDeluxe | yes |
| 17:26.23 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 17:26.31 | leifmadsen | did you enable the rpid stuff as I mentioned? |
| 17:26.36 | leifmadsen | did you do a 'sip reload' after that? |
| 17:26.43 | FlashDeluxe | yes i did that |
| 17:26.58 | FlashDeluxe | ps heres a output of my extensions.conf |
| 17:27.03 | FlashDeluxe | maybe there is sth wrong |
| 17:27.05 | leifmadsen | sorry, I have to go work on other things right now |
| 17:27.51 | FlashDeluxe | doesn`t matter, i will go home now, i ave to get sleep :) thanks for help, i will return tomorrow |
| 17:30.03 | ChrisH | I expect this to be a FAQ, but I did not found the answer :(. Using asterisk 1.4.21.2~dfsg-3+lenny1 I try to signal an incoming call to different clients (Nokia SIP, Gigaset C470IP) which is working via SIP/1&SIP/2 and queues. Answering on one client the remaining ones still show "missed calls". From a tcpdump asterisk just sends a CANCEL. Is it possible to have asterisk send an RFC3326 like "Call was taken somewhere else" string? |
| 17:31.23 | leifmadsen | ChrisH: that exists, but the phone has to support it |
| 17:31.28 | leifmadsen | and I think it might only exist in 1.6.2.x |
| 17:31.54 | leifmadsen | I'm pretty sure the Nokia SIP stack doesn't support that functionality |
| 17:32.38 | ChrisH | leifmadsen: Ah, thats a hint. I will check how to get a 1.6.2.x version on a lenny box. Thanks so far. |
| 17:32.50 | leifmadsen | ChrisH: compiling is the answer |
| 17:33.01 | *** join/#asterisk w4llnu55 (~w4llnu55@dslb-094-217-088-061.pools.arcor-ip.net) |
| 17:33.11 | leifmadsen | 1.4.21 is so crazy old too :) |
| 17:33.20 | leifmadsen | is doing a 1.4.39 release right now |
| 17:34.01 | *** join/#asterisk Denial (Denial@drgi.co.uk) |
| 17:34.05 | ChrisH | leifmadsen: I expected that one.... Should I go for 1.6 or 1.8 for production use? |
| 17:38.45 | leifmadsen | I'm using 1.8 in production, but I can't answer your question |
| 17:38.56 | *** join/#asterisk joel_oliveira (~chatzilla@alpes.nortenet.pt) |
| 17:39.15 | joel_oliveira | hi all |
| 17:40.22 | joel_oliveira | quick and simple question that may not lead to a simple answer: is it possible to remove any type of transcoding on Asterisk 1.6.2.13 ? like having it forcing to negotiate the same codec in both legs of the comunication? |
| 17:40.54 | leifmadsen | joel_oliveira: disallow=all, allow=<codec you want forced> |
| 17:41.09 | leifmadsen | just setup your peers to only allow the codecs you want |
| 17:42.40 | joel_oliveira | leifmadsen: I understand that, but let me try to explain it on other way |
| 17:44.02 | joel_oliveira | friend A, B and Asterisk have all G729 and G711, and I allow them in the sip.conf to everyone. if Asterisk has the priority on G729 then there will be no problem and the call will pass to all of them in G729 |
| 17:44.31 | joel_oliveira | now, if friend A has only G711... the call will be A -> G711 -> Asterisk -> G729 -> B |
| 17:44.57 | joel_oliveira | in the second case I want to force Asterisk that if A sends something in G711 I want to call to B in G711 also |
| 17:45.15 | joel_oliveira | is that possible? I don't want to restrict the peers to a codec only |
| 17:46.05 | joel_oliveira | did I made any sense? :) |
| 17:47.09 | joel_oliveira | I just want to have Asterisk acting as passthrough but negotiating the same codec on both legs of the comunication |
| 17:48.20 | *** join/#asterisk JerJer (~JerJer@asterisk/original-h323-guy/JerJer) |
| 17:51.24 | *** join/#asterisk lfod- (~email@unaffiliated/lfod-) |
| 17:52.00 | *** join/#asterisk Brixius (~kmurphy@PDN-VBA.OnvoyInc.fw.onvoy.net) |
| 17:54.19 | leifmadsen | joel_oliveira: look at the SIP_CODEC() function |
| 17:54.24 | leifmadsen | may or may not do what you want |
| 17:56.08 | joel_oliveira | leifmadsen: I'll look into that, thanks |
| 17:59.22 | cusco | hi |
| 17:59.31 | cusco | is there a function to unset a variable? |
| 17:59.36 | cusco | or how do I unset it? |
| 18:00.54 | cusco | or should I set it to null ?» |
| 18:06.28 | *** join/#asterisk lanning (~lanning@208.87.235.224) |
| 18:10.30 | *** join/#asterisk Defraz (~Defraz@96.18.85.158) |
| 18:10.40 | leifmadsen | cusco: you can't unset a channel variable |
| 18:10.49 | leifmadsen | Set(thisVariable=) |
| 18:11.51 | cusco | nevermind |
| 18:14.53 | *** join/#asterisk shido6 (~shido6@nat/yahoo/x-orzbxtapbfahbkzq) |
| 18:23.22 | *** join/#asterisk russellb (~russell@asterisk/digium-open-source-team-lead/russellb) |
| 18:23.22 | *** mode/#asterisk [+o russellb] by ChanServ |
| 18:25.56 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 18:27.46 | *** join/#asterisk i_heart_asterisk (~chatzilla@firewall-a.buf.ny.i-evolve.net) |
| 18:27.51 | i_heart_asterisk | this is strange, wondering if this has happened to other pple, when i call 905-238-2456 sometimes it goes to Panasonic number other times i get "Welcome to Videotron. This service is no longer available..." |
| 18:30.37 | *** join/#asterisk dmorgga (~dmorgga@c-71-204-89-218.hsd1.ga.comcast.net) |
| 18:31.12 | *** join/#asterisk frigidzephyr (~rnewton@nat/digium/x-xvpwpmsvzqpwgupk) |
| 18:32.30 | *** part/#asterisk dmorgga (~dmorgga@c-71-204-89-218.hsd1.ga.comcast.net) |
| 18:34.32 | cusco | hi... |
| 18:34.39 | Chainsaw | Hello... |
| 18:34.46 | cusco | Im dialing from queue to Local/xxx |
| 18:35.13 | cusco | then in that context _XXX I have some actions and another DIAL(SIP/${EXTEN} |
| 18:35.51 | cusco | because in this particular queue, I am using a call file, when this peer answers the call (last dial) I am yet in another context ... |
| 18:36.10 | leifmadsen | waits for the punchline |
| 18:36.13 | cusco | in this new context ${EXTEN} is no longer 'XXX' |
| 18:36.26 | cusco | how do I get 'XXX' in this latter context? |
| 18:36.27 | leifmadsen | that's because it's the extension the callfile is calling |
| 18:36.37 | cusco | yes |
| 18:36.40 | leifmadsen | Set(__OriginalExtension=${EXTEN}) |
| 18:36.54 | leifmadsen | although that might not even work if you are using a callfile |
| 18:37.01 | cusco | now in that context before the sip dial, I can have it.. and I could pass it in a sip header |
| 18:37.04 | leifmadsen | so you'll probably need to do a SetVar or something in the callfile |
| 18:37.08 | cusco | but is there any other way, besides sip header? |
| 18:37.22 | leifmadsen | you can set variables in the callfile |
| 18:37.26 | cusco | in the call file, I dunno who is going to answer the queue yet... |
| 18:37.38 | leifmadsen | when are you triggering the callfile? |
| 18:37.43 | leifmadsen | I had the impression you were doing that with the Local channel |
| 18:38.20 | cusco | so ... call-file channel goes to queue() --> dial local/XXX -- dialplan---> dial sip/xxx --> final context |
| 18:38.29 | leifmadsen | in 1.8 you might need to use MASTER_CHANNEL() to set a channel variable on the originating channel, but beyond that, you can't pass back information |
| 18:38.42 | cusco | im using 1.6 |
| 18:39.06 | leifmadsen | I don't understand why you need to know the extension number from the other side of the call |
| 18:39.19 | leifmadsen | when you use the callfile and it connects to the queue, it should be connecting the channels together |
| 18:39.29 | leifmadsen | I still don't follow what you're trying to do |
| 18:39.47 | cusco | I need to insert some stuff (in this case 'XXX') in a mysql db |
| 18:39.51 | cusco | only in that final context |
| 18:40.16 | leifmadsen | I don't understand the use of the final context part |
| 18:40.35 | cusco | final contet is specified on the .call file |
| 18:40.36 | leifmadsen | why don't you just do that in the Local channel, or using the .... G() flag I think it is in Dial() |
| 18:40.44 | leifmadsen | I still don't know why you're using a callfile at all |
| 18:40.45 | cusco | G() ? |
| 18:40.50 | leifmadsen | core show application dial |
| 18:41.12 | leifmadsen | it's the ability to trigger a GoSub() or Macro() just prior to connecting to the agent |
| 18:41.17 | cusco | its a... called from a complex php script that needs to generate a call with certain info.. nvm that |
| 18:41.39 | leifmadsen | so just write it in the Local channel or whatever |
| 18:41.55 | cusco | since there is another dial, the channel changtes, right? |
| 18:42.03 | leifmadsen | I presume the "final context" that you refer to is where the callfile says to connect the channel to |
| 18:42.15 | cusco | no |
| 18:42.30 | cusco | ok let me try again |
| 18:42.30 | leifmadsen | ya because I have no idea what you're trying to do |
| 18:43.16 | cusco | .call file has channel that goes to queue AND a context 'BLA' and exten 'BLE' |
| 18:43.41 | eject_ck | Guys, I'm trying to make Meetme workingh |
| 18:43.51 | cusco | queue dials Local/555 on context 'agents' (its not answered yet) |
| 18:43.53 | eject_ck | getting message in cli when I'm trying to call |
| 18:43.55 | eject_ck | <PROTECTED> |
| 18:43.55 | eject_ck | <PROTECTED> |
| 18:43.55 | eject_ck | [2011-01-12 18:43:10] WARNING[1420]: app_meetme.c:774 build_conf: Unable to open pseudo device |
| 18:44.10 | leifmadsen | eject_ck: because you don't have a res_timing_XXX module loaded |
| 18:44.12 | eject_ck | Unable to open pseudo device - what device it mean |
| 18:44.30 | cusco | ael in context 'agents' does stuf... and dial(sip/555) |
| 18:44.36 | eject_ck | leifmadsen: I'm on Xen DomU |
| 18:44.52 | leifmadsen | cusco: you mean your callfile connects the Queue() to [BLA] exten => BLE |
| 18:44.59 | leifmadsen | eject_ck: that does not mean you don't need a timing device |
| 18:45.02 | Katty | welp. |
| 18:45.04 | cusco | softphone 55 answers, and the call goes imediately to ext 'BLE' on context 'BLA' |
| 18:45.07 | Katty | i'm a free woman again. |
| 18:45.08 | cusco | 555 |
| 18:45.15 | leifmadsen | cusco: yes that is what I would expect it to do |
| 18:45.19 | eject_ck | leifmadsen: what device you mean ? |
| 18:45.19 | leifmadsen | Katty: good times |
| 18:45.43 | leifmadsen | eject_ck: I just told you -- res_timing_XXX where XXX is substituted for 'dahdi', 'pthread', or 'timerfd' |
| 18:45.50 | cusco | leifmadsen: ok.... now in ext 'BLE' how can I find 555 ? |
| 18:45.56 | cusco | it could have been 554 or 556 |
| 18:45.57 | leifmadsen | eject_ck: just install DAHDI and use res_timing_dahdi |
| 18:46.19 | cusco | any way to do that besides using sip header?! |
| 18:46.22 | leifmadsen | cusco: in that case I would probably try using MASTER_CHANNEL to write information to that channel |
| 18:46.30 | cusco | 1.6 ? |
| 18:46.32 | leifmadsen | I don't even think a sip header would work there |
| 18:46.37 | leifmadsen | cusco: I don't know what 1.6 means |
| 18:46.39 | leifmadsen | 1.6.2 I presume |
| 18:46.46 | leifmadsen | I don't know when MASTER_CHANNEL was added |
| 18:46.46 | cusco | does asterisk 1.6 has MASTER_CHANNEL ? |
| 18:46.51 | leifmadsen | 1.6 does not exist |
| 18:46.53 | cusco | ok I will check |
| 18:47.11 | cusco | 1.6.2.13 |
| 18:47.14 | cusco | ok I will check |
| 18:47.19 | leifmadsen | yes, check |
| 18:47.40 | leifmadsen | beyond that I can't think of a way of telling the original channel the information you need |
| 18:47.41 | cusco | only in 1.8 |
| 18:47.42 | cusco | :( |
| 18:48.02 | cusco | pbx-cov:/var/spool/asterisk/outgoing_done/audio# asterisk -vrx "core show functions"|grep -i channel |
| 18:48.05 | cusco | CHANNEL |
| 18:48.08 | cusco | CHANNELS |
| 18:48.11 | eject_ck | leifmadsen: I'm using debian lenny packages, unfortunatelly there is no such modules there : |
| 18:48.14 | cusco | isn't there any other way? |
| 18:49.03 | leifmadsen | eject_ck: then you need to install the modules |
| 18:49.10 | leifmadsen | eject_ck: or compile asterisk |
| 18:49.26 | leifmadsen | cusco: there might be, but I can't think of any other way |
| 18:49.49 | cusco | can I use SHARED() ? |
| 18:49.53 | leifmadsen | you can try |
| 18:49.58 | leifmadsen | it probably won't do what you want |
| 18:49.58 | eject_ck | leifmadsen: any luck to get modules for my current debian * ? |
| 18:50.02 | eject_ck | it's 1.4 |
| 18:50.06 | *** join/#asterisk magnusz (~mmg@ool-4a58b348.dyn.optonline.net) |
| 18:50.11 | leifmadsen | eject_ck: I don't use packages -- you need DAHDI |
| 18:50.17 | Chainsaw | 1.6.2 accepted exten => _*TLIO#4XXXX whereas 1.8 does not. |
| 18:50.19 | eject_ck | ok |
| 18:50.20 | leifmadsen | it's missing a timing module |
| 18:50.21 | Chainsaw | Can anyone shed light on why please? |
| 18:50.31 | magnusz | Hey anyone know why i cant type "sip reload" etc |
| 18:50.37 | eject_ck | as I understood I need both dahdi and res_timing_dahdi |
| 18:50.37 | *** join/#asterisk dogonovmax (~Adium@78-106-138-250.broadband.corbina.ru) |
| 18:50.37 | leifmadsen | Chainsaw: what do you mean "does not accept"? |
| 18:50.41 | magnusz | i get a no such command |
| 18:50.48 | leifmadsen | magnusz: you don't have chan_sip.so loaded |
| 18:50.53 | magnusz | yes i do |
| 18:50.57 | Chainsaw | leifmadsen: [2011-01-12 18:49:34] NOTICE[15504]: chan_sip.c:21321 handle_request_invite: Call from '45999' to extension '*TLIO#45999' rejected because extension not found in context 'from-client'. |
| 18:51.00 | magnusz | at least it shows |
| 18:51.12 | Chainsaw | leifmadsen: I can call the nearby *71 extension without issue. |
| 18:51.34 | leifmadsen | Chainsaw: dialplan show from-client |
| 18:51.59 | leifmadsen | Chainsaw: dialplan show *TLIO#45999@from-client" too I think should help for testing |
| 18:52.04 | magnusz | i typed module show like chan and its there leifmadsen |
| 18:52.30 | leifmadsen | magnusz: what version? try 'module reload chan_sip.so" |
| 18:52.49 | leifmadsen | either module is corrupted or it's not loaded |
| 18:52.50 | Chainsaw | leifmadsen: http://www.pastie.org/1453481 |
| 18:53.32 | magnusz | latest ver |
| 18:53.43 | leifmadsen | magnusz: what do you mean by that? |
| 18:53.58 | leifmadsen | latest ver of 1.4 is 1.4.38, latest ver of 1.6.2 is 1.6.2.15, etc... |
| 18:54.11 | leifmadsen | Chainsaw: I don't see anything obviously wrong |
| 18:54.18 | Chainsaw | leifmadsen: I'm glad :) |
| 18:54.19 | leifmadsen | Chainsaw: extensions.conf or extensions.ael ? |
| 18:54.22 | Chainsaw | leifmadsen: extensions.conf |
| 18:54.23 | eject_ck | leifmadsen: got dahdi from backports |
| 18:54.24 | leifmadsen | goes back to work |
| 18:54.37 | *** join/#asterisk timahvo1 (~rogue@41.223.57.74) |
| 18:55.25 | magnusz | i got a warning saying " parse error: no closing ']' line 1 of /etc/asterisk/sip.conf |
| 18:55.45 | Chainsaw | leifmadsen: I will return tomorrow. If it sounds like a genuine bug I'll file a report. |
| 18:55.45 | leifmadsen | magnusz: you have a missing ] on line 1 of sip.conf |
| 18:55.53 | Chainsaw | leifmadsen: Thanks for your time so far :) |
| 18:55.57 | leifmadsen | :) |
| 18:56.06 | leifmadsen | magnusz: the error message was pretty straight forward :) |
| 18:56.14 | leifmadsen | you have an invalid sip.conf |
| 18:56.17 | magnusz | i added the ] like an hour ago and still same error |
| 18:56.23 | magnusz | =/ |
| 18:56.32 | leifmadsen | something else is wrong in the file then |
| 18:56.49 | magnusz | where can i get a fresh sip.conf to look at? |
| 18:56.49 | leifmadsen | you'll have to output it |
| 18:56.51 | *** join/#asterisk iEatChildren (~iEC@204.26.87.226) |
| 18:56.52 | leifmadsen | pastebin |
| 18:56.57 | leifmadsen | sip.conf.sample in the asterisk source |
| 18:57.03 | leifmadsen | configs/ subdir |
| 18:58.38 | cusco | is there a var that contains the briged channel info? |
| 18:59.09 | leifmadsen | https://wiki.asterisk.org/wiki/display/AST/Asterisk+standard+channel+variables |
| 19:00.01 | magnusz | thanks leifmadsen :) , probably happened by editing the conf too many times :P |
| 19:00.09 | magnusz | yeah i got thanks |
| 19:00.26 | *** join/#asterisk lifeforms (~walter@tau.lfms.nl) |
| 19:00.31 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 19:01.39 | *** join/#asterisk chopp (~chopp@unaffiliated/chopp) |
| 19:01.47 | lifeforms | Hi, I've created a macro 'play' which looks at some globals to determine which file to play using Background() ... My problem is now, that if people start pressing a button while this file is still playing, the call is disconnected and I get the message: Invalid extension '1', but no rule 'i' in context 'play' |
| 19:02.34 | lifeforms | This does not seem to be the case when I use Background() directly, but I would like to use some parameters (such as customer type and language) and not retype them so a macro seemed like a good idea. Can I prevent this from happening? |
| 19:02.58 | xSmurf | mroing |
| 19:04.17 | leifmadsen | lifeforms: it's because you're using a Macro() and the DTMF is trying to be matched in the macro itself |
| 19:04.45 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 19:04.55 | leifmadsen | I usually do: Background(${GLOBAL(SOUND_FILES)}/filename_to_play) |
| 19:04.58 | lifeforms | that seems to be the case unfortunately :) and I thought I was so clever and a good coder by doing a macro |
| 19:05.16 | leifmadsen | I don't think the Macro() gives you really much |
| 19:05.20 | lifeforms | that's a good idea :) |
| 19:05.32 | leifmadsen | just use the global variable directly in Background() |
| 19:05.36 | lifeforms | OK :) |
| 19:09.04 | *** join/#asterisk stix (~stix@x1-6-00-1e-2a-87-8d-aa.k1.webspeed.dk) |
| 19:12.16 | joel_oliveira | leifmadsen: sorry to bother you again but according to https://issues.asterisk.org/view.php?id=13243 wasn't the SIP_CODEC_OUTBOUND supposed to be available on version 1.6.2.13? or did I misunderstood it? |
| 19:12.44 | joel_oliveira | the last svnbot message says that it was applied to the 1.6.2 branch on 2009-04-06 |
| 19:13.16 | leifmadsen | depends when 1.6.2.13 was made |
| 19:13.30 | joel_oliveira | hum let me check |
| 19:13.32 | leifmadsen | new features would not have been added to the 1.6.2 branch |
| 19:14.03 | *** join/#asterisk oej (~olle@2001:470:1f15:d79:225:ff:fe44:74ec) |
| 19:14.03 | eject_ck | leifmadsen: do I need make any DAHDI configuration ? |
| 19:14.17 | leifmadsen | joel_oliveira: you didn't read this: Blocked revisions 186624 via svnmerge |
| 19:14.28 | leifmadsen | joel_oliveira: it was explicitly blocked from 1.6.2 |
| 19:14.34 | leifmadsen | eject_ck: no, just load it |
| 19:14.42 | eject_ck | sip*CLI> module load res_timing_dahdi.so |
| 19:14.43 | eject_ck | Unable to load module res_timing_dahdi.so |
| 19:14.43 | eject_ck | Command 'module load res_timing_dahdi.so ' failed. |
| 19:14.55 | eject_ck | btw, I've upgraded to 1.6.2 |
| 19:15.01 | leifmadsen | eject_ck: you're using 1.4, so you don't have res_timing modules -- you just need to run ./configure and 'make install' after you install dahdi |
| 19:15.13 | leifmadsen | you should just change 1 thing at a time |
| 19:15.19 | leifmadsen | close |
| 19:15.21 | *** part/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 19:16.27 | joel_oliveira | leifmadsen: I did read that but I didn't know what that meant :( . So why was it blocked? And if it was blocked how can I put that variable there? changing source code by hand, no? |
| 19:25.36 | citrus2 | has anyone gotten a BRI card working in the US with asterisk? |
| 19:27.39 | i_heart_asterisk | wondering if this has happened to other pple, when i call 905-238-2456 sometimes it goes to Panasonic number other times i get "Welcome to Videotron. This service is no longer available..." |
| 19:27.43 | *** join/#asterisk adr3nalin3 (~Identd@179.sub-174-253-2.myvzw.com) |
| 19:30.09 | *** join/#asterisk ClintGoudie-Nice (~clint@smtp.callware.com) |
| 19:32.58 | adr3nalin3 | Qwell, sorry to bother, i haved used tkd-fender for consulting in the past but i have forgotten his contact details would you be willing to pm me his website? thank you |
| 19:37.15 | adr3nalin3 | is tk still a frequenter of this channel? |
| 19:38.10 | *** join/#asterisk [T]ank- (~T]ank@206.71.78.158) |
| 19:38.29 | [T]ank- | i am looking for a little help on understanding "marked users" in meetme. How do you mark a user? |
| 19:38.58 | [T]ank- | i have a few static rooms created and want them to hear music until one of 5 people enter the room. |
| 19:39.00 | [T]ank- | is that possible? |
| 19:39.04 | [T]ank- | or at least one person |
| 20:29.05 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 20:29.05 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.8.1.1 (2010/12/15), 1.6.2.15 (2010/12/08), 1.4.38 (2010/12/08), *-Addons 1.6.2.2, 1.4.12 (2010/09/22), dahdi-linux 2.4.0 (2010/09/01), dahdi-tools 2.4.0 (2010/09/01), libpri 1.4.11.5 (2010/11/22) -=- Visit the new official Asterisk wiki: wiki.asterisk.org -=- http://xkcd.com/806/ |
| 20:32.32 | *** join/#asterisk DrDamnit (~michael@highpoweredhelp.com) |
| 20:32.45 | DrDamnit | In Asterisk 1.8.1.1, where does the Record() application store the files? |
| 20:35.14 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 20:35.14 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.8.1.1 (2010/12/15), 1.6.2.15 (2010/12/08), 1.4.38 (2010/12/08), *-Addons 1.6.2.2, 1.4.12 (2010/09/22), dahdi-linux 2.4.0 (2010/09/01), dahdi-tools 2.4.0 (2010/09/01), libpri 1.4.11.5 (2010/11/22) -=- Visit the new official Asterisk wiki: wiki.asterisk.org -=- http://xkcd.com/806/ |
| 20:35.32 | *** join/#asterisk atan (~atan@unaffiliated/atan) |
| 20:36.57 | *** join/#asterisk blackhole (~ikpre@unaffiliated/blackhole) |
| 20:38.26 | JunK-Y | DrDamnit: in /var/lib/asterisk/sounds/ (by default) |
| 20:38.36 | [T]ank- | is there a way in meetme() to mark a user without having to use a separate extension? |
| 20:39.15 | DrDamnit | JunK-Y: Is there a reason why when I record it doesn't save? exten => *14, 1, Record(test.wav) isn't producing test.wav. |
| 20:39.31 | DrDamnit | JunK-Y: I get the beep, but htere is no file created. |
| 20:39.55 | JunK-Y | cause you're probably have a default directory defined |
| 20:40.06 | DrDamnit | where does it get defined? |
| 20:40.06 | JunK-Y | in ur linux, type updatedb && locate test.wav |
| 20:40.13 | JunK-Y | you'll find the dir |
| 20:40.14 | DrDamnit | Did that. |
| 20:40.16 | DrDamnit | no file. |
| 20:40.33 | JunK-Y | pastebin ur CLI output |
| 20:41.57 | DrDamnit | http://pastebin.com/y6xp1gkC |
| 20:42.46 | *** join/#asterisk SaiSoma (~chatzilla@client105.jdcc.edu) |
| 20:42.50 | atan | Are there any paid caller ID services that are reliable? |
| 20:42.56 | *** part/#asterisk root52 (~root52@ip70-191-116-76.cl.ri.cox.net) |
| 20:42.59 | atan | Someone I can pass a number to and they pass me back a name. |
| 20:43.42 | blackhole | Hi All, I have a AGI script which uses get_data. Now what is happening is while get_data is streaming the file and user enters some other input (press numbers) it stops playing that file inbetween ... |
| 20:44.01 | blackhole | I want to make sure that file is streamed till end and only then input is taken ... |
| 20:45.52 | JunK-Y | blackhole: use GET OPTION |
| 20:46.18 | JunK-Y | if the received dtmf isnt in ur escapes, it'll continue the streaming |
| 20:47.31 | blackhole | which api? JunK-Y I am little new to AGI |
| 20:47.55 | JunK-Y | blackhole: instead of using GET DATA, use GET OPTION |
| 20:48.53 | JunK-Y | DrDamnit: never saw this. which version? |
| 20:49.11 | DrDamnit | Asterisk 1.8.1.1, release |
| 20:49.14 | JunK-Y | DrDamnit: are you running * as root ? |
| 20:49.21 | DrDamnit | on this box, yes. |
| 20:49.29 | JunK-Y | let me reproduce this |
| 20:52.28 | DrDamnit | JunK-Y: The exact dialplan code i used is: exten => *14,1,Record(testimonial-x.wav) |
| 20:53.00 | SaiSoma | hey guys, since upgrading to 1.8.0, i've been having issues with incoming faxes (grandstream ATAs . .don't shoot me, I know). it worked well in 1.6.x, but incoming always fails now. thoughts? |
| 20:54.13 | DrDamnit | JunK-Y: I also changed the permissions of the directory and the file to 777 trying to ensure it was not a permissions problem. No joy. |
| 20:54.14 | *** join/#asterisk citywok (~kvirc@67-134-194-33.dia.static.qwest.net) |
| 20:54.24 | blackhole | JunK-Y, I don't think you understood my issue. See using Get Data is good for me because i want to return only when $maxdigits have been received, so if i have maxdigits as 10 and i enter 10 it works fine and plays the voice file but if let's say by mistake instead of 10 i have entered 11 digits it actually starts playing the file but cant complete because 1 more char was entered than expected |
| 20:54.47 | blackhole | JunK-Y, If i would use Get Option it will return on every digit i enter |
| 20:54.54 | DrDamnit | blackhole: what language is the agi written in? |
| 20:55.07 | blackhole | DrDamnit, PHP |
| 20:55.20 | DrDamnit | are you using phpagi or coding by hand? |
| 20:55.42 | blackhole | DrDamnit, Sorry it's perl |
| 20:55.48 | blackhole | DrDamnit, Mistakenly wrote PHP |
| 20:55.59 | DrDamnit | oh. Sad christmas. I don't now Perl. :-( |
| 20:56.17 | blackhole | hmm |
| 20:56.23 | JunK-Y | DrDamnit: i dont have that behavior here |
| 20:56.24 | DrDamnit | I use the phpagi class and write everything in PHP. Works great. |
| 20:56.42 | JunK-Y | DrDamnit: pastebin ur asterisk.conf to see ur dir. |
| 20:57.09 | blackhole | JunK-Y, Can you help me? |
| 20:57.15 | JunK-Y | blackhole: GET OPTIOn is for 1 digit only. |
| 20:57.20 | DrDamnit | http://pastebin.com/i61hbUfR |
| 20:57.26 | blackhole | JunK-Y, yes so it's not useful for me |
| 20:57.59 | DrDamnit | blackhole: pastebin your script, and let me look at it. I can read perl, and do plenty of agi work. |
| 20:58.01 | blackhole | JunK-Y, You understand my issue .. after user have entered 10 digits and if message is streaming it shld let the message strream fullly before taking any more digits as input .. |
| 20:58.26 | DrDamnit | blackhole: GET OPTION works like the background() application. |
| 20:58.27 | JunK-Y | u mean the next prompt right? |
| 20:58.36 | DrDamnit | it waits for one digit. |
| 20:58.44 | DrDamnit | you need somehting like READ() but in AGI. |
| 20:58.50 | DrDamnit | paste code. Let me see. |
| 20:58.54 | JunK-Y | DrDamnit: not exactly, it ignores the dtmfs if they are not in the escapes dtmfs. |
| 20:59.14 | DrDamnit | JunK-Y: True. |
| 21:00.29 | JunK-Y | DrDamnit: i dont see any diff with my tests, you're sure in linux, updatedb && locate testimonial-x.wav is empty? |
| 21:00.45 | DrDamnit | JunK-Y. sear on my life. |
| 21:00.52 | DrDamnit | s/sear/swear |
| 21:01.00 | blackhole | DrDamnit, here is it -> http://pastebin.com/tkhTvfWv |
| 21:01.25 | DrDamnit | blackhole: which lines are the issue? |
| 21:01.40 | blackhole | DrDamnit, check $cardinputsoundfile |
| 21:01.44 | blackhole | 108 |
| 21:02.07 | blackhole | it goes to 80 from there |
| 21:02.13 | blackhole | goto AUTHEN |
| 21:02.19 | *** join/#asterisk alphawave (~aw@unaffiliated/alphawave) |
| 21:02.19 | blackhole | and then calls get_data on 82 |
| 21:02.29 | blackhole | now it starts playing message |
| 21:03.03 | DrDamnit | ok. I see what you're oding. |
| 21:03.16 | blackhole | now if user presses any other digit while this is played it stops there |
| 21:03.17 | DrDamnit | you're playing them a message while they re-enter a card number? |
| 21:03.18 | blackhole | :) |
| 21:03.22 | blackhole | yes |
| 21:03.51 | blackhole | and i tell it to only return when it gets 10 numbers |
| 21:04.04 | DrDamnit | well... the easy way out, is to force them to listen to the message then allow them to enter the number. |
| 21:04.16 | DrDamnit | you can't hear a message over the recording anyway. |
| 21:04.18 | blackhole | hmm .. and how can i do that? |
| 21:04.22 | DrDamnit | over the DTMF ... sorry... |
| 21:04.38 | DrDamnit | use stream first, then get data |
| 21:05.06 | blackhole | so how can i force them to listen first and then allow them to enter message? |
| 21:05.15 | DrDamnit | hold on |
| 21:05.18 | DrDamnit | looking something up for you. |
| 21:05.28 | blackhole | ok |
| 21:06.14 | xSmurf | how does one track users that entered a conference with ConfBridge? |
| 21:06.38 | DrDamnit | blackhole: in php, we can do that by passing a variable to the get_data method: $room = $agi->get_data('enter-conf-call-number', 15000, 4); |
| 21:06.42 | xSmurf | MeetMeList doesn't seem to report anything |
| 21:07.00 | *** join/#asterisk JerJer (~JerJer@asterisk/original-h323-guy/JerJer) |
| 21:07.20 | DrDamnit | JunK-Y: Any thoughts on how I can resolve this recording issue? What else should I look for? |
| 21:07.27 | blackhole | DrDamnit, I am using get_data |
| 21:07.34 | blackhole | DrDamnit, It's same way in PHP or perl |
| 21:07.52 | blackhole | DrDamnit, enter-conf-call-number is again the name of file to be streamed\ |
| 21:08.12 | DrDamnit | Oh wait. Brain fart. That script allows people to interrupt. |
| 21:08.38 | DrDamnit | The other option is to break it into two agi scripts, return to diaplan between each, and utilize Playback(). |
| 21:08.44 | DrDamnit | This will ensure they have to listen. |
| 21:10.09 | atan | In features.con in my applicationmap how do I move a call to another context? |
| 21:10.38 | atan | The format is Goto,extension and I can't use Goto(context,extension) whatnot as far as I know |
| 21:11.15 | blackhole | DrDamnit, hmm.. |
| 21:11.21 | DrDamnit | blackhole: I see your exact problem. When users mis-enter their card number, you need to tell them enter it again. |
| 21:11.32 | blackhole | DrDamnit, yes |
| 21:11.36 | DrDamnit | The issue is, that they get impatient, and begin to enter a second time, and stream_file takes the first digit. |
| 21:12.16 | DrDamnit | You're only option that I know of is to return them to dialplan based on the error output, and then have dialplan use Playback() to force them to listen, and then have the dialplan Goto the priority that has the agi script so they can listen again. |
| 21:12.23 | blackhole | DrDamnit, yeh kinda and may be they mistakenly pressed 11 digits instead of 10 and they jst even cant get the message to be listened so a voice comes and fades immediately! |
| 21:12.45 | DrDamnit | I don't think you can 100% agi this because AGI doesn't have a function exactly like Playback() in the dialplan. |
| 21:13.03 | blackhole | DrDamnit, What abt AGI Stream file? |
| 21:13.22 | DrDamnit | Play the given audio file, allowing playback to be interrupted by a DTMF digit. This command is similar to the GET DATA command but this command returns after the first DTMF digit has been pressed while GET DATA can accumulated any number of digits before returning. |
| 21:13.31 | DrDamnit | stream_file: Play the given audio file, allowing playback to be interrupted by a DTMF digit. This command is similar to the GET DATA command but this command returns after the first DTMF digit has been pressed while GET DATA can accumulated any number of digits before returning. |
| 21:13.33 | DrDamnit | sorry |
| 21:13.44 | DrDamnit | http://phpagi.sourceforge.net/phpagi22/api-docs/ |
| 21:15.20 | xSmurf | why not just stream_file in a loop until no digits are entered? |
| 21:15.34 | DrDamnit | That's an idea. |
| 21:15.49 | DrDamnit | It may frustrate the user, but they will have to listen to the whole message.... eventually. |
| 21:15.53 | xSmurf | it's not perfect and maybe even darn right annoying, but it'll ensure the message is played entirely |
| 21:16.32 | blackhole | xSmurf, Hmm but if i have to check if digits are entered or not using get_data which will eventually play two messages one from stream_file and one of get_data file |
| 21:16.40 | blackhole | xSmurf, i tried something like that.. |
| 21:16.45 | DrDamnit | You could even put some logic in there that says; "Please listen to the message in its entirety before re-entering your number" if stream_file returns any digits. |
| 21:16.45 | xSmurf | another idea would be to put your message as a music on hold, choose the class in AGI and hold the call for the duration of the message |
| 21:17.09 | xSmurf | DrDamnit: yeah I was gonna suggest that |
| 21:17.28 | blackhole | xSmurf, How you will loop and check if no digits are entered |
| 21:17.44 | xSmurf | you wouldn't need that |
| 21:17.53 | xSmurf | if you hold the call, digits should be ignore anyway |
| 21:18.07 | blackhole | xSmurf, hmm i want to go with your first idea |
| 21:18.14 | DrDamnit | JunK-Y: I just tried it with a different file name. Still no joy. I also renamed the file just to be sure I was "pointed" to the right directory. When I rename the file, the playback (*15) fails because the file does not exist in any format. |
| 21:18.15 | blackhole | not holding call |
| 21:18.22 | atan | Can you use Dial() to send someone to another extension within the same context? |
| 21:18.34 | DrDamnit | atan: Use goto for that. |
| 21:19.44 | blackhole | xSmurf, |
| 21:19.45 | blackhole | ? |
| 21:19.59 | xSmurf | hmm? |
| 21:20.01 | atan | What's wrong with this line here then? manualfaxrx => ###,peer,Goto,outbound|manualfaxrx|1 |
| 21:20.29 | Kobaz | is there a list of actual changes to dahdi.. http://downloads.asterisk.org/pub/telephony/dahdi-linux-complete/ChangeLog-2.4.0+2.4.0 doesn't really have anything other than, "x.y.z version was released" |
| 21:20.44 | xSmurf | Kobaz: what more do you need? :P |
| 21:21.25 | Kobaz | 'fixed x' 'fixed y' |
| 21:22.29 | Kobaz | it would be easier to look at than code diffs |
| 21:22.29 | xSmurf | I'm kidding |
| 21:22.32 | Kobaz | heh |
| 21:22.51 | DrDamnit | manualfaxrx => ###,peer,Goto(outbound,manualfaxrx,1) |
| 21:23.04 | atan | Well I'm stumped then. DrDamnit, I am trying to let the called SIP phone press ### to transfer the incoming call to the rxfax part. manualfaxrx => ###,peer,Goto,outbound,manualfaxrx,1 is what I have but it just says label outbound not found in extension. Of course, there's no n(outbound) in there =\ |
| 21:23.04 | DrDamnit | atan: manualfaxrx => ###,peer,Goto(outbound,manualfaxrx,1) |
| 21:23.20 | blackhole | xSmurf, How would i achieve using stream_file and checking if digit are entered or not? |
| 21:23.21 | atan | tries that line but thinks the () will throw it off. One sec. |
| 21:23.50 | DrDamnit | atan has never typed core show application goto. |
| 21:24.05 | atan | manualfaxrx => ###,peer,Goto(outbound,manualfaxrx,1) causes the feature to not work at all |
| 21:24.26 | atan | DrDamnit, within applicaionmap I can't use Goto() type stuff since it only accepts it as Goto,options |
| 21:24.26 | DrDamnit | when you say "peer" there, do you mean priority? |
| 21:24.29 | DrDamnit | or extension? |
| 21:24.37 | atan | peer is allowing either party to press ###. |
| 21:24.40 | DrDamnit | oh you're in application map. |
| 21:24.56 | DrDamnit | nevermind then. :-) |
| 21:25.00 | atan | Yeh =\ |
| 21:25.24 | atan | I'm trying to rig it so people can press ### to transfer an inbound fax to the inbound-fax line =) |
| 21:25.29 | blackhole | DrDamnit, xSmurf Can you tell me how can i get data from user without playing any file i just simply want to call some api which waits for x number of digits for user to enter and doesn't streams any file |
| 21:25.41 | Kobaz | Read() |
| 21:26.13 | blackhole | Kobaz, Is it for me? |
| 21:27.11 | *** part/#asterisk wesphillips (~wphillips@64-19-240-7.caprock.net) |
| 21:27.31 | Kobaz | yes |
| 21:28.18 | DrDamnit | JunK-Y: Any thoughts on how I can resolve this recording issue? What else should I look for? |
| 21:28.30 | blackhole | Kobaz, $AGI->read nothing like that? |
| 21:28.37 | blackhole | Kobaz, what exatcly to call I am little new to AGI |
| 21:29.21 | Kobaz | open the asterisk console (asterisk -r) and type: core show application Read |
| 21:30.29 | Kobaz | agi just executes dialplan, so you should be familiar with dialplan and how to get docs on dialplan apps/functions |
| 21:30.45 | DrDamnit | *s |
| 21:30.59 | blackhole | Kobaz, So i cant simply use AGI to get data? |
| 21:31.14 | blackhole | Kobaz, Like i can do using get_data but which must play a file.... |
| 21:31.24 | Kobaz | agi is a transport mechanism for dialplan |
| 21:31.36 | Kobaz | i don't know what get_data is... that's not part of asterisk |
| 21:31.49 | blackhole | hmm ok |
| 21:31.55 | xSmurf | get_data is a method of php agi |
| 21:32.04 | xSmurf | but I'm not sure which agi function it calls |
| 21:32.13 | DrDamnit | blackhole: go here: http://phpagi.sourceforge.net/phpagi22/api-docs/, and click on AGI in the nav. Read all the available methods. |
| 21:32.43 | Kobaz | blackhole: you're using a library to access agi functionality, you'll need to read up on how to use that library... apparently it's phpagi |
| 21:34.31 | DrDamnit | phpagi is an abstraction for dealing with AGI. It's the best / easiest documentation I know of. |
| 21:35.09 | xSmurf | just stumbled on https://github.com/marcelog/PAGI https://github.com/marcelog/PAMI |
| 21:35.13 | xSmurf | looks nice |
| 21:37.05 | JunK-Y | DrDamnit: not quick sure. i'll try to dig it a bit tonight after work |
| 21:37.51 | DrDamnit | JunK-Y: I appreciate it. I am stumped... and I am a dCAP. :-) |
| 21:38.57 | DrDamnit | JunK-Y: It may have something to do with the fact that this was a different box, which was having a CPU overheat problem. So, I migrated the install to a new box, which uses hardware RAID5. I have moved all the asterisk data directories onto the RAID, and used /etc/fstab to mount them back onto the file system where asterisk thinks they should be.... |
| 21:41.20 | *** join/#asterisk luckman212 (~quassel@pool-173-77-253-145.nycmny.fios.verizon.net) |
| 21:45.06 | *** join/#asterisk magnusz (~mmg@ool-4a58b348.dyn.optonline.net) |
| 21:47.05 | *** join/#asterisk Inssomniak (~none@199.19.173.132) |
| 21:47.46 | Inssomniak | hey all, does asterisk depend/use dahdi dummy when its handling a call but not doing any transcoding? (Sip provider -> thru asterisk-> extension) |
| 21:47.56 | Inssomniak | same codec |
| 21:52.51 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 21:54.00 | *** join/#asterisk IsUp (~nocturne@unaffiliated/isup) |
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| 21:57.44 | *** join/#asterisk drift- (ae3001a4@gateway/web/freenode/ip.174.48.1.164) |
| 21:57.47 | drift- | Jan 12 16:41:59] WARNING[28882]: app_dial.c:1275 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown) [Jan 12 16:41:59] WARNING[28882]: pbx.c:2437 __ast_pbx_run: Channel 'SIP/trunk_6969-092ac8f0' sent into invalid extension 'CHANUNAVAIL' in context 'ringroups-custom-3', but no invalid handler |
| 21:57.54 | drift- | what file would i edit to fix this dumb error |
| 21:58.12 | drift- | and why does it happen? my phone goes busy when somone tries calling it :( |
| 21:58.34 | *** join/#asterisk pinoyskull (~pinoyskul@112.198.64.80) |
| 21:59.16 | atan | Well this is a new one for me. "chan_sip.c:19811 handle_response: Remote host can't match request CANCEL to call" |
| 21:59.22 | IsUp | drift-: something "dumb" in your sip.conf and extensions.conf |
| 21:59.39 | drift- | okay |
| 21:59.43 | drift- | i been frustrated all day :\ |
| 21:59.59 | *** join/#asterisk slugs_ (~chatzilla@unaffiliated/slugs) |
| 22:00.18 | *** join/#asterisk chasing`Sol (~rc4@smtp.master-zone.net) |
| 22:00.36 | IsUp | atan: which version of Asterisk u are running |
| 22:00.47 | atan | 1.8.1 |
| 22:01.37 | IsUp | hmm strange, this error (or warning) fixed since 1.4.x as far i know |
| 22:01.50 | IsUp | maybe its about with your provider |
| 22:02.00 | atan | The calls are internal though =) |
| 22:02.05 | IsUp | ah |
| 22:02.10 | atan | How would I get the exact version number? Just to be sure. |
| 22:02.16 | IsUp | core show version |
| 22:02.37 | atan | Yup, it is 1.8.1. Perhaps I should upgrade to 1.8.1.1 |
| 22:02.45 | Inssomniak | OK maybe Ill reword my question, when does asterisk need a timing source? only for playback/recording or does it need it to complete voip only calls with no transcoding? |
| 22:03.36 | atan | IsUp, it's happening when a Cisco IP phone calls itself. |
| 22:03.53 | atan | Previously the call would just go on hold and no troubles would come of it. Now it hangs up on itself =) |
| 22:04.36 | IsUp | atan: paste your sip.conf to pastebin |
| 22:10.05 | atan | http://pastie.org/private/39nbxrywuz6kin3zadvm4w |
| 22:10.18 | atan | IsUp, troubles are then 7960 calls 7960 =) get this cancel message |
| 22:12.44 | IsUp | atan: i dont have any experience on cisco on asterisk but your sip.conf looks ok, try adding "canreinvite=no" on both extensions |
| 22:13.01 | IsUp | maybe something wrong with your phone settings |
| 22:13.50 | IsUp | and try nat=no since they are working locally |
| 22:16.54 | *** part/#asterisk Brixius (~kmurphy@PDN-VBA.OnvoyInc.fw.onvoy.net) |
| 22:19.07 | SaiSoma | anyone have a sec to talk res_fax? |
| 22:19.37 | SaiSoma | having trouble getting ReceiveFAX to work, it's not negotiating T.38 and i'm trying to understand how this works. |
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| 22:21.59 | *** part/#asterisk ClintGoudie-Nice (~clint@smtp.callware.com) |
| 22:23.07 | doctorray | I have an issue where a call routed to multiple extensions is unable to be answered.. they'll pick it up, and it continues to ring everywhere. I'm in the 1.6.1 tree with Polycom phones. I have detailed SIP debugging enabled but haven't made a lot of sense of it. Has anyone else run into this before? |
| 22:23.31 | drift- | how do you restart settings without doing restart now |
| 22:23.33 | drift- | reload sip? |
| 22:23.40 | doctorray | sip reload |
| 22:25.14 | doctorray | 1.6.2.6, sorry |
| 22:25.59 | *** join/#asterisk tzafrir_laptop (~tzafrir@212.179.75.202) |
| 22:28.10 | drift- | [Jan 12 17:14:57] WARNING[29030]: app_dial.c:1275 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown) [Jan 12 17:14:57] WARNING[29030]: pbx.c:2437 __ast_pbx_run: Channel 'SIP/trunk_6969-09b80210' sent into invalid extension 'CHANUNAVAIL' in context 'ringroups-custom-3', but no invalid handler |
| 22:28.26 | drift- | i'm calling diffrent phone # and its trying to read 6969 trunk hrm |
| 22:34.30 | drift- | i got new problem |
| 22:34.44 | atan | drift-, join the club, yo! |
| 22:34.47 | drift- | my line some reason caller id heh |
| 22:34.52 | drift- | doesnt display the correct number :( |
| 22:35.19 | atan | drift-, welcome to voip. Where men are men and numbers are incorrect. |
| 22:35.36 | atan | drift-, if it's any peace of mind inbound calls to my systems from local cellphones all show up with the same number. =( |
| 22:39.07 | IsUp | ~book |
| 22:39.07 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook, or http://ofps.oreilly.com/ |
| 22:39.37 | IsUp | drift-: you should read "book" |
| 22:40.12 | IsUp | atan: you tried nat=no and canreinvite=no ? |
| 22:41.33 | atan | IsUp, sorry to say I have not. I will try canreinvite=no but the device is behind a nat so will leave nat=yes |
| 22:41.47 | IsUp | okay |
| 22:42.47 | atan | I just added it to the sip peer and nothing changed. |
| 22:42.55 | atan | Chaning nat=no disables any calling. |
| 22:43.02 | *** join/#asterisk atommat (~atommat@atommat.plus.com) |
| 22:43.18 | atan | I suppose it's really not a huge issue seeing it's just calling itself. All other calls run just fine as they should. |
| 22:43.18 | IsUp | you always getting "Remote host can't match request CANCEL to call"? |
| 22:43.42 | drift- | ah i fixed it |
| 22:43.53 | IsUp | great to hear that drift- |
| 22:44.34 | atan | IsUp, I get it when the phone tries to call itself. |
| 22:45.01 | atan | I do not get this error when it sends the call off to my trunk or on inbound calls from the trunk. |
| 22:45.19 | drift- | yeah |
| 22:45.32 | drift- | was my context |
| 22:45.38 | drift- | had to change it to proper group |
| 22:47.06 | IsUp | well its not a big issue then but you can try google |
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| 22:50.47 | Katty | mopes a bit |
| 22:51.11 | IsUp | hello Katty |
| 22:53.38 | atommat | are you all people that need help or people that can help? |
| 22:54.29 | xSmurf | can't we be both? |
| 22:55.04 | atommat | good answer |
| 22:55.43 | _Corey_ | Everyone needs help sometimes... :) |
| 22:56.00 | atommat | anyone with experience of bri ? |
| 22:56.40 | atan | _Corey_, some more than others but I'd agree |
| 23:00.54 | _Corey_ | atommat: I would try earlier tomorrow when more of the guys from europe are awake |
| 23:01.16 | atommat | good thinking, tnx |
| 23:01.32 | _Corey_ | (bri is much more common over there) |
| 23:01.56 | atommat | what goes over there? |
| 23:03.07 | _Corey_ | um, well I'm in the US (which I figured was clear enough when I referred to the "guys from europe") so when I said "over there" I meant europe |
| 23:03.36 | _Corey_ | hope that clears things up :) |
| 23:03.41 | *** part/#asterisk kn0x (~pinochle@74.63.68.42) |
| 23:04.33 | atommat | just wondered what was more common in US is bri is mostly european |
| 23:04.39 | atommat | *if |
| 23:04.40 | _Corey_ | ah |
| 23:04.57 | _Corey_ | yeah, it's not seen too much in the US anymore |
| 23:05.06 | _Corey_ | mostly PRI here |
| 23:05.14 | atommat | rgr |
| 23:05.51 | _Corey_ | I honestly couldn't say how common it actually is in Europe, but I know it's at least used |
| 23:06.32 | WIMPy | It depends on where in Europe. |
| 23:06.44 | atommat | seems to be a small to mid business solution in the uk for telelphony |
| 23:07.30 | atommat | back tomorrow, o&o |
| 23:27.03 | *** join/#asterisk hceylan (~hceylan@213.248.153.186) |
| 23:27.32 | hceylan | Guys I am writing a prototype SIP proxy, nothing serious, just a project |
| 23:27.49 | hceylan | I can successfuly establist the session |
| 23:28.15 | hceylan | Then When I try to bridge the RTP sessions, I see no data is sent from either end |
| 23:28.31 | hceylan | I am using connected UDP sockets is that why? |
| 23:28.33 | hceylan | any advice? |
| 23:34.21 | *** part/#asterisk hceylan (~hceylan@213.248.153.186) |
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| 23:59.08 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 23:59.15 | beek | evenin' leifmadsen |
| 23:59.19 | leifmadsen | yo |