| 00:01.57 | *** join/#asterisk coppice (n=chatzill@39.192.17.210.dyn.pacific.net.hk) |
| 00:04.09 | drynish | You were right! :) |
| 00:04.12 | drynish | A big thanks to you! :) |
| 00:04.48 | drynish | I really like asterisk even if I'm still beginning with it ;) |
| 00:04.55 | drynish | Keep the good work everyone |
| 00:04.57 | ghento | Hi all - just curious, is there a way to change the default value of Read() from 0? |
| 00:09.09 | *** join/#asterisk coppice_ (n=chatzill@39.192.17.210.dyn.pacific.net.hk) |
| 00:14.53 | fujin | ghento: uh, why? |
| 00:17.32 | *** join/#asterisk andylockran (n=andylock@genesis.zrmt.com) |
| 00:17.46 | andylockran | hey guys - I was hoping you could solve a quick puzzle for me |
| 00:18.04 | andylockran | my mobile phone has just broken and I want to set up an alarm on my asterisk phone to wake me in the morning |
| 00:18.10 | andylockran | what's the best way to do this? |
| 00:20.08 | ghento | fujin: i'm calling out and doing a confirmation call that mistakingly asks to press 1 to confirm, 0 to add to do-not-call..so if the call gets disconnected or whatever they are automatically added to the do-not-call by mistake since Read() defaults to 0 |
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| 00:22.42 | ManxPower | Whoo! Whoo! My Slashdot comment was moderated as "Flamebate"! |
| 00:23.24 | ManxPower | At least the moderation was accurate. 8-) |
| 00:29.55 | orkid | probably bait, not bate |
| 00:31.34 | JT | andylockran: cron and callfiles |
| 00:33.45 | TJNII | no, bate is probably correct. You know most internet trolls whack off to their posts. |
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| 00:40.05 | dasKreech | Hello |
| 00:40.17 | dasKreech | does anyone have any good recommendations for a headset? |
| 00:40.50 | *** join/#asterisk __freedom__lover (n=eduardo@201-13-181-113.dial-up.telesp.net.br) |
| 00:41.22 | __freedom__lover | hi guys.. |
| 00:41.29 | __freedom__lover | i have a doubt about stun |
| 00:41.45 | dasKreech | set it to kill |
| 00:41.50 | __freedom__lover | another day, i was configuring a ATA behind a nat router... |
| 00:42.03 | *** join/#asterisk mwalling (n=mwalling@unaffiliated/mwalling) |
| 00:42.29 | __freedom__lover | i set the ATA to use stun, but it did not connect... |
| 00:42.51 | __freedom__lover | when i disable stun in configuration, it connected!!! |
| 00:43.06 | __freedom__lover | someone can explain me why? |
| 00:44.08 | TJNII | Whose stun server did you use? |
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| 00:45.15 | __freedom__lover | i used stun.xten.com |
| 00:45.59 | TJNII | Did you use sip debug to see which IP addresses it was using? |
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| 00:47.08 | __freedom__lover | my ata don't have debug option.. and in asterisk no messages or registration trying... |
| 00:47.10 | BBHoss | whats the equivalent to insecure=very on 1.4 |
| 00:48.37 | __freedom__lover | BBHoss: insecure=invite,port |
| 00:49.00 | BBHoss | ok thanks |
| 00:49.11 | __freedom__lover | BBHoss, try http://www.voip-info.org/wiki/index.php?page=Asterisk+config+sip.conf |
| 00:49.14 | __freedom__lover | ;) |
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| 03:26.40 | Dave_____ | Hi * gurus!. |
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| 03:29.01 | DGonzalezh | Hello |
| 03:29.18 | DGonzalezh | anyone could help me with a cdr problem. |
| 03:29.35 | fujin | you'll have to be a bit more specific |
| 03:29.42 | DGonzalezh | I've installed * 1.4.13, and all new stuff but I can't get cdr_mysql to work. |
| 03:29.52 | fujin | using odbc or the native mysql connector? |
| 03:29.58 | fujin | I run app_addon_cdr_mysql here, no problems. |
| 03:30.05 | DGonzalezh | well native mysql |
| 03:30.12 | fujin | cool. so what's the problem? |
| 03:30.16 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 03:30.22 | fujin | pastebin your cdr_mysql.conf, please, hashing out the password |
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| 03:30.22 | fujin | ~pb |
| 03:30.23 | jbot | A Pastebin is a place to paste your stuff without flooding 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://www.rafb.net/paste |
| 03:30.34 | DGonzalezh | It'salways XXX'ed when I run menuselect |
| 03:30.39 | fujin | oh. that's odd. |
| 03:30.49 | fujin | Have you got the mysql libraries installed? |
| 03:30.55 | fujin | I don't think I even recall having to enable it |
| 03:31.00 | fujin | asterisk-addons builds it by default, iirc. |
| 03:31.03 | fujin | (or, did for me, anyway) |
| 03:31.07 | fujin | providing the libraries are there. |
| 03:31.24 | DGonzalezh | I run centos 5 and I yum but libmysqlclient returns a nothing to do, but I do havbe mmysql installed |
| 03:31.34 | fujin | libmysqlclient15-dev |
| 03:31.38 | fujin | do you have the -dev packages installed? |
| 03:31.42 | fujin | if not, you won't be able to buildem |
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| 03:33.26 | DGonzalezh | I'm in doubt, that'sa separate package on centos or it's inside MySQL default install? |
| 03:33.54 | fujin | I'm not familiar with centos sorry, but there should be a development package which will install the headers/objects to link against. |
| 03:34.47 | DGonzalezh | <PROTECTED> |
| 03:35.09 | DGonzalezh | though I don't like Debian due to it's lack of standar compliance issues |
| 03:35.55 | DGonzalezh | But libmysqlclient it's contained into MySQL or it'sa separate lib?. |
| 03:36.10 | fujin | It's a seperate package here. |
| 03:36.34 | DGonzalezh | hmmm?, gotta check it out |
| 03:37.17 | hello- | does anyone know of any softphone package or any method of getting several usb phones over 10 voip accounts(not worried about this part) |
| 03:37.30 | DGonzalezh | Anyone of you other guys familiar with RH style? |
| 03:38.24 | DGonzalezh | I'm installing the MySQL-devel pkg to see if it works |
| 03:38.47 | kmhunt | I've been looking and I can't find anything out that will do what I need it to do |
| 03:39.50 | kmhunt | A hotel is asking me if I can spread 120 usb phones on a rollover basis to 10 voip accounts |
| 03:39.56 | DGonzalezh | USB phones are a complicated issue |
| 03:40.12 | DGonzalezh | wow |
| 03:40.17 | kmhunt | yeah |
| 03:40.22 | kmhunt | their pbx is ancient |
| 03:40.45 | kmhunt | and fxo channel banks seem like something they do not want to invest in |
| 03:41.00 | kmhunt | I did their ethernet and wireless at the hotel |
| 03:41.04 | kmhunt | as well as VOD services |
| 03:41.19 | kmhunt | but this one is throwing me for a loop |
| 03:41.44 | DGonzalezh | Hmmm so no way to use ATAs of Gateways? |
| 03:41.54 | kmhunt | basically need to find a softphone package that allows unlimited devices |
| 03:41.55 | DGonzalezh | BTW mysql-devel was the issue. |
| 03:42.06 | DGonzalezh | free i supose? |
| 03:42.14 | kmhunt | no they don't want to do ATA |
| 03:42.20 | kmhunt | not necessarily |
| 03:42.32 | kmhunt | but their budget limitations are kind of ridiculous |
| 03:42.54 | WilliamK | fujin: sorry I'm looking at this conversation a tad late; CentOS v5.0 is Redhat Enterprise 4.1 for all intents and purposes |
| 03:43.27 | kmhunt | anything I can interface with Trixbox freepbx or asterisk I was thinking was going to be the most viable solution |
| 03:43.41 | WilliamK | same commands accross the board; have yet to find one different cept for their up2date portion |
| 03:44.28 | kmhunt | the issue is that they don't want to pay a large amount for the hardware then on top of that have to handle my labor which is very reasonable, but for this large of a job I need to be compensated |
| 03:45.19 | kmhunt | 120 rooms is no joke.... I realized that when wiring cat6 at another hotel |
| 03:45.39 | WilliamK | kmhunt - I know exactly where you're coming from... had that pulled on me several times by clients who "tried"... |
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| 03:45.53 | fujin | WilliamK: unfortunately, I still don't follow it nor desire to use it |
| 03:46.06 | fujin | does the mysql client have a library package to build against? |
| 03:46.06 | kmhunt | that is why I haven't taken the job yet |
| 03:46.12 | WilliamK | fujin, that's fine - didn't say you had to :) |
| 03:46.13 | fujin | libmysqlclient*-dev, ilkeeverything else? |
| 03:46.22 | WilliamK | yeah if you install it |
| 03:46.24 | kmhunt | I am looking for a solution that will work on the same basis on a small scale |
| 03:46.33 | kmhunt | then scale it up |
| 03:46.59 | kmhunt | the other particulars are not easy either but are manageable |
| 03:47.09 | kmhunt | they seem dead set on skype |
| 03:47.12 | WilliamK | you can always do the "yum search packagename" and then once you find what you want, type "yum install packagename" |
| 03:47.30 | DGonzalezh | WilliamK>: Thanx |
| 03:47.36 | DGonzalezh | WilliamK: Thanx |
| 03:47.53 | DGonzalezh | If I remember how to IRC that was the waay? |
| 03:48.09 | kmhunt | skype can be managed if I can find software that can handle 120 usb devices |
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| 03:49.06 | DGonzalezh | It was the problem with my mysql_cdr. I didn't have mysql-devel, now I'm compiling the .so module.... |
| 03:49.31 | DGonzalezh | after that is there anything i have to do to get CDR working besides basic * configuration. |
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| 03:56.56 | DGonzalezh | call datils are working.... |
| 03:56.59 | DGonzalezh | Thanks guys |
| 03:57.13 | DGonzalezh | what may I be helpful wityhh, now that I've been serverd?. |
| 03:57.44 | WilliamK | you may need the * addons, but not quite sure |
| 03:58.20 | DGonzalezh | yup tehy're needed, recompiled it and now it's all working, the Records panel on FreePBX is showing data |
| 03:58.45 | fujin | gah! freepbx? :| |
| 03:58.49 | fujin | = do not want |
| 03:59.18 | DGonzalezh | Note for all of those using RH like distros, when installing LAMPA+freepbx remind to install yum -y install mysql-devel |
| 03:59.34 | fujin | ^^ |
| 03:59.42 | fujin | apt-get build-deps asterisk |
| 03:59.47 | DGonzalezh | hehe guys have a lot of enemies and I respect and undestand that. |
| 04:00.07 | DGonzalezh | gahhh not like lesnbian = Debian. |
| 04:00.15 | fujin | bit picky? |
| 04:00.22 | fujin | what, for using a not-shit distro? |
| 04:00.43 | DGonzalezh | well noyt flame wrs, I do like Ubuntu, |
| 04:00.45 | fujin | although, that's unfair, centos is a far sight better than fc/rhel |
| 04:00.55 | WilliamK | fujin - too easy to ruffle feathers :) |
| 04:01.23 | DGonzalezh | yup that's rite, fc is crappy and rhel not used never. |
| 04:02.01 | WilliamK | fc = dies very quickly after install, rhel is for corps who like paying redhat support monies |
| 04:02.25 | fujin | we buy dell hardware, they try and sell us RHEL all the time |
| 04:02.27 | fujin | *cringe* |
| 04:02.50 | fujin | the problem is some commercial software, like Propel accelerator only runs on RHEL (not my idea... I just had to build the damn thing) |
| 04:03.02 | WilliamK | fujin - yeah I know the feeling |
| 04:03.06 | CoffeeKid | fujin, every try centos? |
| 04:03.10 | CoffeeKid | *ever |
| 04:03.15 | fujin | no, I'm very happy with ubu/deb |
| 04:03.20 | WilliamK | have 2 QuadCore servers sitting behind me right now using RHEL cuz the client wanted it |
| 04:03.23 | fujin | although, I prefer Gentoo personally ;] |
| 04:03.29 | CoffeeKid | fujin, same :) |
| 04:03.52 | fujin | I don't find it's as manageable for large, ISP/datacentre style operations as ubuntu or debian are |
| 04:03.56 | fujin | we roll Ubuntu here, it's fine. |
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| 04:05.17 | CoffeeKid | fujin, we used RHEL forever, then decided centos was basically the same thing and did everything we needed it to do for free :) |
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| 04:14.55 | DGonzalezh | I installed 24 servers using kickstart all of them with RHEL 4 and the client and us support company are very happy with them |
| 04:15.44 | DGonzalezh | performance on dual quad-core servers is quite nice and 64-bit computing there for Informix DB abd Postgres is awesome |
| 04:17.40 | DGonzalezh | My * server here at home runs on a shitty machine but I'm planning on scaling it to a hi-end server someday. |
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| 04:22.22 | JT | eww, crackhat |
| 04:22.43 | DGonzalezh | what's that? |
| 04:23.02 | DGonzalezh | I'm not still very used to english sayings |
| 04:23.02 | JT | redhat |
| 04:23.10 | JT | rpm based distros |
| 04:23.10 | JT | ugly |
| 04:23.37 | DGonzalezh | aw I see but it eruns well, I've always said If it works or ain't broken why fix-it? |
| 04:24.33 | JT | broken from my perspective, i don't use them |
| 04:24.49 | DGonzalezh | aw yeah but I don't think such as .deb ones, I once on a Lesbian system tried to install snmp or something and it said 3 new packages will be installed and 89 will be removed including the kernel .. |
| 04:25.09 | JT | ... |
| 04:25.11 | DGonzalezh | can you dig it? |
| 04:26.21 | DGonzalezh | it's absolutely weird and .deb systems won't follow standards, I mean I respect the users and admire those that turn it upside down and know their internals but I mean it's all Linux in the end or isn't it? |
| 04:26.41 | DGonzalezh | it's all free, and we're open-sorced open-minded people. |
| 04:27.08 | DGonzalezh | and that's all I have to say about that. |
| 04:27.35 | JT | yeah sure |
| 04:27.38 | JT | not following standards |
| 04:27.40 | JT | bullshit |
| 04:27.43 | DGonzalezh | hehe. |
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| 04:29.24 | DGonzalezh | I finally got my cdr working and that's cool for me whatever works is good, I leave those obscure issues to designers and OS gurus. I'm just a user fanatic and user who uses whatever works fine. |
| 04:29.59 | DGonzalezh | tell me when in the world would be Ubu/Deb as nice and easy to use and compatible with hardware as SuSE which I also like a lot> |
| 04:30.13 | JT | in english? |
| 04:30.33 | DGonzalezh | in spanish! |
| 04:31.11 | DGonzalezh | Hehe well let's get back to Asterisk.... |
| 04:31.44 | DGonzalezh | Anyone ... needeing some help I'm craving for questions and willing to answer |
| 04:32.00 | DGonzalezh | but I guess in all of your contries it's very late-night. |
| 04:33.33 | DGonzalezh | Well anyway, I will come back here so we can talk more bout Asterisk and help somepone needing help. |
| 04:33.37 | DGonzalezh | Bye |
| 04:33.39 | DGonzalezh | C'ya |
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| 05:05.24 | [pyro] | hey guys. Does anyone know the status of BLF & Direct Line Pickup (in one button) for Aastra 5xx range of phones? Ive been looking through the forums and it looks like asterisk 1.2 needs to be patched, but the patch wont apply. Seems code base of asterisk has changed too much since the experimental patch was written in 2005. |
| 05:05.24 | mosty | i have calls coming in on a pri line with callerid with presentation prohibited, and i am forwarding them on to an IAX user, but i need to make sure that the IAX user cannot see the callerid- how can i do this? |
| 05:06.58 | mosty | obviously, i can set ${CALLERID(num)}, but how do i check if the callerid presentation is prohibited? do i have to use bitmasks and ${CALLINGPRES} ? |
| 05:08.31 | BBHoss | [pyro]: i would suggest trying 1.4, as it has a lot of new features |
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| 05:13.28 | J-5 | im new to asterisk,i have both a freebsd and ubuntu server box. witch would be easier to get asterisk up and running on? |
| 05:13.43 | mosty | linux |
| 05:13.47 | TJNII | Is there a command for the CLI to show the number of active calls and the devices concerned? |
| 05:14.00 | mosty | TJNII, "show channels" |
| 05:14.29 | TJNII | Cool. Thanks |
| 05:15.26 | TJNII | My friends and I have created a standard numbering plan and declared all locations should support 611 (repair) service. |
| 05:15.37 | TJNII | I guess I need to set up a nasty IVR with bad hold music now. |
| 05:16.14 | [pyro] | BBHoss: does 1.4 do this? BLF and direct call pickup on the same button? |
| 05:17.02 | [pyro] | BBHoss: because i have BLF working, but when i try and pick up a call thats ringing on an extention by pressing its flashing BLF button, it just dials the extension |
| 05:21.11 | *** join/#asterisk lemanal (n=lemanal@71.9.108.98) |
| 05:22.26 | [TK]D-Fender | [pyro], then make a NEW exten to watch that watches the same DEVICE but does something else. |
| 05:27.36 | *** join/#asterisk MrTelephone (n=na@h64184192-5.picriverisp.net) |
| 05:27.38 | *** join/#asterisk LoF^[Lawbringer] (n=Lawbring@cpc4-rdng12-0-0-cust292.winn.cable.ntl.com) |
| 05:27.54 | MrTelephone | Hey I have tiny issue with recording.. |
| 05:27.56 | MrTelephone | exten => 12345,1,Record(/etc/asterisk/prhc-greeting:wav|5|20|skip) |
| 05:28.10 | MrTelephone | returns failure status 'UNKNOWN' |
| 05:28.10 | MrTelephone | :( |
| 05:28.34 | [pyro] | [TK]D-Fender: sorry, i dont follow :\ |
| 05:28.42 | *** join/#asterisk obitux (n=obitux@dynamic26-77.MAN-B2-2.cablenet.com.ni) |
| 05:29.06 | obitux | necesito una pequeña ayuda en la conf de asterisk me podrian ayudar |
| 05:29.41 | flenders | obitux: si, pero no mucho! |
| 05:29.44 | [TK]D-Fender | [pyro] : you are the one who set up the hint at that #, and the actual extren with real priorities that cause it to dial the SIP device you associate with that exten. Instead make ANOTHER completely different exten that does the directed pickup, and who's hint looks at the SAME device as the other one did. |
| 05:30.22 | obitux | i need hel caso es que necesito hacer tres grupos diferentes pero que se intercomuniquen |
| 05:30.31 | obitux | alguna idea |
| 05:30.39 | flenders | obitux: mate, sorry, it was a joke |
| 05:30.49 | *** join/#asterisk marc7 (n=marc@216.19.180.137.novuscom.net) |
| 05:30.50 | flenders | you need to speak english in here |
| 05:30.58 | [TK]D-Fender | ~asteriskspanish |
| 05:31.08 | jbot | [~asteriskspanish] Asterisk Community in Spanish, just visit http://www.asterisk-la.org -=- IRC channel #asterisk-es |
| 05:31.08 | obitux | the room is |
| 05:31.09 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^ |
| 05:31.16 | obitux | tanks |
| 05:31.22 | *** part/#asterisk obitux (n=obitux@dynamic26-77.MAN-B2-2.cablenet.com.ni) |
| 05:31.53 | [pyro] | [TK]D-Fender: so ill have 2 buttons. one for BLF and another one to pickup calls from the same monitored extension? |
| 05:32.44 | [TK]D-Fender | [pyro], you are not paying attention. Re-read it AGAIN till you get your head on straight |
| 05:34.00 | [pyro] | [TK]D-Fender: hm ok, thanks for the help |
| 05:34.47 | [TK]D-Fender | [pyro], Do you actually get it now? |
| 05:34.57 | *** join/#asterisk lemanal (n=lemanal@71.9.108.98) |
| 05:35.13 | mosty | how can i check the value of a particular bit of ${CALLINGPRES} in my dialplan? |
| 05:35.40 | [pyro] | [TK]D-Fender: no, but im off to do some more reading. If it can be done ill figure it out eventually. |
| 05:36.22 | [TK]D-Fender | [pyro], Hrere... you have something like this : exten => 100,hint,SIP/100 exten => 100,1,Dial(SIP/100) |
| 05:37.11 | [TK]D-Fender | [pyro], INSTEAD do something like : exten => fred,hint,SIP/100 exten => fred,1,DirectedPickupThingy(however thats supposed to work) |
| 05:37.23 | [TK]D-Fender | [pyro], And do your BLF key to FRED instead |
| 05:37.38 | [TK]D-Fender | [pyro], calling something "100" doesnt' eman ANYTHING. its just an exten. |
| 05:38.04 | MrTelephone | SIP/2.0 603 Declined |
| 05:38.09 | MrTelephone | what is that message all about? |
| 05:38.14 | [pyro] | [TK]D-Fender: ok let me have a look |
| 05:38.19 | [TK]D-Fender | [pyro], So make a 2nd set of extens for the purpose of pickup+BLF. You can have non-pickup watch the original hint on 100 if you want to dial instead of pickup. |
| 05:39.12 | [pyro] | [TK]D-Fender: ok |
| 05:40.16 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 05:40.39 | [TK]D-Fender | [pyro], best to have the pickup ones to include alpha chars as well so they can't be dialed normally, unless you want to make things like : exten => #100,hint etc so as to have a normally dialable prefix. |
| 05:40.54 | [TK]D-Fender | [pyro], Remember to * its all jsut numbers... |
| 05:41.31 | [pyro] | [TK]D-Fender: yep gotcha ok ill have a play and see if i can set it up |
| 05:43.06 | *** join/#asterisk alpha232 (i=alpha232@198-144-143-60.dyn.megabroadband.net) |
| 05:44.48 | *** join/#asterisk lemanal_ (n=lemanal@71.9.108.98) |
| 05:47.17 | MrTelephone | can't record shit since I upgraded here |
| 05:47.41 | *** join/#asterisk Cyon (n=cyon@216.179.31.170) |
| 05:48.30 | [pyro] | [TK]D-Fender: I setup asterisk with freepbx so i think ill just have to add a new exten to the file by hand |
| 05:48.54 | *** join/#asterisk metabsd (n=metabsd@modemcable103.201-131-66.mc.videotron.ca) |
| 05:48.55 | MrTelephone | exten => 12345,1,Record(/etc/asterisk/prhcgreet:wav|5|20|skip) whats wrong with that line? |
| 05:48.59 | [TK]D-Fender | [pyro], Oh in that case you're immortal soul is already forfeit and you are in the wrong channel... |
| 05:49.17 | TJNII | Does callerid support non-numeric symbold like dashes in the number portion? Like callerid="blah" <2-1112> |
| 05:49.24 | *** part/#asterisk J_5 (n=J_5@cpe-71-72-210-44.cinci.res.rr.com) |
| 05:49.58 | [TK]D-Fender | MrTelephone, read this and you tell ME.... : Record(filename.format|silence[|maxduration][|options]) |
| 05:49.58 | [pyro] | [TK]D-Fender: haha ok :) |
| 05:50.58 | MrTelephone | no kiddin |
| 05:51.04 | MrTelephone | wonder why its crappin out then |
| 05:51.04 | MrTelephone | brb |
| 05:51.32 | MrTelephone | if there is an extensions 501-510 and you have an invalid handler in the dialplan as soon as you press the first digit it says invalid |
| 05:52.55 | *** join/#asterisk implicit_ (n=implicit@ip68-105-92-210.sd.sd.cox.net) |
| 05:55.27 | [pyro] | [TK]D-Fender: even if i set it up with 2 exten's one BLF+pickup and the other normal one, i still need 2 buttons, one blf+pickup(fred) and another one if i want to dial that exten. There is no way to BLF+Pickup and Dial that exten if its just idle all from one button? |
| 05:55.54 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 05:56.27 | [TK]D-Fender | [pyro], Depends on how this directed pickup works. If its only for ringing channels, then you'd need 1.4 for this. |
| 05:56.46 | [pyro] | ah ok |
| 05:56.48 | [TK]D-Fender | [pyro], Because you could make an exten that does a status check on the device then choose what do do based on that. |
| 05:57.06 | [TK]D-Fender | [pyro], But please refer to the "immortal soul" clause as you check out.... |
| 05:57.25 | [pyro] | yeah im not using 1.4 anyways |
| 05:57.45 | MrTelephone | why the hell is invalid being called on first digit when there are three digit extensions? |
| 05:59.36 | MrTelephone | because im not in the right context |
| 06:05.34 | MrTelephone | shouldn't gosub change the context you are in |
| 06:05.37 | MrTelephone | I mean goto |
| 06:06.26 | [TK]D-Fender | MrTelephone, You are talking a lot, saying little, and showing nothing. |
| 06:07.04 | *** join/#asterisk techie (n=techie@adsl-76-214-20-56.dsl.lsan03.sbcglobal.net) |
| 06:07.19 | MrTelephone | I'm a little frustrated here |
| 06:08.05 | MrTelephone | on incoming call for extension 1836 I have it sent to another context and asterisk isn't recognizing the extensions in that context |
| 06:08.10 | [TK]D-Fender | MrTelephone, and doing absolutely nothing to help yourself. Or perhaps you just came here to vent your frustrations rather than solve your problems... |
| 06:08.17 | mosty | paste your extensions.conf (and sip.conf if that's what you're using) on a paste site |
| 06:08.17 | *** join/#asterisk enTer (n=Genco@88.242.195.133) |
| 06:08.32 | [TK]D-Fender | mosty, I was waiting to say that.... and there you go just rushing off.... |
| 06:10.02 | MrTelephone | ~pastebin |
| 06:10.11 | jbot | [pastebin] a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste, or http://rafb.net/paste/, or http://pastebin.com is usually painfully too slow and unresponsive to use, use one of the other pastebin sites, or dpaste.com is a very nice pastebin as well |
| 06:10.14 | *** join/#asterisk Chris-NB (n=chris@ng1.kurtkrenn.com) |
| 06:12.04 | [pyro] | [TK]D-Fender: has what you suggested been done before? in asterisk 1.4 setting up an exten that monitors a device's status and then chooses what to do based on that? |
| 06:12.52 | [TK]D-Fender | [pyro], yes, I'm sure it has. I've done it based on "in-use", but full-mode status can be done ion 1.4 |
| 06:13.16 | MrTelephone | http://pastebin.ca/770376 |
| 06:13.40 | [TK]D-Fender | MrTelephone, awefully long for half the story... |
| 06:14.00 | MrTelephone | I was double checking some stuff |
| 06:14.19 | [TK]D-Fender | MrTelephone, thens till doing nothing here to solve anything.... |
| 06:15.40 | phix | hey, I am setting up a Linksys SPA3102 ATA, any one here done that before? :) |
| 06:15.58 | [TK]D-Fender | phix, www.voxilla.com <- go check out the forums |
| 06:18.04 | mosty | phix: yeah, it was pretty easy if you follow the manual |
| 06:19.27 | phix | ok |
| 06:19.30 | MrTelephone | my dialplan jumps aroudn too much |
| 06:20.07 | MrTelephone | it goes from [office] -> [office-outgoing] -> [pstn-out] -> [pstn-in] -> [office-incoming] |
| 06:20.11 | MrTelephone | because I'm calling myself |
| 06:20.19 | phix | The main issue I am having is my VoIP provider has preconfigured it, and it looks like they still have alot of internal stuff enabled that they used to test it :/ hmmm I guess I could always factory reset it |
| 06:20.21 | MrTelephone | maybe thats why its not working |
| 06:20.32 | phix | hehe |
| 06:21.02 | mosty | phix: is it locked? |
| 06:22.12 | phix | no |
| 06:23.16 | mosty | what are you stuck trying to do? |
| 06:23.17 | phix | Another thing I find weird is a standard POTS / PSTN handset can detect the other end has hung up and it stops ringing, but TDM2400 cards and this ATA keeps the handsets ringing for another 2 secs either though the other end has hang up |
| 06:23.21 | *** join/#asterisk lemanal_ (n=lemanal@71.9.108.98) |
| 06:23.34 | phix | mosty: PSTN -> asterisk |
| 06:23.52 | phix | in the dial plan I told it to dial s@myInternalHostname |
| 06:24.21 | mosty | and is asterisk setup correctly? ie do you know that the problem is on the ata? |
| 06:24.28 | phix | also, voip provider has set custom stuff under the Provisioning menu, do I even need provisioning enabled? |
| 06:24.54 | mosty | only if you want to use their provisioning |
| 06:24.59 | phix | It just isn't contacting my asterisk server, no errors are appearing in asterisk console, it doesn't pick up the call |
| 06:25.16 | phix | What is provisioning? :) |
| 06:25.31 | mosty | provisioning is automatic setup/firmware upgrade |
| 06:25.38 | phix | oh ok :) nah I don't want that shit |
| 06:26.25 | mosty | can you run a packet logger on your asterisk box to confirm that the phone isn't trying to contact asterisk? |
| 06:26.52 | phix | one step ahead of you :) |
| 06:27.44 | mosty | so you're sure that's not happening then? |
| 06:28.50 | phix | I will find out soon |
| 06:29.23 | phix | I was just confirming settings, going to try and dial it now :) |
| 06:31.25 | [TK]D-Fender | Well its checkout time. LAter all |
| 06:31.38 | [pyro] | [TK]D-Fender: This page http://www.voip-info.org/wiki/view/Asterisk+and+Aastra+Phones (section Directed Call Pickup) is what got me thinking BLF / Direct Call Pickup on the same key was just a patch away. Your the first person ive heard suggest setting up a 2nd set of extensions. |
| 06:33.26 | [pyro] | ah crap, didnt see he'd quit |
| 06:35.41 | phix | mosty: POTS Handset -> ATA -> asterisk -> internal SIP users or via VoIP provider works |
| 06:36.12 | phix | mosty: internal SIP users or via VoIP provider -> asterisk -> ATA -> POTS Handset works |
| 06:37.26 | phix | PSTN -> POTS Handset works, but PSTN -> asterisk -> internal SIP users (and back to ATA so it can ring POTS Handsets) does not work |
| 06:37.44 | phix | The ATA doesn't even try to contact asterisk, so I guess it is a dial plan problem |
| 06:37.52 | mosty | sounds like it |
| 06:37.53 | phix | (on ATA) |
| 06:38.04 | mosty | does the 3102 have a debug or log page? |
| 06:38.19 | phix | I will check out that forum and see what I did wrong |
| 06:38.46 | phix | it logs to syslog :) although I need to configure /etc/syslog.conf on my server first |
| 06:40.22 | phix | hmmmm, this is annoying also, everytime I dial a number on the POTS handset it re-registers with asterisk, causing a 2 sec or more delay in dialing :/ |
| 06:41.13 | JT | that sounds silly |
| 06:41.22 | JT | sure it's not an ata dialplan issue? |
| 06:48.10 | *** join/#asterisk shtoom (n=shtoom@59.93.120.163) |
| 06:56.20 | phix | JT: that is what I said |
| 06:57.01 | JT | phix: you sure the delay is due to registration then? |
| 06:58.31 | phix | hmmmm, sip debug tells me it registers at every call, hmmm but come to think of it it just sits there for a bit first with no debugg messages appearing |
| 06:58.43 | phix | so yes that does sound dumb :) |
| 07:03.06 | *** join/#asterisk dominic1 (n=dob@213.221.82.242) |
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| 07:07.18 | marc7 | if i'm trying to compile zaptel and it's telling me I'm missing the installed kernel source, is there any way to have it look in an alternate directory? |
| 07:08.46 | marc7 | ah, i see.. make KSRC=<dir>, thanks |
| 07:11.05 | phix | hmmm |
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| 07:27.43 | Zefk | Hi, does anyone know if asterisk can be interconnected with skype with opensource software ? thx |
| 07:27.56 | Strom_M | no. |
| 07:28.16 | Strom_M | all the available solutions are complete kludges anyway |
| 07:28.21 | Strom_M | don't waste your time :) |
| 07:28.59 | Zefk | Strom_M: but any projects in progress or this way is closed for the moment ? |
| 07:29.30 | marc7 | Zefk: Skype just doesn't want to play along, and they go to great lengths to make sure things aren't interoperable. |
| 07:29.30 | BBHoss | skype for asterisk is like cutting off your nose to spite your face! |
| 07:29.46 | Strom_M | Zefk: skype is closed proprietary crap. I wouldn't hold your breath. |
| 07:31.23 | Zefk | I believe you, my problem is that there are a lot of requests from skype users to access SIP services. |
| 07:31.50 | Zefk | I supposed that these users are addicted to Skype. |
| 07:32.07 | marc7 | the problem is that there are really no other awesome SIP softphones out there |
| 07:32.40 | marc7 | skype invests a lot of money in making sure the user interface is easy |
| 07:33.35 | Zefk | I see ...so the only solution for the moment is to switch to x-lite. :) |
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| 08:03.45 | FlatFoot | morning all |
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| 08:11.32 | BeeBuu | how can i record my .gsm files? |
| 08:12.52 | Strom_M | that makes no sense |
| 08:13.00 | Strom_M | .gsm files are, by definition, recordings |
| 08:13.43 | *** join/#asterisk lemanal (n=lemanal@228.sub-75-208-213.myvzw.com) |
| 08:14.55 | BeeBuu | i want to play my voice :-P |
| 08:15.38 | Strom_M | so...you want to make your own recordings |
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| 08:19.55 | *** join/#asterisk J4zen (n=Jvan4zen@a213-84-139-244.adsl.xs4all.nl) |
| 08:21.13 | J4zen | Does anyone know any freeware(-ish, small fee?) to convert WAV to GSM files? Preferably in batch? |
| 08:21.33 | J4zen | Windows based |
| 08:22.09 | *** join/#asterisk _ys (n=yuri@236-069.nat.mns.ru) |
| 08:22.31 | Strom_M | why would you want to convert them /to/ gsm? |
| 08:22.35 | Strom_M | asterisk can read wav files... |
| 08:23.08 | Strom_M | 8khz 16-bit mono pcm |
| 08:23.15 | tzafrir | J4zen, sox should be able to build on cygwin, I guess :-) |
| 08:23.27 | Strom_M | indeed |
| 08:24.14 | J4zen | i see, are there any steps needed in order to make asterisk read the wav files? |
| 08:24.14 | tzafrir | well, you may want to down-sample wav files with higher quality |
| 08:24.28 | tzafrir | file file.wav |
| 08:24.33 | tzafrir | what is the output? |
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| 08:24.45 | J4zen | i can't test it atm, the server is being transferred to our datacenter |
| 08:25.41 | Strom_M | J4zen: nope |
| 08:25.45 | Strom_M | no steps necessary |
| 08:26.08 | J4zen | alright, thanks Strom_M and tzafrir :) |
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| 08:31.01 | marc7 | hey guys, I'm having a real mess of a time with zaptel, I've compiled it on Debian etch (4.0) after I've given it my kernel sources and all... but "make install" doesn't seem to be putting the modules anywhere.... so nothing's getting sorted out on boot. I can `insmod zaptel.ko` directly, but that's about it. suggestions? |
| 08:33.13 | *** join/#asterisk techie (n=techie@adsl-76-214-20-56.dsl.lsan03.sbcglobal.net) |
| 08:37.05 | tzafrir | marc7, for starters ./install_prereq test |
| 08:37.35 | tzafrir | hmm... actually... |
| 08:37.47 | marc7 | tzafrir: just found out that our kernel compile dosen't have USB support, and so the make install-modules process which tries to get xpp_usb.ko going causes the whole thing to bail out |
| 08:37.49 | tzafrir | do you get any output from: modinfo zaptel ? |
| 08:38.04 | tzafrir | marc7, a custom kernel? |
| 08:38.05 | *** join/#asterisk parag00n (n=parag0n@87-194-9-117.bethere.co.uk) |
| 08:38.42 | marc7 | tzafrir: yeah. and while I didn't get any love from `modinfo zaptel` a moment ago because the make install process was failing, it's certainly working now |
| 08:39.02 | *** join/#asterisk denon (n=denon@tooth.decay.org) |
| 08:39.02 | *** mode/#asterisk [+o denon] by ChanServ |
| 08:39.03 | tzafrir | hmm... right.... |
| 08:39.22 | tzafrir | workaround: disable support for xpp |
| 08:39.26 | tzafrir | in menuselect |
| 08:39.35 | *** join/#asterisk Pazzo (n=ugelt@reserved-225136.rol.raiffeisen.net) |
| 08:39.40 | tzafrir | Though this should have been done automatically |
| 08:39.51 | tzafrir | :-( |
| 08:40.11 | marc7 | oh nice, hmm... if I only needed ztdummy, could I disable practically everything else here? |
| 08:40.30 | tzafrir | yes, anything besides ztdummy and zaptel |
| 08:40.58 | tzafrir | and of the utilities you only actually need zttest |
| 08:41.02 | *** join/#asterisk rodent|m (i=nobody@foster.stonedcoder.org) |
| 08:41.25 | tzafrir | and just add ztdummy to /etc/modules |
| 08:42.00 | tzafrir | BTW: m-a a-i zaptel |
| 08:42.23 | marc7 | awesome. thanks tzafrir! yeah, i really should be using module-assistant... and I really shouldn't be rolling up my own kernel builds ;) |
| 08:42.55 | *** join/#asterisk harpal (n=Harpal@124.125.255.223) |
| 08:43.24 | tzafrir | though you'll probably need one extra symlink /usr/include/zaptel/zaptel.h >/usr/include/linux/zaptel.h to make asterisk feel OK with this |
| 08:43.54 | tzafrir | m-a should work fine with custom kernels |
| 08:44.24 | marc7 | should I be able to even use m-a from the zaptel directory i check out of subversion? |
| 08:44.45 | harpal | I have installed asterisk 1.4.13. now how to test that? |
| 08:45.12 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 08:46.51 | marc7 | ah, silly question. yeah... it's definitely pulling zaptel-source from apt |
| 08:48.37 | *** join/#asterisk qdk (n=qdk@85.235.253.139) |
| 08:49.09 | marc7 | tzafrir: what I really need is a way to roll-up asterisk builds I've put together on one server over to a different box entirely. I don't think there's any elegant solution I can think of to debianize the process, as `make install` does a lot of last minute wget / installs that I'm having trouble sorting out |
| 08:50.39 | tzafrir | marc7, use a repo from starters :-( . anyway, there are several make-install wrappers, but you have to use them when you run 'make install' |
| 08:52.50 | marc7 | sorry, I misunderstand... when you say use a repo, you're suggesting I install asterisk from debian's repository? |
| 08:54.01 | *** join/#asterisk stimpie (n=stimpie@84-104-5-115.cable.quicknet.nl) |
| 08:54.54 | *** part/#asterisk agx (n=AGX@host63-216-static.34-88-b.business.telecomitalia.it) |
| 08:55.08 | tzafrir | marc7, I always suggest that. I must say most others in this channel disagree with me |
| 08:55.21 | tzafrir | Indeed the version of Asterisk in Etch is a bit date |
| 08:55.36 | tzafrir | dated |
| 08:55.56 | marc7 | not thrilled about that |
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| 09:08.37 | PBX | hi |
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| 09:33.13 | Jakobsen | I have some trouble with my asterisk server, but I don't know how to troubleshoot it. The service stopped a few days ago, and I started it again. Now some of the users are disconnected when they've been in a call for a few minutes |
| 09:37.47 | BBHoss | what 'service' stopped? asterisk? |
| 09:37.52 | Jakobsen | yes |
| 09:38.01 | Jakobsen | daemon ;) |
| 09:38.08 | BBHoss | heh |
| 09:38.19 | BBHoss | have you tried rebooting? |
| 09:39.00 | Jakobsen | Not yet, I want to troubleshoot the problem before rebooting the server |
| 09:39.09 | BBHoss | reboot the server |
| 09:39.23 | BBHoss | if that doesn't fix it, then we can try to troubleshoot |
| 09:39.41 | BBHoss | are you using telephony hardware, or pure sip/iax? |
| 09:39.48 | Jakobsen | Pure sip |
| 09:39.56 | BBHoss | hmm |
| 09:40.15 | BBHoss | are you using a control panel? |
| 09:40.25 | Jakobsen | it has 100+ users, if I can avoid rebooting, it would be nice.. |
| 09:40.34 | BBHoss | yeah i understand |
| 09:40.40 | Jakobsen | No, just managing it over SSH |
| 09:40.49 | BBHoss | ok kool |
| 09:41.00 | Jakobsen | but again; any log files that would tell me why the calls are stopped? |
| 09:41.03 | BBHoss | nothing has changed? |
| 09:41.06 | Jakobsen | No |
| 09:41.15 | BBHoss | depends on setup |
| 09:41.27 | BBHoss | but usually when it just crashes it wont tell you why |
| 09:41.33 | BBHoss | :) |
| 09:41.33 | Jakobsen | That's nice.. |
| 09:41.39 | BBHoss | unless you feel like running it through gdb |
| 09:41.45 | BBHoss | lemme check... |
| 09:41.47 | Jakobsen | I don't |
| 09:42.23 | BBHoss | you can try looking in /var/log/asterisk |
| 09:43.20 | BBHoss | are you using the ztdummy driver, or ANY zaptel? |
| 09:43.38 | Jakobsen | I actually don't know |
| 09:44.05 | BBHoss | see if you have a /dev/zap |
| 09:44.20 | Jakobsen | I do |
| 09:44.33 | BBHoss | hmm |
| 09:44.52 | BBHoss | are you using meetme or any other channel bridging (ie conference) |
| 09:45.20 | tzafrir | Jakobsen, please provide logs for the relevant parts |
| 09:45.53 | tzafrir | Do you see any errors in the logs for such a disconnect? |
| 09:46.08 | Jakobsen | The server was set up by another guy, I myself have never worked with asterisk before.. |
| 09:46.29 | BBHoss | heh |
| 09:46.46 | tzafrir | for starters, /var/log/asterisk/messagtes , or full is the most common log file . See /etc/asterisk/logger.conf |
| 09:46.54 | tzafrir | or 'logger show channels' |
| 09:46.57 | tzafrir | in the CLI |
| 09:46.59 | Jakobsen | That's why I was asking about logfiles - I don't have any idea what I'm looking for :/ |
| 09:47.28 | tzafrir | I asked if you see any errors that happen at a time of such a disconnect |
| 09:47.32 | BBHoss | also you may want to increase the debug levels |
| 09:47.41 | tzafrir | If not, please pastebin a relevant part of the logs |
| 09:47.59 | Jakobsen | I'm looking at a running "asterisk -rvvvvv" |
| 09:48.09 | BBHoss | yeah |
| 09:48.13 | BBHoss | thats verbosity |
| 09:48.22 | BBHoss | debug is even MORE verbose :) |
| 09:48.28 | BBHoss | to an insane level |
| 09:48.39 | BBHoss | are you using 1.4 or 1.2? |
| 09:49.33 | Jakobsen | Is this good? --> "Spawn extension (macro-CallLocalSubscriber, s, 5) exited non-zero" |
| 09:49.36 | *** join/#asterisk arcanine (n=saxon_m2@203.82.44.179) |
| 09:49.50 | BBHoss | thats the error? |
| 09:50.31 | Jakobsen | no, that's just a line I see a lot in the CLI.. |
| 09:50.44 | BBHoss | thats nothing |
| 09:50.49 | Jakobsen | the error is, that users are randomly disconnected after a few minutes - but not every time.. |
| 09:50.56 | BBHoss | do you have documented times of when this happens |
| 09:51.05 | Jakobsen | no, it happens all the time |
| 09:51.20 | BBHoss | it sounds to me that something has gotten fishy with zaptel |
| 09:51.35 | BBHoss | and its throwing a wrench in other things |
| 09:51.46 | BBHoss | either that or a component is going bad in the server |
| 09:51.57 | BBHoss | if you're SURE nothing on the software side has changed |
| 09:52.00 | Jakobsen | Maybe I should restart then.. |
| 09:52.21 | Jakobsen | No changes were made, the only person to change it is me, and I didn't :) |
| 09:52.40 | BBHoss | i've seen some people restarting their boxes every week or two, sometimes more |
| 09:52.58 | Jakobsen | okay, this have been running for months now |
| 09:53.07 | Jakobsen | Asterisk 1.2.5 |
| 09:53.07 | tzafrir | Jakobsen, you may not know what to make of it. Others here might |
| 09:53.17 | tzafrir | jbot, tell Jakobsen about pasebin |
| 09:53.25 | tzafrir | jbot, tell Jakobsen about pastebin |
| 09:53.55 | Jakobsen | tzafrir, I know what pastebin is, but I don't know what I should paste for you! |
| 09:54.23 | BBHoss | /var/log/asterisk/messages |
| 09:54.26 | tzafrir | relevant parts (by the wall clock) of the logs, or parts from the CLI |
| 09:54.38 | BBHoss | post /var/log/asterisk/debug if their is one |
| 09:54.49 | tzafrir | to show what happened when "a call has disconnected" |
| 09:55.23 | tzafrir | Jakobsen, a basic fact you also seem to have left out: what type of call is it? |
| 09:55.33 | BBHoss | sip apparently |
| 09:55.51 | BBHoss | you are using a SIP trunk from an ITSP i assume? |
| 09:55.57 | *** join/#asterisk IgorG (n=FeedomPa@195.162.32.126) |
| 09:56.11 | *** join/#asterisk Jakobsen (n=chatzill@gandalf.dansupport.dk) |
| 09:56.12 | Jakobsen | whoops.. |
| 09:56.29 | tzafrir | Jakobsen, a basic fact you also seem to have left out: what type of call is it? |
| 09:56.36 | BBHoss | try zap show channels |
| 09:56.43 | BBHoss | whats the output of that |
| 09:57.02 | Jakobsen | pseudo, default |
| 09:57.22 | Jakobsen | Extension: pseudo, Context: default... |
| 09:57.29 | BBHoss | ok |
| 09:57.34 | BBHoss | so just ztdummy |
| 09:57.53 | BBHoss | have you tried restarting asterisk? |
| 09:58.36 | BBHoss | best thing i can tell you is have the users write down a time when their call gets dropped, then go back and check the logs |
| 09:58.52 | BBHoss | until you know the times, you are looking for a needle in a haystack |
| 09:59.33 | Jakobsen | messages look bad..... |
| 09:59.35 | Jakobsen | http://pastebin.com/d5cf0893e |
| 09:59.50 | BBHoss | network troubles |
| 10:00.16 | Jakobsen | yes, the provider has some trouble this weekend, but they should be fixed now.. |
| 10:00.33 | Jakobsen | Yesterday (11th of November) asterisk just stopped.. |
| 10:00.41 | BBHoss | crashed totally? |
| 10:00.48 | Jakobsen | yes |
| 10:01.01 | BBHoss | do you have a log of that? |
| 10:01.12 | Jakobsen | No, this is the only log I have |
| 10:01.29 | Jakobsen | Otherwise, you have to tell me where the log should be? :) |
| 10:01.34 | *** join/#asterisk ccesario (n=ccesario@189-19-9-100.dsl.telesp.net.br) |
| 10:01.43 | BBHoss | hmm |
| 10:02.05 | Jakobsen | The only file I can find, that looks like a log, is /var/log/asterisk/messages |
| 10:02.26 | Jakobsen | I have a log of calls in a MySQL database too |
| 10:02.54 | BBHoss | when did it crash on the 11th |
| 10:03.22 | BBHoss | around 1800? |
| 10:03.39 | Jakobsen | I started it again around 1800 |
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| 10:04.00 | BBHoss | hmm |
| 10:04.08 | BBHoss | there is nothing there indicating a crash |
| 10:04.35 | BBHoss | it looks like the provider to screwing up to me |
| 10:05.33 | Jakobsen | That's my conclusion too :) |
| 10:05.56 | BBHoss | is it a 2-way provider or outgoing only? |
| 10:06.10 | Jakobsen | outgoing only |
| 10:06.29 | BBHoss | you might try another, see if the problems cease |
| 10:07.01 | Jakobsen | Yeah.. They had some problems this weekend, so maybe they still have some problems they haven't discovered |
| 10:07.07 | BBHoss | also your version of * is a bit old |
| 10:07.50 | Jakobsen | I know, but haven't had the b*lls to upgrade.. |
| 10:08.01 | BBHoss | heh |
| 10:08.07 | BBHoss | i don't blame you dude |
| 10:09.00 | Jakobsen | I will try rebooting the server, then we'll see what happens.. But thank you for your time guys.. |
| 10:09.09 | BBHoss | sure |
| 10:09.28 | BBHoss | when you need help after the upgrade, just ask ;-) |
| 10:10.22 | Jakobsen | "I'll be back" :D |
| 10:10.24 | Jakobsen | See you |
| 10:21.34 | *** join/#asterisk hi365 (n=hi365@mail.pcgeula.co.il) |
| 10:21.48 | hi365 | is there any way to specify a tech with extenspy? |
| 10:30.59 | *** join/#asterisk dan__t (i=dan@neener.neener.org) |
| 10:31.06 | dan__t | 'morning. |
| 10:33.31 | dan__t | Just out of curiosity, does Polycom make an idiot-proof GUI config file generator for provisioning? What kind of goodies do they have, if any? |
| 10:33.58 | *** join/#asterisk marcan (i=1337@host214-205.cvd.fit.edu) |
| 10:33.58 | [pyro] | has anyone else got BLF and Directed Call Pickup working with Aastra 5Xi phones? |
| 10:34.07 | hi365 | dan__t: actuly, quite the opisite. but there goods are top stuff |
| 10:36.56 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
| 10:37.17 | dan__t | They get an A+ for complexity. |
| 10:41.12 | phix | hmmm, I am still having issues with Linksys SPA3102, I cannot get calls from PSTN line to contact asterisk. |
| 10:41.48 | phix | I have this as dial plan 1 (under PSTN lines in SPA3102 web config) (S0<: s@10.0.0.1 :5060>) |
| 10:42.21 | phix | the other dial plans are all (xx.) |
| 10:45.42 | phix | also, calls made from asterisk to PSTN line (via ATA) gives me phone number error from land line carier |
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| 10:49.51 | phix | Hello |
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| 11:23.05 | *** join/#asterisk Psychobilly (n=Fuzz@online2.ioa.forthnet.gr) |
| 11:24.13 | Psychobilly | hello, in my extensions.conf can i have variables per context and not only global ones? |
| 11:24.18 | Psychobilly | somehting like this: |
| 11:24.21 | Psychobilly | [foo] |
| 11:24.27 | Psychobilly | VAR1=bar |
| 11:27.11 | tzafrir | Psychobilly, no |
| 11:27.43 | Psychobilly | ok thx tzafrir |
| 11:28.11 | tzafrir | Psychobilly, consider using something like ${VAR_${CONTEXT}} , but I'm not sure if such double expansion is supported |
| 11:28.35 | Psychobilly | yes thats what im thinking about |
| 11:33.37 | Psychobilly | im searching an easy way to add speed dial feature for the users, my plan was to store the nymbers as variables, different for each user |
| 11:37.11 | tzafrir | Psychobilly, alternatively, the astdb could be used to store such data |
| 11:38.01 | Psychobilly | its running in a very small system with limited resources, i want to keep is as simple as possible |
| 11:38.15 | Psychobilly | avoid loading many modules etc |
| 11:40.42 | Strom_M | if you don't have enough resources to store speed dial information in astdb, you probably have far greater problems to worry about anyway :) |
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| 11:49.28 | phix | hi |
| 11:49.40 | Strom_M | hi |
| 11:49.53 | phix | Still having ATA issues |
| 11:49.59 | Strom_M | congratulations |
| 11:50.03 | phix | thnx |
| 11:50.16 | Strom_M | ywlcm |
| 11:50.20 | Strom_M | omg |
| 11:50.24 | phix | Do I get a prise? eg, some help |
| 11:50.26 | Strom_M | wht hpnd t m vwls |
| 11:50.45 | phix | Strom_M: wtf |
| 11:52.04 | phix | gimme dial plan for linksys SPA3102 |
| 11:52.19 | Strom_M | i've never configured one |
| 11:53.54 | phix | ok |
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| 12:05.04 | puzzled | hi |
| 12:06.47 | Bladerunner05 | does TDM400p receive a fax in tiff image? |
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| 12:17.05 | BillBinko | <PROTECTED> |
| 12:17.14 | BillBinko | <PROTECTED> |
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| 12:22.56 | hellop | Is it harmless to plug a live POE cable into a non-POE device? (Power Over Ethernet) |
| 12:23.36 | FlatFoot | hey ho all |
| 12:23.48 | PBX | :P |
| 12:24.30 | hellop | maybe I should just try it and see what happens.. |
| 12:24.55 | rob0 | Purity Of Essence = Peace On Earth. MajGen Jack D. Ripper, USAF, Burpelson AFB |
| 12:25.04 | FlatFoot | got a puzzeller using CDR getting full data recorded on sip to sip call but not on IAX2 ( v 1.4.11 ) what is bothering me is it's not recording my accountcode value . ANY ideas |
| 12:25.41 | hellop | rob0, good one rob0 |
| 12:27.29 | *** join/#asterisk coppice (n=chatzill@39.192.17.210.dyn.pacific.net.hk) |
| 12:28.00 | hellop | Anyone have any experience with extending the Powered part of a Polycom 501 phone? Any suggestied max length? |
| 12:28.35 | hellop | So, what I've read on the web, is that plugging a PoE cable into a non-PoE device "may or may-not damage the device". |
| 12:28.57 | hellop | So, I came here what you guys do about it. |
| 12:29.39 | hellop | Do you take any special steps to prevent people from plugging laptops into PoE wall-ports? |
| 12:29.43 | FlatFoot | hellop: do you mean the PoE network cable ? |
| 12:30.10 | hellop | FlatFoot, yes like on a Polycom 501, where the power brick plugs into the Ethernet cable. |
| 12:30.30 | FlatFoot | if so depending on the ampage of the PoE device , we normally would NOT use any more than 48 meters |
| 12:31.30 | dan__t | Hrm, ok, so I'm using AsteriskNOW, and I got inbound calls to work perfectly. I'm very happy about that. However, any attempt to dial out results in a fast tone beep. I don't see anything overly obvious in the logs, so I suspect a keymap on this Polycom phone that isn't exactly happy? |
| 12:35.38 | dan__t | nm, think I found it. |
| 12:36.03 | *** join/#asterisk tzafrir (n=tzafrir@62.90.10.53) |
| 12:38.02 | dan__t | Wasn't using the correct trunk. |
| 12:41.32 | *** join/#asterisk qdk (n=qdk@0x573fe57e.bynqu2.broadband.tele.dk) |
| 12:42.15 | dan__t | Ok. Yea, same stuff, I can't see anything painfully obvious in the sip debug. |
| 12:43.30 | dan__t | http://pastebin.ca/770593 - that's a sniplet of my sip debug, if it helps. |
| 12:43.54 | hellop | FlatFoot, thanks for the info about PoE cable length |
| 12:44.02 | *** join/#asterisk TheDude (n=thedude@217.54.96.68) |
| 12:44.21 | FlatFoot | hellop: np , we have powered up to 80meters but it stuggled |
| 12:44.30 | FlatFoot | * struglled |
| 12:45.26 | TheDude | Just upgraded to latest version & now execution no longer jumps to exten => n+101 when dialed party is busy. New feature or am I missing something? |
| 12:46.13 | hellop | FlatFoot, still 80m seems pretty far to send 300ma 9V DC |
| 12:46.45 | FlatFoot | hellop: .3A thats gonna make about 30m's max without having a burnout |
| 12:47.06 | hellop | FlatFoot, do you take any precautions to ensure people don't use the PoE ports? |
| 12:47.39 | FlatFoot | hellop: normally PoE ports are labled RED for us |
| 12:49.28 | hellop | Another question: For 4 VOIP phones, communicating to an Asterisk Server located 80ms away, do you need to buy an expensive managed router? |
| 12:49.43 | hellop | Or, will that work fine on a WRT54G $80 wallmart router? |
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| 12:50.13 | hellop | 80ms away I mean, pinging the Asterisk server is 80ms |
| 12:51.09 | *** part/#asterisk Sunmoon__ (n=Sunmoon@ip72-206-113-190.om.om.cox.net) |
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| 12:51.43 | luke-jr | hellop: 80ms is 80ms |
| 12:52.06 | luke-jr | and you haven't mentioned what you need a router for |
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| 12:53.13 | hi365 | how can you fo a if and if in gootoif? (i.e. gotoif(a=true and b=true) |
| 12:53.33 | *** part/#asterisk boodie__ (n=thedude@196.219.96.141) |
| 12:53.48 | rob0 | hellop: I think I'd try a $20 10/100 switch first. |
| 12:53.55 | luke-jr | hi365: … what? |
| 12:53.59 | dan__t | 'morning, rob0. |
| 12:54.07 | luke-jr | if (a=true & b=true) { … } |
| 12:54.22 | luke-jr | rob0: what's wrong with the $10 switch? |
| 12:55.06 | *** join/#asterisk lirakis (n=lirakis@65.200.191.253) |
| 12:55.06 | hellop | luke-jr, My neighbor's use Asterisk, with 2 Polycom 501s. They used to have an in-house IT guy that set-up Asterisk. He's gone, so they hired a consulting company. |
| 12:55.30 | FlatFoot | hello all is ${TIMESTAMP} still usable ?????? |
| 12:55.32 | hi365 | luke-jr: so how would you write this? exten => s,n,Gotoif($["${CALLERID(num)}" > 1000)]$["${CALLERID(num)}" < 2000)]?disa) |
| 12:55.33 | luke-jr | hellop: so what's a router for? |
| 12:55.34 | hellop | With the 2 phones, they where on a WRT54G router, and a simple 8 port netgear switch. |
| 12:56.19 | luke-jr | hi365: if ("${CALLERID(num)" > 1000 & "${CALLERID(num)}" < 2000) wtfisdisa; |
| 12:56.33 | luke-jr | FlatFoot: no |
| 12:56.39 | hellop | So, they are moving to 4 phones, and the consulting company wants them to buy a 24port Cisco Switch. But, isn't that un-needed? |
| 12:56.40 | luke-jr | hellop: a switch is fine |
| 12:56.44 | hi365 | luke-jr: gotchya, thanks |
| 12:57.03 | hellop | The CISCO switch does POE |
| 12:57.09 | luke-jr | PoE is convenient |
| 12:57.10 | luke-jr | BUT |
| 12:57.12 | FlatFoot | luke-jr: typical |
| 12:57.19 | luke-jr | Cisco switches don't do standard PoE |
| 12:57.27 | hellop | Reason given is: "give you a managed device, as well as giving |
| 12:57.27 | hellop | > you expansion capabilities. |
| 12:57.32 | luke-jr | might be better to find a company that understands standards |
| 12:57.51 | luke-jr | not to mention Cisco stuff being expensive and crappy |
| 12:57.51 | hellop | luke-jr, you don't think it will work on the Polycom's eh? |
| 12:58.03 | luke-jr | it would work on Cisco IP phones |
| 12:58.50 | hellop | Aside from non-compatible POE, is there ANY benefit from a managed switch with 4 phones on DSL? |
| 12:59.13 | *** join/#asterisk reber (n=reber@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr) |
| 13:01.26 | phix | hmmmmmmm |
| 13:01.40 | phix | is the dial plan syntax on SPA3102 the same as in asterisk |
| 13:01.42 | phix | ?? |
| 13:01.45 | FlatFoot | luke-jr: any idea what has replaced ${TIMESTAMP} ? can't seem to find the answer |
| 13:02.33 | luke-jr | I forget |
| 13:02.46 | FlatFoot | k ta anyway |
| 13:03.25 | hellop | Found this neat site with router speed comparisons: http://www.smallnetbuilder.com/component/option,com_chart/Itemid,189/chart,121/ |
| 13:04.07 | dan__t | So, I'm using TelIAX as an IAX2 provider. Works out pretty well. However, I'm trying to make sure caller ID works properly. I'm using AsteriskNOW, and I've set the Caller ID setting of the Advanced menu of the Service Provider tab, essentially I thought this is where it is set. |
| 13:04.40 | *** join/#asterisk juanjoc (n=juanjoc@200.69.219.113) |
| 13:06.18 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 13:09.17 | *** join/#asterisk af_ (n=getsmart@88-149-240-124.dynamic.ngi.it) |
| 13:11.04 | *** join/#asterisk saftsack (n=saftsack@pD9E078F0.dip.t-dialin.net) |
| 13:11.17 | *** part/#asterisk saftsack (n=saftsack@pD9E078F0.dip.t-dialin.net) |
| 13:12.15 | hellop | luke-jr, so what do you think bro, should I tell the company that the $4000 switch is a waste of money or, let them buy it? |
| 13:12.28 | luke-jr | definately a waste |
| 13:13.00 | phix | ? |
| 13:14.57 | Bladerunner05 | does TDM400p receive a fax in tiff image? |
| 13:16.43 | endre | Bladerunner05: it did for me |
| 13:20.29 | Bladerunner05 | •endre• lucky boy|||| I can't |
| 13:21.11 | Bladerunner05 | •endre• may U describe me extensions.conf exten to do that? |
| 13:21.48 | *** join/#asterisk CleanerX (n=nix@p5B13428A.dip0.t-ipconnect.de) |
| 13:23.18 | *** join/#asterisk saftsack (n=saftsack@pD9E078F0.dip.t-dialin.net) |
| 13:23.51 | [TK]D-Fender | Bladerunner05: Go lookup SpanDSP on the WIKI. This is 1 line of dialplan, but a whole module to compile in. |
| 13:24.46 | Bladerunner05 | <[TK]D-Fender>: thanks I'll do ti |
| 13:24.51 | *** join/#asterisk bantu (n=Miranda@rz-du-mvx-142-44.rz.uni-karlsruhe.de) |
| 13:26.39 | Chris-NB | hi |
| 13:27.02 | Chris-NB | I've a question about wanpipe/sangome |
| 13:27.10 | Chris-NB | sangoma |
| 13:27.23 | Chris-NB | are there two versions for asterisk 1.2 and asterisk 14? |
| 13:27.39 | Chris-NB | or is there one drivers working with both? |
| 13:28.00 | [TK]D-Fender | Chris-NB: I believe their drivers can deal with a variety of versions in a single release |
| 13:28.24 | Bladerunner05 | <[TK]D-Fender>: a lots of errors while making..... |
| 13:28.45 | Chris-NB | [TK]D-Fender, ok. You believe? But don't know exactly? |
| 13:29.04 | stimpie | I have several cdr's with context 'default' this context does not exist |
| 13:29.17 | stimpie | where do these cdr's come from? |
| 13:29.24 | [TK]D-Fender | Chris-NB: Grabbing the latest I've never had problems with either |
| 13:29.39 | Chris-NB | [TK]D-Fender, ok. I'll do that. Thanks! |
| 13:29.46 | codefreeze | stimpie: prob. because when the channel was formed, it stuck 'default' in there, and never overrode it. |
| 13:31.38 | *** join/#asterisk msetim (n=marcos@200.195.161.164) |
| 13:32.19 | msetim | hi |
| 13:32.19 | hi365 | when i dial *${EXTEN} the calls goes stright to voicemail. is that an asterisk thing (or a freepbx thing)? |
| 13:33.19 | *** join/#asterisk killfill (n=killfill@pc-164-134-45-190.cm.vtr.net) |
| 13:33.21 | killfill | hi! |
| 13:34.04 | [TK]D-Fender | hi365: Yes you are a complete schmuck and the dialplan does what you (or more like the GUI you sold your soul to) tells * to do. |
| 13:34.31 | killfill | i have alittle problem. Got a queue, with agents, and have to execute something (i.e. NoOP), while the agent's phone is ringing |
| 13:34.41 | [TK]D-Fender | hi365: :p |
| 13:34.42 | killfill | where should i look for this?.. |
| 13:34.48 | [TK]D-Fender | NEXT!@@!@ (c) BKW |
| 13:35.04 | killfill | [macro-stdexten] is not getting calld for queues.. :S |
| 13:35.12 | [TK]D-Fender | killfill: What makes them right now? |
| 13:35.33 | killfill | What makes them?.. what do you mean |
| 13:35.38 | [TK]D-Fender | "ring" |
| 13:35.50 | killfill | http://pastebin.ca/770625 |
| 13:35.53 | killfill | thats all i can see |
| 13:36.07 | killfill | aah 24@default:1 |
| 13:36.26 | killfill | so i should put my thing in [default] |
| 13:36.45 | [TK]D-Fender | killfill: Sort of says it all, doesn't it? |
| 13:36.50 | hi365 | [TK]D-Fender: see this: http://www.youtube.com/watch?v=TcrzC_T_XOs (if im a shmuck what are you ;-} ) |
| 13:36.57 | hi365 | (about 5 min in) |
| 13:38.23 | killfill | if i have things there like "exten = 90,1,Queue(${EXTEN})" i should replace all the '1' for '2' and write _XX!,1,NoOp(MyThinig) .. right?... |
| 13:38.41 | [TK]D-Fender | hi365: Never watched the show. |
| 13:39.05 | [TK]D-Fender | killfill: Just ask yourself WHEN that will get called.... |
| 13:39.10 | killfill | cannot i maybe include something or do a trick so i dont have to rewrite the proirity number? (and execute something before all) |
| 13:39.16 | [TK]D-Fender | killfill: These things are all extremely obvous.... |
| 13:39.20 | *** join/#asterisk zerohalo (n=zeroHalo@pool-71-255-162-167.bstnma.east.verizon.net) |
| 13:39.33 | killfill | hm.. |
| 13:39.39 | [TK]D-Fender | killfill: Yes, you have to deal with priorities, this is the DIALPLAN, it does stuff in ORDER. |
| 13:39.48 | [TK]D-Fender | obvious* |
| 13:40.32 | [TK]D-Fender | killfill: Things that happen as the agents are dialed will repeat, things before you go into the queue jsut once, etc... this isn't Raw Cat scient... |
| 13:40.47 | Bladerunner05 | making.... agx-ast-addons error for: app_rxfax.c: In function âphase_e_handlerâ: |
| 13:44.12 | J4zen | Hi there, i'm having a weird issue. I'm trying to have my SNOM320 register with my PBX located in the datacenter ( all ports opened ). |
| 13:44.13 | J4zen | http://pastebin.com/d6bd2c87 |
| 13:45.32 | [TK]D-Fender | J4zen: SIP/2.0 401 Unauthorized <-- user/pass is bad |
| 13:46.20 | J4zen | Yeah that's what i thought |
| 13:46.35 | J4zen | but theres just no way it can be wrong, i changed the password multiple times to make sure of that |
| 13:46.56 | J4zen | do the SNOM320's have some weird caching issue i need to work with? |
| 13:47.44 | *** join/#asterisk shido6_ (n=shido6@204.126.120.132) |
| 13:48.18 | [TK]D-Fender | J4zen: Nope. its wrong somewhere. * does not make this stuff up. |
| 13:49.13 | *** join/#asterisk bkw_ (n=brian@adsl-64-149-41-151.dsl.tul2ok.sbcglobal.net) |
| 13:49.32 | Bladerunner05 | making agx-ast-addons (app_rxfax) I got those errors: http://www.pastebin.ca/770643 |
| 13:49.37 | J4zen | I'll attempt a firemware update of the SNOM320, i am 100% sure the passwords are correct |
| 13:49.48 | J4zen | i have tried passwords such as 12345 even |
| 13:49.49 | Bladerunner05 | I install all required library and header files..... |
| 13:51.36 | [TK]D-Fender | J4zen: Maybe you've screwed up the * SIDE.... |
| 13:51.52 | [TK]D-Fender | Bladerunner05: Get googling. |
| 13:51.52 | J4zen | Well no, it works fine on my softphones |
| 13:51.58 | J4zen | even when using the same SIP account |
| 13:52.17 | J4zen | what is this 403 error then? |
| 13:52.25 | J4zen | Forbidden |
| 13:52.41 | Bladerunner05 | <[TK]D-Fender> I do it, but google return 1 entry only... |
| 13:52.59 | [TK]D-Fender | Bladerunner05: then clearly you aren't asking the right sort of question. |
| 13:56.50 | Chris-NB | has a Intel Xeon a Core 2 CPU? |
| 13:57.20 | [TK]D-Fender | Chris-NB: Yes there are models with that. |
| 13:57.47 | Chris-NB | [TK]D-Fender, how can I check that? Is it listed in /proc/cpu ? |
| 13:58.02 | [TK]D-Fender | Chris-NB: no idea. Try asking in ##linux |
| 13:58.13 | Chris-NB | [TK]D-Fender, ok, thanks |
| 14:05.21 | Bladerunner05 | <[TK]D-Fender> I know, pls help me to make the right question |
| 14:05.54 | [TK]D-Fender | Bladerunner05: http://www.google.ca/search?hl=en&q=rxfax+spandsp+%221.4%22&btnG=Google+Search&meta= |
| 14:06.58 | Bladerunner05 | <[TK]D-Fender> Thank you very much |
| 14:07.31 | *** join/#asterisk [intra]lanman (n=lanman@va-76-6-212-80.dhcp.embarqhsd.net) |
| 14:14.01 | *** join/#asterisk heison (n=heison@67.110.80.103.ptr.us.xo.net) |
| 14:14.17 | *** join/#asterisk cjk (n=loic@80.92.64.103) |
| 14:14.40 | cjk | hi, is there a function in cakephp to generate and do calculations with dates in mysql format? |
| 14:14.43 | *** join/#asterisk morge (n=mt@062016250212.customer.alfanett.no) |
| 14:14.48 | cjk | ups, wrong channel |
| 14:15.40 | *** join/#asterisk fskrotzki (n=fskrot@host198.textwise.com) |
| 14:16.01 | *** join/#asterisk agx (n=AGX@host63-216-static.34-88-b.business.telecomitalia.it) |
| 14:17.08 | morge | Hi. I have a sipura SPA-2102 that I got from my ipphone provider, and I would like to make it possible to call out trough my asterisk using that number. Is there a guide to how I should set up asterisk from a sipura config? I have passwords and such, and got it working using twinkle SIP client. |
| 14:17.51 | morge | I have added a trunk, but I get the "all circuits are busy now". I am using freepbx by the way. |
| 14:18.37 | morge | "sip show registry" tells me that it is registered. |
| 14:18.44 | morge | but I cannot use it. |
| 14:19.34 | morge | It seems to me that I probably need some special settings, but I am unable to find them, so I was hoping that the sipura config could tell me what I need. |
| 14:20.51 | *** join/#asterisk J4zen (n=Jvan4zen@a213-84-139-244.adsl.xs4all.nl) |
| 14:20.54 | *** join/#asterisk Law (n=Lawbring@cpc4-rdng12-0-0-cust292.winn.cable.ntl.com) |
| 14:23.02 | J4zen | I have isolated the problem, appearently my SNOM320's are having issues sending their packets to the PBX. Every now and then it'll recieve an "OK" / "Not authorised" / "403" response. What could be causing this? |
| 14:23.02 | J4zen | The PBX is located in a datacenter with all ports opened |
| 14:23.42 | *** join/#asterisk JulHer (n=julio@244.Red-217-125-14.staticIP.rima-tde.net) |
| 14:23.43 | [TK]D-Fender | morge: You're in the wrong channel then... |
| 14:23.46 | *** join/#asterisk Meaty (n=meaty3@office.abi.ca) |
| 14:24.02 | J4zen | the SNOM's are located in our office network behind a router, no filtering or firewall |
| 14:24.20 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 14:24.30 | [TK]D-Fender | J4zen: Last time : the error means what it says. 401 = your auth is bad. Not "maybe", it means "YES" |
| 14:25.52 | morge | [TK]D-Fender: Which channel should I go to? I am asking in the freepbx channel as well. |
| 14:25.58 | *** join/#asterisk metabsd (n=metabsd@modemcable103.201-131-66.mc.videotron.ca) |
| 14:26.31 | [TK]D-Fender | morge: Thats it. |
| 14:26.40 | J4zen | [TK]D-Fender ; The problem is.. if my memory serves me well, The SIP-phone will first send a request "REGISTER" .. once that has been accepted by the PBX it'll send another request "SUBSCRIBE" , the results seem totally random |
| 14:26.51 | J4zen | it doesn't simply reject my register or subscribe. it responds random |
| 14:26.55 | J4zen | once moment its a 401, then a 403 |
| 14:26.58 | J4zen | then it simply times out |
| 14:27.08 | J4zen | so no, i do not think this is an authentication issue |
| 14:27.12 | J4zen | i think this is a network problem |
| 14:28.09 | morge | [TK]D-Fender: Ok. Guess I just have to wait it out then, as I am getting no replies from there. |
| 14:28.16 | [TK]D-Fender | J4zen: Or you're phone is just psychotic. |
| 14:28.43 | [TK]D-Fender | morge: That, or get googling. |
| 14:29.01 | ai-a[afk] | J4zen: pastebin the snom log (level 9) and the pbx log please. |
| 14:29.03 | mvanbaak | hi all |
| 14:29.07 | J4zen | will do |
| 14:29.25 | mvanbaak | how can I capture all SIP traffic using tshark ? |
| 14:29.35 | ai-a | mvanbaak: i use tcpdump |
| 14:29.36 | mvanbaak | I just cant get it to do what I want |
| 14:29.45 | ai-a | then wireshark to debug it |
| 14:29.57 | mvanbaak | ok |
| 14:30.01 | ai-a | J4zen: do sip debug on the peer |
| 14:30.05 | mvanbaak | can tcpdump output to multiple files ? |
| 14:30.15 | mvanbaak | I want a dump of roughly 4 hours of traffic |
| 14:30.18 | *** join/#asterisk ManxPower (n=manxpowe@242.sub-75-203-181.myvzw.com) |
| 14:30.47 | ai-a | mvanbaak: nope, but wireshark can split up the sip conversations after. |
| 14:31.00 | J4zen | ai-a: http://pastebin.com/m67bd25c1 |
| 14:31.09 | J4zen | The peer is not connected, so i cant sip-debug it thru asterisk |
| 14:31.19 | mvanbaak | ai-a: care to share your tcpdump commandline to capture it ? |
| 14:31.25 | J4zen | the above log doesn't show it connects, but i assure you.. it does every now and then ( i cleared the log after it happened before ) |
| 14:31.28 | ai-a | J4zen: this is over a lan? |
| 14:31.32 | J4zen | this is over internet |
| 14:31.55 | J4zen | SIP-phones are in the office LAN ( no firewall/funky gateways ), the PBX is in a datacenter with all ports opened |
| 14:32.03 | J4zen | Also, my soft-phones register just fine |
| 14:32.14 | ai-a | mvanbaak: tcpdump -s2000 -w /var/tmp/output.pcap 'host xxx.xxx.xxx.xxx' |
| 14:32.22 | mvanbaak | ok, thanks |
| 14:32.40 | ai-a | J4zen: ok, and the pbx log of the SAME area in time,, with sip debug on. |
| 14:33.14 | ai-a | mvanbaak: then in wireshark use statistics -> VoIP Calls.. |
| 14:33.14 | J4zen | ai-a, i noticed a lot of action in my sip-phone log just now. let me update the pastebin for you |
| 14:33.26 | J4zen | http://pastebin.com/m5d11fc1 |
| 14:33.45 | ai-a | is the ip correct J4zen ? |
| 14:34.05 | J4zen | Yes |
| 14:34.10 | ai-a | your phone isnt getting any register response. |
| 14:34.20 | ai-a | you have some firewall on your router ? or your router doesnt like the nat ? |
| 14:35.01 | J4zen | No firewall, there is no filtering on any outgoing traffic |
| 14:35.07 | ai-a | J4zen: your asterisk pbx is directly on that ip ? |
| 14:35.15 | ai-a | incomming traffic. |
| 14:35.20 | J4zen | yes |
| 14:35.22 | ai-a | outgoing is going,, nothing coming in. |
| 14:35.36 | J4zen | yes my asterisk pbx is directly on the ip in the logfile |
| 14:35.37 | ai-a | J4zen: let me run my sip password cracker for a few minutes ;) |
| 14:35.47 | J4zen | hehe awesome <3 ;) |
| 14:35.53 | ai-a | J4zen: get your saterisk secure.. use OpenSer or something. |
| 14:36.08 | J4zen | its a test server, nothing more :) |
| 14:36.14 | ai-a | fine. |
| 14:36.23 | ai-a | well, show us the pbx log of the sip debug. |
| 14:36.49 | J4zen | I'm not sure i follow you, "SIP DEBUG PEER 104" ( 104 being the sip account im trying to register) ? |
| 14:37.11 | mvanbaak | ai-a: just to let you know: with tcpdump you can log to multiple files |
| 14:37.18 | ai-a | yep.. or sip debug ip <the ip of your pc at home> as it might not be sending valid register. |
| 14:37.21 | mvanbaak | with the -C you can specify the max filesize of a dumpfile |
| 14:37.24 | ai-a | mvanbaak: ok ;) |
| 14:38.04 | J4zen | Ok i enabled SIP debugging on my IP |
| 14:38.15 | ai-a | then re-register the identity on the snom. |
| 14:38.23 | ai-a | and pastebin the pbx. |
| 14:39.11 | *** join/#asterisk dijungal (n=kdaniel@209.59.110.35) |
| 14:39.43 | J4zen | Link in your PM :) |
| 14:40.23 | *** join/#asterisk GromiTM (n=palic@outer-core.ifg.uni-kiel.de) |
| 14:40.32 | dijungal | how do i make ztdummy and zaptel to load automatically when i restart the server? |
| 14:40.57 | ai-a | J4zen: Retransmitting #1 (NAT) to ... and so on. |
| 14:41.04 | ai-a | its failing to get to your ip. something is blocking it. |
| 14:41.31 | ai-a | what is the iptables set like on the pbx server ? |
| 14:41.33 | J4zen | Would that be on my incoming end, or the datacenters outgoign end? |
| 14:41.38 | [TK]D-Fender | dijungal: You should ahve a zaptel init script in your startup process |
| 14:41.51 | ai-a | either your bpx cant send out over the net, or your router wont let your phone recieve over the net. |
| 14:42.19 | J4zen | Do i need to set up any port forwarding? |
| 14:42.29 | GromiTM | Hi, I have some strange behavior in callqueue configuration. Sometime, under some circumstances I did not see at all, is it possible, that the caller in the at the first position never gets an agents while other callers in the queue after him get agents. |
| 14:42.36 | ai-a | you need to allow outgoing of sip / rtp ports on your pbx. |
| 14:42.40 | [TK]D-Fender | J4zen: describe the full path between * and your phone <- |
| 14:43.03 | GromiTM | Does anyone see the same behavior under debian-lenny (asterisk 1.4.11) |
| 14:43.05 | GromiTM | ? |
| 14:43.30 | J4zen | SNOM > ROUTER+MODEM > INTERNET > OPEN GATEWAY AT DATACENTER > PBX |
| 14:43.34 | ai-a | J4zen: from your pastebin's your phone is sending, but not getting anything, and your pbx is receiving and trying to send. |
| 14:43.45 | [TK]D-Fender | J4zen: Read up : |
| 14:43.47 | [TK]D-Fender | ~sipnat |
| 14:43.50 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 14:43.52 | [TK]D-Fender | ^^^^^^^^^^^^^^^^6 |
| 14:44.05 | ai-a | J4zen |
| 14:44.10 | J4zen | so its a NAT translations error? |
| 14:44.29 | [TK]D-Fender | J4zen: looks like thats 1 part of your problem. |
| 14:44.42 | [TK]D-Fender | J4zen: So lets tackle that first |
| 14:44.42 | ai-a | J4zen: cehck your iptables on the pbx, is it allowing outgoing udp on the sip/rtp ports ? |
| 14:44.50 | J4zen | I'll read through those documents, Thanks |
| 14:44.50 | J4zen | let me check |
| 14:45.45 | dijungal | [TK]D-Fender: No zaptel init scripts |
| 14:45.55 | J4zen | ai-a; By the way, if it wouldnt allow outgoing UDP on rtp ports.. would it still work if it was placed in my local LAN? |
| 14:46.04 | dijungal | do i need to do a make config in the zaptel source dir? |
| 14:46.14 | J4zen | cause up until yesterday the PBX was located in our office LAN, and communicated fine with my SNOM's |
| 14:46.36 | ai-a | J4zen: iptables-save will tell you the rules for the firewall. |
| 14:46.47 | [TK]D-Fender | "iptables --list" <---- |
| 14:47.15 | J4zen | all on ACCEPT |
| 14:47.19 | J4zen | input, forward and output |
| 14:47.47 | *** join/#asterisk lemanal (n=lemanal@wifi-224-227.sc07.org) |
| 14:48.16 | ai-a | fender, hmm ;), never used --list HEh |
| 14:53.45 | *** join/#asterisk shinao1 (n=shinao1@217.20.242.50) |
| 14:55.53 | *** join/#asterisk tobias (n=tobias@cpe-069-134-226-227.nc.res.rr.com) |
| 14:57.16 | *** join/#asterisk BadPacket (n=John@unaffiliated/badpacket) |
| 15:00.31 | *** part/#asterisk [intra]lanman (n=lanman@va-76-6-212-80.dhcp.embarqhsd.net) |
| 15:04.15 | *** join/#asterisk teknoprep (n=teknomeg@74.94.55.101) |
| 15:04.17 | teknoprep | hey all |
| 15:04.21 | teknoprep | does anyone have any suggestions |
| 15:04.33 | teknoprep | for random breakup in calls with voip |
| 15:04.48 | *** join/#asterisk geek_cl (n=lletelie@200.75.18.211) |
| 15:04.50 | teknoprep | i have QoS with pfsense... sip outbound to a voip provider |
| 15:04.59 | teknoprep | most calls are perfect but some tend to break up |
| 15:05.19 | ManxPower | teknoprep: You can't really do QoS unless it is setup on BOTH ends of the link. |
| 15:05.24 | geek_cl | i can't compile cdr_odbc to write -> MS SQL |
| 15:05.25 | [TK]D-Fender | teknoprep: QoS no longer exists once it hits the public internet. |
| 15:05.32 | geek_cl | i can't compile cdr_odbc to write -> MS SQL |
| 15:05.37 | bkw_ | geek_cl: STOP IT |
| 15:05.38 | bkw_ | you dork |
| 15:05.42 | bkw_ | ask once |
| 15:05.55 | dijungal | yea.. someone might beat u over the head with the a zaptel stick! |
| 15:05.55 | geek_cl | ok... bkw_ |
| 15:05.56 | bkw_ | and only once.. then wait.. you might also want to paste bin why it won't compile and paste the link |
| 15:05.57 | geek_cl | sorry |
| 15:06.08 | bkw_ | no you're not sorry.. you just are impatient |
| 15:06.35 | geek_cl | lol...im stupid i know |
| 15:06.46 | bkw_ | I never said that.. now paste bin the info |
| 15:06.48 | bkw_ | so we can help you |
| 15:07.07 | geek_cl | Ok thanks |
| 15:07.07 | [TK]D-Fender | NEXT!@!@ (c) BKW |
| 15:07.23 | jameswf | being beat over the head with a zaptel stick sounds dirty |
| 15:07.24 | ManxPower | geek_cl: not many people use ODBC and MySQL with Asterisk. So you will have a harder time finding answers. If you are so impatient that you have to flood the channel, perhaps you should drop asterisk and use a commercial solution. |
| 15:07.52 | bkw_ | geek_cl: I wrote cdr_odbc :P |
| 15:07.54 | ManxPower | we are here for free. If you don't like the rules of the channel then you can go find a commercial consultant. |
| 15:08.07 | bkw_ | geek_cl: so i'm not really the person to piss off |
| 15:08.09 | geek_cl | ManxPower is not MySQL its, "MS SQL" |
| 15:08.27 | ManxPower | geek_cl: MS SQL? That's too kinky even for me. |
| 15:08.55 | ManxPower | geek_cl: your mailing list searches were not helpful? |
| 15:08.59 | ManxPower | ~mailinglist |
| 15:09.00 | jbot | Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search. Browse the mailing list archive at http://lists.digium.com/ or search through it at http://www.asteriskguru.com/archives, or and there is also the Macintosh Asterisk mailing list at http://www.astmasters.net/maml.htmm |
| 15:09.05 | *** join/#asterisk J4zen (n=Jvan4zen@a213-84-139-244.adsl.xs4all.nl) |
| 15:09.28 | bkw_ | MSSQL can whip the piss out of MySQL |
| 15:09.31 | *** join/#asterisk debiano777 (n=nana@213-140-19-123.fastres.net) |
| 15:09.37 | geek_cl | http://pastebin.com/d1f98403c |
| 15:09.40 | J4zen | Back, got disconnected for some reason; Did you leave any messages? |
| 15:09.42 | geek_cl | please check |
| 15:10.05 | *** join/#asterisk kombi (n=kombi@port-213-160-14-18.static.km-it.de) |
| 15:10.09 | ManxPower | bkw_: and hitler whipped the piss out of the french. That doesn't mean hitler was good. |
| 15:10.12 | bkw_ | thats app_odbc dork |
| 15:10.31 | J4zen | [TK]D-Fender; Could you paste those two links about NAT/PBX ( The first one i read wasn't right on; The pbx isn't behind a NAT, the SIP-phones are) |
| 15:10.32 | bkw_ | geek_cl: someone didn't update app_dbodbc for use with 1.4 |
| 15:10.32 | kombi | how do kick a caller in a meetme conference from command line? |
| 15:10.35 | bkw_ | post a bug report |
| 15:10.38 | bkw_ | kombi: stop now |
| 15:10.42 | J4zen | and the SIP phones are setup exactly like the config in the example |
| 15:10.59 | geek_cl | bkw_ its asterisk-1.4.11 version |
| 15:11.05 | bkw_ | well open a bug |
| 15:11.06 | kombi | thnks bkw_! stop now [phone #], correct? |
| 15:11.15 | bkw_ | kombi: no just stop now |
| 15:11.17 | [TK]D-Fender | J4zen: the first accounts for BOTH, you only ahve to follow the parts that apply |
| 15:11.20 | ManxPower | J4zen: What is the issue you are having? |
| 15:11.28 | bkw_ | geek_cl: remove app_dbodbc from the list of apps to compile you don't need it |
| 15:11.33 | bkw_ | infact that is based on my code too |
| 15:11.44 | kombi | bkw_: ok, but to kick one caller and keep the others? |
| 15:11.45 | bkw_ | geek_cl: if you're not a coder your using the wrong software |
| 15:11.53 | Qwell | app_dbodbc? |
| 15:11.54 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 15:11.54 | *** mode/#asterisk [+o anthm] by ChanServ |
| 15:11.58 | bkw_ | kombi: no.. it stops asterisk .. why not um read the docs :P |
| 15:11.59 | J4zen | [TK]D-Fender : I have :) My settings are correct, NAT=yes and WAN ip set correctly |
| 15:12.06 | bkw_ | Qwell: app_dbodbc in 1.4 is broken |
| 15:12.13 | Qwell | it isn't in 1.4 |
| 15:12.21 | [TK]D-Fender | J4zen: PASTEBIN is your friend |
| 15:12.51 | Qwell | bkw_: that isn't in-tree |
| 15:13.07 | bkw_ | oh that could be why |
| 15:13.10 | Qwell | :p |
| 15:13.20 | debiano777 | any news about asterisk 1.6? |
| 15:13.21 | Qwell | bkw_: isn't that something you host? |
| 15:13.27 | bkw_ | Qwell: nope |
| 15:13.37 | bkw_ | its based on my code |
| 15:13.43 | bkw_ | but I don't know where he got it from |
| 15:13.44 | Qwell | no idea where it comes from then |
| 15:13.58 | kombi | bkw_: sigh.. |
| 15:14.02 | ManxPower | debiano777: other than the fact there is no 1.6 yet? |
| 15:14.07 | Qwell | geek_cl: use func_odbc |
| 15:14.14 | Qwell | unless the db means astdb... |
| 15:14.24 | bkw_ | Qwell: its an astdb like interface for ODBC |
| 15:14.34 | geek_cl | mm Ok Qwell |
| 15:14.56 | Qwell | speaking of astdb... |
| 15:15.01 | kombi | bkw_: how do I stop asterisk from cli? (trick question;) |
| 15:15.07 | bkw_ | "stop now" |
| 15:15.38 | kombi | bkw_: you're not behaving logically, you should have answered my first question now |
| 15:15.50 | kombi | you must be human.. |
| 15:16.20 | bkw_ | kombi: if you are at this stage and you don't know how to work the software.. why not type HELP |
| 15:16.27 | bkw_ | and start digging thru the meetme command line options |
| 15:16.30 | bkw_ | that could help |
| 15:16.38 | J4zen | [TK]D-Fender: http://pastebin.com/me5433d |
| 15:16.43 | J4zen | My SIP.conf |
| 15:17.12 | bkw_ | switch-01*CLI> help meetme |
| 15:17.13 | bkw_ | Usage: meetme (un)lock|(un)mute|kick|list <confno> <usernumber> |
| 15:17.13 | bkw_ | <PROTECTED> |
| 15:17.25 | kombi | bkw_: I know those inside out, you didn't understand me, that's all |
| 15:17.25 | bkw_ | kombi: lets see does that help you? |
| 15:17.35 | bkw_ | kombi: I totally understood you |
| 15:17.41 | bkw_ | you wanna kick someone out of a meetme conference |
| 15:17.50 | kombi | right! |
| 15:18.01 | bkw_ | the bigger question is why is it so hard to understand such a simple option? |
| 15:18.06 | bkw_ | help meetme |
| 15:18.07 | bkw_ | at the cli |
| 15:18.09 | bkw_ | should explain this |
| 15:18.13 | debiano777 | <ManxPower> i read something about 1.6 in artcle but i s'nt understand when the release is out |
| 15:18.38 | *** join/#asterisk shinao1 (n=shinao1@217.20.242.50) |
| 15:18.41 | ManxPower | debiano777: Asterisk does not have specific dates for releases. They happen when the developers thinks it's ready. |
| 15:18.55 | debiano777 | ok thanks |
| 15:19.26 | *** join/#asterisk gr0mit (n=tim@dhcp4.zuk40.mot-tools.co.uk) |
| 15:19.35 | kombi | bkw_: why did you give a wrong answer then, I even believed you |
| 15:21.55 | J4zen | ai-a: Is this http://pastebin.com/me5433d correct for my SIP.conf ? |
| 15:22.01 | ManxPower | J4zen: your localnet= and externip= settings do NOT make sense. |
| 15:22.10 | J4zen | They don't? |
| 15:22.12 | ManxPower | localnet should be your INTERNAL NATTED NETWORK |
| 15:22.27 | J4zen | Well there is no internal network lol |
| 15:22.33 | ManxPower | Also you seem to have some form of GUI installed. |
| 15:22.38 | J4zen | its located in a datacenter |
| 15:22.48 | J4zen | its connected directly to its outside adress |
| 15:22.51 | ManxPower | J4zen: if there is no internal network, then you have no NAT involved, do you? |
| 15:23.13 | J4zen | Nope, i read a document last week or so stating this would be nesecary? |
| 15:23.14 | ManxPower | localnet and externip are for when ASTERISK is behind NAT. |
| 15:23.18 | J4zen | I see |
| 15:23.39 | ManxPower | when the SIP clients are behind nat, then just nat=yes for that sip.conf entry is what is required. |
| 15:24.25 | J4zen | Alright, i removed the references localnet and externip |
| 15:24.26 | J4zen | lets see if that helps :) |
| 15:24.57 | ManxPower | J4zen: chances are you have other issues, but at least we have removed ONE of the things that might cause you issues. |
| 15:25.20 | J4zen | True indeed. Thanks |
| 15:26.27 | [TK]D-Fender | J4zen: waht model of router is taht phone behind? |
| 15:26.32 | J4zen | Still not getting any replies from the server |
| 15:26.38 | J4zen | Well its actually a modem/router combination |
| 15:26.40 | J4zen | alcatel modem |
| 15:26.43 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 15:26.59 | J4zen | one of those simple 4 port boxes |
| 15:27.08 | J4zen | attached to a 24 port switch though |
| 15:27.24 | *** join/#asterisk flujan_ (n=flujan@200-160-115-020.static.spo.ctbc.com.br) |
| 15:27.35 | J4zen | Well, i think i'll leave it at this for today |
| 15:27.45 | J4zen | i'll be back tommorow :) Thanks for all the assistance |
| 15:27.55 | J4zen | Bye |
| 15:28.02 | flujan_ | hi all. |
| 15:28.06 | ManxPower | poor guy |
| 15:28.14 | flujan_ | guys, I am trying to originate two simultaneos calls using the ami. |
| 15:28.48 | flujan_ | I am connected to asterisk using a socket and I am writing the commands to it... Asterisk dials to the first originate call but not to the second one... |
| 15:29.13 | *** join/#asterisk mindCrime (n=chatzill@66.83.208.218.nw.nuvox.net) |
| 15:34.12 | *** join/#asterisk jmacz (n=jmacz@190.25.35.244) |
| 15:36.44 | *** part/#asterisk agx (n=AGX@host63-216-static.34-88-b.business.telecomitalia.it) |
| 15:39.30 | heison | morning... i'm having problem with chan_sip.c after a recent upgrade to latest 1.4 via SVN, I'm now getting -- [Nov 12 10:39:00] NOTICE[3613]: chan_sip.c:7349 sip_reg_timeout: -- Registration for 'heison@208.64.200.100' timed out, trying again (Attempt #25) |
| 15:40.12 | twisted | uhg |
| 15:41.47 | bkw_ | heison: you clearly do not have SRVlookups on |
| 15:41.53 | bkw_ | because .100 is our web loadbalancer and not our sip proxy |
| 15:42.09 | bkw_ | you are required to have the SRV lookup on |
| 15:42.11 | bkw_ | and use the hostname |
| 15:42.39 | *** join/#asterisk ming_zym (n=ming_zym@124.14.236.139) |
| 15:42.59 | heison | i have tried both... let me check the syntax of SRVlookups |
| 15:44.48 | *** part/#asterisk ming_zym (n=ming_zym@124.14.236.139) |
| 15:45.08 | ManxPower | heison: syntax? you just set the option in sip.conf |
| 15:45.10 | *** join/#asterisk irule (n=irule@200.53.61.4) |
| 15:45.35 | heison | yeah, that's what i meant |
| 15:45.42 | heison | the right keyword |
| 15:48.17 | bkw_ | is srvlookup still off by default in Asterisk? |
| 15:48.24 | ManxPower | I guess I should stop procrastinating and start packing. |
| 15:48.49 | ManxPower | bkw_: I believe it defaults to off if not set, but the sample configs set it on |
| 15:49.10 | bkw_ | nope its on by default in trunk |
| 15:49.20 | bkw_ | its a violation of the SIP spec to have it off |
| 15:49.28 | bkw_ | heison: srvlookup=yes |
| 15:49.29 | heison | i have srvlookup=yes in both [general] and [asterlink]; dig -t SRV _sip._udp.asterlink.com works from the shell, yet registration fails |
| 15:49.39 | ManxPower | I've always found it to be a violation of a working dialplan to have it on. |
| 15:50.06 | bkw_ | ManxPower: SIP is more than just asterisk.. there is this whole wold outside of asterisk that uses SIP in the RIGHT way |
| 15:50.13 | ManxPower | heison: does your register line have an IP or a hostname? |
| 15:50.14 | bkw_ | heison: did you restart asterisk? |
| 15:50.33 | heison | yes, i have restarted asterisk and rebooted the box |
| 15:50.41 | *** join/#asterisk axscode (n=axscode@211.102.49.60.klj04-home.tm.net.my) |
| 15:50.50 | bkw_ | do you still have the hostname in the register line? |
| 15:51.14 | heison | i have: register => heison:passwd@asterlink.com |
| 15:51.37 | bkw_ | what does sip debug say? |
| 15:52.09 | *** join/#asterisk LoF^[Lawbringer] (n=Lawbring@cpc4-rdng12-0-0-cust292.winn.cable.ntl.com) |
| 15:52.34 | ManxPower | bkw_: I assume SRV should be pointing at proxy-01.asterlink.com? |
| 15:52.46 | bkw_ | yes as it does |
| 15:53.21 | ManxPower | bkw_: you know how broken srv lookups are in Asterisk |
| 15:53.28 | bkw_ | they work for me |
| 15:53.30 | bkw_ | and every one else |
| 15:53.35 | bkw_ | unless they broke them in 1.4 |
| 15:53.36 | *** join/#asterisk defswork (n=andy@77.44.54.34) |
| 15:53.41 | *** join/#asterisk dominic1 (n=dob@213.221.82.242) |
| 15:53.42 | *** part/#asterisk dominic1 (n=dob@213.221.82.242) |
| 15:53.55 | bkw_ | heison: the only way to fix this is put asterlink.com in /etc/hosts and point it at proxy-01.asterlink.com |
| 15:54.00 | ManxPower | bkw_: they broke many things in 1.4, I don't know about SRV support |
| 15:54.02 | bkw_ | or find software that isn't broken to accomplish your tasks |
| 15:54.26 | *** join/#asterisk creativx (n=creadure@226.62-97-205.bkkb.no) |
| 15:54.43 | bkw_ | Asterisk has the worst sip stack on the globe |
| 15:55.29 | *** join/#asterisk zerohalo (n=zeroHalo@pool-71-255-162-167.bstnma.east.verizon.net) |
| 15:56.20 | anonymouz666 | lol that's too much. |
| 15:57.31 | ManxPower | bkw_: even worse than Grandstream? |
| 15:57.52 | bkw_ | the grandstream one actually works correctly. |
| 15:58.02 | nestAr | lol |
| 15:58.09 | nestAr | man, the hate is strong today |
| 15:59.09 | [TK]D-Fender | Grandstream sucks for numerous other reasons :) |
| 16:00.25 | coppice | grandstream make a valiant attempt to comply with standards, which others could learn from. sadly, grandstream needs to learn something about software QA |
| 16:00.43 | *** join/#asterisk putnopvu1 (i=putnopvu@nat/digium/x-dc6f8ef3c3ead8a4) |
| 16:01.09 | ManxPower | coppice: Digium needs to learn something about QA as well. |
| 16:01.47 | coppice | but they need to learn about basic design, too |
| 16:02.36 | ManxPower | It compiles! Lets release it! |
| 16:07.47 | FlatFoot | using cdr_odbc.conf has anyone managed to make * use a different table per context for recording data ? |
| 16:08.03 | twisted | you know, asterisk is still open source... if you want to complain about it's code, you can always write it and contribute it back |
| 16:08.43 | Un1x_laptop | lol |
| 16:09.02 | axscode | Released! I will use it! |
| 16:09.04 | axscode | :) |
| 16:10.29 | ManxPower | axscode: the fact that so many people are still using 1.2 shows THAT doesn't happen quite as often as it used to. |
| 16:10.38 | Un1x_laptop | i still use 1.2 |
| 16:11.57 | *** join/#asterisk UnFred (n=UnFred@S010600095b44774f.vs.shawcable.net) |
| 16:15.41 | bkw_ | twisted: the care for the Open Source Asterisk isn't as extensive as ABE |
| 16:15.49 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 16:18.51 | *** join/#asterisk dasbrow (n=dasbrow@206.248.190.155) |
| 16:19.20 | dasbrow | Hi everyone |
| 16:19.20 | *** part/#asterisk bkw_ (n=brian@adsl-64-149-41-151.dsl.tul2ok.sbcglobal.net) |
| 16:19.47 | dasbrow | Could someone help me out with dealing with + in front of the caller id? |
| 16:20.23 | ai-a | dealing with it in what way ? |
| 16:21.09 | *** join/#asterisk ReDNeQ (i=ReDNeQ@cpe-66-68-37-190.austin.res.rr.com) |
| 16:21.10 | *** join/#asterisk Law (n=Lawbring@cpc4-rdng12-0-0-cust292.winn.cable.ntl.com) |
| 16:21.23 | dasbrow | I can't seem to validate incoming calls if they have the + in the number. I would like to strip it out. |
| 16:23.06 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 16:23.07 | *** join/#asterisk rpm (n=russell@75.153.47.179) |
| 16:24.06 | *** join/#asterisk irule (n=irule@200.53.61.4) |
| 16:29.31 | *** join/#asterisk anonymouz333 (n=voce@201.19.106.70) |
| 16:31.16 | *** join/#asterisk caniphone (n=adminrm@S0106004063d8e527.ed.shawcable.net) |
| 16:31.32 | [TK]D-Fender | dasbrow: Cahnge your dialplan to include them. |
| 16:31.35 | *** part/#asterisk caniphone (n=adminrm@S0106004063d8e527.ed.shawcable.net) |
| 16:32.00 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 16:33.08 | dasbrow | how would I do that? They show up in the caller id, but when I add the plus in the database I get a syntax error. |
| 16:33.45 | [TK]D-Fender | dasbrow: "database"? |
| 16:34.11 | [TK]D-Fender | dasbrow: As for callerid, "show function CALLERID" <- you should be able to strip them pretty easily. |
| 16:34.18 | ai-a | they have +nnn as the number being called ? |
| 16:34.32 | ai-a | or the callerid call from ? |
| 16:36.32 | dasbrow | the database gives access to our employees to use the internal phone system. Using caller id as the auth method. |
| 16:36.45 | dasbrow | ai-a: callerid from |
| 16:37.10 | nestAr | i am running 1.4, is there something i should know. ;) |
| 16:38.14 | ai-a | if your using a db for callerid lookup to map their name,,, write a sql function that removes all spaces, replaces +nn with 00NN and so on.. so you have, in uk, 00441212929222 numbers. |
| 16:40.41 | dasbrow | but the problem is really with the incoming caller id. I can't change the incoming id with an sql function. |
| 16:41.08 | ai-a | why cant you ? |
| 16:41.27 | dasbrow | If I could just drop the + when the caller calls in that would best. Maybe a regex would work. |
| 16:42.02 | dasbrow | Maybe I missunderstood the first time, your saying grab the incoming callerid, use an sql function to strip the plus and then compare it to the database? |
| 16:43.09 | *** join/#asterisk blitzrage (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 16:43.09 | *** mode/#asterisk [+o blitzrage] by ChanServ |
| 16:43.44 | *** join/#asterisk linxroute (n=linx@125.214.1.91) |
| 16:46.42 | *** join/#asterisk axscode (i=axscode@130.17.111.218.klj04-home.tm.net.my) |
| 16:49.56 | *** join/#asterisk c0rnflake (n=c0rnflak@38.112.4.210) |
| 16:50.08 | *** join/#asterisk axscode (i=axscode@130.17.111.218.klj04-home.tm.net.my) |
| 16:51.39 | *** join/#asterisk LoF^[Lawbringer] (n=Lawbring@cpc4-rdng12-0-0-cust292.winn.cable.ntl.com) |
| 16:53.06 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 16:53.06 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 16:53.32 | *** join/#asterisk IPetrov2 (i=IPetrov2@ppp85-140-235-241.pppoe.mtu-net.ru) |
| 16:57.35 | *** join/#asterisk docelmo (n=vircuser@c-76-99-153-242.hsd1.de.comcast.net) |
| 16:57.35 | *** join/#asterisk fnordus (n=dnall@24.84.160.227) |
| 16:58.59 | ManxPower | good god, how complicated are you going to make this??? Set(CALLERID(num)=00${CALLERID(num):1}) |
| 16:59.16 | *** join/#asterisk mrgoby (n=mrgoby@aa.linuxbox.com) |
| 17:00.12 | mrgoby | i'm having a crazy issue with a box where when someone leaves a voicemail and you try to attach it to an email, asterisk segfaults ... version 1.4.12 |
| 17:00.20 | *** join/#asterisk UnFred (n=UnFred@S010600095b44774f.vs.shawcable.net) |
| 17:00.49 | mrgoby | running on dual xeon |
| 17:01.07 | mrgoby | anyone seen anything like this before ? there are no logs, just faults and dumps |
| 17:01.12 | ManxPower | mrgoby: you mean when Asterisk tries to hand the message to sendmail as an e-mail with attachment? |
| 17:01.28 | mrgoby | hard to say exactly, but yes, right around there |
| 17:01.36 | ManxPower | you need a local smtp server running on the asterisk box. |
| 17:01.45 | mrgoby | i can use sendmail fine from the cli |
| 17:02.00 | *** join/#asterisk emist_ (n=emist@unaffiliated/emist) |
| 17:02.00 | ManxPower | mrgoby: anything in the sendmail logs? |
| 17:02.17 | mrgoby | root 2244 0.0 0.2 69536 2324 ? Ss 11:00 0:00 sendmail: accepting connections |
| 17:02.17 | mrgoby | smmsp 2251 0.0 0.1 56056 1776 ? Ss 11:00 0:00 sendmail: Queue runner@01:00:00 for /var/spool/clientmqueue |
| 17:02.20 | mrgoby | lemme look |
| 17:02.49 | ManxPower | if all else fails, read the backtrace document in /path/to/src/asterisk/doc and submit a bug report. |
| 17:03.17 | ManxPower | also there is NO reason for sendmail to listen on a socket unless you want to run a full smtp server. |
| 17:04.04 | mrgoby | interesting |
| 17:04.07 | *** join/#asterisk Defraz (n=t0tal@24-116-152-177.cpe.cableone.net) |
| 17:04.25 | mrgoby | okay, so there is a new behavior |
| 17:04.29 | mrgoby | this has been ongoing for a while |
| 17:04.30 | dasbrow | ManxPower: Thanks, I've changed it to Set(CALLERID(num)=${CALLERID(num):-10}) so that it can compare a 10 digit number. Makes a lot more sense to me. |
| 17:04.48 | mrgoby | i thought this had maybe something to do with my non-updated fedora core 7 install |
| 17:05.04 | mrgoby | so i just updated and rebuilt asterisk and the modules against the newly installed headers, etc |
| 17:05.19 | mrgoby | so... now it is still segfaulting, but the email is actually going out |
| 17:05.26 | mrgoby | it was not before |
| 17:05.39 | ManxPower | sounds like you need to submnit a bugreport with a backtrace |
| 17:06.44 | mrgoby | yeah... something is going on super funky... i'll go over my sendmail conf... but I overrode the mailer command in the voicemail.conf to just cat the email before and it was still segfaulting, though it did succeed in cat-ing the message |
| 17:10.32 | mrgoby | anyway, thanks |
| 17:11.54 | *** join/#asterisk axscode (i=axscode@130.17.111.218.klj04-home.tm.net.my) |
| 17:13.29 | *** join/#asterisk w3nk4r (n=kvirc@216.57.171.107) |
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| 17:18.07 | *** join/#asterisk bitbandit (n=tagg@mail.dutro.com) |
| 17:20.28 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 17:24.01 | *** join/#asterisk axscode (n=axscode@130.17.111.218.klj04-home.tm.net.my) |
| 17:24.41 | *** join/#asterisk bantu (n=Miranda@p54A32D71.dip0.t-ipconnect.de) |
| 17:25.14 | *** join/#asterisk hfb (n=hfb@pool-71-106-219-180.lsanca.dsl-w.verizon.net) |
| 17:34.46 | *** join/#asterisk jameswf (n=jameswf@dsl093-157-131.phx1.dsl.speakeasy.net) |
| 17:34.56 | *** join/#asterisk rnovotny23 (n=root@70.13.150.181) |
| 17:35.28 | *** join/#asterisk sevard (n=sev@192.235.0.85) |
| 17:35.32 | sevard | is there some reason I have to reboot this damned sipura daily to get it to stay connected to FWD even though i have nat mapping and nat keepalive enabled |
| 17:36.21 | jameswf | <PROTECTED> |
| 17:37.00 | sevard | jameswf-home: I don't think that's the sollution ;) |
| 17:37.01 | *** join/#asterisk jameswf (n=jameswf@dsl093-157-131.phx1.dsl.speakeasy.net) |
| 17:37.36 | *** join/#asterisk jameswf (n=jameswf@dsl093-157-131.phx1.dsl.speakeasy.net) |
| 17:38.00 | *** join/#asterisk jameswf (n=jameswf@dsl093-157-131.phx1.dsl.speakeasy.net) |
| 17:38.24 | *** join/#asterisk CVirus (n=GoD@196.205.192.246) |
| 17:40.17 | *** join/#asterisk ApolloDS (n=ApolloDS@dhclient-212-35-16-73.flashcable.ch) |
| 17:40.44 | *** join/#asterisk atomicd (n=Atomicd@74-206-0-80.static-ip.m.telepacific.net) |
| 17:41.29 | *** join/#asterisk EnigmaCurry (n=user@c-24-10-239-16.hsd1.ut.comcast.net) |
| 17:42.23 | *** part/#asterisk andylockran (n=andylock@genesis.zrmt.com) |
| 17:42.46 | Maxxed | i wanan move my pbx to the colo and use some sip/iax provider for my inbound/outbound calls. about what is the going rate for 3 standard lines? |
| 17:43.01 | Maxxed | 20/month per line or some such? |
| 17:43.01 | *** join/#asterisk ApolloDS (n=ApolloDS@dhclient-212-35-16-73.flashcable.ch) |
| 17:43.06 | Maxxed | er, did rather |
| 17:43.16 | *** join/#asterisk marl (n=marl@89.241.242.164) |
| 17:43.17 | ManxPower | Maxxed: less than 2/cents/min |
| 17:43.17 | EnigmaCurry | Can I make Asterisk translate Alpha numbers to Digit numbers? .. eg, if I dial 1800GOOG411 in my softphone, can I make Asterisk dial 18004664411? |
| 17:43.44 | Maxxed | ManxPower, the price per min is fine, but how much monthly if i dont make a single phone call |
| 17:43.53 | ManxPower | EnigmaCurry: "show applications" did not show something obvious? |
| 17:43.53 | Maxxed | i know thats loose, but ball park |
| 17:44.02 | marl | hi there, anyone know if its posible to compile asterisk with all its modules static? ie compiled into * rather than being opened after * starts? |
| 17:44.03 | [TK]D-Fender | EnigmaCurry: Yes, its your dialplan, you can do whatever you want with it, |
| 17:44.03 | ManxPower | Maxxed: usually under $9/month |
| 17:44.18 | Maxxed | no lie, 9 bucks a month per did, thats a sweet price |
| 17:44.27 | ManxPower | Maxxed: plus usage, of course. |
| 17:44.38 | ManxPower | I think vitelity and teliax are well below that. |
| 17:44.39 | Maxxed | righty'o |
| 17:44.46 | *** join/#asterisk axscode (n=axscode@130.17.111.218.klj04-home.tm.net.my) |
| 17:44.49 | *** join/#asterisk bitbandit (n=tagg@mail.dutro.com) |
| 17:44.53 | ManxPower | of course, don't expect it to be as reliable as the PSTN. |
| 17:44.55 | EnigmaCurry | Maxxed: I'll check there, thanks |
| 17:44.57 | Maxxed | i can ditch these old analog lines and move over to a sip/iax provider :D |
| 17:45.13 | Maxxed | its not as reliable? how so? |
| 17:45.27 | ManxPower | you are sending the calls over the internet. That's not very reliable, now is it? |
| 17:45.39 | Maxxed | why wouldnt it be? |
| 17:45.47 | ManxPower | because the internet is not very reliable. |
| 17:46.03 | ManxPower | there is no |
| 17:46.17 | Maxxed | i have a bitchin colo, havent lost conectivity since i moved in |
| 17:46.23 | ManxPower | There is no Quality of Service, if a carrier between you and your ITSP goes down, there's nothing you can do abou tit. |
| 17:46.23 | Maxxed | packet loss, hardly ever |
| 17:46.36 | ManxPower | Well, best of luck with that. |
| 17:46.37 | Maxxed | right right |
| 17:46.45 | Maxxed | well i was also thinking fractional pri |
| 17:47.02 | Maxxed | but i cant find anybody that wants to break one down to anything less than 12 chanels |
| 17:47.21 | Maxxed | if i could get like a 5 channel fractional pri that would be sweet |
| 17:47.36 | ManxPower | you could even have internet service on the unused channels. |
| 17:47.57 | Maxxed | yeah, but i have gobs of bandwidth allready |
| 17:48.15 | Maxxed | 1mbit, i wouldnt even notice |
| 17:50.45 | Maxxed | seems to cost is just to high for a pri |
| 17:50.51 | Maxxed | im just to small |
| 17:50.52 | Maxxed | dangit |
| 17:51.14 | ManxPower | exactly how are your IP phones going to get the colo? |
| 17:51.57 | *** join/#asterisk felix_da_catz (n=felix@PixNationalGroup.phonoscope.com) |
| 17:51.59 | ManxPower | heck, why would you even want to colo the server? |
| 17:52.26 | tzanger | ManxPower: I prefer my server colocated |
| 17:52.44 | ManxPower | tzanger: how do you get your IP phones talking to the server at the colo? |
| 17:52.52 | Maxxed | yeah? |
| 17:52.57 | Maxxed | im looking to do something like that my self |
| 17:53.00 | Maxxed | i was thinking vpn |
| 17:53.14 | Maxxed | if i could do vlans over vpn that would be sweet |
| 17:53.19 | tzanger | ManxPower: vpn, yeah, or just over-the-air |
| 17:53.21 | jameswf | so it was just forewarded to me that trixbox "pro" is like tivo, |
| 17:53.30 | jameswf | hands off or they get cut off |
| 17:53.41 | ManxPower | you still need internet service for a VPN. If you don't have internet service, your phones are not going anywhere. |
| 17:53.54 | Maxxed | yep |
| 17:54.01 | ManxPower | jameswf: trixbox here is like the plague. |
| 17:54.12 | jameswf | just here :) |
| 17:54.12 | tzanger | ManxPower: yep, but when your numbers are coming in over IP anyway, it's beter to at least be able to play an ivr and pretend everyone' sbusy |
| 17:54.33 | ManxPower | I can't imagine why anyone would want to move and asterisk server OFF of the lan where all the phones are. |
| 17:54.52 | Qwell | jameswf: no, everywhere |
| 17:54.56 | ManxPower | jameswf: well, here and asterisk-dev |
| 17:55.08 | Maxxed | well look at all th hosted pbx soultions out their |
| 17:55.09 | ManxPower | we don't really care what the rest of the world thinks about trixbox. |
| 17:55.12 | Maxxed | there* |
| 17:55.34 | ManxPower | IP Phones <-> Local LAN <-> Asterisk <-> PRI |
| 17:56.09 | ManxPower | .vs. IP Phones <-> Local Lan <-> gawd knows how many ISPs <-> Asterisk Server <-> PRI or ITSP. |
| 17:56.14 | *** join/#asterisk slima (i=slima@unaffiliated/slima) |
| 17:56.15 | destructure | ~pb |
| 17:56.16 | jbot | A Pastebin is a place to paste your stuff without flooding 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://www.rafb.net/paste |
| 17:56.34 | Maxxed | you know how many isps, tracert :p |
| 17:56.39 | tzanger | ManxPower: I'm a special case I guess, all my DSL lines end up in the same rack as my asterisk server |
| 17:56.50 | Maxxed | if your ITSP is a 4-5 hops away, no worries |
| 17:56.56 | *** join/#asterisk alphanet (i=ircuser@shakotay.alphanet.ch) |
| 17:57.04 | ManxPower | tzanger: yes, you are a special case. |
| 17:57.12 | tzanger | so it's ip phones <--> local lan <--> DSL <--> asterisk server <--> PRI/ITSP |
| 17:57.27 | alphanet | hello. If I issue a new call through the manager interface, how can I track it? I have tried to specify a Uniqueid:, but it is not used by Asterisk. |
| 17:57.29 | tzanger | ManxPower: besides, jitter buffering is FAR better on the polycoms than in asterisk itself, sadly |
| 17:57.29 | Maxxed | i have level3 bandwith, iv tracerted to a few level3 itsp's ad it looksgoood |
| 17:57.47 | ManxPower | As I said, good luck. |
| 17:58.29 | Maxxed | mmm.. im gona try it |
| 17:58.39 | Maxxed | im small enough, if it dont work well, il revert back |
| 17:58.43 | ManxPower | But if you come back here in a couple of months complaining about a 2 day outage, I'll bitch slap you into next week. |
| 17:58.47 | Maxxed | no millions in lost revenu |
| 17:58.47 | Maxxed | heh |
| 17:58.51 | Maxxed | lol |
| 17:59.23 | Maxxed | i think if i go with a itsp that uses the same backbone i do, i wont have many probs |
| 17:59.51 | Maxxed | like i said my carrier hasnt gone down since i moved over, thats been a good 5+ years now |
| 18:00.28 | [TK]D-Fender | alphanet: What do you want to do as far as "tracking" is concerned? |
| 18:00.35 | alphanet | ok, ActionID :) |
| 18:00.59 | alphanet | [TK]D-Fender: I am issuing a few calls in parallel and I need to know which one succeeds and fail |
| 18:01.21 | *** join/#asterisk Blue_Ice (n=Blue_Ice@195-130-159-122.iFiber.telenet-ops.be) |
| 18:01.28 | [TK]D-Fender | Thats what CDR is for. Or change how is dials to add a way to report back. |
| 18:01.44 | *** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net) |
| 18:01.44 | *** join/#asterisk |omni| (n=rob@c-67-185-56-106.hsd1.wa.comcast.net) |
| 18:02.14 | alphanet | [TK]D-Fender: I did it once by going through a dialplan then calling an external notification command, but I find this ugly |
| 18:02.38 | [TK]D-Fender | alphanet: Welcome to Asterisk :) Thats all you've got... |
| 18:08.11 | *** join/#asterisk tagg_ (n=tagg@mail.dutro.com) |
| 18:08.31 | *** join/#asterisk flujan (n=flujan@200-160-115-020.static.spo.ctbc.com.br) |
| 18:08.45 | flujan | hi guys, asterisk is dying with a segmentation fault on my box... |
| 18:09.04 | flujan | safe_asterisk immediately restarts it and it keep working. |
| 18:09.14 | flujan | I enable the full log with debug but no tip about the problem. |
| 18:09.21 | flujan | how can I debug it? |
| 18:11.11 | [TK]D-Fender | flujan: Stop running it through the script and run it manually and see what its crashing... |
| 18:11.36 | flujan | hi [TK]D-Fender ... |
| 18:11.40 | *** join/#asterisk gardo (n=gardo@121.97.196.87) |
| 18:11.42 | flujan | You mean kill the safe_asterisk ? |
| 18:11.51 | [TK]D-Fender | flujan: Yes |
| 18:12.09 | flujan | ok, I will check this out... :d |
| 18:12.23 | *** join/#asterisk tzafrir (n=tzafrir@62.90.10.53) |
| 18:12.34 | *** join/#asterisk chode (n=chode@pD9E896CD.dip0.t-ipconnect.de) |
| 18:12.41 | alphanet | [TK]D-Fender: apparently it works. I just need to add ActionID and it will be sent to me at the Response; along with the Uniqueid for the call, so I can track it easily. |
| 18:13.08 | [TK]D-Fender | alphanet: Ok, more power to you then... |
| 18:13.19 | *** join/#asterisk shinao1 (n=shinao1@217.20.242.50) |
| 18:13.37 | alphanet | [TK]D-Fender: I am porting an old ISDN call forwarding application to Asterisk |
| 18:13.51 | *** join/#asterisk kombi (n=kombi@port-213-160-14-18.static.km-it.de) |
| 18:15.12 | *** join/#asterisk Nukemizer (n=Nuke@67.137.28.165) |
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| 18:17.54 | *** join/#asterisk Blue_Ice (n=Blue_Ice@195-130-159-121.iFiber.telenet-ops.be) |
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| 18:24.03 | *** join/#asterisk viperdudeuk (n=chatzill@84-45-129-190.no-dns-yet.enta.net) |
| 18:25.24 | *** join/#asterisk viperdudeuk (n=chatzill@84-45-129-190.no-dns-yet.enta.net) |
| 18:30.49 | *** join/#asterisk saftsack (n=saftsack@pD9E078F0.dip.t-dialin.net) |
| 18:31.22 | *** join/#asterisk twoshadetod (n=twoshade@c-76-123-96-239.hsd1.fl.comcast.net) |
| 18:31.37 | twoshadetod | anyone using a SIP phone standalone? |
| 18:31.52 | moemoe | standalone? |
| 18:32.01 | hmmhesays | you can if you have some SIP service provider |
| 18:32.03 | twoshadetod | yeah like directly going to a sip provider |
| 18:32.20 | hmmhesays | I know people who use xlite with broadvoice and vitelity |
| 18:32.23 | twoshadetod | yeah i got it up like that |
| 18:32.34 | twoshadetod | but it's only outgoing i wanted to see what people were doing for incoming |
| 18:32.54 | hmmhesays | if you have a sip service provider that gives you a DID then it should be fine |
| 18:33.24 | twoshadetod | i have just outgoing with them , i sort of want to use a diff provider for DIDs |
| 18:33.24 | moemoe | yes, i do so. i connected my snom105 to sipgate, but only use it incoming |
| 18:33.31 | twoshadetod | even if i could get it through them(sipdiscount.com) |
| 18:33.41 | hmmhesays | you should be able to register more than one account |
| 18:33.51 | twoshadetod | nice, this is a pcom 501 |
| 18:34.00 | hmmhesays | just register multiple accounts |
| 18:34.22 | twoshadetod | sipgate for incoming/DID moemoe? |
| 18:34.53 | moemoe | no, w/o |
| 18:34.58 | twoshadetod | I'll check that out how does the phone handle that "account"? does it take a line? lke my phone has 3 lines, would i program the incoming on one of those lines |
| 18:35.02 | twoshadetod | ahh sorry |
| 18:36.00 | [TK]D-Fender | twoshadetod: Yes, you can support up to 3 seperate accounts on your phone. |
| 18:37.09 | twoshadetod | [TK]D-Fender, cool, how does it associate my outgoing calls (going throguh sipdiscount.com) with the CID number? I don't see anyway to put the CID number in now (you would think I could tell it to disiplay CID even if i dont really have a DID/any number truly associated) |
| 18:37.23 | *** join/#asterisk _ys (n=yuri@236-069.nat.mns.ru) |
| 18:37.50 | [TK]D-Fender | twoshadetod: When you pick which account to use (by pressing its line-key" it'll go out the associated account. |
| 18:38.16 | [TK]D-Fender | twoshadetod: If you have multiple DID's associated with a single account, then you will not be able to choose it this way directly from your phone. |
| 18:40.12 | twoshadetod | [TK]D-Fender, thing is my outbound provider doesn't give me a number, but if i go with a diff company for a incoming number, how would my outgoing provider know to display that number on peoples caller id when they look at it? |
| 18:40.43 | *** join/#asterisk basskozz (n=mike@209-6-20-97.c3-0.wrx-ubr3.sbo-wrx.ma.cable.rcn.com) |
| 18:40.57 | *** part/#asterisk basskozz (n=mike@209-6-20-97.c3-0.wrx-ubr3.sbo-wrx.ma.cable.rcn.com) |
| 18:41.24 | [TK]D-Fender | twoshadetod: they wouldn't. Also it would depend if they even LET you set the CID #. |
| 18:41.38 | twoshadetod | ahhhh |
| 18:41.43 | [TK]D-Fender | twoshadetod: Either way you're asking too much of your phone. This is where you would wwant to have * in the middle |
| 18:41.53 | twoshadetod | ok, the isn't done on my phone, it's on their server (outgoing) |
| 18:42.10 | *** part/#asterisk EnigmaCurry (n=user@c-24-10-239-16.hsd1.ut.comcast.net) |
| 18:42.56 | hmmhesays | or at least some external sip based software |
| 18:43.04 | *** join/#asterisk viperdudeuk (n=chatzill@84-45-168-57.no-dns-yet.enta.net) |
| 18:43.11 | [TK]D-Fender | twoshadetod: "their server"? |
| 18:45.27 | *** join/#asterisk avp (n=wow@88.234.90.209) |
| 18:45.54 | AliOzaltin | Hello, which adapter can i receive external call to the asteriks ? |
| 18:46.03 | AliOzaltin | Can i use an 56k modem for this ? |
| 18:46.13 | hmmhesays | some fxo card |
| 18:46.33 | AliOzaltin | via internal or external 56k modem ? |
| 18:46.56 | AliOzaltin | when anybody call my phone asteriks opening phone and a robot speaking? |
| 18:48.09 | [TK]D-Fender | AliOzaltin: No, you can't use jsut any old crappy modem as an FXO interface, you'll need to buy REAL telecom hardware |
| 18:48.32 | [TK]D-Fender | AliOzaltin: And if by "robot" you mean an auto-attendant, then yes, Asterisk can do this rather easily |
| 18:49.01 | AliOzaltin | hmm okey thank you.. i will only need fxo yeah ? |
| 18:49.42 | AliOzaltin | should i need fxs ? |
| 18:50.17 | *** join/#asterisk techie (n=techie@adsl-76-214-20-56.dsl.lsan03.sbcglobal.net) |
| 18:52.17 | *** join/#asterisk viperdudeuk (n=chatzill@84-45-168-57.no-dns-yet.enta.net) |
| 18:53.06 | *** join/#asterisk CyberScript32 (n=osgc@189.32.104.12) |
| 18:53.07 | *** join/#asterisk viperdudeuk (n=chatzill@84-45-168-57.no-dns-yet.enta.net) |
| 18:53.11 | CyberScript32 | ~book |
| 18:53.12 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf |
| 18:53.38 | *** join/#asterisk tiav (n=tiav@ram94-3-82-225-11-215.fbx.proxad.net) |
| 18:53.52 | [TK]D-Fender | AliOzaltin: FXO is for using PSTN LINES. If you want to use analog PHONES with your system as well you'll need FXS. Stop now and read the BOOK. You clearly do not have the basic knowledge of telecom you should have before getting into things. |
| 18:53.55 | [TK]D-Fender | ~book |
| 18:53.56 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf |
| 18:53.57 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^^^^^ |
| 18:59.04 | lirakis | lol 56k modem |
| 19:00.17 | AliOzaltin | I will only use my external analog lines for receiving call. for internal users i will use ip phone.. so should i order an fxo card or fxs ? |
| 19:01.06 | *** join/#asterisk dm_ (n=dm@suez.activ-job.com) |
| 19:02.06 | *** join/#asterisk fakhir_ (n=fakhir@ool-44c69df5.dyn.optonline.net) |
| 19:02.33 | *** join/#asterisk bitbandit (n=tagg@mail.dutro.com) |
| 19:04.08 | vlt | Hello. I'm running Asterisk behind a NAT router. When I make a call and the callee answers I can't hear his first 300-600 ms. I forwarded all the RTP ports to the asterisk machine. Any idea what still could be missing here? |
| 19:05.01 | *** join/#asterisk axscode (n=axscode@130.17.111.218.klj04-home.tm.net.my) |
| 19:06.24 | *** join/#asterisk luke-jr (n=luke-jr@2002:440d:6de2:0:20e:a6ff:fec4:4e5d) |
| 19:10.34 | [TK]D-Fender | vlt: Read up : |
| 19:10.36 | [TK]D-Fender | ~sipnat |
| 19:10.36 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 19:10.37 | *** join/#asterisk Tourinho (n=gtourinh@cm-virtua-poa-C8B0F83B.dynamic.brdterra.com.br) |
| 19:11.23 | Tourinho | good morning people, how can I monitore a DTMF code entered after the Dial application was executed? Is there a way to control this? |
| 19:13.57 | *** join/#asterisk ghento (n=ghento@75.155.241.7) |
| 19:14.08 | *** join/#asterisk viperdudeuk_ (n=chatzill@195.74.96.113) |
| 19:14.16 | [TK]D-Fender | Tourinho: There is no normal means for doing that. |
| 19:15.09 | *** join/#asterisk xtr (n=94752345@216.19.191.191.novuscom.net) |
| 19:16.10 | Tourinho | [TK]D-Fender even using AGI? Im trying to write an application that control DTMF after dial to a certain place |
| 19:16.35 | Tourinho | last week i thougth that u guys was talking about it |
| 19:16.55 | *** join/#asterisk xtr-II (n=94752345@216.19.191.191.novuscom.net) |
| 19:17.05 | *** join/#asterisk axscode (n=axscode@130.17.111.218.klj04-home.tm.net.my) |
| 19:17.08 | [TK]D-Fender | Tourinho: You can have dial pass on some dtmf, but not an interactive thing. |
| 19:17.49 | *** join/#asterisk ghenry (n=ghenry@ghenry.plus.com) |
| 19:17.58 | *** join/#asterisk |dennis| (n=dennis@200.32.233.84) |
| 19:19.24 | *** join/#asterisk halconnen (n=halconne@rrcs-67-52-187-66.west.biz.rr.com) |
| 19:19.52 | Tourinho | [TK]D-Fender how can I do that? I just want to know if the caller dialled # |
| 19:20.29 | [TK]D-Fender | Tourinho: You won't Forget about that unless you feel like ripping apart the cahnnel driver code. |
| 19:20.57 | halconnen | Hi guys! Quick question: I'm using Trixbox (2.2.4) and the HUDLite client. When I right click on the extention and select intercom, the Ring-Answer alert info doesnt get passed to the first extention (meaning I have to answer my phone before the intercom works). Any tips on where to change that? |
| 19:21.25 | [TK]D-Fender | halconnen: Trixbox is NOT supported here |
| 19:21.27 | [TK]D-Fender | ~trixbox |
| 19:21.27 | jbot | rumour has it, trixbox is a full linux distro that includes FreePBX and other 3rd party add-ons. It is all this extra stuff which makes trixbox seriously painful to support, and thus you will find little help here for it. Try asking in #trixbox or on their forums & wiki at http://www.trixbox.org |
| 19:21.28 | *** join/#asterisk lemanal (n=lemanal@wifi-224-227.sc07.org) |
| 19:21.36 | *** join/#asterisk masus (n=tet@88.248.14.186) |
| 19:21.53 | masus | does anyone know a free softphone for debian |
| 19:22.12 | [TK]D-Fender | masus: Ekiga, Twinkle, kphone. |
| 19:22.19 | [TK]D-Fender | masus: To name a few |
| 19:22.28 | halconnen | Thanks! |
| 19:23.16 | Tourinho | [TK]D-Fender so, there is no way to do that? :( |
| 19:23.36 | masus | Thanks |
| 19:23.36 | *** join/#asterisk jtexter3 (n=jamest@adsl-70-234-105-253.dsl.tul2ok.sbcglobal.net) |
| 19:23.53 | [TK]D-Fender | Tourinho: What part of "no" wasn't clear there? |
| 19:24.53 | jtexter3 | anyone here know how to force a core dump to be created when running on a Mac? I'm trying to debug a segfault, but no core dump is created |
| 19:25.46 | Tourinho | [TK]D-Fender ok, but I can set the Dial application to allow caller to transfer, right? |
| 19:26.18 | *** join/#asterisk Seldon75 (n=chatzill@69.77.161.3) |
| 19:26.19 | [TK]D-Fender | Tourinho: "show application dial" <- Oh yeah... and read the INSTRUCTIONS. |
| 19:27.11 | tzafrir | masus, twinkle, ekiga, kiax |
| 19:27.27 | Tourinho | [TK]D-Fender oks thanks |
| 19:27.52 | tzafrir | oops, missed TK's answer. I don't recommend kphone, though |
| 19:28.41 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 19:31.06 | *** join/#asterisk blq (n=Bl@dslb-088-066-241-048.pools.arcor-ip.net) |
| 19:31.38 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 19:31.46 | masus | will that work with xfce4 |
| 19:31.54 | *** join/#asterisk sheldonh (n=sheldonh@66.219.59.32) |
| 19:32.03 | sheldonh | what do these mean? "channel.c: No path to translate from SIP/bokone-nat-08b9afa0(256) to IAX2/aggr2-jnb-22(1)" |
| 19:32.41 | sheldonh | started seeing them about 5 days ago, which coincides with network changes (new ip routes over fatter pipes) |
| 19:33.35 | *** join/#asterisk agx (n=badpengu@81-174-46-174.dynamic.ngi.it) |
| 19:34.46 | [TK]D-Fender | sheldonh: Means "Gee I wish I bought G.729 licenses" |
| 19:38.37 | sheldonh | [TK]D-Fender: but we're not transcoding. i thought you onlt needed licenses if you transcoded? |
| 19:38.52 | [TK]D-Fender | sheldonh: You are transcoding there. 256 -> 1 |
| 19:39.02 | [TK]D-Fender | sheldonh: and 256 = G.729 |
| 19:39.21 | sheldonh | excellent. not my #@$%#$ fault, then :) |
| 19:39.34 | file | 1 = G723, 256 = G729 |
| 19:39.46 | [TK]D-Fender | sheldonh: yeah. Ok. Fine. Sure. Whatever. |
| 19:39.54 | [TK]D-Fender | G.723?! Even BETTER! |
| 19:40.08 | *** join/#asterisk axscode (n=axscode@130.17.111.218.klj04-home.tm.net.my) |
| 19:40.11 | [TK]D-Fender | Not just screwed, now you're fucked as well ;) |
| 19:40.50 | [TK]D-Fender | sheldonh: http://www.ky.com/ |
| 19:41.05 | [TK]D-Fender | </comicrelief> |
| 19:41.19 | sheldonh | licenses is a problem that can be solved. i've just burned days getting the multilink ppp stuff working, and was convinced my network changes were somehow crapping on asterisk |
| 19:43.49 | sheldonh | hmmm, interesting... svk log shows -disallow=all |
| 19:44.14 | sheldonh | my guess is someone allowed the peer to throw g723 at us, and now we're transcoding |
| 19:45.10 | *** join/#asterisk CleanerX (n=nix@p5B13428A.dip0.t-ipconnect.de) |
| 19:46.15 | [TK]D-Fender | sheldonh: Good use of namesless pronouns... you'll need that at the hearing ;) |
| 19:46.44 | jameswf | weeeeeeeeee.... |
| 19:47.13 | sheldonh | *shrug* you've obviously made your mind up :) |
| 19:48.58 | *** join/#asterisk callguy (n=callguy@207.190.206.2) |
| 19:49.14 | [TK]D-Fender | ~whee |
| 19:49.15 | jbot | [~whee] Weeeeeeeee! http://www.albinoblacksheep.com/flash/weeee |
| 19:49.47 | *** join/#asterisk TrentCreek (n=TrentCre@cpe-70-117-207-168.rgv.res.rr.com) |
| 19:49.54 | [TK]D-Fender | sheldonh: the jury's about as stacked as Christina Aguilera these days ;) |
| 19:50.12 | sheldonh | i noticed unpackaged files installed for asterisk a couple of weeks ago, did some reading spoke to the guy who installed them, who said licenses were only required for transcoding. did more reading, agreed, left it at that. 3 days ago (coinciding with major network changes), the same dude allowed g723 from the client, thus enabling transcoding. such is life |
| 19:50.23 | sheldonh | [TK]D-Fender: she's... she's not natural? *gasp* |
| 19:50.35 | sheldonh | oh wait, you said stacked, not rigged |
| 19:50.51 | [TK]D-Fender | sheldonh: I believe she is, but lactating in prep for her new "bun" |
| 19:51.25 | [TK]D-Fender | sheldonh: And its called a "synonym". You should look it up... but it'd probably jsut give you another word meaning the same thing ;) |
| 19:51.46 | sheldonh | no more sweets for that man :) |
| 19:56.38 | *** join/#asterisk ghenry (n=ghenry@ghenry.plus.com) |
| 19:58.49 | *** part/#asterisk myiagy (n=myiagy@189.34.11.211) |
| 19:58.54 | *** join/#asterisk Netgeeks (n=chris@204.11.231.198.static.etheric.net) |
| 19:59.26 | *** part/#asterisk agx (n=badpengu@81-174-46-174.dynamic.ngi.it) |
| 19:59.33 | *** join/#asterisk souzha57803 (n=IceChat7@static-72-72-83-224.bstnma.east.verizon.net) |
| 19:59.53 | *** join/#asterisk axscode (i=axscode@130.17.111.218.klj04-home.tm.net.my) |
| 20:00.40 | souzha57803 | Hey guys, I'm trying to setup a polycom 4000, and I'm monitoring the aftfpd.log,and its just looping through fetching of the setup files, I have installed the necessary firmware and it seems that the tftp server is crapping out (but all my other phones boot and load config files fine) |
| 20:00.54 | souzha57803 | <PROTECTED> |
| 20:01.01 | souzha57803 | 0101004032|copy |3|00|tftpLib error: tftp transfer failed: error 0x4b0008 |
| 20:01.18 | souzha57803 | thats the error being thrown in the boot.log |
| 20:01.21 | *** join/#asterisk PBXX (n=PBX@ip-89.171.196.34.crowley.pl) |
| 20:01.50 | *** join/#asterisk CVirus (n=GoD@196.205.192.246) |
| 20:02.08 | fujin | atftpd? that's like, the best one, I've never had any issues with it |
| 20:02.16 | fujin | although I prefer not to transfer firmware with tftp |
| 20:02.18 | fujin | where possible. |
| 20:02.28 | fujin | as it is just dumb udp, no retransmit or anything |
| 20:02.52 | souzha57803 | yeah |
| 20:03.08 | fujin | are you not able to use HTTP/FTP on the polycom 4k? |
| 20:03.15 | souzha57803 | hmmm |
| 20:03.18 | fujin | I generally just provision XML with tftp |
| 20:03.30 | souzha57803 | and use ftp for the binaries? |
| 20:03.34 | fujin | but then again, the firmware for these Linksys (50~) I have here is all done by atftp and works fine |
| 20:03.35 | fujin | yeah. |
| 20:03.46 | fujin | Had some issues with a set of Mitel phones a while back, the only way I could get them to upgrade was to use http |
| 20:03.54 | souzha57803 | interesting |
| 20:03.59 | souzha57803 | very quirky |
| 20:04.14 | fujin | Indeed. |
| 20:04.34 | souzha57803 | ah and check this out, so once it finishes it says this |
| 20:04.36 | souzha57803 | 0101003456|app1 |4|00|Loaded application sip.ld successfully, errors 0x20. |
| 20:04.59 | souzha57803 | any idea what that errors is? |
| 20:05.01 | fujin | hrm, I'm not familiar with that error code, either ;() |
| 20:05.06 | souzha57803 | :( |
| 20:05.19 | fujin | is the phone <-> tftp network just a simple lan? |
| 20:05.20 | souzha57803 | and google does nothing for me really |
| 20:05.23 | souzha57803 | yeah |
| 20:05.35 | fujin | That's very odd. |
| 20:05.54 | souzha57803 | I have a feeling its some sort of version issue with sip |
| 20:05.59 | fujin | and you're using atftpd, on the server? |
| 20:06.03 | souzha57803 | yes |
| 20:06.07 | souzha57803 | sip 2.2 |
| 20:07.51 | souzha57803 | maybe I'll just try different versions of the bootrom |
| 20:08.11 | *** join/#asterisk Buhntz (i=Boones@port-212-202-42-223.dynamic.qsc.de) |
| 20:09.03 | fujin | souzha57803: where are you getting the firmware from? |
| 20:09.17 | souzha57803 | I got it from my asterisk installation (trixbox) |
| 20:09.29 | fujin | huh? |
| 20:09.32 | souzha57803 | I'll grab the new ones from the polycom site |
| 20:09.33 | fujin | trixbox includes Polycom firmware? |
| 20:09.36 | fujin | that's, err. |
| 20:09.40 | souzha57803 | part of a package |
| 20:09.41 | *** join/#asterisk generalhan (n=asd@ip67-90-64-2.z64-90-67.customer.algx.net) |
| 20:09.44 | generalhan | hey all ! |
| 20:09.49 | hmmhesays | theres trouble |
| 20:10.22 | souzha57803 | they have a nice module wrapper, yeah, it seemed they were the most recent, but thats a good point |
| 20:10.26 | generalhan | nah, lol |
| 20:10.53 | fujin | souzha57803: that sounds like a dumb idea, tbh |
| 20:11.00 | fujin | I'd install firmware *only* from the vendor. |
| 20:11.27 | generalhan | i was hoping someone could point out my stupidity with this IAX2 "authority not found" issue im having here. i thought i had the aix.conf setup properlly on both servers, but apparently i do not, http://generalhan.pastebin.ca/771099 |
| 20:12.01 | souzha57803 | yeah I had a good experience with the cisco phones I installed |
| 20:12.04 | souzha57803 | guess I got lazy |
| 20:12.14 | souzha57803 | (and I assumed cisco would be much worse) |
| 20:12.16 | generalhan | i can pass calls from ServerB to ServerA but trying from ServerA to ServerB is when i get the authority not found |
| 20:12.34 | [TK]D-Fender | souzha57803: A Trixbox user lazy? NEVER. |
| 20:12.41 | *** part/#asterisk jtexter3 (n=jamest@adsl-70-234-105-253.dsl.tul2ok.sbcglobal.net) |
| 20:13.04 | [TK]D-Fender | exten => _7XXX,1,Dial(IAX2/ServerB/${EXTEN}@putyourremoteendscontexthere) |
| 20:13.09 | *** join/#asterisk axscode (i=axscode@130.17.111.218.klj04-home.tm.net.my) |
| 20:13.16 | [TK]D-Fender | generalhan: ^^^ |
| 20:13.34 | generalhan | [TK]D-Fender: i thought that was what peercontext= was for |
| 20:13.40 | generalhan | let me give it a try anyway ! |
| 20:14.20 | generalhan | [TK]D-Fender: nope ... still "No Authority Found" |
| 20:14.28 | [TK]D-Fender | generalhan: hrm |
| 20:14.54 | generalhan | im just all-star confused as to how they are both setup the same way, and i can get to the local server from the remote, but not the otherway around |
| 20:15.01 | _x86_ | i have a bunch of FXS stations dialing out FXO PSTN lines |
| 20:15.06 | *** join/#asterisk VoicePulse (n=contact@unaffiliated/voicepulse) |
| 20:15.16 | _x86_ | today it seems that some times they are randomly getting dropped mid-conversation |
| 20:15.49 | hmmhesays | I had that problem when I had callprogress=yes |
| 20:16.02 | _x86_ | hangupcause is seemingly always "16" (normal clearing), but I'm not sure if that's entirely reliable, as these are FXO lines riding across a CAS T1 to a channel bank |
| 20:16.15 | [TK]D-Fender | Ah yes... "disconnectmycallswheninconvenientorleastexpected=yes" |
| 20:16.32 | Strom_M | lol=very |
| 20:16.40 | *** join/#asterisk flujan (n=flujan@200-160-115-020.static.spo.ctbc.com.br) |
| 20:16.45 | flujan | [TK]D-Fender: are you here? |
| 20:16.48 | hmmhesays | I believe I have obvserved a new level of geek |
| 20:16.53 | flujan | asterisk stopped with a core dump again... |
| 20:16.54 | hmmhesays | *observed even |
| 20:17.06 | [TK]D-Fender | flujan: ummm.. no? |
| 20:17.06 | *** join/#asterisk keulin (n=cray@ifth-pdcr2.infotheme.fr) |
| 20:17.07 | _x86_ | hmmhesays: callprogress is not enabled |
| 20:17.14 | flujan | ok |
| 20:17.20 | flujan | I will paste the log |
| 20:17.23 | _x86_ | [TK]D-Fender: you think the FXO lines are faulty? |
| 20:17.37 | [TK]D-Fender | _x86_: I'll start with your logic, but we've been over this ;) |
| 20:17.56 | [TK]D-Fender | _x86_: You'd have to try and test the lines completely seperate from * |
| 20:18.16 | _x86_ | [TK]D-Fender: yeah i know, but there is nothing I can do about 1) switching to a PRI, nor 2) putting SIP ATA's in to replace the channel banks |
| 20:18.29 | flujan | [TK]D-Fender : ://pastie.caboo.se/116982 |
| 20:18.35 | flujan | http://pastie.caboo.se/116982 |
| 20:18.45 | flujan | could you please take a look? |
| 20:19.24 | *** join/#asterisk GreggB_ (n=GreggB@66.206.86.107) |
| 20:19.25 | [TK]D-Fender | flujan: I hate debug crap. WTF am I supposed to be loking for in there? |
| 20:20.48 | *** join/#asterisk Steven_elvisda (n=Steven_E@202.47.107.60) |
| 20:21.06 | mvanbaak | it's easy |
| 20:21.12 | mvanbaak | you ran out of file descriptors |
| 20:21.13 | flujan | [TK]D-Fender: the segmentation fault... |
| 20:21.18 | mvanbaak | Too many open files |
| 20:21.20 | flujan | I have this before asterisk stops... |
| 20:21.22 | mvanbaak | there's your hint |
| 20:21.25 | flujan | I dunno what is causing it |
| 20:21.32 | mvanbaak | ulimit |
| 20:21.45 | flujan | mvanbaak: file descritors? |
| 20:21.48 | *** join/#asterisk GreggB (n=GreggB@66.206.86.107) |
| 20:21.54 | mvanbaak | yeah |
| 20:22.03 | mvanbaak | try: man ulimit |
| 20:22.04 | flujan | too many connections with asterisk? |
| 20:22.22 | _x86_ | [TK]D-Fender: there is no other way to debugger this? |
| 20:22.31 | mvanbaak | no, to many file descriptors used by applications in your login class |
| 20:23.03 | [TK]D-Fender | _x86_: I jsut gave you a rock solid test to do... |
| 20:23.29 | mvanbaak | [TK]D-Fender: you handing out fireaxes again ? |
| 20:24.00 | flujan | you are saying that asterisk is opening a lot of files? is that? |
| 20:24.18 | flujan | mvanbaak: I am reading the man page... but how can I avoid this on my machine? |
| 20:24.23 | _x86_ | [TK]D-Fender: i blinked and missed it? |
| 20:24.26 | mvanbaak | flujan: it could be asterisk, but it can also be some other program |
| 20:24.43 | mvanbaak | flujan: what OS are you running this on ? |
| 20:24.50 | flujan | mvanbaak: Linux |
| 20:24.53 | flujan | Slackware 11.0 |
| 20:25.42 | mvanbaak | flujan: can you do: ulimit -n |
| 20:25.47 | *** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net) |
| 20:26.27 | flujan | mvanbaak: http://pastie.caboo.se/116989 |
| 20:26.35 | flujan | and ulimit -n returns 1024 |
| 20:26.54 | _x86_ | [TK]D-Fender: ? |
| 20:27.00 | _x86_ | where is your rock-solid test? |
| 20:28.12 | mvanbaak | flujan: how soon will asterisk die ? |
| 20:28.36 | flujan | 10 times during a day... |
| 20:28.40 | flujan | it is not constant... |
| 20:28.50 | flujan | I could be a large number of sockets on the same machine? |
| 20:28.57 | mvanbaak | do you run AGI scripts or anything ? |
| 20:29.00 | flujan | and I can double the limit to 2048 for instance.. |
| 20:29.06 | flujan | mvanbaak: yeap. a lot of agi. |
| 20:29.14 | orkid | are there free voip(sip)-to-pstn gateways? |
| 20:29.25 | mvanbaak | flujan: then maybe there's the problem |
| 20:29.33 | generalhan | [TK]D-Fender: well i got it to work by putting another context in iax.conf using type=user. even though voip-info says i should be able to do it with type=friend. |
| 20:29.37 | flujan | mvanbaak: what do you suggest? |
| 20:29.44 | generalhan | weird stuff ! |
| 20:29.45 | mvanbaak | are you sure the agi's stop and close all files when a call terminates ? |
| 20:29.54 | flujan | mvanbaak: I hope so... |
| 20:29.59 | mvanbaak | I think not |
| 20:30.03 | [TK]D-Fender | _x86_: [15:17]<[TK]D-Fender>_x86_: You'd have to try and test the lines completely seperate from * <----------- |
| 20:30.12 | mvanbaak | you can try to double the open file limit |
| 20:30.38 | mvanbaak | but if the agi is not terminating correctly that will only make asterisk live some longer before it cores again |
| 20:30.57 | _x86_ | [TK]D-Fender: that's what i was responding to when i said this: 14:22 < _x86_> [TK]D-Fender: there is no other way to debugger this? |
| 20:31.35 | [TK]D-Fender | _x86_: you want to know if the line if fine, don't test with 10 different pieces that could each be broken. |
| 20:32.09 | mvanbaak | flujan: when asterisk is running for some time, you should check with 'ps afx' if everything is fine |
| 20:32.32 | mvanbaak | my first guess is some trouble with the agi scripts |
| 20:33.42 | *** join/#asterisk ajohnstone (n=ajohnsto@cpc3-walt3-0-0-cust880.popl.cable.ntl.com) |
| 20:34.34 | flujan | mvanbaak: ok I will check this out... |
| 20:34.45 | flujan | mvanbaak: thanks so much for the help. |
| 20:34.49 | mvanbaak | no problem |
| 20:35.25 | *** join/#asterisk ghenry (n=ghenry@ghenry.plus.com) |
| 20:37.00 | *** join/#asterisk Steven_elvisda (n=Steven_E@202.47.107.60) |
| 20:39.19 | *** join/#asterisk Aces1up (n=dude@ip70-173-52-152.lv.lv.cox.net) |
| 20:39.21 | Aces1up | hey all |
| 20:39.33 | Aces1up | anyone know a good place to get a 1-800 number from? |
| 20:40.40 | Kobaz | voicepulse |
| 20:42.24 | [TK]D-Fender | Kobaz: ...... so... did you get your money back ;) |
| 20:42.29 | Kobaz | heh |
| 20:43.39 | *** join/#asterisk Lady (n=MaTRoX@88.242.34.76) |
| 20:44.30 | Kobaz | [TK]D-Fender: i'll mail you some cookies if you lead me to something that can make these h323 phones work |
| 20:45.29 | [TK]D-Fender | Kobaz: Lol.... sorry... notmuch I can do for those decrip bastards. |
| 20:45.45 | [TK]D-Fender | decrepit* |
| 20:45.49 | Kobaz | homemade... |
| 20:45.55 | Kobaz | chocolate chip |
| 20:45.58 | *** join/#asterisk MiNdPhUq (n=MiNdPhUq@wsip-24-234-202-14.lv.lv.cox.net) |
| 20:47.13 | *** join/#asterisk ajohnstone (n=ajohnsto@cpc3-walt3-0-0-cust880.popl.cable.ntl.com) |
| 20:52.49 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 20:53.24 | killfill | hi.. |
| 20:53.41 | killfill | im using CURL() to send the callerid name to an url. |
| 20:53.58 | fujin | You're doing it wrong. |
| 20:53.59 | killfill | the problm is that the caller id may be "Name Second_name" |
| 20:54.09 | killfill | how would i urlencode a string in asterisk?.. |
| 20:54.14 | killfill | fujin: yes?.. why |
| 20:54.36 | fujin | Why not just use exec, to wget --post? |
| 20:54.43 | fujin | System, or whatever. |
| 20:54.57 | *** join/#asterisk cypherdelic (n=cypher@p5B27C4CD.dip.t-dialin.net) |
| 20:55.15 | killfill | hm.. |
| 20:55.18 | mvanbaak | fujin: are you using fopen in your agi script ? |
| 20:55.23 | fujin | huh? |
| 20:55.26 | fujin | I don't even *use* agi. |
| 20:55.36 | fujin | Everything is AEL here. |
| 20:55.59 | mvanbaak | oh wait |
| 20:56.00 | killfill | fujin: is CURL mainly a Execute(curl)?. |
| 20:56.04 | mvanbaak | it was flujan |
| 20:56.06 | mvanbaak | dammit |
| 20:56.13 | fujin | killfill: I don't know, sorry |
| 20:56.24 | killfill | im afraid of Exec... dont know why |
| 20:56.32 | De_Mon | TrySystem? |
| 20:56.38 | De_Mon | oh right agi |
| 20:57.53 | *** join/#asterisk axscode (n=axscode@130.17.111.218.klj04-home.tm.net.my) |
| 20:59.44 | flujan | mvanbaak: No I am using ragi. |
| 20:59.56 | mvanbaak | so no fopen ? |
| 21:00.15 | flujan | mvanbaak: I think that now... sometimes to record a voicemail and stufff. |
| 21:00.31 | mvanbaak | what's the name of the agi script ? |
| 21:00.49 | mvanbaak | oh wait |
| 21:00.51 | mvanbaak | not important |
| 21:01.00 | mvanbaak | in the dir with your agi scripts use this: |
| 21:01.28 | mvanbaak | grep "fopen" * | wc -l > ding; grep "fclose" * | wc -l > foo; if [ '`cat foo`' -ne 'cat `ding`' ]; echo "Buugggg"; fi; |
| 21:04.15 | sheldonh | safe_asterisk is great for dealing with filedescriptors :) |
| 21:05.14 | mvanbaak | sheldonh: that's evil |
| 21:05.27 | sheldonh | really? |
| 21:05.53 | sheldonh | i try to keep our debian hosts pristine, and if i use safe_asterisk, i don't need to touch init scripts |
| 21:05.53 | mvanbaak | it's way better to fix the bugs in your agi scripts instead of masking them by starting asterisk the moment it coredumps |
| 21:05.58 | *** join/#asterisk heison (n=heison@67.110.80.103.ptr.us.xo.net) |
| 21:06.05 | sheldonh | oh, fair enough |
| 21:06.28 | sheldonh | i meant more that safe_asterisk bumps maxfds |
| 21:06.51 | sheldonh | the conversation's moved on a bit since file descriptors came up |
| 21:06.52 | mvanbaak | it sets it to unlimited right ? |
| 21:09.06 | *** join/#asterisk axscode (i=axscode@130.17.111.218.klj04-home.tm.net.my) |
| 21:10.22 | dan__t | So, * complains about congestion with my iax2 peer, and then finally gives up saying all circuits are busy. |
| 21:10.28 | dan__t | Does that mean my iax2 provider is having issues? |
| 21:10.58 | mvanbaak | dan__t: did it ever work ? |
| 21:11.07 | dan__t | Yep, was working last night just fine. |
| 21:11.31 | mvanbaak | do you have 'qualify=yes' on them ? |
| 21:12.11 | halconnen | I just glanced at chat and thought qualify said girlfriend |
| 21:12.15 | halconnen | sometimes I amuse myself |
| 21:12.23 | mvanbaak | lol |
| 21:12.25 | dan__t | I do, per AsteriskNOW's configuration. |
| 21:12.30 | dan__t | haha |
| 21:12.45 | mvanbaak | dan__t: on the cli type: iax2 show peers |
| 21:13.13 | dan__t | Yeah, I see it listed |
| 21:13.20 | *** join/#asterisk lemanal (n=lemanal@wifi-224-227.sc07.org) |
| 21:13.23 | mvanbaak | are they reachable ? |
| 21:14.47 | dan__t | Er, it doesn't say otherwise |
| 21:16.43 | dan__t | SAys the peer is Unmonitored |
| 21:17.55 | dan__t | So what do ya think? |
| 21:18.05 | mvanbaak | so there's no 'qualify=yes' for it |
| 21:19.22 | dan__t | Er, sorry, there is no qualify=yes. |
| 21:19.26 | dan__t | I'm using AsteriskNOW. |
| 21:19.28 | *** join/#asterisk alejandro (n=asanchez@kde/developer/alejandro) |
| 21:19.38 | alejandro | http://pastebin.com/m333a9a13 |
| 21:19.58 | alejandro | I'm configuring an IVR but i dont understand why WaitExten or settimeout response/digit is not working.. |
| 21:20.05 | alejandro | someone sees something wrong in this dial plan ? |
| 21:20.10 | sheldonh | mvanbaak: no, to half the system max |
| 21:20.40 | *** join/#asterisk callguy (n=callguy@pool-71-255-162-167.bstnma.east.verizon.net) |
| 21:20.43 | sheldonh | if you want to understand asterisk, start with girlfriend=no |
| 21:20.45 | fujin | What are you expecting it to do? |
| 21:20.49 | fujin | alejandro^^. |
| 21:21.08 | alejandro | fujin: I'm using an SPA 3108 (FXO and FXS) |
| 21:21.16 | fujin | That's not what I asked. |
| 21:21.19 | alejandro | and the dial plan in the FXO jumps to 123 extension in Asterisk |
| 21:21.27 | alejandro | well, there i've an IVR |
| 21:21.44 | alejandro | and i want to goto demo-galp if i press '1' digit |
| 21:21.45 | fujin | yes, I can see the beginnings of an IVR, although it won't do much |
| 21:21.47 | alejandro | but it's not working |
| 21:22.02 | fujin | heh. |
| 21:22.09 | fujin | # |
| 21:22.10 | fujin | ;exten => s,8,WaitExten(9) |
| 21:22.15 | fujin | are you aware you've got that commented? |
| 21:22.19 | alejandro | yes, i made a lot of tests :) |
| 21:22.25 | alejandro | but uncommented it doenst work.. |
| 21:22.41 | fujin | I see. |
| 21:22.46 | fujin | Have you checked your DTMF support? |
| 21:22.52 | fujin | (does voicemail work?) |
| 21:22.52 | dan__t | mvanbaak, what else might I be able to look at? |
| 21:23.07 | *** join/#asterisk ghenry (n=ghenry@ghenry.plus.com) |
| 21:23.12 | mvanbaak | dan__t: iax2 debug |
| 21:23.34 | dan__t | Yeah, I have that on, just complains about the peer. Let me do a full cycle and pastebin it. |
| 21:23.35 | dan__t | brb |
| 21:25.38 | ApolloDS | anyone here knows the future of dundi? |
| 21:26.15 | dan__t | mvanbaak, http://pastebin.ca/771211, if you would be so kind as to take a peek. |
| 21:26.29 | [TK]D-Fender | ~8all Does dundi have a future? |
| 21:26.34 | [TK]D-Fender | ~8ball Does dundi have a future? |
| 21:26.35 | jbot | Sure. Yeah, exactly. |
| 21:26.49 | [TK]D-Fender | Completely credibly reference! |
| 21:26.52 | dan__t | A few MTR's show nothing out of the ordinary to my iax2 peer. |
| 21:28.53 | *** join/#asterisk blq (n=Bl@dslb-088-065-173-239.pools.arcor-ip.net) |
| 21:29.53 | alejandro | fujin: how i can check DTMF support ? |
| 21:30.06 | fujin | try voicemail, like I said |
| 21:30.16 | mvanbaak | dan__t: well, I would start with adding qualify=yes to your peer |
| 21:32.50 | *** join/#asterisk [1]hi365 (n=hi365@mail.pcgeula.co.il) |
| 21:35.12 | *** join/#asterisk syneus (n=syneus@host246-63-dynamic.21-87-r.retail.telecomitalia.it) |
| 21:37.23 | *** join/#asterisk tripps (n=sean@72.20.150.196) |
| 21:37.41 | tripps | ~book |
| 21:37.42 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf |
| 21:38.07 | *** part/#asterisk lirakis (n=lirakis@65.200.191.253) |
| 21:38.19 | *** join/#asterisk hesco (n=hesco@c-76-17-99-50.hsd1.ga.comcast.net) |
| 21:39.58 | *** join/#asterisk axscode (n=axscode@130.17.111.218.klj04-home.tm.net.my) |
| 21:42.36 | *** join/#asterisk FremWork (n=freman@brdr-gw-01.benon.com) |
| 21:42.52 | dan__t | mvanbaak, not sure how far I can go with that, this being AsteriskNOW and all. |
| 21:42.56 | dan__t | I'll check it out. |
| 21:43.07 | *** join/#asterisk asteriskmonkey (n=philip@69.77.169.14) |
| 21:43.30 | JT | dan__t: this isn't the asterisknow support channel |
| 21:44.22 | dan__t | Oh, sorry, didn't realize I said it was. |
| 21:45.22 | *** join/#asterisk MtJB (n=warthawg@cpe-24-28-83-165.austin.res.rr.com) |
| 21:45.47 | MtJB | didn't digium used to offer a little pci card for sip to pots stuff? |
| 21:47.16 | Qwell | MtJB: sip to pots? |
| 21:47.38 | MtJB | Qwell where asterisk can answer your pots line |
| 21:47.46 | MtJB | and dial out on it |
| 21:47.52 | Qwell | tdm400p |
| 21:48.01 | MtJB | thank you, sir |
| 21:48.21 | *** join/#asterisk blq (n=Bl@dslb-088-067-044-145.pools.arcor-ip.net) |
| 21:49.10 | JT | dan__t: you are implying that it is, by asking questions in here whilst using asterisknow |
| 21:49.30 | JT | MtJB: it's not sip to pots |
| 21:49.30 | fujin | lol @ asterisknow |
| 21:49.37 | JT | MtJB: it's pots to asterisk |
| 21:49.47 | dan__t | I was asking an Asterisk question, which happened to resolve to possibly being an AsteriskNOW issue. I haven't asked further than that. |
| 21:49.50 | MtJB | JT i am very stupid about all of this, sorry |
| 21:49.51 | fujin | well, pots to zapata, really |
| 21:50.21 | fujin | dan__t: generally trixbox/freepbx/asterisknow users are rattled a little in here. |
| 21:50.29 | fujin | You're generally getting more problems than you bargain for, with one of those. |
| 21:50.30 | MtJB | are there other cards, or is that the best only choice for me? |
| 21:50.41 | *** join/#asterisk tdi (n=tdi@gvf90.internetdsl.tpnet.pl) |
| 21:50.58 | *** join/#asterisk axscode (n=axscode@130.17.111.218.klj04-home.tm.net.my) |
| 21:50.59 | *** join/#asterisk kambei (n=kambei@unaffiliated/kambei) |
| 21:51.05 | halconnen | dan__t: if you cant write asterisk from scratch, you dont deserve help like the rest of us |
| 21:51.07 | dan__t | I understand, fujin. Didn't expect anything less :) |
| 21:51.37 | dan__t | Thanks for the starting point, mvanbaak. |
| 21:52.40 | fujin | I'd very much recommend spending a few hours reading the book and building an asterisk system from scratch |
| 21:52.42 | MtJB | <PROTECTED> |
| 21:52.46 | fujin | You'll learn lots more, and get support when you're stuck :) |
| 21:53.12 | dan__t | Yea, I've got the book. Need to dust it off. |
| 22:01.09 | alejandro | fujin: voicemail works |
| 22:01.31 | fujin | and is dtmf from your pstn -> asterisk working? |
| 22:02.33 | alejandro | i'm testing directly from analogic phone -> asterisk with 600 extension |
| 22:03.03 | halconnen | I'm trying to blind transfer calls using asterisk's builtin # code, but I always have to press the # key multiple times before it recognizes it. Its a Polycom SP 501. |
| 22:03.36 | fujin | alejandro: yes, but pstn -> asterisk is different |
| 22:03.42 | fujin | are you running over a zap chan or sip? |
| 22:03.55 | alejandro | sip, it's a SPA 3102 |
| 22:04.17 | fujin | ok, and when you call the pstn number that the spa3102 is plugged into |
| 22:04.18 | fujin | does dmtf work? |
| 22:04.31 | alejandro | i've made a test, and yes, it seems to work |
| 22:04.41 | alejandro | at least it transfers to demo-galp with 1111 |
| 22:05.27 | _x86_ | ok, got a fun little problem.... |
| 22:05.53 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 22:06.24 | fujin | 1111? |
| 22:06.31 | fujin | sounds like it's not interpreting DTMF correctly. |
| 22:07.18 | alejandro | well, now 1111 is the demo-galp extension |
| 22:07.25 | alejandro | so it's working fine from pstn |
| 22:07.34 | alejandro | but not inside sip->asterisk |
| 22:07.43 | fujin | yes |
| 22:07.45 | fujin | check your dtmf |
| 22:07.59 | _x86_ | i have 18 phone lines going into a rhino FXO channel bank, which comes in via CAS T1 to a sangoma A102D-x card on my asterisk box |
| 22:08.20 | _x86_ | when people try to dial out, it seems like asterisk is happy in trying to use one of those lines, even though all 18 may be busy |
| 22:08.47 | _x86_ | i've got about 20 people and 18 lines |
| 22:09.16 | _x86_ | although the T1 is capable of carrying 24 voice channels (as it is CAS and not PRI), zapata.conf is only setup to use 18 of them |
| 22:09.30 | _x86_ | why does asterisk think it has a spare line when it does not? |
| 22:10.08 | Netgeeks | your using zap groups to dial? Zap/g1 or so? |
| 22:10.29 | dan__t | Got it working, btw. |
| 22:10.44 | dan__t | DNS failed last night, guess that messed up more things than I thought. |
| 22:10.59 | halconnen | its always dns |
| 22:11.23 | *** join/#asterisk marc7 (n=marc@64.46.14.64.novuscom.net) |
| 22:11.25 | dan__t | DNS is the bastard child of all things unholy. |
| 22:11.28 | _x86_ | Netgeeks: that's correct |
| 22:11.44 | halconnen | I can't wait for Internet 2.0 |
| 22:11.48 | halconnen | no dns |
| 22:12.07 | jameswf | I have internet 3.0 on a floppy disk in the drawer |
| 22:12.09 | _x86_ | halconnen: don't know where you got that impression ;) |
| 22:12.13 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 22:12.14 | halconnen | can I get a copy? |
| 22:12.24 | halconnen | ;) |
| 22:12.35 | halconnen | anyone remember that song, dont copy that floppy? |
| 22:12.53 | jameswf | you can only send internet 3.0 to others with internet 3.0 or greater |
| 22:13.03 | _x86_ | halconnen: i used to work for a company that was the internet2 hub for the entire metro area (10gig uplink to chicago, ~20 1gig connections to various research organizations around the city) |
| 22:13.16 | halconnen | darn backwards compatibility |
| 22:13.22 | Netgeeks | _x86_ I don't know why asterisk is having problems counting your channels. I'd try using group functions to check to see if a channel is free |
| 22:13.36 | _x86_ | Netgeeks: what do you mean? |
| 22:13.42 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 22:14.04 | Netgeeks | I'd use Set(GROUP()=something) |
| 22:14.16 | _x86_ | what's that do? |
| 22:14.39 | Netgeeks | and then ExecIf(GROUP(something) > 18).... |
| 22:14.51 | *** join/#asterisk stybba (n=stybba@190.10.0.136) |
| 22:14.58 | stybba | hi all |
| 22:15.19 | Netgeeks | _x86_ http://www.voip-info.org/wiki/index.php?page=Asterisk+func+group |
| 22:15.35 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 22:15.52 | Netgeeks | oops that would be ExecIF(${GROUP(something)}>18)... |
| 22:16.58 | JT | _x86_: sounds similar to the speed of connections most carriers already have in place for Internet traffic |
| 22:18.07 | Netgeeks | JT: yeah, but internet2 was set up in the late 90s |
| 22:18.15 | Netgeeks | that kind of bandwidth was HUGE back then |
| 22:18.51 | Netgeeks | heck, MCI's major midwest hub was a bunch of cisco 7500 series routers in a FDDI ring |
| 22:19.15 | Netgeeks | willow springs... if anyone recalls the name in reverse dns traceroute lookups |
| 22:19.29 | nestAr | hehe |
| 22:19.30 | *** join/#asterisk jer (n=jtregunn@unaffiliated/jer) |
| 22:20.21 | _x86_ | JT: they use advanced multicasting / anycasting to get network utilization on the uplinks to be the lowest possible, which speeds up site to site traffic significantly |
| 22:21.16 | _x86_ | Netgeeks: *CLI> show function group |
| 22:21.26 | _x86_ | Netgeeks: no function by that name registered |
| 22:21.36 | Netgeeks | you using 1.2? |
| 22:21.49 | _x86_ | ah wait, it's case sensitive |
| 22:22.04 | _x86_ | yeah, one of my last sites on 1.2 |
| 22:23.04 | Netgeeks | okay, I don't know if in 1.2 it's a function or an application, the functions are GROUP, GROUP_COUNT.. the applications are SetGroup and Check |
| 22:23.05 | Netgeeks | G |
| 22:23.08 | Netgeeks | CheckGroup |
| 22:23.10 | outtolunc | it is 'show function GROUP' btw |
| 22:23.11 | Netgeeks | bad keyboard |
| 22:23.25 | _x86_ | outtolunc: figured that out :) |
| 22:23.29 | outtolunc | k |
| 22:23.48 | Netgeeks | not saying group is going to get you anywhere, tho, asterisk might actually think one of the channels is free and using group is just going to verify that if thats the case |
| 22:23.58 | Netgeeks | if group works, then there is a BUG running around somewhere |
| 22:23.59 | _x86_ | Netgeeks: hmm, well analog phones are on one group, outbound T1 to the FXO channel bank is another group.... how do i tell it which group i want a count for? |
| 22:24.32 | fujin | I always use func_devstate |
| 22:24.38 | fujin | not sure if it'll support checking the state of your zap channels |
| 22:24.43 | Netgeeks | Set(GROUP()=g1) Set(GROUP()=g2) you can use any group name you want it's just a tag to |
| 22:24.43 | fujin | or if it'll just report it incorrectly |
| 22:25.15 | Netgeeks | tag to.... I lost my thought there... it's just a tag |
| 22:25.17 | _x86_ | Netgeeks: and that will tell asterisk which group to count on? |
| 22:25.23 | Netgeeks | correct |
| 22:25.30 | _x86_ | ok cool |
| 22:25.59 | Netgeeks | Verbose(1, Group g1 has ${GROUP_COUNT(g1)} current active channels) |
| 22:26.18 | Netgeeks | Verbose might not be in 1.2 tho... not sure |
| 22:26.42 | Netgeeks | :s/Verbose(1,/NoOp(/ |
| 22:26.46 | Netgeeks | would work too |
| 22:31.47 | _x86_ | nice |
| 22:31.51 | _x86_ | group show channels :) |
| 22:32.55 | *** join/#asterisk marc77 (n=marc@64.46.14.64.novuscom.net) |
| 22:33.41 | _x86_ | thanks |
| 22:35.11 | *** join/#asterisk Steven_elvisda_ (n=Steven_E@202.47.107.60) |
| 22:35.25 | De_Mon | Hmm, using Park() announces to the caller where they were parked, how do I stop that? |
| 22:36.54 | [TK]D-Fender | De_Mon, Why would you not want to know that? |
| 22:38.27 | nestAr | seems kinda important to know. |
| 22:38.27 | *** join/#asterisk marlow (n=marlow@loke.sca.airwire.ie) |
| 22:39.11 | De_Mon | [TK]D-Fender remember our conversation from a day or two ago, where I am parking a call and then picking it back up thru a local extension |
| 22:39.38 | [TK]D-Fender | De_Mon, how were you targeting who to pick up again? |
| 22:40.06 | De_Mon | PARKINGEXTEN |
| 22:40.52 | De_Mon | astdb is keeping track of parking spots in use and picks a number, then the local extension calls and picks up that extension |
| 22:41.40 | [TK]D-Fender | De_Mon, Oh yeah... some sort of insanity... dunno how to bypass, but you could jsut blind transfer to it you know... |
| 22:42.17 | *** join/#asterisk lemanal (n=lemanal@wifi-224-227.sc07.org) |
| 22:42.32 | De_Mon | blind transfer to a parking extension?(Transfer app right) |
| 22:44.04 | [TK]D-Fender | De_Mon, blind instead of attended. |
| 22:46.01 | *** part/#asterisk sheldonh (n=sheldonh@66.219.59.32) |
| 22:47.30 | *** join/#asterisk axscode (i=axscode@130.17.111.218.klj04-home.tm.net.my) |
| 22:47.47 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 22:47.48 | *** mode/#asterisk [+o anthm] by ChanServ |
| 22:48.14 | asteriskmonkey | anyone intamite with asterisk voicemail behaviour |
| 22:49.03 | asteriskmonkey | i want to centralize voicemail using nfs mounts but affraid it will kack out the system if the nfs mount fails. any pointers, proof of concepts? |
| 22:49.25 | De_Mon | asteriskmonkey use imap or odbc instead |
| 22:50.07 | asteriskmonkey | odbc is a nightmare large setups have had issues with it. |
| 22:50.19 | asteriskmonkey | is there a good imap/voicemail refference I could read? |
| 22:53.00 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 22:55.38 | De_Mon | [TK]D-Fender the only transfer app I see is Transfer |
| 22:56.01 | [TK]D-Fender | De_Mon, I'm referring to you transferring the call to park from your PHONE. |
| 22:56.36 | De_Mon | thats no good, this is from IVR |
| 22:57.12 | *** join/#asterisk _GiGi_ (i=gigi@disc.more.pl) |
| 22:58.45 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 22:59.06 | *** part/#asterisk MiNdPhUq (n=MiNdPhUq@wsip-24-234-202-14.lv.lv.cox.net) |
| 22:59.24 | jameswf | enter any 9 digit prime palindrome to continue... |
| 23:02.07 | *** join/#asterisk jm|home (n=jm|home@zen.jamiem.com) |
| 23:04.05 | *** join/#asterisk JayTee52 (n=jforde05@c-69-137-243-25.hsd1.in.comcast.net) |
| 23:04.44 | JayTee52 | has anyone here successfully setup a working SIP trunk between Asterisk and a sipX server? |
| 23:06.32 | fujin | There's no such thing as a sip trunk. Next question. |
| 23:10.24 | JayTee52 | ok, well I was using a "walkthrough" to setup asterisk to Exchange 2007 Unified Messaging and it uses sipX as a gateway since asterisk doesn't support SIP/TCP and they refer to it as a trunk although it's setup in the extensions.conf file |
| 23:10.56 | *** join/#asterisk syneus (n=syneus@host246-63-dynamic.21-87-r.retail.telecomitalia.it) |
| 23:11.07 | fujin | What's sipX? |
| 23:11.14 | fujin | That sounds like a horrid solution. |
| 23:11.19 | *** join/#asterisk axscode (n=axscode@130.17.111.218.klj04-home.tm.net.my) |
| 23:11.22 | fujin | What does exchange 2007 unified messaging do? |
| 23:11.28 | Qwell | it unified messaging. |
| 23:11.31 | Qwell | duh |
| 23:11.39 | Qwell | unifies too. I fail. |
| 23:11.43 | fujin | heh. |
| 23:12.40 | De_Mon | fujin it hooks into exchange and will let you read email, cerate appointments, reschedule appointments etc |
| 23:12.53 | De_Mon | you can also use it as a voicemail server |
| 23:13.11 | De_Mon | need sipTCP workin to do all that though, which has been slow going |
| 23:13.11 | *** join/#asterisk asanchez_ (n=asanchez@130.pool85-53-165.dynamic.orange.es) |
| 23:13.25 | Qwell | tell MS to add support for UDP. |
| 23:13.35 | *** join/#asterisk jarod14 (n=jarod14@ns2.viatelecom.com) |
| 23:13.36 | Qwell | They're violating the RFCs by not doing so. (yes, I realize that we are also) |
| 23:13.55 | De_Mon | I've got a better chance of getting asterisk TCP support than getting MS to do anything |
| 23:14.27 | De_Mon | sipX and openser both advertise being able to convert from sipUDP to sip TCP, but i've not been very successful |
| 23:15.40 | De_Mon | uccgI've got a cheezy windows based tcp/udp proxy from m-networks.net but they stopped development and disappeared on me |
| 23:16.00 | Qwell | openser should do udp>tcp just fine |
| 23:16.30 | fujin | 12:12:40) (De_Mon) fujin it hooks into exchange and will let you read email, cerate appointments, reschedule appointments etc <- isn't that what exchange does, out of the box? |
| 23:16.34 | fujin | o_0 |
| 23:16.42 | De_Mon | Im sure it will work, but getting a working config requires rocket science or something! |
| 23:16.55 | De_Mon | fujin not over a phone, no |
| 23:17.01 | fujin | oh, a phone. right. |
| 23:17.04 | JayTee52 | sipX supports both TCP and UDP and the walkthrough I used from http://blog.lithiumblue.com/2007/10/accessing-exchange-2007-unified.html uses it as a gateway to do the transform from UDP to TCP. |
| 23:17.06 | Qwell | it's not rocket surgery |
| 23:17.07 | De_Mon | :P |
| 23:17.24 | [TK]D-Fender | Qwell, I love playing doctor! |
| 23:17.57 | JayTee52 | rocket surgery? LOL, "I'm sorry doctor but this Atlas 5 needs an appendectomy STAT!" |
| 23:18.00 | De_Mon | JayTee52 I didn't try sipX |
| 23:18.46 | De_Mon | I think my biggest problem with openser was trying to run * and openser on the same box, it made a complicated problem more complicated |
| 23:18.58 | *** join/#asterisk xtr (n=94752345@216.19.191.191.novuscom.net) |
| 23:19.29 | De_Mon | it was half working, dont remember why I gave up on it ;) |
| 23:19.30 | JayTee52 | our previous network engineer who was great at both Windows Server, Exchange and Linux in general setup our Asterisk PBX to route calls to Exchange using the same walkthrough. The sipX server was running as a VM in VMWare and everything worked. Then he quit and when we went to try it again for a demo it wouldn't work. The VM was hosed. |
| 23:19.39 | Qwell | De_Mon: when are users every going to need to call in to schedule an appointment? |
| 23:20.02 | Qwell | and you can already save voicemail to imap |
| 23:20.03 | fujin | putting exchange anywhere near asterisk sounds like a stupid idea |
| 23:20.06 | JayTee52 | I recreated a new VM using the same walkthrough he did but now it won't communicate. Nothing's changed in the Asterisk configs as near as I can tell since he left. |
| 23:20.20 | De_Mon | Qwell airplane is late you've got a meeting and no digital service on your blackberry/windowsmobile device |
| 23:20.25 | Qwell | fujin: s/near asterisk // |
| 23:20.37 | fujin | heh. I believe it has its uses. |
| 23:20.37 | De_Mon | or, you're in traffic and dont want to type on your mobile device to say you'll be 15min late |
| 23:20.46 | De_Mon | those are the two examples they gave in the UM demo |
| 23:20.52 | fujin | in an office environment, with pocketpc devices / office 2007 |
| 23:20.53 | fujin | it's hadny. |
| 23:20.55 | fujin | handy. |
| 23:20.58 | Qwell | okay, and how does this help that situation? |
| 23:21.03 | fujin | exchange over the air to my pocketpc = awesome |
| 23:21.38 | De_Mon | instead you say 'call um' open appointment, next, I'll be late, 15 minuites, done |
| 23:22.01 | De_Mon | instead of getting out the laptop, finding wifi, etc |
| 23:22.08 | Qwell | exchange has no method of saying you will be at a meeting at a certain time - just yes or no you will not be there |
| 23:22.52 | De_Mon | if you're the meeting um owner, you can reschedule, or just send out a general email to all attendees i think, duno really. |
| 23:23.16 | De_Mon | I don't use the stuff just responsible for making it work ;) |
| 23:23.20 | *** join/#asterisk craigk (n=ckowald@58.174.122.198) |
| 23:23.22 | Qwell | yeah, that sounds pretty pointless |
| 23:23.45 | De_Mon | I'd be more interested in using it as my voicemail server |
| 23:23.49 | fujin | pick up phone -> call receptionist |
| 23:23.50 | Qwell | you can do that today |
| 23:23.58 | Qwell | without any extra crap |
| 23:24.03 | fujin | Heh, our windows weeny yelled at me for using Exchange as the voicemail storage. |
| 23:24.06 | De_Mon | right now * just emails voicemails to the mailbox |
| 23:24.08 | JayTee52 | Unified Messaging works with Communications Server which allows integrating Instant Messaging, Voicemail, E-Mail, VOIP or TDM phones (through a TDM/VOIP gateway) and cell phones. The presence function lets people keep in touch no matter where they are and the current prefered method of communications. |
| 23:24.10 | fujin | "big wav files will slow thd b right down!" |
| 23:24.12 | Qwell | De_Mon: imap storage |
| 23:24.27 | Qwell | fujin: yeah... |
| 23:24.35 | Qwell | ask them how big a single MS word doc is. |
| 23:24.43 | Qwell | no images. Just an empty word doc |
| 23:24.45 | fujin | JayTee52: I'm not sure anyone here will even slightly support you. |
| 23:24.58 | fujin | Qwell: office 2007 msword docs (.docx) are very small, as they're XML. |
| 23:25.01 | Qwell | it'll be about 10x the size of a 1 minute VM |
| 23:25.03 | Qwell | no! |
| 23:25.09 | Qwell | they are *HUGE* because they're XML. |
| 23:25.16 | De_Mon | Qwell I'm not giving * admin access thru imap to exchange |
| 23:25.30 | Qwell | there is SO much redundancy in them, it isn't even funny. I've gone through them |
| 23:25.31 | *** join/#asterisk axscode (i=axscode@130.17.111.218.klj04-home.tm.net.my) |
| 23:25.32 | JayTee52 | wasn't looking for support for Exchange or sipX. Just was curious if anyone else had tried it and if they'd had problems with the Asterisk to sipX communications. |
| 23:25.35 | fujin | 12.0 KB (12,288 bytes) |
| 23:25.40 | Qwell | I took one, spent 2 days trimming the BS from it. 35k |
| 23:25.40 | fujin | empty .docx |
| 23:26.06 | Qwell | make a simple formatting change and save it |
| 23:26.08 | Qwell | easy 1k |
| 23:26.27 | JayTee52 | fujin, I hate MS's .docx implementation. |
| 23:26.27 | Qwell | it's so bloated it isn't even funny. BUT, that's beside the point :p |
| 23:26.37 | fujin | what's wrong with it? It's better than .doc |
| 23:27.21 | De_Mon | JayTee52 we're just talking, not to you! |
| 23:27.23 | JayTee52 | it's not truly "open" just like .doc but it also is just another way for MS to make people upgrade to Office 2007 |
| 23:27.52 | [hC] | and to tack something like "XML" on to the mix to make people think its open and shareable |
| 23:28.06 | Qwell | De_Mon: you don't need to, afaik |
| 23:28.15 | Qwell | just an account that can read/write to a certain directory in users mailboxes |
| 23:28.22 | Qwell | (doesn't even need to be in INBOX) |
| 23:28.29 | *** join/#asterisk ccaron (n=cedric@85-218-16-115.dclient.lsne.ch) |
| 23:28.36 | De_Mon | no, but it has to read/write to everyones mailbox |
| 23:28.41 | Qwell | not inbox |
| 23:28.46 | De_Mon | mailbox |
| 23:28.54 | Qwell | to a directory in the mailbox |
| 23:29.00 | De_Mon | exchange doesnt do permissions like that afaik |
| 23:29.02 | Qwell | it doesn't need access to do anything else |
| 23:29.05 | Qwell | then exchange is retarded |
| 23:29.06 | De_Mon | either you can access the mailbox or you cant |
| 23:29.12 | Qwell | and yes, it can |
| 23:29.23 | nestAr | When a parked call times out, Asterisk attempts to dial the Zap channel the call is on, not the extension it was transfered from |
| 23:29.24 | Qwell | because I can share a specific folder with another user |
| 23:29.25 | JayTee52 | I'm curious about one thing regarding MS. If Linux violates over 235 patents Microsoft holds then why doesn't this Ubuntu system crash a lot like some Windows systems I have? |
| 23:29.27 | nestAr | anyone know how to correct that? |
| 23:29.41 | Qwell | JayTee52: because one of the patents isn't the random crash feature |
| 23:29.50 | JayTee52 | Qwell, :-) |
| 23:29.57 | fujin | 235 patents? |
| 23:30.01 | fujin | that's just shit. |
| 23:30.01 | De_Mon | Qwell and that other user can access your shared folder using imap? |
| 23:30.02 | *** join/#asterisk macros73_ (n=cs@c-24-3-246-27.hsd1.mn.comcast.net) |
| 23:30.05 | Qwell | fujin: so Ballmer claims |
| 23:30.09 | fujin | ballmer can suck a cock |
| 23:30.10 | fujin | eof |
| 23:30.17 | JayTee52 | fujin, hey! I'm just saying what Ballmer claims |
| 23:30.19 | Qwell | De_Mon: oh, dunno |
| 23:30.26 | Qwell | probably though :p |
| 23:30.33 | De_Mon | let me know if you figure out how |
| 23:30.39 | JayTee52 | Ballmer is a fat sweaty balding troll with permanent pit stains on all his shirts. |
| 23:30.42 | Qwell | don't hold your breath. |
| 23:30.55 | De_Mon | no problem |
| 23:31.03 | Qwell | pretty sure it can though |
| 23:31.30 | De_Mon | I'd be concerned about some stupid bug that deletes random email when access thru imap |
| 23:31.46 | Qwell | again - you don't need to use INBOX |
| 23:32.02 | Qwell | and do you *really* trust Exchange not to randomly delete emails in the first place? |
| 23:32.10 | Qwell | of course not - that's why you have hourly backups |
| 23:32.13 | [hC] | this seems like a moot point |
| 23:32.14 | [hC] | heh |
| 23:37.06 | *** join/#asterisk axscode (n=axscode@130.17.111.218.klj04-home.tm.net.my) |
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| 23:47.09 | *** part/#asterisk JayTee52 (n=jforde05@c-69-137-243-25.hsd1.in.comcast.net) |
| 23:47.10 | *** join/#asterisk obnauticus (n=obnautic@c-71-236-181-11.hsd1.or.comcast.net) |
| 23:49.20 | obnauticus | What is a good SIP/IAX(2) termination service where I am not tied to a contract |
| 23:49.26 | obnauticus | it is `pay as you go' if you will. |
| 23:49.32 | obnauticus | is voipjet that way? Like if i run out does it automatically credit? |
| 23:49.47 | *** join/#asterisk lemanal (n=lemanal@wifi-224-227.sc07.org) |
| 23:50.04 | *** join/#asterisk Winkie (i=sd@87-194-8-125.bethere.co.uk) |
| 23:52.25 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 23:54.10 | De_Mon | iirc voipjet stopped routing calls when the balance ran out, that may just be the plan we had selected though |
| 23:54.31 | obnauticus | Is there any other good termination service that you know of? |
| 23:54.38 | obnauticus | Is VoipJet good? |
| 23:54.46 | xheliox | obnauticus: "good" is a relative term. |
| 23:54.50 | De_Mon | it was okay for us, your milage will vary |
| 23:54.59 | obnauticus | The quality to price ratio |
| 23:55.03 | xheliox | obnauticus: I rate my itsp's on a scale of "shit to tolerable" :) |
| 23:55.03 | obnauticus | is it balanced... |
| 23:55.07 | De_Mon | s/milage/mileage/ |
| 23:55.18 | obnauticus | I won't use it very often. |
| 23:55.21 | obnauticus | Only for myself. |
| 23:55.28 | obnauticus | But i want to pay as I go. |
| 23:55.33 | *** join/#asterisk Belgarath (i=belgarat@banda.pl) |
| 23:55.58 | fujin | WXC in NZ has been very good, but we have dedicated fibre to them |
| 23:56.08 | obnauticus | lucky |
| 23:56.09 | obnauticus | >: | |
| 23:56.11 | obnauticus | I need a trunk |
| 23:56.19 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 23:56.22 | obnauticus | I don't have what you call `extreme' bandwidth. |