| 00:06.10 | pabelanger | doug: are you using trunk? |
| 00:06.56 | doug | i think so. |
| 00:07.27 | doug | URL: http://svn.asterisk.org/svn/asterisk/trunk |
| 00:07.30 | doug | yup, would seem to be. |
| 00:08.15 | pabelanger | well, if you are seeing chan->__do_not_use_name you must be. It's in the process of opaque-ifying ast_channel, so there might be some bugs |
| 00:08.46 | pabelanger | https://reviewboard.asterisk.org/r/1655/ |
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| 00:10.33 | doug | yeah, i figured that already. |
| 00:10.49 | doug | "There would be many benefits..." |
| 00:10.52 | doug | that's not you writing, is it? |
| 00:11.00 | pabelanger | no |
| 00:11.47 | doug | cool, then you won't be insulted when i laugh |
| 00:12.20 | pabelanger | problem? |
| 00:12.53 | doug | no, it's just silly to have accessors for perfectly functional fields. |
| 00:13.15 | doug | i can't think of when i've ever run into a situation and thought "wow, if only this had accessors instead of a field..." |
| 00:13.32 | pabelanger | meh |
| 00:14.08 | doug | but i've run into 29384928 instances where accessors interfered with stuff |
| 00:14.19 | doug | poor or difficult-to-understand implementations, etc. |
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| 02:54.04 | Psych0 | Anyone have experience configuring FXO? |
| 02:54.20 | p3nguin | What kind of card? |
| 02:54.30 | Psych0 | OpenVox a400p |
| 02:54.36 | Psych0 | $ FXO modules |
| 02:54.39 | Psych0 | 4 |
| 02:54.54 | *** part/#asterisk doug (doug@breakout.horph.com) |
| 02:55.22 | Psych0 | I don't have an incoming route set up yet, but I do hear the "no service message" when I call the PSTN line |
| 02:55.39 | Psych0 | With an outbound route setup and a SIP phone on the system |
| 02:56.03 | Psych0 | The receiving end rings, and can answer which stop the ring out on the Asterisk CLI |
| 02:56.08 | p3nguin | We don't know what "incoming route" means. |
| 02:56.10 | Psych0 | But it just sounds like an open mic |
| 02:56.39 | p3nguin | nor "outbound route" |
| 02:57.16 | p3nguin | Pastebin your zap/dahdi config files. |
| 02:57.26 | Psych0 | Aw crap, I've been thrust into this position, and I'm not very familiar with any of it |
| 02:57.31 | Psych0 | K |
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| 03:00.48 | Psych0 | http://pastebin.com/u79Hshe9 |
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| 03:11.16 | p3nguin | I'm confused by the double context settings in zapata-auto.conf. |
| 03:12.57 | WIMPy | Seen that before. Some of the odd GUI scrips seems to generate that. |
| 03:13.14 | Psych0 | I hadn't noticed that. It was auto generated and said not to hand edit.... so I didn't really look into it |
| 03:13.32 | Psych0 | Should I comment out the default lines? |
| 03:14.16 | WIMPy | The are complete sense- and harmless. |
| 03:14.37 | Psych0 | Oh, OK |
| 03:18.41 | Psych0 | Hmmm, openvox's documentation mentions zapata-channels.conf..... I don't even have that file. |
| 03:19.15 | Psych0 | Oh, NVM, it just has what I have in zapata-auto |
| 03:19.18 | [TK]D-Fender | Not important. |
| 03:19.23 | [TK]D-Fender | chan_dahdi.conf is what is important |
| 03:19.43 | [TK]D-Fender | Well.. in your case it's referring to deprecated stuff... |
| 03:19.49 | [TK]D-Fender | What ver of * are you on? |
| 03:19.58 | Psych0 | Their sample has context from-pstn instead of from-zaptel on the FXO lines |
| 03:20.42 | Psych0 | 1.4.22-4 |
| 03:20.52 | [TK]D-Fender | ew |
| 03:21.34 | Psych0 | I'm not the one choosing versions |
| 03:21.47 | Psych0 | I'm just trying to get the card working in the box |
| 03:23.33 | [TK]D-Fender | Files looks fine as is so far |
| 03:23.48 | [TK]D-Fender | Go setup your inbound route |
| 03:23.54 | Psych0 | Hey, should extensions.conf have from-zaptel included? |
| 03:24.54 | [TK]D-Fender | Psych0: You are using FreePBX as it says, right? |
| 03:25.32 | Psych0 | It's a Trixbox 2.6.2.3 box..... I'm gueesing that's built on, stole, or utilizes freePBX |
| 03:25.37 | [TK]D-Fender | Psych0: Then don't mess with the dialplan. |
| 03:25.43 | Psych0 | OK |
| 03:31.26 | Psych0 | Alright, 2 dumb questions..... 1 how do I confirm which port the line is in (or does it matter)? and Do I do this in "inbound routes" or "Zap DIDs"? |
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| 03:34.25 | Psych0 | NVM on 2, I found instructions on a simple route for all un routed incoming |
| 03:36.02 | [TK]D-Fender | Psych0: Zap DID's is important if you want to treat your lines seperately on incoming |
| 03:37.09 | Psych0 | I THINK that the client just want them to all ring as one incoming line |
| 03:37.38 | Psych0 | They have 4 exisint PSTN lines and they don't want to have to notify all their clients of changes |
| 03:40.30 | Psych0 | Well, I don't have access to any of the SIP phones, so I set the inbound to put on hold. I hear the music well enough. |
| 03:40.52 | Psych0 | I guess I'm going to have to wait til tomorrow when I have access to a phone at the other end too. |
| 03:41.00 | Psych0 | Thanks guys |
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| 05:39.52 | shamelessn00b | where can I find the changelog for asterisk 10.x |
| 05:40.18 | p3nguin | http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-10.0.0 |
| 05:41.42 | shamelessn00b | thanks p3nguin |
| 05:42.06 | shamelessn00b | nice native XMPP support |
| 05:44.06 | shamelessn00b | confbridge got an overhaul as well |
| 05:50.47 | cstachris | what does native XMPP support mean? a native channel like chan_sip but for XMPP? |
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| 06:09.50 | LTxda | hi all. Does anyone know of a way to configure a user's voice mailbox to delete messages when they reach x days old? |
| 06:09.58 | LTxda | via addon or script? |
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| 06:27.02 | srbh | Can I modify the outgoing SDP for originated calls dynamically? |
| 06:27.39 | shamelessn00b | cstachris: yeah you can send XMPP messages from the dialplan now |
| 06:28.50 | cstachris | shamelessn00b: thanks - i haven't tried 10 because it's not long term support - still getting used to 1.8 stuff - trying to find it's weaknesses |
| 06:29.40 | shamelessn00b | I'm also using 1.8 in production, and extensively using AMI, confbridge app is really very catchy though |
| 06:30.47 | shamelessn00b | srbh: NAT issues?? |
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| 06:31.52 | cstachris | shamelessn00b: AMI hey.... set enabled=no then do a manager reload. do you get manager still enabled and opening on a random port (netstat -ptan | grep asterisk)? I just opened a bug a day or two ago about this - be interesting if you get the same result |
| 06:33.11 | shamelessn00b | cstachris: lemme see if I can check |
| 06:33.36 | cstachris | cool |
| 06:36.13 | shamelessn00b | 1.8.7.1 |
| 06:36.30 | shamelessn00b | and 1.8.2.3 |
| 06:36.54 | shamelessn00b | what version did you test on |
| 06:37.04 | cstachris | mine is 1.8.8.1 |
| 06:37.19 | cstachris | i've also got tlsenabled=yes |
| 06:37.29 | cstachris | which might be buggering up AMI reading it's config file |
| 06:38.36 | shamelessn00b | don't have 1.8.8.1 on any server I'm afraid |
| 06:52.22 | srbh | shamelessn00b: I have my own RTP server which I need to integrate for a client |
| 06:52.51 | srbh | shamelessn00b: so I want to handle RTP myself and use asterisk as a SIP stack as well as a dialplan executor |
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| 07:14.12 | cstachris | canreinvite = yes in sip.conf |
| 07:14.23 | cstachris | maybe the name has changed in 1.8 configs |
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| 07:28.23 | kaldemar | LTxda: they are just files on disk, a delete and rename on the files will do it, IIRC. |
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| 07:36.22 | shamelessn00b | hey kaldemar o/ |
| 07:37.15 | kaldemar | shamelessn00b: hi |
| 07:50.37 | *** join/#asterisk schmidts (~schmidts@Lo20-rt01.lm33.sil.at) |
| 07:50.39 | schmidts | good morning |
| 07:53.11 | wdoekes2 | ~ask |
| 07:53.11 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 07:53.21 | wdoekes2 | (not directed at anyone here) |
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| 08:00.50 | ollii | moin ! |
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| 08:37.12 | Defraz | trying to dial a toll free number then wait for a few seconds and dial an extension. |
| 08:37.30 | Defraz | exten => 123,1,Answer exten => 123,2,Wait(2) exten => 123,3,dial(Local/8008193902@outbound-allroutes,60,D(wwwwww85597)) |
| 08:37.36 | Defraz | SHouldn't that work? |
| 08:38.05 | Defraz | if I dial 123 it would wait 2 seconds then dial the toll free number then extension 85597? |
| 08:43.37 | ChannelZ | not entirely sure if the D arg understands w |
| 08:43.41 | kaldemar | Defraz: it would wait 2 seconds, then dial Local/8008193902@outbound-allroutes, wait for an answer and then send 85597 as DTMF after a 2,5 s break. |
| 08:44.33 | Defraz | hmmm wierd it isn't working. |
| 08:44.38 | Defraz | I must be missing something |
| 08:46.24 | kaldemar | D() does understand w. |
| 08:46.44 | Defraz | yea that is what I read. |
| 08:46.58 | Defraz | that is a two second pause correct |
| 08:47.14 | kaldemar | maybe the key is what you have behind 8008193902@outbound-allroutes |
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| 08:48.49 | kaldemar | 3 s if you have six w's. my eyes only caught five the first time. :P |
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| 08:57.45 | *** join/#asterisk qakhan (~qakhan@182.185.173.165) |
| 08:57.49 | qakhan | hi all |
| 08:58.36 | qakhan | i want to setup a message. if caller dial noexisting ext in asterisk then message play "invalid Ext" |
| 08:58.46 | qakhan | anyone can help me? |
| 09:08.16 | schmidts | qakhan IMHO there is the "i" extension especially for this |
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| 09:09.43 | *** join/#asterisk qakhan (~qakhan@182.185.199.48) |
| 09:10.39 | qakhan | i want to setup a message. if caller dial noexisting ext in asterisk then message play "invalid Ext" |
| 09:11.22 | schmidts | qakhan exten => i,1,Playback(Wrong_extension_dialed.wav) |
| 09:12.41 | kaldemar | i is only used in certain situations, it is not a catch-all-invalid. |
| 09:13.20 | kaldemar | the certain situations being applications that allow to enter extensions. |
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| 09:26.32 | schmidts | kaldemar thanks i wasnt sure about this ;) |
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| 09:31.43 | wdoekes2 | schmidts: s/IMHO/IIRC/ ;) |
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| 09:35.41 | schmidts | :P |
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| 09:49.05 | aberrios | Anyone using the R-Series stuff from Digium? I have a query about this section in the Manual: The following components are continuously replicated between nodes. |
| 09:49.05 | aberrios | Asterisk configuration files, voicemails, AstDB, and logs |
| 09:49.10 | *** join/#asterisk screenn (~screenn@178.151.86.196) |
| 09:50.16 | aberrios | If AstDB is replicated between nodes. When a switchover happens does the R-Series force a restart of Asterisk so it will re-read the latest AstDB or does it work by simply starting up * on the backup node |
| 09:51.21 | *** join/#asterisk bbkk (~chatzilla@124-168-183-154.dyn.iinet.net.au) |
| 09:54.14 | bbkk | hi, today I installed asterisk@home via a live CD and got it running -- currently I have my asterisk box; a windows box running xlite and an android phone running sipdroid. In coming calls to sipdroid from xlite works nicely. However outgoing calls from sipdroid to xlite simply hangs up |
| 09:54.30 | bbkk | or rather goes to voice mail -- immediately |
| 09:55.04 | aberrios | Ah found the answer. 4. The host software on PBX B starts Asterisk. So * is dormant on the secondary node and is loaded after the failure of the primary node. |
| 09:55.05 | bbkk | i tried googling; but can't find a proper hit, so i'm suspecting that it could be something very trivial - that i have missed |
| 09:55.50 | aberrios | bbkk, paste the CLI output of the outgoing call to a pastebin |
| 09:55.54 | bbkk | i don't think i have any NAT issues - as its all on the same local lan |
| 09:56.58 | bbkk | aberrios, i'm a total newbie.. i'm just reading up the docs -- just don't know how to get the CLI output which I have seen others paste in the asterisk forums |
| 09:57.26 | bbkk | let me search for a sipdroid debug option |
| 09:58.17 | aberrios | bbkk, on the box, get into the Asterisk Console by typing "rasterisk" on the Linux CLI, not type "core set vervobe 10". Now when in the asterisk CLI make the problematic call. You'll see the console scroll stuff. Paste this into pastebin |
| 09:58.26 | aberrios | not*now |
| 09:58.39 | aberrios | verbobe*verbose |
| 09:58.45 | aberrios | keyboard=arsey |
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| 10:01.07 | bbkk | got it... any pastebin i can use? |
| 10:01.17 | bbkk | or would any do? |
| 10:01.25 | aberrios | any will do |
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| 10:02.38 | bbkk | http://pastebin.com/meqvzTvK |
| 10:02.40 | *** join/#asterisk qakhan (~qakhan@203.130.22.202) |
| 10:02.44 | qakhan | i want to setup a message. if caller dial noexisting ext in asterisk then message play "invalid Ext" |
| 10:05.00 | aberrios | bbkk, do a "sip show peers" and pastebin that |
| 10:05.39 | qakhan | aberrios let me clear 1 thing |
| 10:05.43 | *** join/#asterisk tamiel (~tamiel@85-171-171-179.rev.numericable.fr) |
| 10:05.51 | qakhan | i have ext 1001 to 1010 |
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| 10:06.15 | bbkk | http://pastebin.com/7bCwZc88 |
| 10:06.29 | shamelessn00b | qakhan: search for the keywords invalid extension asterisk |
| 10:06.45 | qakhan | but if someone call 1011 or 1012 etc then message play "invalid ext" |
| 10:06.52 | aberrios | bbkk, extension 200 doesn't seem to be registered correctly. Double check your x-=lite settings |
| 10:07.24 | aberrios | qakhan, yes what the other guys said. You need the invalid extension. It is well documented. Check Digiums Asterisk wiki |
| 10:07.30 | bbkk | thanks.. |
| 10:07.58 | aberrios | bbkk, you'll now when it registers properly because you should see it on the asterisk CLI |
| 10:08.32 | bbkk | i can make calls from ext200 (xlite) to ext300 (sipdroid) .. even if i switch the extensions.. it would still behave the same.. anyway.. i'll look at the xlite settings now..thanks |
| 10:09.00 | aberrios | bbkk, you may have "accept guest calls" set to on. |
| 10:09.20 | aberrios | or whatever that setting is. means phones can make SIP calls without being registered |
| 10:10.44 | bbkk | brilliant.. |
| 10:10.54 | bbkk | it worked.. i haven't registered within xlite |
| 10:11.05 | bbkk | now it works |
| 10:11.11 | aberrios | ;) |
| 10:11.16 | bbkk | thank you very much aberrios |
| 10:11.28 | aberrios | bbkk, get another android with sip droid and try a video call =D |
| 10:11.33 | *** join/#asterisk Kernel_Core (~IceChat7@h-213-136-53-142.na.cust.bahnhof.se) |
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| 10:11.37 | aberrios | thats fun. ish. |
| 10:11.41 | Kernel_Core | hi all |
| 10:11.56 | bbkk | thanks again |
| 10:13.32 | Kernel_Core | anybody knows , after you save from makemenuselect , where does it write the saved config ? |
| 10:15.56 | Goldwing | check the menuselect directory in the asterisk source dir. |
| 10:17.49 | qakhan | aberrios let me clear myself. i made exts 1001-1010. if someone dial other then 1001-1010 then message play "ext not exist" |
| 10:18.48 | aberrios | qakhan, yes, we've all told you: you need the i extension. Search Digiums Asterisk Wiki for "invalid extension" or "i extension" |
| 10:19.03 | Kernel_Core | Goldwing: I don't see any related file there. |
| 10:19.15 | *** join/#asterisk Neptu (~Neptu@mail.avtech.aero) |
| 10:19.52 | qakhan | i know about i extention |
| 10:20.27 | qakhan | but dont know how to use in dialplan |
| 10:21.20 | qakhan | exten => _10XX,1,Dial(SIP/${EXTEN},25,t) |
| 10:21.21 | qakhan | exten => _10XX,n,VoiceMail(${EXTEN}@exi,u) |
| 10:21.21 | qakhan | exten => _10XX,n,PlayBack(vm-goodbye) |
| 10:21.21 | qakhan | exten => _10XX,n,HangUp() |
| 10:21.26 | qakhan | here is my dial plan |
| 10:21.55 | aberrios | qakhan, like schmidts said "exten => i,1,Playback(Wrong_extension_dialed.wav)" |
| 10:23.04 | qakhan | where i put exten => i,1,Playback(Wrong_extension_dialed.wav)? |
| 10:23.53 | aberrios | qakhan, usually at the end. play with it, see what happens. |
| 10:24.10 | kaldemar | qakhan: i extension will not work for you. and your dialplan is destined for trouble. you should not match numbers that you don't have. |
| 10:24.56 | *** join/#asterisk master_of_master (~master_of@p57B55B13.dip.t-dialin.net) |
| 10:25.08 | qakhan | kaldemar how can i resolve this issue |
| 10:25.16 | kaldemar | qakhan: if you have 1001-1010, then match those numbers only. exten => _100[1-9] and exten => 1010 |
| 10:26.11 | qakhan | but if i have exts like 1001-1010 and 1034 and 1028 |
| 10:26.14 | kaldemar | then you can have exten => _10XX,1,Playback(pbx-invalid), same => n,Hangup |
| 10:26.30 | kaldemar | qakhan: then you make more extensions. |
| 10:27.09 | qakhan | is there any option in sip.conf which check does ext exist? |
| 10:27.37 | kaldemar | qakhan: your problem is in dialplan, not in sip.conf. |
| 10:28.03 | qakhan | ok plz help me |
| 10:28.19 | kaldemar | qakhan: i already told you what to do in detail. go do it. |
| 10:29.26 | qakhan | sir i have to make ext rendomly. like i told you. 1023, 1087, 1055 |
| 10:29.42 | qakhan | then how can i do this |
| 10:31.33 | kaldemar | qakhan: i already told you. make more extensions. |
| 10:32.03 | qakhan | its to much work |
| 10:32.08 | qakhan | is there any way out |
| 10:32.22 | kaldemar | qakhan: consider another profession then. |
| 10:34.15 | wdoekes2 | qakhan: check ${HANGUPCAUSE} after the Dial() to a non-existent account |
| 10:42.33 | *** join/#asterisk shamelessn00b (~bwahahaha@host-130-net-107-160-119.mobilinkinfinity.net.pk) |
| 10:43.43 | *** join/#asterisk Faustov (madrid@gentoo/user/faustov) |
| 10:43.53 | qakhan | wdoekes2 how i use in dialplan |
| 10:44.03 | phix | hey hey |
| 10:44.05 | phix | I am back |
| 10:44.19 | phix | snom 300 or polycom SP312>? |
| 10:47.01 | aberrios | phix, i can tell you i have lots of 331s only trouble is the 2.5mm jack for the headsets keeps breaking. However there is a 331 revision that uses an RJ port for headset instead of 2.5mm. |
| 10:47.17 | phix | 331s what is that? |
| 10:47.22 | phix | polycom? |
| 10:47.27 | aberrios | Polycom Sooundpoint 331 sorry |
| 10:47.46 | aberrios | But I've never used Snom. So i cant really say which is better |
| 11:01.04 | *** join/#asterisk akv (~akv@lnxbx.dk) |
| 11:02.04 | akv | hey, is it possible to have queue use a ringingtone instead of music when a user is on hold? In denmark, you cannot use MOH without paying royaltees, even tough it were music you made yourself... |
| 11:08.58 | *** join/#asterisk Faustov (madrid@gentoo/user/faustov) |
| 11:14.13 | *** join/#asterisk Nasga (~Nasga@239.212.10.93.rev.sfr.net) |
| 11:18.45 | *** part/#asterisk bbkk (~chatzilla@124-168-183-154.dyn.iinet.net.au) |
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| 12:09.25 | shamelessn00b | akv you could make the user hang up, and call them back |
| 12:09.32 | shamelessn00b | instead of putting them on wait |
| 12:11.33 | *** join/#asterisk imox (~imox@91-64-181-64-dynip.superkabel.de) |
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| 12:50.38 | *** join/#asterisk jkroon (~jkroon@dsl-244-35-76.telkomadsl.co.za) |
| 12:51.37 | jkroon | hi all, i'm wondering if it's possible to do disallow=all; allow=g729; allow=alaw and then in the dialplan (before answer) force the codec in the incoming channel to either g729 or alaw, in my case based on the actual dialed number? |
| 12:52.03 | *** join/#asterisk OfferL (504a7b08@gateway/web/freenode/ip.80.74.123.8) |
| 12:52.15 | OfferL | hello everybody! |
| 12:52.55 | shamelessn00b | jkroon: is the number being dialed from the same trunk? |
| 12:53.02 | OfferL | I need a genious that will be able to tell me if I can route G729 between 2 ext in my asterisk if they have it but the server doesn't. |
| 12:53.19 | OfferL | (or - if someone can please help me install g729) |
| 12:53.54 | jkroon | shamelessn00b, all incoming calls are coming from the same source IP yes, or the same "trunk" as you would say, via the same peer definition yes. |
| 12:54.50 | jkroon | I need to manipulate the incoming call codec based purely on the number dialed. I peer with the specific partner purely on an IP basis |
| 12:55.02 | jkroon | OfferL, yes you can. |
| 12:55.23 | jkroon | but you can't record it, capture it, manipulate it or do anyting else "funny". |
| 12:57.35 | kaldemar | jkroon: i think that would require a re-invite since the call between asterisk and the endpoint is already set up. see what setting SIP_CODEC does to the whole call. |
| 12:57.42 | shamelessn00b | jkroon: you want to choose the codec for outgoing calls? |
| 12:58.30 | jkroon | kaldemar, if the channel was already Up I would agree. as it stands I can manipulate this after receiving the INVITE and replying with 100 Trying, before even sending a 18X Ringing -or 200 OK response. |
| 12:58.31 | kaldemar | OfferL: that's called passthrough. |
| 12:58.44 | jkroon | shamelessn00b, no, for incoming :). |
| 12:59.14 | shamelessn00b | For incoming calls you can simply define a separate uLaw only context in your sip.conf |
| 13:00.42 | jkroon | shamelessn00b, uhu? |
| 13:01.07 | jkroon | how. because the ONLY non-variable is the source IP of the initial INVITE. |
| 13:02.11 | OfferL | Thanks. I have a provider that can route calls into my system only with G729. My softphone is x-lite, and it suppost to support it. The problem is I don't have PINGs between clients and outside world (3 NICs on asterisk), so I'll have to route the calls via the server but without "understanding" them. |
| 13:02.18 | OfferL | How can I configure it? |
| 13:02.30 | jkroon | x-Lite does NOT support G.729, you need to upgrade to eyeBeam |
| 13:02.39 | OfferL | :-( |
| 13:03.06 | OfferL | Ok. So It seems I'll have to install G729 on the asterisk and let it parse the calls. Right? |
| 13:03.37 | OfferL | (btw- my Nokia VoIP client suppose to support 729?) |
| 13:04.39 | jkroon | OfferL, depends on the exact phone, my E52 did not support G729, my N9 does. |
| 13:04.52 | OfferL | I have N97. |
| 13:04.56 | *** join/#asterisk Faustov (madrid@gentoo/user/faustov) |
| 13:05.01 | OfferL | I'm looking in the menus right now.. |
| 13:05.07 | *** join/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
| 13:05.55 | *** join/#asterisk rvleij (~robin@square.fx-services.com) |
| 13:06.15 | jkroon | nokia won't explicitly enable/disable it, your phone will simply advertize it, or not. |
| 13:06.23 | OfferL | I see. |
| 13:07.02 | kaldemar | OfferL: directmedia=no for the peers in sip.conf |
| 13:07.03 | OfferL | Anyway, maybe you genious can help me with G729 installation on the asterisk box and then I hope everything will work ok? |
| 13:07.51 | OfferL | directmedia=no is "trunk based", not "route based". right? |
| 13:08.22 | jkroon | kaldemar, very little info on that SIP_CODEC variable that I can find via google but it looks like the right thing to go after, thanks. |
| 13:08.35 | OfferL | (I mean, if trunk set to =no, it will be always for that trunk) |
| 13:09.36 | OfferL | so if directmedia=no, there is no need to have the same codecs on both sides? |
| 13:10.44 | kaldemar | OfferL: there is no such thing as trunk=no in sip.conf. nor do "trunk based" or "route based" mean anything. |
| 13:11.28 | *** join/#asterisk cnu (cnu@the.ultimate.lamer.la) |
| 13:11.28 | OfferL | so if I set directmedia=no it means any call from any extension or any trunk will be no directmedia? |
| 13:11.47 | jkroon | kaldemar, SIP_CODEC_INBOUND seems to be the "correct" variable nou. |
| 13:12.15 | *** join/#asterisk [TK]D-Fender (~aoulton@216.191.106.162) |
| 13:12.18 | kaldemar | jkroon: good to know. |
| 13:12.53 | kaldemar | OfferL: if you set it under [general] and don't override it under any device. |
| 13:13.18 | OfferL | I see. Thanks. |
| 13:13.33 | OfferL | So now I just need help installing G729 on my system... |
| 13:14.02 | schmidts | wow wikipedia went just offline for SOPA strike, even i am with them it scared me for a second ;) |
| 13:14.34 | jkroon | OfferL, instructions on the digium site is very good. |
| 13:14.45 | jkroon | http://store.digium.com/productview.php?category_id=5&product_code=8G729CODEC - start here. |
| 13:16.40 | OfferL | 1. There is no free solution? 2. Will it work on Trixbox also? |
| 13:16.56 | kaldemar | G.729a is not free. |
| 13:17.19 | OfferL | and G729 (without a)? |
| 13:17.48 | kaldemar | not free either. |
| 13:17.58 | [TK]D-Fender | OfferL, NO FREE G.729 |
| 13:18.45 | jzaw | remember OfferL you have to hav a licence at both ends |
| 13:18.56 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 13:18.56 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 13:19.03 | OfferL | :-( |
| 13:19.16 | [TK]D-Fender | He's looking to bridge 2 PBX's? |
| 13:19.53 | jzaw | he might not be |
| 13:20.05 | jzaw | the linksys pap2 has 1 license in it |
| 13:20.17 | jzaw | so i can use it with my * |
| 13:20.32 | jzaw | if my * had a licence |
| 13:20.37 | kaldemar | [TK]D-Fender: no, provider that only supports "G729" |
| 13:20.47 | [TK]D-Fender | Saying "both ends" like that largely implies that's the topic. because every hardware vendor that supports it has paid theirs and nothing is required of the end-user |
| 13:20.55 | *** join/#asterisk Faustov (madrid@gentoo/user/faustov) |
| 13:20.58 | jzaw | sure |
| 13:21.01 | jzaw | but theres another cost |
| 13:21.18 | jacc0 | hehehe, lefgozer: http://www.nujij.nl/algemeen/dronken-man-vs-overvaller-opmerkelijk-video-zie.15322231.lynkx#axzz1jnu2yDdT |
| 13:21.28 | jzaw | transcode from g729 to alaw or ulaw if your sip provider doesnt do g729 |
| 13:21.42 | jacc0 | sorry wrong channel |
| 13:22.28 | OfferL | My problem is opposite. My provider has only 729. I don't have it. |
| 13:22.47 | jzaw | aye |
| 13:23.11 | jzaw | besides my saying both ends need a lincence doesnt preclude the use of vendor's kit or providers who already have a licence |
| 13:23.53 | jzaw | OfferL my experience of use of g729 is that the sound quality really is nice and it really really does save b/w |
| 13:24.12 | jzaw | so $10ish isnt bad |
| 13:26.18 | [TK]D-Fender | IAX2 +GSM saves more bandwidth than SIP+G.729 as of your 2nd channel.... |
| 13:26.43 | jzaw | aye but gsm ... urrgh |
| 13:26.49 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 13:26.51 | jzaw | barely legible to my old ears |
| 13:27.58 | [TK]D-Fender | majority of people consider GSM6.0 & G.729 to be about the same. |
| 13:28.03 | [TK]D-Fender | 6.10* |
| 13:28.31 | *** join/#asterisk jermey_g (~test@gw-sthlm01.rebtel.com) |
| 13:30.28 | *** join/#asterisk kaushal (~kaushal@49.248.16.122) |
| 13:30.38 | rvleij | jacc0: goeie :-) |
| 13:30.46 | jermey_g | helo |
| 13:30.54 | jkroon | [TK]D-Fender, GSM compresses better than g729? i thought they were both 8Kbps which means that on IAX/2 trunked they should be exactly the same? |
| 13:31.59 | [TK]D-Fender | jkroon, No, they aren't the same BW. GSM 6.10 = 13kbps, G.729 = 9kbps |
| 13:33.24 | jkroon | [TK]D-Fender, then I don't understand your statement re IAX2 +GSM saving more handwidth than ... never mind, you're comparing IAX2+GSM to SIP+G.729, and then yes, IAX2 + almost any codec kicks SIP in terms of BW. |
| 13:34.00 | [TK]D-Fender | jkroon, chan_bigprint ;) |
| 13:34.23 | jkroon | yea yea ... i already feel like a n00b after realizing my mistake ... |
| 13:35.05 | [TK]D-Fender | jkroon, s'ok, didn't take you long and you didn't jump off the deep end. Caffeine time :) |
| 13:35.23 | jkroon | sounds like a good plan that one. |
| 13:36.30 | [TK]D-Fender | Caffeine is always a good plan. |
| 13:39.23 | *** join/#asterisk garymc (~chatzilla@81.138.225.164) |
| 13:39.31 | *** join/#asterisk s[X] (~s_x_@ppp59-167-157-96.static.internode.on.net) |
| 13:40.18 | *** join/#asterisk Naikrovek (~mbluth@unaffiliated/naikrovek) |
| 13:41.16 | *** join/#asterisk generalhan (~generalha@about/windows/staff/generalhan) |
| 13:44.26 | beaver_rrr | what's the problem with this extension? http://pastebin.com/zyu5TZyG it wont convert wav to mp3 :( |
| 13:44.38 | Naikrovek | do you guys use Request Tracker, or something like it, to handle IT request/problem management? |
| 13:44.57 | ollii | otrs |
| 13:45.50 | jermey_g | my asterisk sends an INVITE with credentials in sip header WWW-Authenticate towards the remote peer |
| 13:45.56 | *** join/#asterisk timahvo1 (~rogue@197.176.227.144) |
| 13:46.16 | Naikrovek | beaver_rrr: why are you converting to mp3? the .wav files are going to be small. i don't nkow lame though, is it expecting a certain type of input? also you have no error handling. call a script or something that can leave the original .wav unmodified if lame fails |
| 13:46.21 | jermey_g | can i make it use header Authorization: instead of WWW-Authenticate. Eyebeam uses this header Authorizaton. |
| 13:46.42 | Naikrovek | ollii: thank you |
| 13:47.03 | beaver_rrr | Naikrovek: not small enough. Also i'm using flash-mp3 player for simple online service to listen calls |
| 13:47.34 | Naikrovek | well run lame manually and see what it bitches about |
| 13:47.37 | jermey_g | pls answer me or else i have to change the remote ser :( |
| 13:47.39 | Naikrovek | that'll be your answer |
| 13:49.28 | beaver_rrr | Naikrovek: hah, it bitches about nothing, it just works when i run it manually :( |
| 13:50.21 | *** join/#asterisk vipkilla (~t_dot_zil@unaffiliated/t-dot-zilla/x-2830497) |
| 13:50.23 | Naikrovek | beaver_rrr: then asterisk isn't calling it; you've got a problem in your dialplan. why are you using mixmonitor instead of system |
| 13:51.34 | Naikrovek | eh nevermind. i dont' know asterisk well enough to help you, honestly. |
| 13:51.39 | Naikrovek | but your problem is in the dialplan |
| 13:51.43 | Naikrovek | i just don't knwo what it is |
| 13:53.00 | ollii | where is your cli output if you execute this part of dialplan |
| 13:55.22 | Naikrovek | beaver_rrr: ^^^ what he said |
| 13:55.30 | beaver_rrr | yes, copypasting it |
| 13:56.41 | beaver_rrr | http://pastebin.com/cnKPKQQM that's how it looks |
| 14:01.06 | phix | http://en.wikipedia.org/wiki/Main_Page |
| 14:01.15 | ollii | line 5 and line 6 look strange |
| 14:03.51 | *** join/#asterisk MarKsaitis (~MarKsaiti@195.59.185.18) |
| 14:05.14 | *** join/#asterisk serafie (~erin@nat/digium/x-mwbediugkdlzwguh) |
| 14:06.02 | *** join/#asterisk pigpen (~mark@fw.seamans.cc) |
| 14:09.40 | *** join/#asterisk mjordan1 (~mjordan@nat/digium/x-ihwotcswkmvowzkz) |
| 14:11.09 | Katty | drags in |
| 14:11.45 | fenrus | mornin |
| 14:11.48 | *** part/#asterisk mjordan1 (~mjordan@nat/digium/x-ihwotcswkmvowzkz) |
| 14:11.52 | WIMPy | Happy internet censorship day |
| 14:11.54 | beaver_rrr | gonna fix tomorrow ._. |
| 14:13.15 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-akbuhzfjlvtyjjnq) |
| 14:13.15 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 14:13.30 | pigpen | WIMPy, you can't say "censorship" online. It is censored. |
| 14:13.49 | pigpen | shit. I just did it. |
| 14:14.01 | *** join/#asterisk newbe (~Newb@c-98-226-119-225.hsd1.il.comcast.net) |
| 14:14.18 | WIMPy | I guess freenode has to be blocked now. |
| 14:14.41 | pigpen | Heh, it already is by my isp. I can only connect to one node?in Australia |
| 14:15.02 | pigpen | Just so everyone knows: It is GVTC in South Texas. Call them and bitch. |
| 14:15.02 | Nugget | Don't mess with Texas. |
| 14:15.26 | pigpen | Texas is censored. We all carry guns remember. |
| 14:17.20 | Katty | just got out of texas |
| 14:17.39 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 14:17.39 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 14:17.45 | pigpen | Traitor. ;-) |
| 14:17.47 | pigpen | haha |
| 14:18.22 | [TK]D-Fender | Thre is this comedy routine about a girl dating foreign men, one of whom was about to take their citizenship test for the US. He said "2nd amendment, is most important amendment". She says "Really? More import than the 1st amendment protecting free speech?". He says "Man who has gun can say whatever he wants..." |
| 14:18.24 | Katty | what is GVTC |
| 14:19.15 | pigpen | Katty, GVTC is Guadalupe Valley Telephone Cooperative (I think) http://www.gvtc.com |
| 14:19.26 | pigpen | It is an ISP/Telco in Central/South Texas. |
| 14:20.07 | Katty | and why are we calling them to bitch again |
| 14:20.27 | pigpen | they block freenode. |
| 14:21.01 | pigpen | my guess is that I am the only one on GVTC that is actually signed into freenode. |
| 14:21.32 | pigpen | but I am a persistentant bastard and I love you guys to much! |
| 14:23.41 | Katty | ah..i see |
| 14:24.04 | newbe | they sound like a really nice ISP |
| 14:24.21 | newbe | really ahead of their time :p |
| 14:29.13 | generalhan | i cant seem to get the app_mysql add-on installed for 1.8.8.1 on CentOS 6.2. anyone else had similar issues and/or know of a possible solution ? |
| 14:29.48 | jacc0 | did you : make menuselect |
| 14:29.55 | [TK]D-Fender | generalhan, that is the outcome, where can we see the actual point of failure? |
| 14:29.56 | jacc0 | en select the module? |
| 14:30.01 | Katty | my asterisk does not work at all how to fix plz?? |
| 14:30.08 | generalhan | lol |
| 14:30.28 | generalhan | i was hoping that it was a common problem that i couldnt find a solution to ... thats why i didnt go into detail out of the gate |
| 14:30.47 | chuckf | Katty: turn the computer on |
| 14:30.54 | jacc0 | hug it, kis it, dance for it and sing for it; that usualy helps |
| 14:31.12 | Katty | what do u mean turn computer on?? |
| 14:31.23 | jacc0 | @generalhan: did you 'make menuselect' ? |
| 14:31.30 | jacc0 | or did you apt-get install asterisk |
| 14:31.47 | generalhan | when i take a look at it via menuselect, the app_mysql its 'X's out. it says at the bottom that it depends on 'mysqlclient(E)' but i cant seem to find a package for CentOS6.2 |
| 14:31.51 | chuckf | Oh sorry, first purchase the computer and when it arrives come back and we'll talk |
| 14:32.15 | Katty | *hee* |
| 14:32.18 | Katty | haven't heard that one yet |
| 14:32.26 | Katty | gives chuckf shiny blue ribbon |
| 14:33.12 | chuckf | hehe |
| 14:33.34 | [TK]D-Fender | generalhan, go install it. the package names are the same as always... |
| 14:33.40 | [TK]D-Fender | mysql-dev , etc |
| 14:34.16 | newbe | how about yum install mysql-devel ? |
| 14:34.44 | generalhan | [TK]D-Fender: the only mysql packages listed that i dont have installed are -embedded and -test |
| 14:35.07 | jacc0 | let me se what i use in debian |
| 14:35.34 | jacc0 | <PROTECTED> |
| 14:35.43 | jacc0 | that is what i use in debian |
| 14:36.06 | generalhan | in older version there was a mysqlclient. but not in 6.2 ... that i can find |
| 14:36.50 | jacc0 | and you could run ./configure after adding one of the dependencies |
| 14:41.00 | generalhan | bah ... |
| 14:41.14 | *** join/#asterisk adeel|work (~adeel@unassigned-220.80.183.216.net.blink.ca) |
| 14:42.07 | generalhan | jacc0: based on your libmysqlclient i took a look at some other packages hoping that there was something else that would work ... dont know why, but the libdbi and dependencies solved it ! app_mysql is now available ! |
| 14:42.55 | adeel|work | my SIP provider is sending me incoming calls from a random assortment of ip's inside a single subdomain ( /27 block); is there a way to get * to allow sip connections from that range for the peer? |
| 14:43.15 | adeel|work | i was thinking of setting the host=dynamic, and use the permit/deny fields |
| 14:43.19 | [TK]D-Fender | adeel|work, "permit / deny |
| 14:43.46 | *** join/#asterisk topro (~quassel@host-62-245-142-50.customer.m-online.net) |
| 14:45.09 | adeel|work | iirc, the order matters right? it should be deny first, and then a permit? |
| 14:45.27 | *** join/#asterisk _Corey_ (~chatzilla@mail1.infradapt.com) |
| 14:46.48 | topro | Hi, I'm running an asterisk 1.6 box to connect my local sip phones to an isdn line. Everything works quite fine but setting different callerid for different sip phones on outbound calls |
| 14:48.03 | [TK]D-Fender | adeel|work, yes, and yes. |
| 14:48.11 | topro | in my extensions.conf I got exten => _X.,1,Set(CALLERID(all)=somename <12345>) within [outgoing_calls] context |
| 14:48.50 | adeel|work | [TK]D-Fender, hmmm...ok, thanks....it seems to hit the * box, but doesn't seem to match the right context and is throwing it to my garbage collector (for SIP connections) |
| 14:49.03 | WIMPy | topro: Make sure the numbers are in the right format. Setting prolocaldialplan=unknown in chan_dahdi.conf will make that easier. |
| 14:49.39 | topro | this works for setting the same callerid for all phones but whatever I try I can't figure out how to set callerid depending on which phone makes the call, did a lot of search but no luck so far |
| 14:50.20 | WIMPy | topro: Set the callerid for the peers in sip.conf. |
| 14:51.14 | WIMPy | topro: Or of you want different caller IDs internally, you will have to do it manually. |
| 14:51.21 | topro | WIMPy: and not set it in extensions.conf at all? |
| 14:52.00 | topro | WIMPy: doing it manually is what I want, but how to? |
| 14:53.22 | WIMPy | No, not in extensions.conf. And what kind of manual do you want? |
| 14:54.23 | topro | You told me if I want different callerid's internally I will have to do it manually. that is what I need to have, so how to do it manually is what I couldn't figure out |
| 14:55.37 | WIMPy | I think the easiest way would be to set a channel variable in the sip peers containing the external CID. |
| 14:57.10 | topro | any hint on where I can find an example or something similar |
| 14:58.15 | WIMPy | Look for setvar in sip.conf. Then just use that variable in your dialplan in the place you already have. |
| 14:59.34 | topro | ok, gonna see if I can manage. thanks a lot |
| 15:01.30 | adeel|work | hmmm...it seems like the only way * matches a context (for a peer) is via the host=foo field |
| 15:01.48 | jzaw | re asterisk and xmpp (jabber) JabberStatus() interogates the status of a jid |
| 15:02.00 | jzaw | is there anyway to change the status of a jid if say hes on a sip call etc |
| 15:03.07 | [TK]D-Fender | topro, "SetVar=OutboundNumber=12345" , "SetVar=OutboundName=John Smith", ${OutboundNumber} , ${OutboundName} |
| 15:03.11 | ollii | for ejabberd exists a module which listens to ami and sets approriate status |
| 15:03.16 | [TK]D-Fender | topro, for your sip.conf peer |
| 15:03.29 | [TK]D-Fender | adeel|work, type=user |
| 15:03.42 | WIMPy | I'm pretty sure, name won't get anywhere. |
| 15:03.43 | adeel|work | [TK]D-Fender, just figured that out actually |
| 15:04.46 | jzaw | ah ollii so its more a function of ejabberd? |
| 15:06.01 | ollii | i'm using ejabberd to set status and asterisk jabbermessage to send xmpp messages to the approriate user for missed calls e.g |
| 15:07.35 | *** join/#asterisk timahvo1 (~rogue@41.90.5.117) |
| 15:07.51 | ollii | http://www.acipia.fr/community/asterisk/asterisk-ejabberd-mod-client-asterisk/ |
| 15:08.50 | *** join/#asterisk JohnnyAsterisk (~JohnnyAst@86-46-224-65-dynamic.b-ras1.pgs.portlaoise.eircom.net) |
| 15:12.49 | jzaw | thanks ollii i found that page will look into it |
| 15:13.16 | jzaw | i already send messages .... clid and time stamp to _a_ user |
| 15:13.24 | ollii | if you're using a newer version of ejabberd it works relatively fine.. |
| 15:13.37 | jzaw | now id like to change that status so that if i pick up a call or make one |
| 15:13.44 | jzaw | ppl can see im busy or free |
| 15:15.13 | jzaw | ollii: i presume mod_shared_roster is an ejbabberd module? |
| 15:15.28 | ollii | yes, its default, just has to be activated |
| 15:15.38 | jzaw | nods :) |
| 15:19.16 | topro | WIMPy: FYI got the callerid thing working with a combination of setvar in sip.conf and gotoif/got in extensions.conf. thanks again for your help |
| 15:19.34 | topro | s/got/soto/ |
| 15:20.11 | topro | infobot that was not exactly the replacement I meant, but nevermind |
| 15:20.23 | jzaw | ollii: just uncomment {mod_shared_roster,[]}, |
| 15:20.23 | jzaw | <PROTECTED> |
| 15:21.18 | OfferL | Hi again! |
| 15:21.32 | OfferL | I've just finished installing my G729, and it works!! |
| 15:21.40 | OfferL | Now I have another little problem. |
| 15:22.17 | jacc0 | just put on a dry pents |
| 15:22.18 | WIMPy | topro: Why do you need a gotoif? |
| 15:22.24 | [TK]D-Fender | To see if it's set |
| 15:22.31 | [TK]D-Fender | Because I'm betting it won't always be |
| 15:22.53 | OfferL | Something in my trunk config is not OK, so I can make calls (outbound), but if "anonymous sip call" general option is set to 'no' - The incoming calls get "not in service" (context external-sip). |
| 15:23.12 | OfferL | How can I fix the trunk definitions so it'll not be unknown sip caller? |
| 15:23.13 | OfferL | thanks! |
| 15:23.41 | topro | WIMPy: at the moment I only want a single phone to use a different callerid the all the others so I set a custom var in sip.conf, check for its value in extensions.conf and if it is set setting callerid to the other value. that way I don't have to have the var set/updated on all sip peers |
| 15:24.06 | WIMPy | ah |
| 15:25.37 | OfferL | Anyone, please? |
| 15:26.37 | [TK]D-Fender | OfferL, #freepbx <- |
| 15:26.40 | [TK]D-Fender | ~freepbx |
| 15:26.40 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 15:27.30 | OfferL | It's Trixbox, but I can edit the .conf files themselve. no? (so it's the sam asterisk general problem). Can I post my settings here? |
| 15:28.23 | Katty | i'mmmm the very model of a modern major general |
| 15:28.29 | Katty | i've information animal vegetable and mineral |
| 15:28.31 | *** join/#asterisk jrose_atDigium (~jrose_atD@nat/digium/x-itxpxwcwjkxpumea) |
| 15:28.37 | Katty | i know the kings of england and i quote the facts historical |
| 15:28.44 | Katty | from marathon to waterloo in order categorical! |
| 15:29.53 | *** join/#asterisk [TK]D-Fender (~aoulton@216.191.106.162) |
| 15:30.04 | [TK]D-Fender | <OfferL> It's Trixbox, but I can edit the .conf files themselve. no? <- NO |
| 15:31.40 | [TK]D-Fender | OfferL, When you join #freepbx and make a pastebin of your failed incoming call at verbose 10 & SIP DEBUG enabled along with your trunk config (masking only the secret) I will continue assiting you in there |
| 15:31.51 | [TK]D-Fender | ~pb |
| 15:31.51 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com , http://pastebin.us , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ ; or install pastebinit with yum or aptitude. |
| 15:31.53 | [TK]D-Fender | ^^^^ |
| 15:34.25 | *** join/#asterisk gego (~quassel@b238085.customer.hansenet.de) |
| 15:34.54 | OfferL | D-Fender - You're my hero! I'm going there! |
| 15:36.16 | jzaw | ollii: just to confirm |
| 15:36.16 | jzaw | if i have an extension 204 |
| 15:36.32 | jzaw | and i want to change asterisk@mydomain.co.uk status to busy when 204 is busy |
| 15:36.34 | jzaw | i just put |
| 15:36.44 | jzaw | <PROTECTED> |
| 15:36.57 | jzaw | in the ejabberd conf ? |
| 15:38.57 | ollii | not exactly |
| 15:39.21 | ollii | {"SIP/peername","jabbername","jabber.domain.tld}, |
| 15:39.46 | OfferL | D-fender: No one in #freepbx :-( |
| 15:39.48 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 15:39.55 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 15:40.47 | ollii | you should see something like this in ejabberd.log: I(<0.27384.1>:mod_client_asterisk:50) : Utilisateur "jabbername" enregistre et pas au telephone. |
| 15:40.52 | ollii | yes its french in the log -.- |
| 15:49.34 | jacc0 | " No one in #freepbx" ?? |
| 15:49.44 | [TK]D-Fender | Disregard. |
| 15:50.37 | *** join/#asterisk singler (~singler@84.15.129.49) |
| 15:59.44 | *** join/#asterisk chendy (~Alex_Chan@113.97.239.64) |
| 16:01.17 | LTxda | kaldemar, but is there a way to automate it or will I need to write a script to do it? |
| 16:01.33 | LTxda | i guess if I read between the lines there is no addon for this or a settings, it just needs to be scripted. |
| 16:03.26 | *** join/#asterisk OfferL (504a7b08@gateway/web/freenode/ip.80.74.123.8) |
| 16:04.17 | OfferL | D-fender: I've moved to #freepbx, posted the files, and didn't see that I'm connected nor you're there. |
| 16:04.37 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 16:04.37 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 16:05.05 | jzaw | [15:39:21] <ollii> {"SIP/peername","jabbername","jabber.domain.tld}, |
| 16:05.05 | jzaw | thanks |
| 16:08.22 | *** join/#asterisk afink (~afink@207.106.66.194) |
| 16:09.14 | afink | Hello everyone, does anyone have any experience with Yealink and paging? I'm just wondering how well Yealink supports asterisk's page function. |
| 16:10.26 | [TK]D-Fender | afink, *'s "page" isn't what does anything |
| 16:10.51 | [TK]D-Fender | afink, "paging" as most will implement is jsut sending an auto-answer header with the call to the device |
| 16:11.00 | [TK]D-Fender | afink, And has nothing to do with the actual Page app |
| 16:11.13 | [TK]D-Fender | afink, Go see what format it supports and just set it. |
| 16:11.46 | *** join/#asterisk navaismo (~navaismo@fixed-203-96-202.iusacell.net) |
| 16:12.30 | afink | I see, so not much of an issue as long as you know what your're doing? Is getting it working device dependent? |
| 16:14.28 | afink | or see what format it (the phone) supports and set it? |
| 16:14.44 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 16:17.20 | [TK]D-Fender | afink, which if any format it accepts |
| 16:18.03 | [TK]D-Fender | http://www.google.ca/#sclient=psy-ab&hl=en&source=hp&q=yealink+auto+answer+sip+header&pbx=1&oq=yealink+auto+answer+sip+header&aq=f&aqi=&aql=&gs_sm=e&gs_upl=1252l5199l0l5677l30l16l0l5l5l1l1125l5234l0.3.7.4.1.7-1l18l0&bav=on.2,or.r_gc.r_pw.r_cp.,cf.osb&fp=a28f50725e2c54b8&biw=1600&bih=927 |
| 16:18.55 | jzaw | ollii: strange issue .... i can see in my asterisk console that ejabberd registers to the management interface and immediately it seems that ejabberd stop listening on 5222 so asterisk says no connection |
| 16:19.19 | jzaw | <PROTECTED> |
| 16:19.46 | jzaw | Jan 18 16:19:23] ERROR[24803]: res_jabber.c:3187 aji_initialize: JABBER ERROR: No Connection |
| 16:19.47 | jzaw | have you seen this before? |
| 16:20.40 | jzaw | netstat -an -W | grep 5222 shows nothing |
| 16:20.55 | jzaw | netstat -an -W | grep 5038 |
| 16:21.20 | jzaw | tcp 0 0 y.y.y.y:49092 x.x.x.x:5038 ESTABLISHED |
| 16:22.12 | *** join/#asterisk dobby156 (~joe@79.135.102.10) |
| 16:22.13 | dobby156 | hi |
| 16:22.15 | navaismo | ~pb |
| 16:22.15 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com , http://pastebin.us , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ ; or install pastebinit with yum or aptitude. |
| 16:22.59 | *** join/#asterisk moobius (~mobius@84.254.40.156) |
| 16:23.18 | *** join/#asterisk Ad-Hoc (~nimbus@athedsl-388756.home.otenet.gr) |
| 16:23.32 | dobby156 | if I Gosub in dailplan code, and someone hangs up does it execute the hangup in both the context called and/or the context that called it |
| 16:23.36 | dobby156 | thank you |
| 16:24.19 | moobius | hello there. I am having a problem with asterisk 1.8.8.1 and privacy headers. I am receiving a call from a remote peer and I want to change the CALLERID(num-pres) to prohib_passed_screen. However although the remote-party-id is being set properly my from tag is completely striped out and replaced the (name) and (num) with asterisk |
| 16:24.57 | moobius | this is different behaviour than the 1.6.2 branch |
| 16:26.08 | moobius | trustrpid = yes on both pears and i've tried with both yes,pia on the sendrpid. Even if I set explicitely the callerid(num)/callerid(name) it still keeps on changing them to asterisk |
| 16:26.56 | OfferL | Hello everyone. |
| 16:27.00 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 16:27.19 | OfferL | First I want to tell to the world that D-fender is a genious and a hero! |
| 16:28.06 | OfferL | second: I have a little general question: does asterisk supports SMS sending? (if so, I'll start looking for gateways - hardware or sites..) |
| 16:28.22 | leifmadsen | OfferL: pretty much, no |
| 16:29.00 | leifmadsen | OfferL: on a very specific circuit it could, but you're highly likely not running that type of circuit (see app_sms -- this is not a general SMS application) |
| 16:32.18 | moobius | OfferL: usually sms is working over smsc gateways in which case you should look into kannel, or via SS7 you could check the asterisk implementation of it, or via a web service, which is usually easier :) |
| 16:33.28 | OfferL | The reason I'm wondering about the SMS thing, is that someone trying to sell me a piece of hardware with a SIM card in it, that makes GSM-SIP gateway into asterisk. That's simple. But also he sais that his device "supports SMS" and I'm trying to figure out how can one deply it at all... |
| 16:34.51 | moobius | OfferL: most such devices do gsm-sip in which case you can use them as peers for your asterisk and provide some api for the sms'es. SIP support MESSAGE, in theory a SIP MESSAGE to a GSM endpoint would not be difficult with some intermediate custom development |
| 16:35.19 | moobius | OfferL: but you end up sending sms'es at retail prices, which sucks |
| 16:36.31 | moobius | OfferL: you could probably use chan_mobile and a standard gsm phone for that matter without spending $500+ for a gsm-sip gw |
| 16:36.45 | *** join/#asterisk teddymills (~teddy@mail1.decisioningsolutions.com) |
| 16:36.58 | moobius | depending on the number of calls you are willing to send through it anyway |
| 16:37.09 | *** join/#asterisk pzyko (~pzyko@static-72-70-44-39.bstnma.fios.verizon.net) |
| 16:37.38 | *** join/#asterisk asilva (~asilva@gandalf.ai.unesp.br) |
| 16:37.43 | pzyko | I am having some problems with the endpoint manager in PBX in a flash... hopefully this is the right place to ask |
| 16:38.00 | [TK]D-Fender | ~freepbx |
| 16:38.00 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 16:38.03 | [TK]D-Fender | ^ |
| 16:38.07 | pzyko | aha, thanks! |
| 16:38.16 | ollii | jzaw: 2.1.2-2ubuntu0.1 is my version...that works well... |
| 16:38.36 | ollii | have to go by now...work day over! nice evening ;-) |
| 16:39.17 | moobius | leifmadsen: hey there! how is the shift8 development going on there? I need to add a showcase :) |
| 16:40.10 | *** join/#asterisk singler (~singler@84.15.129.49) |
| 16:40.16 | asilva | leifmadsen: heya!! :) |
| 16:42.19 | *** join/#asterisk Meaty (~meaty@office.abi.ca) |
| 16:44.27 | leifmadsen | moobius: heh I haven't really don't too much with it in a while now :) |
| 16:44.35 | leifmadsen | s/don't/done/g |
| 16:45.09 | moobius | leifmadsen: same here :) it takes more time to build a ui that to build a library :P |
| 16:45.32 | Meaty | Hi Everybody! Is there a good way to custom voicemail in asterisk 1.8? |
| 16:46.10 | [TK]D-Fender | Meaty, that is "custom voicemail"? |
| 16:46.25 | leifmadsen | [TK]D-Fender: I'm sure he means customized menus |
| 16:46.31 | leifmadsen | Meaty: there is not unless you change the code |
| 16:46.37 | leifmadsen | there is no templating system for voicemail right now |
| 16:46.45 | Meaty | change number options, and remove options from menu ? |
| 16:47.23 | [TK]D-Fender | Meaty, vi app_voicemail.c |
| 16:47.53 | moobius | Meaty: you could already re-record the prompts and leave the options out :P |
| 16:49.26 | Meaty | [TK]D-Fender: I did it in past, it's tiring when im upgrading asterisk |
| 16:50.32 | [TK]D-Fender | Meaty, There is still no config file to specify what you want. |
| 16:50.33 | Meaty | moobius: I thought of that |
| 16:50.41 | Meaty | Thx to all |
| 16:51.19 | afink | [TK]D-Fender: just saw, thank you |
| 16:51.25 | Meaty | I just wanted to be sure |
| 16:52.41 | Meaty | Good day EVERYBODY! |
| 16:52.46 | moobius | bye Meaty |
| 16:53.52 | moobius | did by any change anyone had any enlightenments on my issue above? I am thinking of opening an issue with it |
| 16:59.25 | *** join/#asterisk l2trace99 (~jr@rrcs-71-43-104-238.se.biz.rr.com) |
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| 17:08.25 | jermey_g | Hi, does asterisk support Authorization header or it only uses www-authenticate or proxy-authenticate header depending upon if its a 401 or 407 response code? |
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| 17:14.16 | mwolf9 | Hello. I am working with the AMI to get the current state of the Queues and update the state as new events come in. I am currently using the QueueStatusAction to get the initial state of the queue via the QueueParamsEvent. The issue I am having is that the only post initalization event that I can use to update the state of the queue is the QueueSummaryEvent however this is an event generated by an action.. Is there any other event that will be fired when |
| 17:14.17 | mwolf9 | the queue statistics change that does not need to be instantiated by an action? |
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| 17:25.43 | Gn0m | hi |
| 17:26.01 | Gn0m | I have a question regarding upgrading from asterisk 1.6 to 1.8 |
| 17:26.52 | Gn0m | I'm on a ubuntu box and I'm considering updating using the apt-get repository provided by digium |
| 17:27.11 | Gn0m | I'd just like to know how much config files are compatible between both versions |
| 17:27.35 | Gn0m | which is to say, I'd like to have an idea of the ammount of work required to get a working configuration after the update |
| 17:29.02 | [TK]D-Fender | Gn0m, upgrade.txt <- tells you what changed. |
| 17:29.36 | Gn0m | should have looked for that one |
| 17:29.40 | Gn0m | thanks :) |
| 17:32.05 | jermey_g | I am kinda a stuck with this |
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| 17:51.31 | leifmadsen | anyone know the option to specify the SIP signal/listening port on a polycom device? |
| 17:54.14 | jermey_g | leifmadsen:doesnt it have a web gui or sth? |
| 17:54.27 | leifmadsen | jermey_g: of course, but who uses a web gui to configure their phones? |
| 17:54.36 | leifmadsen | and nevermind, I got this to work without the port changes |
| 17:54.44 | leifmadsen | I need food as I can't think straight anymore |
| 17:55.10 | jermey_g | we are in the same boat. i skipped my lunch and i wanna eat this keyboard |
| 17:55.37 | jermey_g | brb |
| 17:55.56 | *** join/#asterisk d_preston215 (~chatzilla@50-73-214-237-philadelpia.hfc.comcastbusiness.net) |
| 17:57.10 | d_preston215 | Is there a way to for asterisk to detect a beep after a persons voicemail greeting? |
| 17:57.41 | p3nguin | You can use AMD and WaitForSilence together to do what you're probably trying to do. |
| 17:58.35 | d_preston215 | I have that going, but its hit or miss with messages playing before the voicemail recording actually begins. |
| 17:58.44 | *** join/#asterisk chuckf (~chuckf@ubuntu/member/chuckf) |
| 17:59.11 | p3nguin | Use two WaitForSilence. |
| 18:00.11 | d_preston215 | Why would that work better than just using a timeout with WFS? |
| 18:01.16 | p3nguin | I think it is described in the book. |
| 18:03.23 | p3nguin | With the proper silence duration and timeout on each of multiple WaitForSilence()s, you can account for silence before the outgoing message plays, the silence after it before the beep, and some silence after a beep. Then you record your voicemail for their mailbox. |
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| 18:31.40 | *** join/#asterisk rossand (~aross@foundation-yow.eclipse.org) |
| 18:34.31 | rossand | Hi, I'm using asterisk 1.6.2.22. I am setting variables in extensions.conf: joe=500 |
| 18:34.31 | rossand | This is working great in extensions.conf, but I'd like to have the value available to me in meetme.conf. I'm sorry to ask as this is no doubt very simple. I've tried SetGlobalVar and other examples I found online to no avail. Advice or links are greatly appreciated. Thanks |
| 18:40.15 | *** part/#asterisk rossand (~aross@foundation-yow.eclipse.org) |
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| 19:15.57 | tonsofpcs | Hey, I'm looking to have someone call into my SIP system here and I'm looking for a lightweight, easy-to-configure (or no-need to configure -- registration is unnecessary) Windows client to give them. Any suggestions? |
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| 19:51.59 | *** join/#asterisk fprior (be0bbc86@gateway/web/freenode/ip.190.11.188.134) |
| 20:00.31 | fprior | hi all: is there someone who could run Asterisk 1.8 with LinkSys SPA400 ? |
| 20:02.59 | p3nguin | fprior: Sure, it speaks SIP. |
| 20:03.13 | eZz | hm? what's a problems ? |
| 20:04.37 | p3nguin | I'm not seeing any problems. |
| 20:04.52 | fprior | for last 2 years my Asterisk 1.4 worked fine with SPA400; updating to 1.8, SPA400 enter in a reboot cycle: every 60 seconds gateway restart |
| 20:05.15 | eZz | turn on debug |
| 20:05.40 | p3nguin | I don't know of a way Asterisk could make the device restart. |
| 20:06.01 | eZz | i'm agree with that too |
| 20:08.45 | fprior | in group asterisk-es (http://tinyurl.com/8xyp5nm) the problem was discussed but with no solution; in the opinion of someone, there is "something" different in Asterisk 1.8 not compatible with LinkSys |
| 20:09.13 | p3nguin | I disagree. |
| 20:09.31 | p3nguin | It is not the responsibility of Asterisk to restart the device. |
| 20:09.44 | p3nguin | The device communicates by SIP. |
| 20:10.04 | fprior | the product SPA400 is EOL, no solution from Cisco. |
| 20:10.38 | fprior | p3nguin : I agree with you but, any idea for test the problem ? |
| 20:10.46 | eZz | 22:05 < eZz> turn on debug |
| 20:10.55 | p3nguin | What did you change on the device? |
| 20:11.05 | p3nguin | Something must have changed to cause a different behavior. |
| 20:11.25 | p3nguin | If you don't know, consider resetting the device to factory settings and then reconfigure it for your needs. |
| 20:11.31 | eZz | maybe you've changed a sip peers/users/password/etc |
| 20:11.46 | p3nguin | Even that should not make the device restart. |
| 20:12.26 | eZz | if every 60 seconds gateway restart - maybe this is caused by invalid attempts to register |
| 20:12.35 | eZz | check your peers and device info |
| 20:12.53 | eZz | basically turn on sip debug and see what's going on |
| 20:12.55 | p3nguin | I suppose that could be how that device operates. It should be easy to check. |
| 20:14.23 | fprior | p3nguin, I've checked configuration; I can make and receive calls. After 60 sec (from last restart cycle), call drop |
| 20:15.35 | eZz | maybe codecs incompability |
| 20:18.44 | p3nguin | Does it restart every 60 seconds, even if you are not on a call? |
| 20:18.52 | fprior | news: I changed an option in SPA400, from "static Address" to "Discover Dinamically" in thi page http://tinyurl.com/7w4zarz . Now, gw not restart; I can do one call fine. the second call return "Got SIP response 503 "Service Unavailable" back from 192.168.0.xxx:5060", "SIP/spa400e-000001b2 is circuit-busy " |
| 20:20.20 | *** join/#asterisk fofware (~fabian@host71.190-137-122.telecom.net.ar) |
| 20:22.57 | *** join/#asterisk TimeRider (steve@188-220-34-144.dsl.cnl.uk.net) |
| 20:24.41 | p3nguin | You have one SIP account per line on the device? |
| 20:25.34 | *** join/#asterisk felimwhiteley (~quassel@089-101-203026.ntlworld.ie) |
| 20:27.09 | fprior | p3nguin , only one account for 4 lines |
| 20:27.47 | p3nguin | When you send a call through the device, is it responsible for choosing which line to send the call out? |
| 20:28.12 | *** join/#asterisk felimwhiteley (~quassel@089-101-203026.ntlworld.ie) |
| 20:31.17 | fprior | p3nguin, I can select line, if number is 843575, I call L1843575 to select first channel |
| 20:32.39 | *** join/#asterisk ndespres (~ndespres@208.103.80.226) |
| 20:32.48 | rossand | Is there a way to change pins on a meetme conference like there is for voicemail? |
| 20:33.16 | p3nguin | I do not have usage experience with that device, so I don't know its capabilities. But your message indicates that the device will not accept more than one call at a time. |
| 20:33.25 | *** join/#asterisk [sr] (~kvirc@pal-213-228-140-150.netvisao.pt) |
| 20:36.15 | *** join/#asterisk Nasga (~Nasga@AAmiens-157-1-151-136.w83-192.abo.wanadoo.fr) |
| 20:36.24 | fprior | p3nguin: with * 1.4 I can use 4 line in the same time. testing 1.8 I'm waiting to end first call, before call follow number |
| 20:36.46 | p3nguin | It isn't a limit set by 1.8. |
| 20:37.02 | p3nguin | It might be something you have configured while using 1.8, but it isn't a setting imposed by 1.8. |
| 20:38.27 | p3nguin | I think it is a setting in the device. |
| 20:43.43 | *** join/#asterisk moy (~moy@173.239.155.74) |
| 20:44.11 | fprior | p3nguin, any idea to follow the problem ? |
| 20:46.57 | Nasga | fprior: then with asterisk 1.2 you should be able to use 16 lines at the same time :p |
| 20:47.20 | p3nguin | haha |
| 20:47.20 | *** join/#asterisk autofsckk (~autofsckk@unaffiliated/autofsckk) |
| 20:49.14 | fprior | Nasga: :p |
| 20:53.55 | eZz | btw it there any solution to voice recognition getting worked with asterisk ? |
| 20:54.22 | eZz | *is |
| 20:54.27 | p3nguin | Only third party apps. |
| 20:54.48 | eZz | do you know which ones except microsoft's tts ? |
| 20:54.49 | *** join/#asterisk singler (~singler@84.15.177.252) |
| 20:54.54 | [TK]D-Fender | eZz, Lumenvox . |
| 20:54.59 | p3nguin | Check out lumenvox. |
| 20:55.09 | eZz | thanks, will check it |
| 20:57.57 | fprior | eZz : Google's ASR, I've not tested but you can find an howto here: http://zaf.github.com/asterisk-speech-recog/ |
| 20:59.12 | p3nguin | I can't find pricing on lumenvox's site. |
| 21:02.29 | p3nguin | That is probably the #1 reason I don't buy things. No advertised pricing, I move on to another product. |
| 21:04.19 | _Corey_ | fprior: http://www.lumenvox.com/Store/ |
| 21:04.44 | p3nguin | Of course Digium sells it for $720. |
| 21:05.58 | eZz | hm I don't think I would like use any of google services |
| 21:06.56 | p3nguin | Why would someone want to pay for software on a per-month basis? |
| 21:07.28 | _Corey_ | Well, if I'm not planning on using it past the upcoming election or something like that ... |
| 21:08.06 | _Corey_ | (we're evaluating their new AMD app for a robocalling customer) |
| 21:08.52 | _Corey_ | They used to have a nicely priced "development kit" product so you could play with the engine with a little support.. not sure what happened to it |
| 21:09.00 | _Corey_ | i think it was only $150 or something |
| 21:09.02 | eZz | i was changed asterisk's AMD to sangoma's CPA/NCA engine... |
| 21:10.11 | p3nguin | _corey_: I don't see a link to their store anywhere on their site. How would a person navigate to the store without prior knowledge of the link to the store page? |
| 21:10.41 | _Corey_ | uh, good question... one of their guys e-mailed it to me :) |
| 21:11.21 | p3nguin | Ridiculous. |
| 21:12.00 | _Corey_ | Something's definitely up with those guys... I had a sales rep e-mail me and then a couple days later I get another e-mail from their Chairman saying the guy was gone and he was going to help me instead |
| 21:12.07 | p3nguin | It's like people making web sites never have a single thought about an end user actually using their site. |
| 21:12.42 | p3nguin | It's so easy to get fed up with people. |
| 21:13.12 | p3nguin | It's been how many months now waiting to get an account with that Altus Carrier company? |
| 21:13.40 | *** join/#asterisk timeshell (~chatzilla@gw.lusi.on.ca) |
| 21:13.41 | p3nguin | Their customer service is a pile of shit. |
| 21:13.45 | timeshell | ping leifmadsen |
| 21:14.04 | leifmadsen | timeshell: pong |
| 21:14.15 | timeshell | News for 1.8.9.2 on main site gives "You are not authorized to access this page" |
| 21:14.31 | p3nguin | Probably because it does not exist. |
| 21:14.31 | timeshell | Sorry, 1.8.9.0 RC2 |
| 21:14.55 | leifmadsen | timeshell: you'll have to ping mjordan about that -- my thought is that he probably hasn't hit the Published button yet. |
| 21:15.13 | *** join/#asterisk rjvvliet (~rjvvliet@178-85-122-187.dynamic.upc.nl) |
| 21:15.14 | timeshell | mjordan ping ^^^^^ |
| 21:15.24 | timeshell | :) |
| 21:15.53 | leifmadsen | I'm not doing the releases anymore |
| 21:18.02 | p3nguin | By the way, it's over two months, for those interested. |
| 21:19.59 | mjordan | timeshell: try now :-) |
| 21:20.07 | mjordan | leifmadsen: looks like I forgot step 42b |
| 21:20.21 | leifmadsen | mjordan: I've forgotten that step a few times myself |
| 21:28.28 | *** join/#asterisk chasing`Sol (~cS@smtp.master-zone.net) |
| 21:35.50 | rossand | for meetme, is there a way to have the person entering the admin code be the marked user? |
| 21:36.45 | rossand | I see an example of doing it with another extension. I'd prefer to do it with the same extension if possible. |
| 21:37.16 | *** join/#asterisk BJD10 (~ben@c-98-246-210-98.hsd1.or.comcast.net) |
| 21:41.32 | rjvvliet | rossand: can't you use the 'a' and 'A' options at the same time when joining? (sorry new to meetme...) |
| 21:41.40 | *** join/#asterisk ketas (~ketas@2001:ad0:91f:0:290:27ff:fea6:a66a) |
| 21:42.43 | rossand | rjvvliet: sure. However unless I have a dedicated extension for the admin, I can't think of how to get it to apply to admin users and not regular ones. |
| 21:43.44 | rossand | What I'm aiming for is non-admin users to sit with music on hold until the admin user arrives. Both would use the same extension, but different passcodes. |
| 21:44.33 | rjvvliet | rossand: yep, was thinking of that too.. , do you have some selection menu in front in witch you can build the options..... |
| 21:45.05 | rjvvliet | rossand: Ac differtend passcodes, what if you build the meetme OPTIONS var depending on the password ? |
| 21:46.24 | rjvvliet | rossand : mmmm missed where the pin is entered ,.... sorry.. its while calling meetme() |
| 21:46.53 | *** join/#asterisk libryder (~david@209.33.214.243) |
| 21:46.58 | rossand | rjvvliet: that's a thought, I might be able to hack that with authenticate. I expecting that's what the admin passcode was for. :-) |
| 21:47.29 | libryder | hey guys, know of any way to parse canadian zipcodes entered with DTMF tones? i'm not bringing up much in my searches |
| 21:48.22 | rossand | libryder: Canadian postal codes have letters of course. How would one specify the letter vs. the number, hit it twice/three times? |
| 21:48.24 | rjvvliet | rossand: nah Admin mode is for the menu options inside meeme() if i;am correct... ,but yep creating a dialplan infront validating the PIN and than reusing is would be an option. |
| 21:49.30 | libryder | rossand: is it possible that the keypad combinations are unique? i was thinking of something like predictive text where it can figure out what you are spelling based on the possible combinations of the keypad |
| 21:50.03 | rossand | rjvvliet: Thanks. I'll explore that. Very much appreciated. |
| 21:50.47 | rjvvliet | libryder: sorry to barge in, but if i'am correct DTMF is only 0 trough 9 and *# and ABCD, so i think building a complete dialplan would be needed. |
| 21:51.23 | rossand | libryder: Theoretically possible, but wow... challenging. You'd want to have an index on all possible postal codes and try and narrowing it down. You may need to offer a choice if you can't find a unique one. |
| 21:51.37 | rjvvliet | rossand : good luck.... hope i was helpfull, first 10min on IRC ;-) EVER........ |
| 21:52.22 | rossand | rjvvliet: Honoured to be here for this great moment. :-) |
| 21:54.00 | libryder | rjvvliet: it's ok... each key has a corresponding 3-4 letters and i believe there is also a pattern of number-letter in the postal code |
| 21:54.10 | libryder | but i'm having a tough time finding info on it |
| 21:54.40 | libryder | A1A 1A1 ... so 3 letters and 3 numbers |
| 21:55.09 | libryder | i can do the actual parsing with rails, then i'll pass the zipcode to google to get geo-coordinates |
| 21:56.00 | rjvvliet | libryder: Yep, you mean the ABC on the digit key 1 when you look at your cell phone?, i think only the phone itself lknows that you are dailing B when pushing the 1 twice, when pushing it 'incall' it will only dail 11 to the other side.. |
| 21:56.53 | libryder | i found something helpful here: http://blogs.adobe.com/formfeed/2008/11/validating_a_postal_code.html |
| 21:57.04 | rjvvliet | libryder: i think you will need a dialplan witch will filter those Key presses and builds the letter from it witch you can send to an AGI script |
| 21:59.09 | rjvvliet | libryder: Aa nice, i tink before looking for validation check if you can have asterisk detect a Postalcode by entering DTMF, i think that there you challenge is. |
| 21:59.31 | libryder | rjvvliet: exactly |
| 22:00.20 | rjvvliet | libryder: so witout regards if the postalcode is correct, and i think that this wil be quite some dialplan bilding.. ;-) (don't we love asterisk) |
| 22:00.50 | libryder | rjvvliet: i'm actually writing the dialplan in ruby using adhearsion :) |
| 22:01.05 | *** join/#asterisk s[X] (~s_x_@eth589.qld.adsl.internode.on.net) |
| 22:01.31 | libryder | i suspect that the key combinations for post codes are unique |
| 22:01.36 | rjvvliet | libryder: mmmmm have heard of both.... never looked at it ;-), hope you'll get the job done ;-) |
| 22:02.53 | rjvvliet | libryder: i wish you luck.. i sit this one out... |
| 22:03.16 | libryder | thanks :) i'll post back when/if i figure it out |
| 22:25.55 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 22:28.33 | *** join/#asterisk AkkerKid (~AkkerKid@50-73-146-237-ip-static.hfc.comcastbusiness.net) |
| 22:28.43 | *** part/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
| 22:29.04 | AkkerKid | Good afternoon everyone! Happy SOPA day! |
| 22:29.49 | eZz | :-D |
| 22:29.52 | WIMPy | BTW: Did the strike have any impact on traffic? |
| 22:30.16 | AkkerKid | Would anyone be able to point me towards information that could help me route calls based on their CID showing up in a SQL database with certain criteria? |
| 22:30.35 | eZz | i hope nobody like level3 or something like that will be involved with |
| 22:31.37 | AkkerKid | I'm sure we'll see some usage graphs for the sites like wiki... |
| 22:32.53 | ks3 | AkkerKid, Have you looked at func_odbc? |
| 22:32.56 | eZz | netcraft |
| 22:34.20 | navaismo | AkkerKid: a PHPAGI script maybe... |
| 22:34.34 | navaismo | s/PHPAGI/AGI/ |
| 22:35.04 | *** join/#asterisk shamelessn00b (~bwahahaha@host-130-net-107-160-119.mobilinkinfinity.net.pk) |
| 22:35.17 | AkkerKid | ... a bot that understands conversational english? |
| 22:35.51 | ks3 | AkkerKid, No, a bot that understands regular expressions |
| 22:36.01 | *** join/#asterisk tamiel (~tamiel@85-171-171-179.rev.numericable.fr) |
| 22:36.12 | WIMPy | ... in a certain form. |
| 22:36.53 | AkkerKid | I'm just looking for project names. I can take it from there. So basically what I should look up are func_odbc in an Asterisk context, netcraft, and AGI scripts. |
| 22:37.21 | *** join/#asterisk Nasga (~Nasga@AAmiens-157-1-151-136.w83-192.abo.wanadoo.fr) |
| 22:37.48 | AkkerKid | So I've got a starting point. Thanks, all! |
| 22:38.16 | AkkerKid | infobot: what's pi/0? |
| 22:39.30 | navaismo | ~pi/0 |
| 22:39.36 | navaismo | ?pi/0 |
| 22:39.47 | navaismo | ~book |
| 22:39.47 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 22:39.55 | navaismo | ~LOL |
| 22:39.56 | infobot | hmm... lol is stands for Laughing Out Loud. It is grammatically incorrect to use LOL in the first person; use 'heh' or 'haha' instead. If you want to use LOL, do '/me lol' instead. |
| 22:40.17 | navaismo | hmmmm |
| 22:40.35 | AkkerKid | quite. |
| 22:40.46 | AkkerKid | THanks again! |
| 22:40.47 | eZz | Little Old Lady |
| 22:40.49 | eZz | :-D |
| 22:40.59 | AkkerKid | vamooses |
| 22:41.49 | eZz | http://acronyms.thefreedictionary.com/LOL |
| 22:42.53 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 22:44.59 | navaismo | are playing with the bot |
| 22:46.08 | *** join/#asterisk afink (~afink@wsip-98-190-183-147.om.om.cox.net) |
| 22:48.08 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-akbuhzfjlvtyjjnq) |
| 23:07.24 | p3nguin | ezz: http://www.facebook.com/pages/Adults-thinking-LOL-means-Lots-Of-Love/238515634547 |
| 23:07.38 | *** join/#asterisk GGD (~deberle@pool-173-72-204-39.clppva.fios.verizon.net) |
| 23:11.07 | eZz | ;) |
| 23:26.56 | *** join/#asterisk autofsckk (~autofsckk@unaffiliated/autofsckk) |
| 23:43.44 | *** join/#asterisk logicwrath (~no@c-68-62-24-205.hsd1.mi.comcast.net) |
| 23:46.05 | *** join/#asterisk NightMonkey (~NightMonk@pdpc/supporter/professional/nightmonkey) |