| 00:09.44 | *** join/#asterisk micols (n=mio@rlogin.dk) |
| 00:11.14 | *** join/#asterisk micols (n=mio@rlogin.dk) |
| 00:16.07 | *** join/#asterisk MikeJ (n=MikeJ@freeswitch/developer/mikej) |
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| 00:19.30 | *** join/#asterisk SwK (n=SwK@freeswitch/developer/swk) |
| 00:20.17 | areay | i've got a sip trunk connected and when i dial the number from another phone it's busy...but i can't make or receive any external calls through asterisk |
| 00:25.51 | *** part/#asterisk jcoffi (n=jcoffi@75.147.155.89) |
| 00:27.31 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 00:27.31 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 00:30.51 | *** join/#asterisk nemik (n=cyanact@c-67-173-76-34.hsd1.il.comcast.net) |
| 00:32.00 | nemik | so would be nice if there was another Festival()-like command that executed a script that returned text, kinda like AGI instead of just taking in text to the Festival cmd |
| 00:34.09 | *** join/#asterisk MrNaz (n=mrnaz@ppp118-208-237-246.lns10.mel6.internode.on.net) |
| 00:36.24 | Ritzerisk | anyoneknow of good sip providers |
| 00:41.06 | lmadsen | Ritzerisk: unlimitel is awesome |
| 00:42.08 | *** join/#asterisk Kisu (n=k@daniel1117.broker.freenet6.net) |
| 00:43.34 | *** join/#asterisk f0ner00t (i=f0ner00t@c-67-187-154-111.hsd1.ca.comcast.net) |
| 00:45.10 | f0ner00t | Hello Q: If I set up Asterisk on my box is there an service that provide dialin numbers where a T1 is not needed? |
| 00:47.05 | *** join/#asterisk phix (n=threat@123-243-44-131.tpgi.com.au) |
| 00:48.18 | areay | im running asteriskNOW but its cutting and disconnecting calls |
| 00:48.19 | f0ner00t | Anybody? |
| 00:50.08 | *** join/#asterisk ingenius (n=alektro@111-197-235-201.fibertel.com.ar) |
| 00:52.48 | *** join/#asterisk killown (n=ukendt@unaffiliated/killown) |
| 00:55.04 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-6016e565c6564d91) |
| 00:56.28 | areay | every time i take in incoming call through asterisk it disconnects within 10 seconds... anyone got any ideas? |
| 00:57.59 | *** join/#asterisk CrazyTux (n=brandon@c-98-196-6-54.hsd1.tx.comcast.net) |
| 00:58.53 | jblack | areay: Learn to talk really, really fast? |
| 00:59.23 | areay | jblack, lol |
| 00:59.23 | jaytee | hahahaa |
| 00:59.26 | jblack | Do you mean the phone only rings for ten seconds, or what? |
| 00:59.31 | areay | nah it connects |
| 00:59.39 | areay | and i can make weird noises down the line for 10 seconds |
| 00:59.43 | areay | but then it cuts me off |
| 01:00.04 | jblack | no idea. maybe there's a firewall in the way screwing things up |
| 01:00.19 | areay | dunno where would i find logs |
| 01:00.29 | jaytee | /var/log/asterisk/messages |
| 01:00.34 | areay | sweet |
| 01:00.43 | jblack | you'd look at sip debug, iax debug, maybe there's an rtp ebug to |
| 01:00.44 | *** join/#asterisk axisys (n=axisys@ip68-98-177-71.dc.dc.cox.net) |
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| 01:00.53 | *** join/#asterisk asterisk` (n=asterisk@unaffiliated/adrianxxx) |
| 01:00.59 | asterisk` | hi there |
| 01:02.06 | asterisk` | hi there i try to install the g729 codec. |
| 01:02.14 | asterisk` | and i got some errors.. |
| 01:02.41 | asterisk` | [Mar 10 09:57:59] WARNING[5102]: loader.c:359 load_dynamic_module: Error loading module 'codec_g729a.so': /usr/lib/asterisk/modules/codec_g729a.so: cannot restore segment prot after reloc: Permission denied |
| 01:02.42 | asterisk` | [Mar 10 09:57:59] WARNING[5102]: loader.c:653 load_resource: Module 'codec_g729a.so' could not be loaded. |
| 01:03.45 | areay | jblack, im getting "hanging up call" in my logs |
| 01:04.39 | asterisk` | someone can help me please ? |
| 01:04.43 | areay | jblack, http://paste.ubuntu.com/129598/ <-- here's the last few lines |
| 01:07.07 | *** join/#asterisk Kumbang (n=unknown@125.163.83.153) |
| 01:07.43 | jaytee | asterisk`, not running * as root I take it |
| 01:07.43 | *** join/#asterisk killown (n=ukendt@unaffiliated/killown) |
| 01:08.29 | f0ner00t | What service can I use to take inbound calls on my asterix server which is free? |
| 01:08.31 | asterisk` | yes i'am running as root |
| 01:08.42 | asterisk` | bin |
| 01:08.48 | asterisk` | uid=0(root) gid=0(root) groups=0(root),1(bin),2(daemon),3(sys),4(adm),6(disk),10(wheel) context=root:system_r:unconfined_t:SystemLow-SystemHigh |
| 01:09.35 | f0ner00t | Anybody know a good server? |
| 01:09.38 | asterisk` | jaytee can you help me please |
| 01:10.05 | jaytee | asterisk`, well either the file permission isn't set or you don't have a valid license for the codec. g729 isn't free |
| 01:10.37 | asterisk` | i know that |
| 01:10.41 | asterisk` | i just buy the licence |
| 01:10.44 | asterisk` | and is active |
| 01:11.19 | asterisk` | i dont know what kind of module i need form my machine |
| 01:11.25 | *** join/#asterisk thelordmortis (n=lordmort@124-169-248-10.dyn.iinet.net.au) |
| 01:11.30 | [TK]D-Fender | f0ner00t: www.ibm.com . They make great servers |
| 01:11.31 | asterisk` | is CentOS release 5.2 (Final) |
| 01:11.50 | jaytee | asterisk`, well then it should work. I don't use 729 though so I'm not much help with it's specific issues but the message you got clearly indicates a permissions issue. |
| 01:11.52 | bougyman | asterisk`: you think SE might be stopping you? |
| 01:11.53 | asterisk` | model name : Intel(R) Core(TM)2 Quad CPU Q6600 @ 2.40GHz |
| 01:11.59 | bougyman | that seems like you're in an SELinux context. |
| 01:12.14 | jaytee | that was my next question, do you have selinux enabled? if so disable it and try reloading |
| 01:14.08 | *** join/#asterisk outtolunc (n=me@c-71-198-197-201.hsd1.ca.comcast.net) |
| 01:14.32 | *** join/#asterisk astetrg (n=asterisk@unaffiliated/adrianxxx) |
| 01:14.36 | astetrg | back |
| 01:15.25 | *** join/#asterisk k-man (n=jason@unaffiliated/k-man) |
| 01:15.50 | k-man | appart from google talk, does asterisk talk to any of the other non SIP voip services? |
| 01:16.08 | [TK]D-Fender | k-man: Treid reading the book? |
| 01:16.10 | [TK]D-Fender | tried* |
| 01:16.12 | russellb | asterisk supports 7 voip protocols |
| 01:16.33 | [TK]D-Fender | russellb: Not in the capacity he's looking for :) |
| 01:16.54 | asterisk`` | i try to install the codec_g729a-1.4_3.0.3-core2.tar.gz |
| 01:16.57 | asterisk`` | but dont load.. |
| 01:17.03 | asterisk`` | and codec_g729a-1.4_3.0.3-i686.tar.gz |
| 01:17.15 | [TK]D-Fender | asterisk`pastebin the complete output of "asterisk -gvvvvvvvvvvvvvvvvc" |
| 01:17.20 | asterisk`` | ok |
| 01:17.36 | asterisk`` | what do you need of paste bin ... ? |
| 01:17.38 | russellb | asterisk``: you should run the "benchg729" tool. It will tell you which module to use. |
| 01:17.48 | asterisk`` | ok |
| 01:17.49 | russellb | not only based on compatibility, but based on which one will run the fastest. |
| 01:18.03 | asterisk`` | Recommended flavor for this system is 'athlon-xp' with an average of 361 milliseconds. |
| 01:18.18 | *** join/#asterisk viq_ (n=viq@unaffiliated/viq) |
| 01:18.50 | asterisk`` | i put the athlon-xp |
| 01:18.50 | *** join/#asterisk asterisk` (n=asterisk@unaffiliated/adrianxxx) [NETSPLIT VICTIM] |
| 01:18.50 | *** join/#asterisk Woody2143 (n=Brandon_@machine76.Level3.com) [NETSPLIT VICTIM] |
| 01:18.50 | *** join/#asterisk KavanS (i=proxy@static-71-117-242-28.ptldor.dsl-w.verizon.net) |
| 01:18.51 | *** join/#asterisk eppigy (n=Dave@plasticlobster.com) [NETSPLIT VICTIM] |
| 01:18.51 | *** join/#asterisk Maliuta (n=scooby@kiev.lusan.id.au) [NETSPLIT VICTIM] |
| 01:18.51 | *** join/#asterisk yang (i=yang@CAcert/Assurer/pdpc.supporter.base.yang) |
| 01:18.51 | *** join/#asterisk viq (n=viq@unaffiliated/viq) |
| 01:18.51 | asterisk`` | and still dont work |
| 01:18.51 | *** join/#asterisk f0ner00t (i=f0ner00t@c-67-187-154-111.hsd1.ca.comcast.net) [NETSPLIT VICTIM] |
| 01:18.51 | [TK]D-Fender | asterisk``: PASTEBIN please |
| 01:18.51 | [TK]D-Fender | ~pb |
| 01:18.52 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines 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, or , http://bin.cakephp.org/ |
| 01:18.58 | k-man | [TK]D-Fender: where should I look in the book for that info? |
| 01:19.21 | [TK]D-Fender | k-man: one of those nice introduction chapters that tells you all the wonderful things * can talk to |
| 01:19.25 | f0ner00t | [TK]D-Fender: I have a server already. I just need a free inbound service to hook it to. |
| 01:19.53 | *** join/#asterisk orkid (n=orkid@unaffiliated/orkid) |
| 01:20.01 | jaytee | I have driving gloves, I just need a nice free sports car to use them with |
| 01:20.02 | [TK]D-Fender | f0ner00t: www.ipkall.com |
| 01:20.15 | f0ner00t | [TK]D-Fender: Dank you! |
| 01:20.21 | asterisk` | http://pastebin.com/m37b1cd64 |
| 01:20.48 | [TK]D-Fender | asterisk`: that is not what I asked for |
| 01:23.13 | *** join/#asterisk killown (n=ukendt@unaffiliated/killown) |
| 01:23.59 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.static.bezeqint.net) |
| 01:24.05 | k-man | f0ner00t: are you in Aus? |
| 01:24.05 | asterisk` | what do you need. ? |
| 01:24.08 | k-man | Australia i mean? |
| 01:25.28 | [TK]D-Fender | <PROTECTED> |
| 01:25.47 | *** join/#asterisk Kisu (n=k@daniel1117.broker.freenet6.net) |
| 01:27.37 | *** join/#asterisk coppice (n=chatzill@46.166.17.210.dyn.pacific.net.hk) |
| 01:27.55 | k-man | well, if you are in Aus, you can get a very cheap indial for $5/year from MyNetFone |
| 01:30.24 | *** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim) |
| 01:30.43 | f0ner00t | k-man. Nope in the us.. |
| 01:31.13 | f0ner00t | [TK]D-Fender: It says I have to register my sip ip phone? is thi scorrect? |
| 01:31.46 | [TK]D-Fender | f0ner00t: .....huh? |
| 01:32.15 | [TK]D-Fender | f0ner00t: They send an un-auth'd call directly to your server, or to a service of your choice. |
| 01:32.27 | f0ner00t | [TK]D-Fender: I'm at ipkall.com It says here's how it works: 1. Register your sip phone with a VOIP services like freeworkdialup.com or www.mutualphone.com is that correct? |
| 01:32.36 | f0ner00t | I don't exactly understand. |
| 01:32.45 | [TK]D-Fender | f0ner00t: That, or point it directly to your server |
| 01:32.47 | f0ner00t | I understand SIP.. Just not what they are looking for me to do. |
| 01:33.15 | [TK]D-Fender | f0ner00t: they do a direct URI dial, there is no registering with them to tell them where to place the call to. |
| 01:33.16 | *** join/#asterisk ecret (n=ecret@CPE001e9002348e-CM001225d8ab30.cpe.net.cable.rogers.com) |
| 01:33.40 | [TK]D-Fender | f0ner00t: So your server needs a fixed host or has to register to a service that will accept their URI dial. |
| 01:33.54 | [TK]D-Fender | f0ner00t: these days ekiga.net would probably be the best choice. |
| 01:34.04 | [TK]D-Fender | f0ner00t: since FWD is no longer "free" |
| 01:34.21 | f0ner00t | I haven't installed the software yet. I'm just looking into it. |
| 01:34.36 | f0ner00t | I work for a server provider that does sip.. Just haven't had my own hands on sip. |
| 01:35.15 | f0ner00t | I just want inbound not outbound on it for now. |
| 01:35.23 | f0ner00t | But Ekiga looks pretty cool! |
| 01:35.36 | [TK]D-Fender | f0ner00t: "the software"? What software? |
| 01:35.57 | f0ner00t | I'm going to install astrixs. |
| 01:36.03 | f0ner00t | opps asterisk. |
| 01:36.07 | [TK]D-Fender | f0ner00t: Never heard of it |
| 01:36.22 | [TK]D-Fender | f0ner00t: Go right ahead. |
| 01:36.23 | f0ner00t | [TK]D-Fender: Thats why your in the channel! |
| 01:36.42 | [TK]D-Fender | f0ner00t: yes... I can spell and conjugate. |
| 01:36.56 | [TK]D-Fender | Excellent reasons |
| 01:37.29 | f0ner00t | [TK]D-Fender: So when registering for ipkall their is a place for sip phone number what should that be? |
| 01:38.03 | [TK]D-Fender | f0ner00t: Whatever number you need to dial to that host for the call to get to you |
| 01:38.34 | f0ner00t | And where would I get that info? |
| 01:38.44 | [TK]D-Fender | f0ner00t: that depends how you set it up. |
| 01:39.02 | [TK]D-Fender | f0ner00t: the # is the exten to dial against whatever server you point them towards |
| 01:39.24 | [TK]D-Fender | f0ner00t: If you register to a service like ekiga.net then you point them to your account # there. |
| 01:39.57 | [TK]D-Fender | f0ner00t: If you point them directly to your server then you tell them the # you want them to dial and set up an extension in your dialplan to match. |
| 01:40.06 | *** join/#asterisk path_ (n=path_@pc-15-190-86-200.cm.vtr.net) |
| 01:41.02 | asterisk` | [Mar 10 10:37:51] WARNING[5492]: translate.c:175 framein: no samples for alawtolin |
| 01:41.08 | asterisk` | ? |
| 01:41.25 | lmadsen | anyone know how to export the XML documentation from asterisk trunk? |
| 01:42.58 | russellb | lmadsen: it's automagic. |
| 01:43.12 | russellb | lmadsen: it is generated at compile time. it is placed in doc/<something>.xml i thin |
| 01:43.20 | russellb | and installed in /usr/lib/asterisk/documentation (i think) |
| 01:43.28 | lmadsen | ah... I found an old post from you now :) |
| 01:43.37 | russellb | RTFML! |
| 01:43.45 | lmadsen | :D |
| 01:45.21 | f0ner00t | [TK]D-Fender: How do I point it directly to my server/ |
| 01:45.27 | f0ner00t | I guess that is what I'm asking. |
| 01:45.44 | lmadsen | /var/lib/asterisk/documentation :) |
| 01:45.46 | [TK]D-Fender | f0ner00t: there are some nice clear BLANKS to fill in on their site for your IP. |
| 01:46.58 | f0ner00t | So the Sip phone number would be the IP? |
| 01:48.37 | [TK]D-Fender | f0ner00t: no, that is the EXTENSION you want it to dial at the server |
| 01:49.20 | f0ner00t | I don't see a place for IP address. |
| 01:49.41 | *** join/#asterisk maxagaz (n=maxagaz@soho2.i-xanadu.com) |
| 01:50.02 | f0ner00t | Okay cool cool. |
| 01:56.30 | f0ner00t | Goodnight everyone |
| 01:58.24 | *** join/#asterisk killown (n=ukendt@unaffiliated/killown) |
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| 02:35.15 | areay | i just had my voip semi-working with SIP, tried IAX which wouldn't work at all, and now I can't even get SIP working again |
| 02:35.22 | areay | i'm using exactly the same settings as before |
| 02:35.54 | areay | my voip provider told me to edit sip.conf and extensions.conf |
| 02:36.07 | [TK]D-Fender | areay: Clearly it is NOT the same |
| 02:36.52 | areay | [TK]D-Fender, i know... its hard because it seems the instructions on my provider's site are for an older version maybe |
| 02:37.17 | areay | [TK]D-Fender, i had to edit users.conf last time, not sip.conf or extensions.conf |
| 02:37.33 | areay | so i did it again but its broken |
| 02:37.38 | [TK]D-Fender | areay: Had to? No. Cose to? Apparently |
| 02:37.46 | [TK]D-Fender | Chose* |
| 02:38.14 | areay | i'm just trying to get it to work... last time i had it connected with sip it kept dropping phone calls |
| 02:38.40 | areay | so i figured i'd try iax... and now it's completely unusable... i've completely reconfigured it from scratch and it won't work |
| 02:39.26 | drmessano | [TK]D-Fender |
| 02:39.53 | [TK]D-Fender | areay: how do "from scratch" and "same settings" coexist"? |
| 02:40.02 | [TK]D-Fender | drmessano |
| 02:40.27 | drmessano | Remember my problem the other night where my PBX was off, the fans werent running, couldnt access it on the network, and my phones wouldnt register? |
| 02:40.29 | drmessano | I think I may have a power issue |
| 02:40.59 | [TK]D-Fender | drmessano: Grab your flashlight and check :p |
| 02:41.05 | drmessano | hahaha |
| 02:41.10 | jaytee | areay, if you used the example in "the book" for IAX the examples have their contexts backwards. |
| 02:41.22 | *** join/#asterisk killown (n=ukendt@unaffiliated/killown) |
| 02:41.26 | areay | [TK]D-Fender, after i couldn't get it working with iax, i changed the settings back to what they were before (with sip).... it wouldn't work so i reinstalled asterisknow and followed the same (as far as i remember) steps i did before with the sip configuration... |
| 02:41.33 | drmessano | Jaytee, you missed it |
| 02:41.46 | areay | jaytee, nah i'm following this: http://www.voiptalk.org/products/asterisk-sip-trunk-setup.html |
| 02:41.48 | *** join/#asterisk apeiron (n=apeiron@c-76-124-252-61.hsd1.pa.comcast.net) |
| 02:41.57 | *** join/#asterisk Zippoman (n=chatzill@adsl-69-234-110-253.dsl.irvnca.pacbell.net) |
| 02:42.00 | jaytee | missed what? another 30 minute * setup and don't give me any shit? |
| 02:42.27 | drmessano | PM'ed it |
| 02:43.27 | [TK]D-Fender | areay: Here's a tip : enable SIP debug at CLI and realize the difference between telling us "I did everything like they said" and "actually looking at configs and what's happening at CLI" |
| 02:43.37 | [TK]D-Fender | ~pb |
| 02:43.38 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines 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, or , http://bin.cakephp.org/ |
| 02:43.43 | [TK]D-Fender | areay: PASTEBIN is your friend. |
| 02:43.58 | [TK]D-Fender | areay: remember to mask posswords, but nothing else |
| 02:44.09 | apeiron | recently signed up for Broadvoice, running Asterisk for his home phone. :D |
| 02:44.53 | *** join/#asterisk jsoftw (n=Administ@unaffiliated/jsoft) |
| 02:45.02 | jsoftw | Anyone running pabix in a flash in xen? |
| 02:45.24 | areay | [TK]D-Fender, i'm just using the asterisknow vmware image for testing... i need to know how to set this up quickly for my provider... i can pastebin my users.conf, sip.conf, and extensions.conf if u want... those are the only settings i've changed |
| 02:46.10 | [TK]D-Fender | areay: I would be paying very close attention to the SECOND part I asked for. Actual debug of CALL ATTEMPTS |
| 02:46.46 | areay | i don't understand how to get the info you're after |
| 02:47.56 | [TK]D-Fender | areay: got to * CLI. Enable SIP DEBUG. Cut & pastebin. The end. |
| 02:48.18 | areay | wtf is cli? |
| 02:48.29 | mmlj4 | well, if you're using asterisknow or any other cookie-cutter distro, you're better off never touching the config files manually |
| 02:48.56 | [TK]D-Fender | areay: Not knowing this part is like asking "what's a steering wheel" when you go to sit in the driver's seat of a car. |
| 02:49.00 | mmlj4 | areay: command-line interface |
| 02:49.07 | areay | look i run a small business from home and i just want a nice lady to answer the phone when we're closed... i'm not a phone engineer |
| 02:49.19 | [TK]D-Fender | areay: the lovely terminal interface that shows you what is HAPPENING on your server |
| 02:49.22 | mmlj4 | areay: then use the GUI exclusively |
| 02:49.43 | mmlj4 | and quick mucking about in the filesystem |
| 02:50.02 | areay | i've set all the settings i can in the gui |
| 02:50.11 | *** join/#asterisk pmhaddad (n=pmhaddad@24-247-41-171.dhcp.mrqt.mi.charter.com) |
| 02:50.15 | areay | and it wouldn't work... the only time i had it working was when i edited users.conf |
| 02:50.28 | [TK]D-Fender | areay: *NOW is not for you. Just go download and install Trixbox and be done with it. |
| 02:50.56 | areay | trixbox... i've heard of that... thanks for the pointer i'll check it out |
| 02:51.09 | mmlj4 | [TK]D-Fender: you're implying *now is harder than trix-for-kidsbox? |
| 02:51.43 | [TK]D-Fender | mmlj4: Less complete in its hand-holding. And no, I'm not implying it. I'm now telling you to your face :) |
| 02:52.04 | *** join/#asterisk Kumba_ (n=james@198.92.99.232) |
| 02:52.05 | mmlj4 | fair enough |
| 02:52.07 | areay | what's the difference anyway... can i run trixbox on vmware? |
| 02:52.15 | [TK]D-Fender | areay: same thing |
| 02:52.16 | mmlj4 | yes, areay |
| 02:52.38 | Kumba_ | is dial(sip/provider/${EXTEN}) treated the same way inside asterisk as dial(sip/${EXTEN}@provider) ? |
| 02:53.31 | [TK]D-Fender | areay: http://trixbox.org/ <- There, And they have their own channel when you get stuck : #trixbox . Have fun. |
| 02:53.37 | [TK]D-Fender | Kumba : Should |
| 02:53.49 | Kumba_ | Hmmm, ok... |
| 02:56.14 | drmessano | lol |
| 03:01.56 | carrar | w00fters |
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| 03:25.03 | *** join/#asterisk mcab (n=mb@mostly-harmless.ca) |
| 03:32.19 | *** join/#asterisk maxagaz (n=maxagaz@soho2.i-xanadu.com) |
| 03:33.27 | *** join/#asterisk dalbaech (n=dalbaech@knight.darkvoip.net) |
| 03:34.15 | dalbaech | I just had to check out the code tonight.... |
| 03:34.16 | dalbaech | meh. |
| 03:34.16 | areay | [TK]D-Fender, how is this supposed to be easier? you've pointed me in the direction of an overly complex gui and told me to go to a support channel with 22 idle people in it |
| 03:34.27 | dalbaech | russellb: it's all your fault! :P |
| 03:34.41 | thehar | ewww trixbox |
| 03:34.48 | areay | [TK]D-Fender, the digs i could take, but blatantly wasting my time? |
| 03:34.58 | areay | [TK]D-Fender, why do you even come here? |
| 03:35.15 | areay | [TK]D-Fender, obviously it's not to help people... and you're too much of a smartass to ask anyone else for help |
| 03:35.15 | [TK]D-Fender | areay: You want something that "just works" |
| 03:35.29 | [TK]D-Fender | areay: its is a COMPLETE install. No mucking around. Fill in a few blanks and go |
| 03:35.38 | [TK]D-Fender | areay: *NOW's solution is NOT |
| 03:35.58 | areay | [TK]D-Fender, it's useless... i got *now working myself |
| 03:36.22 | areay | [TK]D-Fender, i could spend a week with this trixbox crap |
| 03:36.23 | [TK]D-Fender | areay: Congratulations then. |
| 03:37.29 | areay | i come here asking for help and all i get is attitude with a side helping of RTFM |
| 03:38.14 | [TK]D-Fender | areay: No, I pointed you to a solution that will handle to probable extent to which you will care to configure things, and I see you spent the entire sub-hour since I brought it up harldy looking at it |
| 03:38.22 | [TK]D-Fender | areay: You are the one with an attitude probelm. |
| 03:39.11 | [TK]D-Fender | areay: You know it all already so I guess I won't bother trying to convince you otherwise. |
| 03:39.12 | *** join/#asterisk inv_arp (n=junya@b07s03mr.corenetworks.net) |
| 03:39.35 | [TK]D-Fender | areay: But best of luck with your continued endeavors. |
| 03:39.58 | areay | i told you my level of experience and you sent me in the wrong direction. good job. |
| 03:40.27 | [TK]D-Fender | areay: what "level of experience". Nothing gave any serious indication of any. |
| 03:40.42 | [TK]D-Fender | areay: And what makes Trixbox a "wrong direction"? |
| 03:41.23 | areay | i said i'm not an expert... i'm just following a guide on my sip provider's website |
| 03:41.30 | [TK]D-Fender | areay: And please cool it already. We've tried pointing you where we figured you would best end up with your minimal answering machine-grade service. |
| 03:41.35 | jsoftw | Perhaps you were heading for the kitchen for a feed, and instead went to work on trixbox. In such a case, trixbox would be a step in the wrong direction. |
| 03:42.54 | [TK]D-Fender | areay: And what does "jsut following what my provider told me" say? Are they going to tell you how to do everything you want to do? Or jsut the minimum to have a peer you could use once configuring so much else? |
| 03:43.20 | areay | i can figure out the IVR stuff |
| 03:43.24 | [TK]D-Fender | areay: We did not see your configs. Or the actual errors if any. Yet you rave at us. |
| 03:43.26 | *** join/#asterisk CunningPike (n=arodgers@S01060014bf81366b.vc.shawcable.net) |
| 03:44.16 | areay | i'm not raving at anyone... i was pointing out that you belittled me and sent me away to the noob channel without fully considering what i had to say |
| 03:44.29 | areay | and pointing out that i didn't appreciate it. |
| 03:44.30 | [TK]D-Fender | areay: Belittled? How so? |
| 03:45.05 | *** join/#asterisk tecnico (n=tecnico@75.76.169.148) |
| 03:45.11 | [TK]D-Fender | areay: And I can already see how little consideration you've given to the alternative. "Outright dismissal" seems to sum it up. |
| 03:45.44 | areay | yeah because i asked a simple question and you've told me to go back to the drawing board because i'm too stupid |
| 03:46.26 | [TK]D-Fender | areay: That assessment of my evaluation is incorrect. |
| 03:46.30 | [TK]D-Fender | areay: Any more words you'd like to put in my mouth? |
| 03:46.46 | areay | <[TK]D-Fender> areay: PASTEBIN is your friend. |
| 03:46.59 | [TK]D-Fender | areay: That doesn't say "you're stupid" to me. |
| 03:47.00 | areay | TK]D-Fender> areay: how do "from scratch" and "same settings" coexist"? |
| 03:47.12 | areay | everything says "you're stupid" |
| 03:47.23 | [TK]D-Fender | areay: that seems to say "please show us what you've got and not spam in the channel while doing it" |
| 03:48.41 | [TK]D-Fender | areay: and You say "same settings" and then something that does not imply a serious level of confidence that you did indeed recreate everything the way it was. Especially when if everything is the same as it was it would work the way it used to. Thats the laws of physics. |
| 03:49.06 | [TK]D-Fender | areay: Doing the same thing = get the same results. Different results = differnt situation. |
| 03:49.10 | areay | of course |
| 03:49.23 | areay | but stating the obvious when someone's asking a question doesn't work |
| 03:49.27 | [TK]D-Fender | areay: So again, this was not a persectution. |
| 03:49.46 | [TK]D-Fender | areay: You are overly sensitive and I don't feel like taking the fall for your frustrations. |
| 03:50.14 | areay | i didn't come here to argue with you... i came here to set up a sip trunk |
| 03:50.36 | *** join/#asterisk killown (n=ukendt@unaffiliated/killown) |
| 03:50.45 | [TK]D-Fender | areay: Well you seem to have acheived both. Again, congratiulations on the latter. |
| 03:51.20 | areay | actually it's been a complete waste of time, and i'm no further than i was four hours ago |
| 03:52.23 | [TK]D-Fender | [23:36]<areay>[TK]D-Fender, it's useless... i got *now working myself |
| 03:53.02 | [TK]D-Fender | areay: I had this funny thought that "working" and the "in my face" tone to mean you did it and somehow in spite of us |
| 03:53.15 | *** join/#asterisk whirrclickk (n=wolthuis@mimezine.com) |
| 03:53.19 | [TK]D-Fender | areay: So what does this new state of "working" mean? |
| 03:53.46 | areay | [TK]D-Fender, i was referring to before, when i had it working with a sip connection |
| 03:53.56 | areay | [TK]D-Fender, well at least connecting anyway |
| 03:54.38 | areay | [TK]D-Fender, i meant there was no way i would be getting trixbox working myself because the gui is more complicated than config files in the first place |
| 03:55.02 | areay | [TK]D-Fender, which defeats the object of having a gui in the first place |
| 03:55.13 | [TK]D-Fender | areay: Have to admit, thats not something I hear almost ever |
| 03:55.45 | [TK]D-Fender | areay: So, go to * CLI and look at the SIP debug for your failed call attempt and pastebin it. |
| 03:55.58 | [TK]D-Fender | areay: at Linux CLI : "asterisk -rvvvvvvvvvvv" |
| 03:56.15 | [TK]D-Fender | areay: and then "sip set debug on" |
| 03:56.21 | [TK]D-Fender | areay: and try a call. |
| 03:57.23 | [TK]D-Fender | areay: And since you're going down the route of doing it yourself, go grab the Book and spend some time later learning how * works. |
| 03:57.25 | [TK]D-Fender | ~book |
| 03:57.26 | jbot | i heard book is probably Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://tfot.leifmadsen.com or see ~buybook |
| 03:57.49 | areay | i skimmed over that the other day |
| 03:58.17 | areay | it pretty much tells you if you're using *now to skip the entire book |
| 03:58.19 | jaytee | skimming is like reading Cliff notes for a final exam. |
| 03:59.15 | areay | jaytee, agreed... but i wasn't looking for a qualification... i just wanted the thing up and running... i learn quicker through experience than by reading |
| 03:59.16 | [TK]D-Fender | areay: I'm sure you spent jsut as long reading it as following previous advice. |
| 04:00.53 | areay | [TK]D-Fender, if i wanted to set up 1000 lines for a fortune 500 company i'd be studying a lot more than i am now... but what i want to do is pretty simple... once i get it basically set up i can learn the different features as and when i want to use them |
| 04:01.26 | areay | all i'm trying to do is get these settings to work in *now --> http://www.voiptalk.org/products/asterisk-sip-trunk-setup.html |
| 04:01.46 | [TK]D-Fender | areay: Yes, but at the same time your saying " i just want a nice lady to answer the phone when we're closed" doesn't say "I want to learn Asterisk". A gui will let you fill in a few blanks for provider setting, and they usually even give specific instructions for this. Set up a defau;t route in the GUI toa VM box and you're done. |
| 04:02.13 | [TK]D-Fender | areay: Fo someone who didn't want to be a phone engineer, you sure turn a mean 180 |
| 04:02.44 | [TK]D-Fender | areay: But I have also jsut given you specific instruction to follow to try to debug this. Please go and do them. |
| 04:03.08 | areay | ok brb |
| 04:06.06 | whirrclickk | does anyone know of if its possible to use MRCP with asterisk? |
| 04:06.26 | areay | [TK]D-Fender, im doing this from a fresh vmware image too so it'll take a couple mins to reconfigure... |
| 04:06.38 | k-man | whirrclickk: what is MRCP? |
| 04:06.56 | whirrclickk | http://tools.ietf.org/html/rfc4463#section-3.2 |
| 04:06.59 | [TK]D-Fender | areay : Didn't you just say you put everything back? |
| 04:07.33 | whirrclickk | k-man: TTS/AVR interface standard. |
| 04:07.53 | k-man | whirrclickk: oh |
| 04:08.02 | k-man | whirrclickk: what is TTS/AVR? |
| 04:08.13 | [TK]D-Fender | whirrclickk: No. |
| 04:08.22 | k-man | text to speech? |
| 04:08.25 | [TK]D-Fender | .... |
| 04:08.25 | areay | [TK]D-Fender, it only takes a min to set up... i just wanna make sure im doing it right |
| 04:08.26 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 04:08.27 | whirrclickk | aye |
| 04:08.37 | k-man | whirrclickk: ah, ok |
| 04:08.46 | k-man | whirrclickk: well - i have no idea ;) |
| 04:09.02 | whirrclickk | [TK]D-Fender: no, you don't know? or no, it cant? :) |
| 04:09.16 | [TK]D-Fender | whirrclickk: No, * does not do this. |
| 04:09.20 | whirrclickk | drat |
| 04:12.10 | k-man | i have read that asterisk is not a SIP Peer - is that a good/bad or indifferent feature of asterisk? |
| 04:12.18 | k-man | err |
| 04:12.22 | k-man | s/peer/proxy |
| 04:12.27 | k-man | sorry for the slip |
| 04:12.55 | jblack | It's just what it is. |
| 04:13.09 | jblack | it can be used as a sip intermediary, btw |
| 04:14.21 | [TK]D-Fender | o>O |
| 04:17.04 | areay | [TK]D-Fender, i tried "sip set debug on" and its the wrong syntax... it wants an ip |
| 04:17.14 | areay | or a peername |
| 04:17.30 | [TK]D-Fender | areay: "sip set debug" |
| 04:17.40 | areay | ah sorry |
| 04:18.13 | [TK]D-Fender | areay: No need, syntax has changed, I do mix these up often enough |
| 04:18.22 | *** join/#asterisk zenfox (n=steven@c-98-213-240-12.hsd1.il.comcast.net) |
| 04:18.34 | *** join/#asterisk killown (n=ukendt@unaffiliated/killown) |
| 04:19.37 | areay | [TK]D-Fender, ok it's kicking some stuff out now |
| 04:20.18 | areay | "scheduling destruction of sip dialog..." |
| 04:21.30 | [TK]D-Fender | areay: Should be getting a LOT more than that on a call attempt |
| 04:21.31 | areay | its having trouble registering to voiptalk.org |
| 04:21.36 | areay | i am |
| 04:21.43 | [TK]D-Fender | areay: that last message can be ignored. |
| 04:21.54 | areay | hold up i'll go use my "friend" |
| 04:22.06 | [TK]D-Fender | Excellent idea. |
| 04:23.51 | areay | [TK]D-Fender, i'll have to screenshot it i can't get it to transfer the clipboard data out of vmware player |
| 04:24.19 | [TK]D-Fender | areay: Connect to it via SSH from another session / PC |
| 04:24.32 | [TK]D-Fender | areay: 1 screen doesn't give enough. |
| 04:24.42 | [TK]D-Fender | areay: You'll need a lot more scroll-back |
| 04:24.47 | areay | [TK]D-Fender, i tried but it was giving me an error... hold up |
| 04:25.33 | [TK]D-Fender | areay: use "exit" to get back to Linux CLI. then "chkconfig sshd on" , "service sshd start" |
| 04:26.05 | areay | the ssh connection was fine... just the asterisk command giving me errors... http://paste.ubuntu.com/129634/ |
| 04:26.50 | k-man | what distro is areay using? |
| 04:27.18 | areay | ubuntu on my machine... im not running asterisk on ubuntu tho |
| 04:28.49 | k-man | what distro is asterisk running on? |
| 04:28.50 | areay | *now... i forgot to sudo i got it workin now |
| 04:29.51 | *** join/#asterisk Coolthreads (n=Coolthre@203-97-238-71.cable.telstraclear.net) |
| 04:31.14 | [TK]D-Fender | areay: "admin" doesn't have rights to * |
| 04:31.38 | k-man | what distro does asterisknow run on? |
| 04:31.46 | [TK]D-Fender | k-man: He's on the OLD one. |
| 04:32.03 | [TK]D-Fender | areay: You very seriously should be on 1.5 Beta.... |
| 04:32.14 | [TK]D-Fender | areay: That looks like the old rPath junk they dumped |
| 04:32.38 | areay | shit i thought i was using the new one... i should make my own vmware image really |
| 04:32.58 | [TK]D-Fender | areay: look at the COPYRIGHT date for a clue |
| 04:33.10 | [TK]D-Fender | areay: 1.4.0 is ancient |
| 04:33.27 | areay | damn |
| 04:33.37 | [TK]D-Fender | areay: and words fail me as to just how bad the first few releases are. |
| 04:33.38 | Kobaz | i have a completely random question... how would one detect if someone is calling from a payphone |
| 04:34.01 | [TK]D-Fender | Kobaz: "module load res_psychic.so" |
| 04:34.05 | *** join/#asterisk jsgoecke (n=jsgoecke@c-67-180-103-93.hsd1.ca.comcast.net) |
| 04:34.12 | Kobaz | yes, that would work lovely |
| 04:34.21 | Kobaz | i only i could use that module for everything |
| 04:34.47 | areay | there's only one *now virtual machine on the vmware site |
| 04:35.15 | *** join/#asterisk CrazyTux (n=brandon@c-98-196-6-54.hsd1.tx.comcast.net) |
| 04:35.30 | areay | [TK]D-Fender, http://paste.ubuntu.com/129635/ |
| 04:36.05 | [TK]D-Fender | areay: Found no matching peer or user for '192.168.1.100:5097' <- first it can't ID the incoming caller |
| 04:36.25 | [TK]D-Fender | areay: Looking for 908702404040 in default (domain 192.168.1.68) -- SIP/2.0 404 Not Found |
| 04:37.02 | [TK]D-Fender | areay: Next it can't find a match for "908702404040" in [default] (which is where the call lands in your dialplan due to your SIP config). |
| 04:37.41 | [TK]D-Fender | areay: The mere fact you even get the call does suggest that if you had to register to them, you have got that part working at the very least. |
| 04:38.06 | areay | oh this is me dialling out from a softphone |
| 04:38.21 | areay | to an outside line |
| 04:38.31 | [TK]D-Fender | areay: AH, I did "skim" that myself... |
| 04:38.40 | areay | lol |
| 04:39.02 | [TK]D-Fender | areay: All this time you described what would effectively be an "answering machine". This does not seem to be your approach |
| 04:39.38 | areay | [TK]D-Fender, i know... but the incoming and outgoing are using the same trunk |
| 04:39.52 | [TK]D-Fender | areay: And indeed is is very bad that your softphone is not matching a peer on your system. Did you try to set one up for it? |
| 04:40.05 | [TK]D-Fender | areay: Your problem has nothing to do with your provider. |
| 04:40.08 | areay | [TK]D-Fender, nah i just set up an extension and password |
| 04:40.21 | areay | [TK]D-Fender, when i dial my voip number from a regular phone i just get a busy signal |
| 04:40.21 | [TK]D-Fender | areay: Your problem is currently solely between * and your phone |
| 04:40.31 | [TK]D-Fender | areay: Try again now. |
| 04:40.39 | [TK]D-Fender | areay: Since SIP debug is enabled |
| 04:40.46 | [TK]D-Fender | areay: perhaps you'll see something |
| 04:40.47 | *** part/#asterisk SparFux (n=raoul@e182020098.adsl.alicedsl.de) |
| 04:40.59 | areay | nothing at all... |
| 04:41.13 | areay | if asterisk isn't running i get an error message when i dial that number tho |
| 04:41.13 | *** part/#asterisk Mog (n=mog@c-68-62-218-186.hsd1.al.comcast.net) |
| 04:41.17 | areay | so asterisk must be doing something |
| 04:42.18 | k-man | theres 2 mistakes i made when setting up asterisk. 1. my voip provider didn't give me a DID when I thought they did - so i always got engaged signals until i found that out and fixed it. 2. i made a mistake in the password in the sip register line - so nothing worked until i fixed that either |
| 04:42.26 | k-man | both were hard to track down |
| 04:42.59 | [TK]D-Fender | areay: What is the networking between * and the internet? |
| 04:43.03 | *** join/#asterisk mintos (n=mvaliyav@nat/redhat-in/x-728a0695ce50235c) |
| 04:43.42 | areay | a router/firewall... it's not direct |
| 04:43.50 | areay | but it worked before |
| 04:44.00 | [TK]D-Fender | areay: * needs all sorts of ports forwarded to it, config file settings, etc. |
| 04:44.23 | [TK]D-Fender | areay: I hate to say this is going to be a very long process. |
| 04:44.34 | areay | i know how to forward ports... i just need to know which ones |
| 04:44.47 | [TK]D-Fender | ~sipnat |
| 04:44.47 | 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 |
| 04:44.53 | [TK]D-Fender | areay: First link. |
| 04:45.07 | [TK]D-Fender | areay: Here, for some "inspiration" : |
| 04:45.10 | [TK]D-Fender | ~jerjerguide |
| 04:45.11 | jbot | [~jerjerguide ] Jeremy McNamara's quick sample guide to setting up * : http://www.jeremy-mcnamara.com/2007/02/26/how-to-configure-asterisk-your-first-installation/ |
| 04:45.53 | [TK]D-Fender | areay: fix the fact that Ekiga is not authing to a peer first. |
| 04:46.20 | areay | how do i do that? |
| 04:46.30 | [TK]D-Fender | areay: Then confim the dialplan in the context it points to. then fix your NAT setup and try to integrate the bits your provider told you to do for the basics |
| 04:46.35 | areay | jerry mcnamara link is broken btw |
| 04:46.48 | [TK]D-Fender | areay: You are starting practically from scratch.... you have a LOT of learning to do.... |
| 04:47.32 | [TK]D-Fender | areay: http://74.125.95.132/search?q=cache:x-SVjZ-02u8J:www.jeremy-mcnamara.com/2007/02/26/how-to-configure-asterisk-your-first-installation/+http://www.jeremy-mcnamara.com/2007/02/26/how-to-configure-asterisk-your-first-installation/&hl=en&ct=clnk&cd=1&gl=ca |
| 04:47.33 | areay | [TK]D-Fender, i had monkeys screeching down my phone earlier |
| 04:47.50 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 04:48.03 | [TK]D-Fender | areay: Ignore everything except the extensions, sip, & voicemail bits |
| 04:48.26 | areay | [TK]D-Fender, it was all working perfectly (for the first 30 seconds of the call)... if only i didnt mess it up |
| 04:48.42 | [TK]D-Fender | areay: Your peer (if you have one) can't ID the phone and the call is being allowed in un-authed. Not a good thing normally. |
| 04:49.11 | Kobaz | used that line because 1-866-IDOLS-13 is owned by a phone sex operation which promises to connect listeners to a "nasty girl" for $3.99 a minute. |
| 04:49.14 | Kobaz | er |
| 04:49.14 | [TK]D-Fender | areay: well this is going to be an extreme hand-holding process from the looks of it, and my time is almost up for tonight |
| 04:49.33 | Kobaz | Voters for final performer Alexis Grace, the 21-year-old single mother from Memphis, Tenn., were directed to 1-866-IDOLS-36. The singing contest - which has 13 finalists this season instead of the usual dozen - used that line because 1-866-IDOLS-13 is owned by a phone sex operation which promises to connect listeners to a "nasty girl" for $3.99 a minute. |
| 04:49.38 | Kobaz | there you go |
| 04:49.41 | areay | [TK]D-Fender, its frustrating me because i had it working before and i barely changed anything |
| 04:49.52 | k-man | what is it with jbot talking about soviet russia? |
| 04:50.04 | Kobaz | in soviet russia, asterisk calls you! |
| 04:50.42 | [TK]D-Fender | k-man: In-joke that only Sargun and I get. |
| 04:51.08 | k-man | [TK]D-Fender: ok |
| 04:51.12 | Sargun | Yeah, exactly. |
| 04:51.15 | [TK]D-Fender | (apparently) |
| 04:51.37 | [TK]D-Fender | Sargun: You are done playing with my bitch, right? :) |
| 04:51.41 | Sargun | hehe |
| 04:54.02 | joako | Does anyone use a Nokia WiFi phone with asterisk? |
| 04:54.26 | [TK]D-Fender | joako: I've head of the E61/62 being used with *. Sucks hard with NAT though. |
| 04:54.37 | [TK]D-Fender | (or so people have consistently claimed) |
| 04:55.31 | [TK]D-Fender | heard* |
| 04:56.38 | joako | That sort of answers my question. I have an E71 with massive NAT issues |
| 04:56.56 | joako | And I thought the E62 was a Cingular version w/o WiFii..... |
| 04:57.45 | joako | But OTOH VoIP calls over 3G on the E71 work *surprisingly* well |
| 05:02.33 | [TK]D-Fender | joako: Lower your expectations far enough and lots of things will surprise you. |
| 05:04.39 | joako | lol over the 3G I was able to make a VoIP call while I drove around without it breaking up, I was expecting it to not be usable at all |
| 05:04.52 | k-man | thats cool |
| 05:05.04 | k-man | i see there is a SIP client for the iphone now |
| 05:05.14 | k-man | anyone tried that? |
| 05:11.56 | drmessano | Does it look like a sunflower? |
| 05:21.17 | *** join/#asterisk orkid (n=orkid@unaffiliated/orkid) |
| 05:22.03 | *** join/#asterisk Pan3D (n=Pan3D@node2.sensoryresearch.net) |
| 05:30.43 | *** part/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-6016e565c6564d91) |
| 05:30.58 | *** join/#asterisk [T]ank (n=ckwall@c-71-199-25-239.hsd1.ut.comcast.net) |
| 05:44.45 | *** join/#asterisk MmixX (n=mmixx@61.14.191.142) |
| 05:45.11 | *** join/#asterisk ghento (n=ghento@d75-157-199-106.bchsia.telus.net) |
| 05:46.56 | [T]ank | I am trying to place a call from my softphone and I am getting errors that the call failed with a cause 99. configs are here: http://pastebin.ca/1357999 |
| 05:47.04 | [T]ank | can anyone see what might cause this? |
| 05:47.50 | [T]ank | if I do a console dial and that same number, the call will go through just fine. |
| 05:49.28 | ecret | one of the svn sources on http://www.asterisk.org/developers/get-source seem to be down : svn checkout http://svn.digium.com/svn/libpri/trunk libpri Can someone else verify please? |
| 05:50.30 | apeiron | trunk is gone, but the parent directory works. |
| 05:51.08 | ecret | should site be updated? i will just use http://svn.digium.com/view/libpri/branches/1.4/ |
| 05:51.10 | apeiron | damnit.us, heh, great hostname. |
| 05:52.30 | [T]ank | If I google:   -- Channel 0/1, span 1 got hangup, cause 99 I find others who have had the same issue, but no resolution. |
| 06:03.14 | [T]ank | anyone have any ideas at all as to why I cant get my call to work when using a softphone? |
| 06:03.48 | *** join/#asterisk l0gis1c (n=logisic@adsl-75-18-166-189.dsl.pltn13.sbcglobal.net) |
| 06:09.10 | *** join/#asterisk fiddur (i=fiddur@c042.rit.se) |
| 06:12.42 | [T]ank | where is everyone? Its dead |
| 06:18.54 | *** join/#asterisk _gm (n=gmustafa@115.186.106.37) |
| 06:21.10 | *** join/#asterisk thansen (n=thansen@7.247.sfcn.org) |
| 06:22.08 | *** join/#asterisk TrentCreek (i=kvirc@adsl-70-254-122-233.dsl.hrlntx.swbell.net) |
| 06:22.48 | TrentCreek | why would my box be asking for callers to record their voice befor progressing the call? |
| 06:24.41 | jblack | you're not sending them where you think. check your dialplan, and look at asterisk -r output with debug and verbose bumped up |
| 06:25.34 | TrentCreek | okay..i got it...it was screening the calls..I disabled it |
| 06:28.03 | *** join/#asterisk jeffgus (n=jeffgus@green.zimage.com) |
| 06:29.32 | TrentCreek | thanks |
| 06:30.34 | *** join/#asterisk SkywaIker (n=pirch@58.147.17.166) |
| 06:39.32 | *** join/#asterisk goodjoke (i=4a4359ab@gateway/web/ajax/mibbit.com/x-262e198dfb1b6744) |
| 06:45.29 | *** join/#asterisk BeeBuu (n=beebuu@59.38.96.249) |
| 07:04.29 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 07:20.04 | Kumba_ | I wonder if you can license some george carlin stand-up one-liners for inclusion into the open-source sounds project... |
| 07:20.53 | j_kroon | hi, is it possible to at the end of each call for each channel log the jitter, latency and packet loss for each of the two channels (or one of them at least)? |
| 07:26.25 | stabler | what the best way to add a 2 sec pause in my ivr? |
| 07:26.47 | stabler | ive done it before but cant seem to find the syntax i used before |
| 07:31.01 | *** join/#asterisk ghento (n=ghento@d75-157-199-106.bchsia.telus.net) |
| 07:31.23 | ecret | I setup asterisk and was hoping someone could try to see if it works. 99.246.65.73 dial 1000. Thanks |
| 07:33.32 | stabler | ecret: do you not have an inbound/outbound trunk |
| 07:34.37 | ecret | stabler just inbound i think. I did the default install and asteriskgui. I am able to call it locally adding in 1000 as the call # |
| 07:35.17 | *** join/#asterisk xrmx__ (n=rm@host128-22-dynamic.15-87-r.retail.telecomitalia.it) |
| 07:35.39 | stabler | do you have a DID? |
| 07:36.28 | ecret | yes, but its not setup, i am trying to figure out how to configure it. I think I did it correctly but its not calling my asterisk so was hoping to see if someone outside localhost could call |
| 07:38.17 | ecret | the site is not very helpful for the did provider. They ask for URI or public IP and offer no tips or other info. Tried my IP so hope someone with a SIP phone can try to call me |
| 07:38.24 | stabler | is your asterisk box behind a router? |
| 07:38.42 | ecret | yes , i dmz'd though and the 5060 port is open |
| 07:39.09 | stabler | did you setup nat? |
| 07:39.09 | stabler | ~sipnat |
| 07:39.10 | 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 |
| 07:40.34 | *** join/#asterisk dandate2 (n=dandate2@c-67-169-101-3.hsd1.ca.comcast.net) |
| 07:40.37 | ecret | i am dmz'ing though, dont think I need to nat |
| 07:41.00 | dandate2 | my speed test sayts i get 16000kb down and 4000kb up, is this enough to handle 10 sales reps with concurrent calls? |
| 07:41.02 | stabler | dmzing is not good for security |
| 07:41.38 | *** join/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 07:42.19 | stabler | dandate2: what codec/ |
| 07:43.11 | dandate2 | ulaw |
| 07:43.17 | ecret | stabler ok thanks i will setup NAT . Were you able to connect? |
| 07:43.31 | *** join/#asterisk mosty (n=mosty@213-66-224-163-no22.tbcn.telia.com) |
| 07:43.51 | stabler | ecret: i didnt try |
| 07:44.04 | stabler | i dont currently have a phone setup to make a direct ip call |
| 07:45.47 | stabler | dandate2: you should be fine |
| 07:46.45 | stabler | dandate2: i believe you need 82.1 kbps per call |
| 07:46.49 | stabler | if i remember correctly |
| 07:47.28 | dandate2 | k 16000kb is business class cable right? |
| 07:47.45 | dandate2 | i ordered the ugprade a couple days ago but wasn't able to test my speed before hand so i dno if its improved or not |
| 07:47.53 | dandate2 | but i was getting choppyness before at that load so i ordered the upgrade |
| 07:48.06 | stabler | that would all depend on provider and there package levels |
| 07:48.31 | dandate2 | they said it would be 16mb download, thats what 160000kb comes out to right? |
| 07:48.34 | stabler | i have a 15mb residental connection |
| 07:48.46 | dandate2 | are you using cablehack? |
| 07:48.54 | stabler | dandate2: correct |
| 07:49.07 | dandate2 | damn how do i get involved with that |
| 07:49.18 | dandate2 | do i need to order their modem mod? |
| 07:49.29 | stabler | modem mod? |
| 07:49.39 | dandate2 | i saw config scripts and actual modems at cablehack |
| 07:49.46 | dandate2 | didn't know where to begin heh |
| 07:50.12 | stabler | uhh if its for a production environment i wouldnt run anything hacked |
| 07:50.34 | dandate2 | well could i throw up a second cable modem based off the TV route? |
| 07:50.40 | stabler | production environments you should always remain legit |
| 07:52.05 | stabler | if you get caught doing something illegal itll come back on the company |
| 07:52.18 | stabler | and put there reputation on the line |
| 07:52.29 | stabler | which in you in turn caused |
| 07:52.32 | stabler | not a good idea |
| 07:52.48 | dandate2 | word |
| 07:53.00 | stabler | *which in turn you caused |
| 07:53.36 | stabler | its getting late =/ |
| 07:53.49 | stabler | im not making sense anymore |
| 07:53.50 | stabler | lol |
| 07:53.54 | dandate2 | i know |
| 07:54.13 | dandate2 | i guess i shall risregard cable hack for now unless i go relaly underground |
| 07:55.13 | stabler | i would give you current connection a whirl |
| 07:55.50 | stabler | it should be very close to working fine |
| 07:56.12 | stabler | as far as i know |
| 07:56.36 | dandate2 | yeah i hope its already been upgraded |
| 07:56.48 | dandate2 | i am trying to get onto an open box with wireless but can never obtain ip |
| 07:56.59 | *** join/#asterisk h-idrisi (n=h-idrisi@86.60.52.157) |
| 07:57.03 | stabler | hah |
| 07:57.24 | stabler | 4mb upload sounds like buisness class |
| 07:57.29 | dandate2 | listening to my reps its been sad there was so much chopyness the last days |
| 07:57.35 | dandate2 | alright |
| 07:58.19 | stabler | so the call quality has been bad with the current setup |
| 07:58.40 | dandate2 | yeah i dont know about now though i just did the speed test |
| 07:58.46 | *** join/#asterisk botox93 (n=botox93@213.221.82.242) |
| 07:58.46 | dandate2 | we took on a lot of new reps and the choppyness got bad |
| 07:58.56 | stabler | oh |
| 07:59.11 | dandate2 | but i ordered business class cable 2 days ago and it might already be installed i think |
| 07:59.16 | stabler | you could always get two connectinos and load balance them |
| 07:59.45 | dandate2 | they said thursday morning but i think it might have kicked in |
| 08:00.07 | dandate2 | thats the only solution anyone could tell me as far as that heh |
| 08:00.08 | stabler | or T1 :D |
| 08:00.27 | dandate2 | how fast is t1 usually and how much |
| 08:00.35 | stabler | expensive |
| 08:00.55 | stabler | around 300/month |
| 08:00.59 | dandate2 | dammn |
| 08:01.02 | *** join/#asterisk thansen (n=thansen@7.247.sfcn.org) |
| 08:01.09 | stabler | how many users total do you have |
| 08:01.15 | jblack | and that's large cities. small cities, it can go up to 700 or so |
| 08:01.20 | dandate2 | about 15 |
| 08:01.28 | dandate2 | bu we bring in thousands of calls a day also |
| 08:01.41 | jblack | You could fit that on a dedicated cablemodem |
| 08:01.46 | stabler | yea your connection is alittle weak for 15 concurrent calls |
| 08:02.00 | dandate2 | should i put the cable directly to the box then |
| 08:02.25 | stabler | how many connections do you have? |
| 08:02.27 | jblack | that's what dedicated means. It depends upon the bandwidth of the provider, and what codecs you chose. |
| 08:02.41 | stabler | hes using g.711 |
| 08:02.46 | dandate2 | the box is hooked up to a router with all its ports full |
| 08:03.19 | stabler | hes working with a 16/4 connection |
| 08:03.26 | stabler | cable provider |
| 08:03.57 | *** join/#asterisk brunner (n=chris@24.214.202.118) |
| 08:04.03 | stabler | so youre using one 16/4 connection to serve 15 reps with internet access and serve for 15 concurrent calls? |
| 08:04.19 | dandate2 | the reps are over sees |
| 08:04.21 | dandate2 | but yes |
| 08:05.00 | stabler | dandate2: but the connection you have only is used for the asterisk box.. right? |
| 08:05.16 | dandate2 | no its shared with my other boxes that dn't use it for much |
| 08:05.20 | mosty | dandate2, any reason you can't use g729? |
| 08:05.34 | dandate2 | don't know much about g729 |
| 08:05.38 | jblack | Try gsm. That's very good bang for the buck. |
| 08:06.04 | dandate2 | connection goes to router which directs to the * box |
| 08:06.12 | mosty | gsm doesn't cost anything, but more phones support g729 |
| 08:06.13 | dandate2 | router also shares with a couple pcs for internet use |
| 08:06.27 | stabler | change codecs may help alot |
| 08:06.38 | stabler | g.711 is alittle harsh on bandwidth |
| 08:06.41 | stabler | as far as i know |
| 08:06.47 | dandate2 | gee will that be stable? |
| 08:07.00 | dandate2 | where do i learn about changing codecs |
| 08:07.54 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 08:08.51 | *** join/#asterisk Limeni (n=nkostani@smtp.one2play.hr) |
| 08:10.23 | mosty | dandate2, disallow=all then allow=gsm in your sip.conf |
| 08:10.28 | mosty | then do a sip reload |
| 08:10.48 | mosty | but beware that not all phones support gsm |
| 08:10.54 | stabler | youll have to let you reps know though.. depending on what phones they use |
| 08:11.23 | stabler | may have to change settings on there phones |
| 08:11.36 | dandate2 | they all use x-lite |
| 08:11.39 | dandate2 | will that be fine? |
| 08:11.48 | dandate2 | i am readign now and have much realized that i do need to change codec |
| 08:11.54 | dandate2 | since all my reps work off the WNA |
| 08:11.55 | dandate2 | WAN |
| 08:11.55 | stabler | yes |
| 08:12.09 | stabler | g.711 is pretty much fail |
| 08:12.12 | dandate2 | whats the license fee on g729? |
| 08:12.15 | mosty | dandate2, some older versions of x-lite support gsm |
| 08:12.32 | dandate2 | hm so if i setup gsm my reps using xlite won't beable to use? |
| 08:12.35 | mosty | dandate2, on the asterisk side, g729 costs 10 USD per simultantous call |
| 08:12.45 | mosty | simultaneous, even |
| 08:12.57 | dandate2 | ?? |
| 08:13.00 | [T]ank | when using a callfile to place a call, if I set the field callerid: the call will fail with a cause 99. But if I take out that line, the call goes through just fine. |
| 08:13.08 | [T]ank | How do I set caller id in a callfile? |
| 08:14.05 | stabler | dandate2: just use gsm if cost is an issue |
| 08:15.18 | dandate2 | alright |
| 08:15.27 | dandate2 | all my workers use newer x-lite download though will they all crash? |
| 08:15.36 | dandate2 | i just set gsm in the trunk settings then nothing else? |
| 08:15.55 | mosty | recent free versions of x-lite don't support gsm |
| 08:16.32 | stabler | dandate2: how do you find a sweet job such as yours |
| 08:16.34 | [T]ank | dandate2: I thing zoiper supports it |
| 08:17.13 | [T]ank | dandate: yes, zoiper does support it |
| 08:17.19 | dandate2 | gotta make it |
| 08:17.19 | dandate2 | find a product, learn how to sell it and resell it online, over the phone, however you can get the money |
| 08:17.29 | dandate2 | so i should dig through the oldapps.com and find an old x-lite? |
| 08:17.31 | *** join/#asterisk tamiel (n=tamiel@213.30.183.226) |
| 08:17.39 | dandate2 | zoiper will take my gsm calls then, and that is free and works like x-lite? |
| 08:17.55 | [T]ank | no... zoiper is a softphone... like x-lite |
| 08:18.02 | [T]ank | sorry, misread... |
| 08:18.07 | dandate2 | lol right |
| 08:18.11 | [T]ank | yes, it works LIKE xlite |
| 08:18.19 | *** join/#asterisk Weazelon (n=deazel@mail2.tikalnetworks.com) |
| 08:18.20 | dandate2 | so i tell all my reps to get zoiper and i'm set, is the configuration all confusing or same? |
| 08:18.28 | [T]ank | thought you said WITH xlite |
| 08:18.36 | [T]ank | been up way to late tonight |
| 08:18.40 | stabler | dandate2: sounds cool.. great idea |
| 08:18.46 | [T]ank | super easy config |
| 08:18.50 | [T]ank | download it and check it out |
| 08:19.04 | [T]ank | windows, linux and mac versions I believe |
| 08:19.17 | [T]ank | and doesnt crash like xlite sometimes does |
| 08:19.21 | stabler | lol |
| 08:19.33 | dandate2 | cool |
| 08:19.40 | stabler | ive been trying to find asterisk work for a while with no luck :( |
| 08:19.42 | dandate2 | gunna try this gsm thing out |
| 08:19.59 | [T]ank | one thing i like about zoiper is it can be installed to a usb key and the settings stay local to that directory. So its a portable softphone as well |
| 08:20.18 | stabler | [T]ank: good to know |
| 08:20.20 | stabler | thats sweet |
| 08:20.51 | [T]ank | use a autorun.inf file and it autolaunches when you plug it in. Softphone for idiot users :-D |
| 08:20.55 | Weazelon | by the way for zoiper |
| 08:20.58 | Weazelon | if you use linux |
| 08:21.05 | Weazelon | i totally suggest using 2.07 |
| 08:21.13 | Weazelon | the 2.09 has a wierd crash issue |
| 08:21.19 | Weazelon | using ubuntu myself... |
| 08:21.31 | stabler | linux is all i use |
| 08:21.36 | [T]ank | is 2.09 released? |
| 08:21.39 | stabler | my favorite flavor is unbuntu |
| 08:21.43 | stabler | :D |
| 08:22.20 | Weazelon | yea 2.09 is available for download, but its crap imo |
| 08:22.29 | [T]ank | looks like it. |
| 08:22.47 | Weazelon | for linunx that is... windows has no prob as far as i checked |
| 08:23.02 | Weazelon | but its still a windows.... yuck... |
| 08:23.03 | [T]ank | when do you experience the crash? is there a trigger? |
| 08:23.19 | Weazelon | yea, its called "double click to open the damn thing" |
| 08:23.21 | Weazelon | >< |
| 08:23.24 | [T]ank | lol |
| 08:23.25 | stabler | lmao |
| 08:23.38 | [T]ank | im gonna test it out really quick.... |
| 08:23.46 | *** join/#asterisk h-idrisi (n=h-idrisi@86.60.52.157) |
| 08:23.50 | WeazelON | good luck ;D |
| 08:24.24 | WeazelON | u might not experinence it right away though.. it might work the first few times, but as soon as you see "Zoiper got a sigkill something" know that this is the time the test is over ^_^ |
| 08:24.52 | [T]ank | how many times would you say it would take? |
| 08:24.56 | [T]ank | so far its working for me |
| 08:25.16 | WeazelON | as i said, it looked promising the first couple of tries, but then suddenly it flipped |
| 08:25.42 | stabler | sounds like windows in general |
| 08:25.56 | WeazelON | i know.. but its on a linux... wierd.. |
| 08:26.19 | stabler | well im refering to the windows OS |
| 08:26.20 | stabler | lol |
| 08:26.43 | stabler | looks promising the first couple tries but suddenly flips |
| 08:27.14 | [T]ank | heh |
| 08:30.24 | [T]ank | so far so good... I'll keep running with it. |
| 08:30.50 | [T]ank | so, anyone here know anything at all about callfiles? |
| 08:31.06 | [T]ank | I can make them work so long as they do not include a callerid: line |
| 08:32.12 | *** join/#asterisk Avinoash (n=kai@mail2.tikalnetworks.com) |
| 08:32.20 | Avinoash | helo world |
| 08:33.03 | stabler | ahh |
| 08:33.39 | WeazelON | Hello |
| 08:33.50 | dandate2 | damn zoiper won't open or close heh |
| 08:33.56 | WeazelON | >< |
| 08:34.34 | [T]ank | if you cant open it, how do you know you cant close it? |
| 08:34.34 | WeazelON | dandate2, did your problem got fixed since yesturday ? |
| 08:37.15 | mosty | [T]ank, can you pastebin a sample call file? |
| 08:41.27 | dandate2 | yes weazel i had to hire them over at freepbx |
| 08:41.33 | dandate2 | actually the problem was within my router |
| 08:42.07 | dandate2 | broken settings, had configured the router to be 192.168.1.1 and also 192.168.1.100 which was also the localnet for the pbx |
| 08:42.19 | WeazelON | ohhhhh..... |
| 08:43.05 | *** join/#asterisk ReDNeQ (n=ReDNeQ@70.114.226.70) |
| 08:43.07 | WeazelON | well glad to hear its ok now |
| 08:43.13 | dandate2 | yeah it was an ightmare! |
| 08:43.29 | WeazelON | how fast was it fixed? |
| 08:43.45 | [T]ank | mosty: http://pastebin.ca/1358062 |
| 08:45.18 | WeazelON | [T]ank, are you using ubuntu by the way ? {regarding the zoiper} |
| 08:45.34 | [T]ank | not for asterisk... but for desktop |
| 08:46.00 | [T]ank | mosty: What i get when I have the CallerID: line is is a - Channel 0/1, span 1 got hangup, cause 99 error |
| 08:46.14 | [T]ank | if I take out the callerid line it goes through no problem at all |
| 08:46.15 | dandate2 | i got zoiper, i am using a sip inbound calling and iax outbound calling, do i need to make both sip and iax accounts? |
| 08:46.47 | mosty | [T]ank, is the caller id number valid for that zap channel? |
| 08:46.50 | WeazelON | dandate2, i'm gussing you are refering to the trunk |
| 08:46.57 | [T]ank | mosty: yes |
| 08:47.04 | dandate2 | no no in the zoiper setting |
| 08:47.34 | WeazelON | no what i mean is, when you say "Iax outbound calling" its through the IAX trunk right ? |
| 08:47.37 | mosty | [T]ank, as a workaround, you could try setting a channel variable with the callerid, and use that channel variable in your starting context to Set(CALLERID(number)=XXX) |
| 08:47.53 | dandate2 | yes |
| 08:48.00 | dandate2 | i use an iax trunk for outgoing calls |
| 08:48.21 | WeazelON | well then it doesnt matter which type of account u use |
| 08:48.25 | WeazelON | since its only the extension |
| 08:50.26 | WeazelON | the calls you'll make will start with sip and if outgoing it'll go using sip to the pbx and iax outgoing on the trunk itself |
| 08:50.42 | *** join/#asterisk DarkRift (n=dark@65.92.166.190) |
| 08:50.49 | WeazelON | althoguh if you plan on saving some bandwidth, you better off using iax |
| 08:50.58 | WeazelON | since its only in need of one initiation packet |
| 08:51.09 | WeazelON | and NOT every single call as for SIP |
| 08:52.18 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-8f70e84ec9f38ed7) |
| 08:54.11 | *** join/#asterisk joelbryan (n=joelbrya@203.177.143.137) |
| 08:54.26 | dandate2 | do i want to set allow=gsm on my iax trunk as well as sip? |
| 08:56.33 | WeazelON | allow=gsm means less bandwidth on the trunk itself not the extension's codec... |
| 08:57.59 | WeazelON | which means if you allow=ulaw or alaw which is the highest quality codec, your softphone will use the first priority of the codec in his list that fits the trunk's settings, in this case is ulaw or alaw if i remember ur trunk |
| 08:58.03 | dandate2 | hmm my phone system stopped working when i set allow=gsm |
| 08:58.17 | dandate2 | when i dialed in it said theire was nothing in existance here |
| 08:58.57 | dandate2 | disallow=all |
| 08:58.57 | dandate2 | allow=ulaw,alaw |
| 08:59.14 | dandate2 | was my settings that worked, when changing to gsm i can no longer receive inbound calls |
| 09:01.54 | *** part/#asterisk [T]ank (n=ckwall@c-71-199-25-239.hsd1.ut.comcast.net) |
| 09:02.53 | WeazelON | Zoiper shouldnt be a problem when using allow=gsm |
| 09:03.07 | WeazelON | the problem maybe with your sip provider who doesnt allow that codec |
| 09:03.10 | WeazelON | dont forget |
| 09:03.18 | WeazelON | Trunk settings must be identical at both sites |
| 09:03.19 | WeazelON | which means |
| 09:03.27 | WeazelON | your site and your sip provider's |
| 09:03.38 | WeazelON | unless its half duplex. |
| 09:06.35 | dandate2 | i see |
| 09:06.47 | dandate2 | i'm using didforsale.com would i have to contact them to know if they allow gsm? |
| 09:07.16 | mosty | besides just trying it, yes |
| 09:08.20 | WeazelON | tbh i believe you can try " allow=g729 " |
| 09:08.30 | WeazelON | thats another compressed codec which most can allow |
| 09:08.39 | dandate2 | i thought you had to purchase that no? |
| 09:08.52 | WeazelON | well that is totaly depending the sip provider i guess |
| 09:09.23 | WeazelON | i myself use a PRI line that is connected to the PBX and i handle all the codecs myself |
| 09:09.29 | WeazelON | so i can allow whatever i need. |
| 09:09.36 | dandate2 | hmm still failed |
| 09:10.02 | WeazelON | if you allow=g729 |
| 09:10.06 | WeazelON | it may not be enough |
| 09:10.36 | WeazelON | since you need to see if in the extension field of the FreePBX it has a disallow all , and in that case you need to add g729 in the allow field |
| 09:10.53 | WeazelON | otherwise, even though the trunk is set to take codec g729, the extension has it disabled, and will fail |
| 09:11.02 | dandate2 | i see |
| 09:11.07 | dandate2 | well the calls are suppsoed to go to queue |
| 09:11.18 | dandate2 | so any other setting but ulaw,alaw seems to be not even connecting to that |
| 09:14.50 | *** join/#asterisk KermitTheFragger (n=KermitTh@118-197.bbned.dsl.internl.net) |
| 09:15.52 | dandate2 | where can i find a did provider that will allow gsm? |
| 09:16.13 | WeazelON | i see, you are reffering solely for the inbound. well your inbound is sip provider only i'm afraid. |
| 09:17.15 | WeazelON | you can explore the terf of maybe adding a PRI card to the pbx and turn yourself into a self working PBX provider |
| 09:18.09 | *** join/#asterisk pcrack (n=pcrack@121.58.195.10) |
| 09:18.48 | pcrack | hi i have an asterisk hosted outside..my ip phones inside my LAN invironment having some problems its said congestion... |
| 09:19.23 | pcrack | i have a siproxd installed on my firewall to help my IP phone for NAT problems on sip |
| 09:20.01 | pcrack | any one can help me |
| 09:20.27 | pcrack | ? |
| 09:21.44 | *** join/#asterisk aiksa[LV] (n=aiks@mx.fiveplus.lv) |
| 09:21.52 | aiksa[LV] | Hi everyone. |
| 09:22.08 | WeazelON | heya |
| 09:22.46 | aiksa[LV] | I occasionally see a calls stuck at a Busy() application for ever. How could I catch them from AMI interface to do a forced doft hangup? |
| 09:23.43 | aiksa[LV] | that relates to Congestion() as well disregarding of channel type (Untill now I have seen it on ZAP, SIP and IAX) |
| 09:24.56 | aiksa[LV] | i could theoretically do a cron with "asterisk -rx "show channels" | grep Busy" but i consider this to be dirty workaround hack which I would rather avoid. |
| 09:25.05 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 09:25.12 | aiksa[LV] | any other ideas |
| 09:25.28 | aiksa[LV] | and how it gets into this status anyway> |
| 09:25.43 | aiksa[LV] | remote party not correctly signalling end of call? |
| 09:30.07 | *** join/#asterisk luca`gervasi (n=ashura@host218-170-dynamic.16-87-r.retail.telecomitalia.it) |
| 09:30.18 | Ashetic1 | Hallo |
| 09:31.02 | Ashetic1 | how can i have a verbose like the one you see on the second post here? http://www.voipuser.org/forum_topic_3784.html . I setup verbose 9999 and debug 9999...but it doesn't work. (Ast 1.6.0.4) |
| 09:32.58 | aiksa[LV] | Ashetic1: what is that part which is shown on that linmk but which you are not able to see? |
| 09:33.00 | *** join/#asterisk h-idrisi (n=h-idrisi@86.60.35.71) |
| 09:33.16 | aiksa[LV] | most of that looks pretty like standart verbose output |
| 09:33.46 | Ashetic1 | the part which says "-- Executing SetVar" and so on |
| 09:34.07 | Ashetic1 | i can call some extension from my phone, but the "verbose" output is not so verbose at all... |
| 09:35.34 | aiksa[LV] | dont know works for me verbosity of 4 |
| 09:35.37 | Ashetic1 | this is the only output i get when using "core set verbose 9999; core set debug 9999" and dialing in or out happens: "== Using SIP RTP CoS mark " |
| 09:36.19 | aiksa[LV] | Are you sure the output is routed to cli for you |
| 09:36.33 | aiksa[LV] | what does core show verbose says? |
| 09:36.46 | Ashetic1 | using agi set debug on, i can see the verbose debug of my agis |
| 09:37.02 | Ashetic1 | no such command |
| 09:37.30 | Ashetic1 | core show settings: |
| 09:37.37 | Ashetic1 | Verbosity: 2147483647 |
| 09:37.38 | Ashetic1 | <PROTECTED> |
| 09:39.22 | aiksa[LV] | strange |
| 09:39.35 | aiksa[LV] | nevertheless there is a configuration file /etc/asterisk/logger.conf |
| 09:39.58 | aiksa[LV] | which controllos what gets displayed where. take a look at it perhaps it has some strange settings |
| 09:40.14 | Ashetic1 | thanks... reading it :D |
| 09:42.05 | *** join/#asterisk ghenry (n=ghenry@ghenry.plus.com) |
| 09:45.12 | *** join/#asterisk dlu_dus (n=dlu@83.125.8.7) |
| 09:47.09 | dlu_dus | hi all |
| 09:52.52 | dlu_dus | is libss7 broken? |
| 09:53.37 | dlu_dus | i have tried to instal it like the readme in lib_ss7 explained and get a chan_dahdi error on making asterisk |
| 09:53.53 | dlu_dus | chan_dahdi.c: In function ‘ss7_reset_linkset’: |
| 09:53.54 | dlu_dus | chan_dahdi.c:9548: warning: passing argument 2 of ‘isup_grs’ makes pointer from integer without a cast |
| 09:53.56 | dlu_dus | chan_dahdi.c:9548: error: too many arguments to function ‘isup_grs’ |
| 09:53.57 | dlu_dus | chan_dahdi.c: In function ‘ss7_linkset’: |
| 09:54.15 | dlu_dus | and many more "too many argument" errors |
| 09:54.59 | Ashetic1 | http://rafb.net/p/azdK2W43.html <--- inbound calls doesn't hit the "inbound" context...they just get rejected... why? |
| 10:05.07 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
| 10:22.20 | *** join/#asterisk WeazelON (n=deazel@mail2.tikalnetworks.com) |
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| 11:00.32 | merlan8282 | hiho |
| 11:00.53 | merlan8282 | I have a problem with snom360 and BFL. |
| 11:02.03 | merlan8282 | All works well, but when one does a blind transfer and this called third person is busy, the LED on the monitoring snom360 keeps blinking until I reboot it. |
| 11:02.13 | merlan8282 | No change when user hangs up, and so on. |
| 11:02.22 | merlan8282 | Does anyone have a clue ? |
| 11:02.22 | dude7064 | The calling card companies usually have one single number for customers to use when calling,, to avoid having a busy signal, I am guessing they route the calls somehow to somewhere else,, but do they require one single phone line for every call ? meaning that if they had 200 simultaneous calls, they would need 200 different phone lines ?? |
| 11:03.45 | merlan8282 | BLF* I mean. |
| 11:25.18 | *** join/#asterisk Pazzo (n=ugelt@reserved-225136.rol.raiffeisen.net) |
| 11:31.02 | angryuser | merlan8282, how do you do your blind transfer ? * key code or transfer button ? |
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| 12:22.24 | merlan8282 | angryuser: through asterisk, with *[keycode] |
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| 12:23.02 | *** join/#asterisk SG25 (n=info@p50907EF9.dip.t-dialin.net) |
| 12:23.52 | *** join/#asterisk leikh (n=chatzill@port162.ds1-amb.adsl.cybercity.dk) |
| 12:24.18 | SG25 | hi there |
| 12:24.42 | leikh | I'm looking for 'Register expires' in sip.conf, but it isn't there. Wrong place? |
| 12:24.45 | merlan8282 | Mmm, found this, i'll try it out |
| 12:24.46 | merlan8282 | http://jkroon.blogs.uls.co.za/uncategorized/blf-asterisk-reloads-and-sip-registry |
| 12:25.04 | SG25 | i have a little question.. |
| 12:25.53 | Gido-E | ~ask |
| 12:25.54 | jbot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 12:26.17 | SG25 | i want to use a pbx at my home.. max. 5 sip phones, 1 isdn out/in line, but i have only a 350mhz, 512 ram pc here.. |
| 12:26.30 | Chainsaw | SG25: Compared to most embedded systems, that's quite a powerhouse. |
| 12:27.05 | Chainsaw | SG25: For up to 8 extensions I don't see a problem there. |
| 12:27.06 | SG25 | i read asterisk need min. 500mhz so? can i use it? |
| 12:27.50 | Chainsaw | Where did you read that? |
| 12:28.12 | SG25 | website and in the oreilly book |
| 12:28.39 | leikh | SG25: my asterisk is running on an nslu2 with just 266 MHz :) |
| 12:29.38 | Chainsaw | (Which is a Linksys device) |
| 12:29.38 | SG25 | what codec are you all using? ulaw? gsm? |
| 12:29.38 | SG25 | i think i want alaw.. because im in europe |
| 12:29.47 | Chainsaw | You want whatever your devices can support, basically. |
| 12:30.09 | merlan8282 | gsm is good too |
| 12:30.25 | SG25 | hm ok.. |
| 12:30.47 | leikh | SG25: I'm using G711u (ulaw) |
| 12:30.57 | Chainsaw | GSM is computationally more expensive. |
| 12:31.11 | SG25 | leikh: with how many calls? :) |
| 12:31.16 | Chainsaw | So with the more minimal machine you'll be using, probably ulaw/alaw. |
| 12:31.44 | leikh | SG25: max 3 |
| 12:33.41 | SG25 | hm.. i need 4.. maybe i must test that.. |
| 12:35.11 | SG25 | my 350 mhz pc is all ready in use a my router.. so.. ya.. thx all.. i will test it |
| 12:35.30 | *** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
| 12:35.36 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 12:35.39 | SG25 | is there a guide how i can connect a "AVM Fritz!Card" (euroisdn card 1port) whit asterisk? |
| 12:36.05 | SG25 | with* sry |
| 12:36.16 | jaytee | google is your friend |
| 12:36.30 | SG25 | no normaly he dont like me ;) |
| 12:36.47 | leikh | when using asterisk as a sip client, is 'register expires: 3600 sec' the correct setting for sip.conf? or should I use maxexpiry and minexpiry values? |
| 12:38.44 | mosty | leikh, "register expires" is not a valid setting in sip.conf |
| 12:39.06 | mosty | http://www.voip-info.org/wiki-Asterisk+config+sip.conf |
| 12:39.21 | leikh | mosty: thx :) |
| 12:39.27 | SG25 | what version would you recommand to use with my 350mhz machine? 1.4? |
| 12:40.07 | merlan8282 | at least |
| 12:42.48 | SG25 | k thx for now.. bye bbl |
| 12:43.56 | [TK]D-Fender | ... |
| 12:44.00 | jaytee | 350mhz? wow! what a speed demon |
| 12:44.37 | leikh | mosty: is defaultexpirey the equivalent of register expires? |
| 12:44.44 | [TK]D-Fender | exorcises SG25's "server". |
| 12:44.46 | [TK]D-Fender | THE POWER OF CHRIST COMPELS THEE! THE POWER OF CHRIST COMPELS THEE! THE POWER OF CHRIST COMPELS THEE! |
| 12:45.21 | *** join/#asterisk bird_of_Luck (n=melifaro@melifaro.astel.ru) |
| 12:46.07 | *** join/#asterisk frk2 (n=frk2@zivios/member/fkhan) |
| 12:46.51 | jaytee | hahaha |
| 12:49.01 | mvanbaak | hhmm, I have a party soon. Can I hire you as entertainer [TK]D-Fender ? |
| 12:49.10 | mosty | leikh, kind of, i suppose |
| 12:50.18 | leikh | mosty: my voip-provider blocked my ip-address because asterisk was registering all the time. They tell me to set it to 3600... |
| 12:51.29 | jaytee | [TK]D-Fender, I know I can get pri debug info to dump to a file but other than the CLI how can I get sip debug info to go to a file. Does it already go to a history log somewhere? I can't find anything in either the book or the WIKI that's current for 1.4. Just a mention of LOG_DEBUG for 1.2. |
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| 12:52.18 | merlan8282 | jaytee: oh yes, i've got the smae problem, with incoming SMses |
| 12:52.46 | merlan8282 | I only see incoming SMS whether on the CLI or trough telnet if i'm connected to my server |
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| 12:53.39 | mosty | try defaultexpirey and run a packet tracer to confirm that it's not re-registering too quickly |
| 12:53.52 | jaytee | I've been using a separate console instance and piping it's output through TEE to a file for now but that seems too roundabout. why didn't they put a sip set debug file command in 1.4 like they did for pri debug? |
| 12:57.11 | *** join/#asterisk bob_slacker (n=root@189.27.17.113.dynamic.adsl.gvt.net.br) |
| 12:57.34 | bob_slacker | hello! :D ne1 uses VICIDIAL ? |
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| 13:03.04 | *** mode/#asterisk [+o Mog] by ChanServ |
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| 13:11.55 | *** join/#asterisk telnettech (i=telnette@gw.percipia.com) |
| 13:12.02 | stefanlsd_ | Can i use blind pickup *8 for sip calls? |
| 13:14.13 | *** join/#asterisk fskrotzki (n=fskrotzk@host198.textwise.com) |
| 13:14.18 | merlan8282 | stefanlsd_: sure |
| 13:15.00 | [TK]D-Fender | merlan8282: * does not support SMS receipt or "SIP messaging" |
| 13:15.07 | *** join/#asterisk Katty (n=asterisk@mail.copi-rite.com) |
| 13:15.14 | [TK]D-Fender | jaytee: Dunno |
| 13:15.26 | [TK]D-Fender | mvanbaak: My rates are very accessible :) |
| 13:15.32 | *** join/#asterisk axisys (n=axisys@ip68-98-177-71.dc.dc.cox.net) |
| 13:15.38 | merlan8282 | [TK]D-Fender: mmm, why then can I send SMS with my DuoGSM card ? |
| 13:15.43 | merlan8282 | Maybe it's part of Bristuff ? |
| 13:16.10 | [TK]D-Fender | merlan8282: * can SEND SMS on certain channel types, but thats it. |
| 13:16.19 | merlan8282 | ok. |
| 13:16.23 | *** join/#asterisk bminish (n=bminish@2001:770:180:0:219:d1ff:fe80:ea64) |
| 13:16.42 | jaytee | [TK]D-Fender, I figured it out, at least as far as dumping debug info to a separate file other than messages but I think I'd need to modify my logger.conf file everytime I wanted to turn on sip debug or turn it off again so I don't end up with a terabyte of crap I don't need. |
| 13:18.55 | [TK]D-Fender | jaytee: rotating logs |
| 13:19.15 | *** join/#asterisk bob_slacker (n=root@189.27.17.113.dynamic.adsl.gvt.net.br) |
| 13:19.23 | jaytee | rotating logs |
| 13:19.28 | jaytee | logger rotate |
| 13:20.17 | mvanbaak | [TK]D-Fender: ;) |
| 13:21.04 | *** join/#asterisk Khratos (n=khratos@190.166.103.111) |
| 13:21.08 | jaytee | [TK]D-Fender, sure that's fine, but that won't keep me from running out of disk space, I'd still have to disable debug logging or would doing a sip set debug off stop any logging to that file? |
| 13:21.47 | merlan8282 | In fact, my bristuffed asterisk is able to receive SMSes, but only visible in the console. And they're not decoded, they're shown in "PDU" format. |
| 13:22.19 | *** join/#asterisk fcois93 (n=Administ@bagnolet.acropolistelecom.net) |
| 13:22.29 | fcois93 | anyone know SS7 ? |
| 13:22.46 | fcois93 | I need to know all the 3 first digist possible |
| 13:25.40 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 13:25.58 | espent | hey |
| 13:26.08 | espent | this shit is sent to asterisk from my sip-server: http://rafb.net/p/kJWotd68.html |
| 13:26.22 | *** join/#asterisk jjshoe (n=jjshoe@h69-129-142-83.mdsnwi.tisp.static.tds.net) |
| 13:26.55 | espent | why isnt that read as hangup, and then could be catched by extension h, which in my situation runs a DeadAgi-script |
| 13:27.43 | [TK]D-Fender | espent: If an INVITE is getting refused then there is no accepted call to hangup. |
| 13:27.49 | *** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim) |
| 13:30.55 | *** join/#asterisk UQlev (n=kvirc@91.184.221.31) |
| 13:31.18 | espent | [TK]D-Fender: hm it is not refused, it just says internal server error after the invite is completed |
| 13:32.08 | espent | [TK]D-Fender: i think it happens because i started 90 channels, and all the bandwidth got used up |
| 13:33.55 | [TK]D-Fender | espent: You'd have to show actual CLI to back up the call's overall process |
| 13:34.54 | ThoMe | is it posible sip over secure phone connection? |
| 13:35.15 | *** join/#asterisk ltd (n=z@pat.transact.net.au) |
| 13:35.17 | *** join/#asterisk drfreeze (n=Jim@207.191.114.82) |
| 13:35.21 | drfreeze | Morning |
| 13:35.51 | WeazelON | morning idd |
| 13:37.14 | drfreeze | I was at an office the other day and noticed a phone system where the staff would call a phone and instead of ringing, the speaker would activate |
| 13:37.51 | drfreeze | Anyone know what that feature is called? |
| 13:38.10 | [TK]D-Fender | drfreeze: "auto answer" |
| 13:38.53 | *** join/#asterisk mbranca (n=matteo@2001:1418:130:0:21e:8cff:fe51:5b05) |
| 13:40.59 | WeazelON | you can check the feature for it "Paging & intercom" at Freepbx gui |
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| 13:45.25 | *** mode/#asterisk [+o putnopvut] by ChanServ |
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| 13:53.31 | espent | [TK]D-Fender: show actual CLI? you mean turn on debug on the channel? problem is that this only occurs when i got about 40+ channels running, so its a bit difficult to find out whats going wrong |
| 13:54.01 | [TK]D-Fender | espent: Well we have nothing solid to look at. We really can't comment at this point. |
| 13:54.46 | *** join/#asterisk coppice (n=chatzill@46.166.17.210.dyn.pacific.net.hk) |
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| 13:58.32 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 13:58.32 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 13:58.36 | lmadsen | ~book |
| 13:58.36 | jbot | somebody said book was probably Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://tfot.leifmadsen.com or see ~buybook |
| 13:58.50 | *** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose) |
| 13:59.30 | lmadsen | jbot: no, book is Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 13:59.31 | jbot | lmadsen: okay |
| 14:01.14 | *** join/#asterisk RobertLaptop (n=rmiddle@63.68.135.4) |
| 14:01.52 | *** join/#asterisk nima0102 (n=nima@91.98.193.4) |
| 14:02.18 | [TK]D-Fender | ~book |
| 14:02.19 | jbot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 14:02.27 | apeiron | ~buybook |
| 14:02.28 | jbot | You can buy "Asterisk the Future of Telephony" at http://www.oreilly.com/catalog/9780596510480/ so go buy it SERIOUSLY |
| 14:02.45 | lmadsen | [TK]D-Fender: I changed the link to the HTML version |
| 14:03.23 | [TK]D-Fender | lmadsen: I know... and I host a mirror of it ;) |
| 14:05.49 | lmadsen | have you seen the new site though? |
| 14:06.08 | lmadsen | we generate our own html now from the svn docbook sources |
| 14:07.40 | drfreeze | [TK]D-Fender: thanks |
| 14:08.28 | *** join/#asterisk arpu (n=arpu@chello062178159144.10.14.univie.teleweb.at) |
| 14:08.29 | [TK]D-Fender | lmadsen: Nope, will have to get around to that |
| 14:09.00 | *** join/#asterisk n3hxs (n=HAMming@63.68.135.4) |
| 14:10.25 | lmadsen | [TK]D-Fender: we're using the stuff from the subversion guys now, so we can generate some nice looking html :) |
| 14:10.54 | [TK]D-Fender | lmadsen: As long as I can wget-mirror it, its all good :) |
| 14:11.13 | lmadsen | most likely should be able to :) |
| 14:11.24 | lmadsen | so that's your "update" notice :) |
| 14:15.36 | *** join/#asterisk tzafrir_laptop (n=tzafrir@local.xorcom.com) |
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| 14:16.39 | Arsenick- | Hi all, I have a polycom related question if anobody can help... I want my phone to get their config from a ftp server provided by dhcp via the boot-server option but in the option 66 string I provide the username/password ( ftp://user:pass@10.10.0.2") but the phone don't use it.. I tried to reset to factory default but the phone still don't want to log into my ftp server, is there a special setting for this kind of set |
| 14:16.39 | Arsenick- | up to work ? |
| 14:16.49 | Arsenick- | j #trixbox |
| 14:17.01 | Arsenick- | sry.. |
| 14:17.12 | espent | [TK]D-Fender: anyway, generally, if a channel is just terminated, without any hangup, is there a posibility to catch it in my extension-table? |
| 14:17.20 | jjshoe | polycom has an option in it for specifying the user/pass i thought |
| 14:17.41 | Arsenick- | yup |
| 14:18.11 | *** join/#asterisk axisys (n=axisys@155.70.141.45) |
| 14:18.25 | Arsenick- | but i don't really want to pass phone by phone to change the username apssword... in a lot of howto I've read they apss the username apssword in the url like I said and nobody talk about special setting on the phone.. |
| 14:18.58 | jjshoe | so temporarily change the user/pass to whatever is already in the phones |
| 14:19.51 | Arsenick- | yup I know this option but I'm trying to find out why he don't read my boot-server string correctly.. |
| 14:19.54 | *** join/#asterisk harry__ (n=h@imperialglamour.com) |
| 14:20.04 | espent | [TK]D-Fender: not extension, but dialplan i mean |
| 14:20.04 | *** join/#asterisk [intra]lanman (n=intralan@freeswitch/developer/intralanman) |
| 14:21.22 | Arsenick- | jjshoe, anyway thx for helping |
| 14:21.52 | *** join/#asterisk orkid (n=orkid@unaffiliated/orkid) |
| 14:22.09 | *** part/#asterisk harry__ (n=h@imperialglamour.com) |
| 14:24.09 | orkid | I have a weird issue that I can't find on google. Recently (within a day or so) I've been getting weird 'beeps' when I'm on a zap channel (digium FXO pci card) via my ATA.. both incoming and outgoing... it does not happen with calls via sip alone, so I'm ruling out the ATA. Any ideas? |
| 14:24.47 | orkid | try a different pci slot? :) |
| 14:25.10 | orkid | (I've change motherboards recently.. actually this started to happen after a motheboard change probably) |
| 14:25.37 | jjshoe | message waiting? |
| 14:26.38 | Chainsaw | jjshoe: That's supposed to be an intermittent interruption in the dial tone though. Should not occur during a call. |
| 14:26.45 | orkid | no, i don't have that service. it actually sounds like the beep that our phone makes when you dial a button (not dtmf, and not like it really says much, unless you knows how a uniden cordless from a while ago sounds :) |
| 14:27.45 | Chainsaw | orkid: It's worth checking whether a BIOS upgrade is available for said mainboard. |
| 14:27.48 | orkid | and it's not periodic really, but it's the same pitch all the time. and also sometimes the call will not go through, though console says Zap-1 'answers' the call is silent. |
| 14:28.09 | Chainsaw | orkid: What OS are you on? Linux? |
| 14:28.14 | orkid | Chainsaw: will do, but i doubt it. it's a k8v-x out of production. Could it be that it doesn't like a certain pci slot? |
| 14:28.21 | orkid | yes, ubuntu hardy server. |
| 14:28.44 | Chainsaw | orkid: If said slot forces it to share resources with other hardware, yes, that wouldn't help. |
| 14:29.10 | orkid | I'll check on that, since I've had issues with this card and a certain PCI slot on another mobo |
| 14:29.20 | orkid | Thanks |
| 14:29.27 | Chainsaw | Generally the lowest one on the board is shared with on-board devices. |
| 14:29.34 | Chainsaw | (Assuming a tower case, standing upright) |
| 14:29.50 | Chainsaw | It's worth checking in /proc/interrupts. |
| 14:30.05 | Chainsaw | In my case, the wctdm driver has the interrupt line to itself: |
| 14:30.07 | Chainsaw | <PROTECTED> |
| 14:30.13 | orkid | I also went from a 'real' CPU (P3) to a low cache one, a duron. would that help? |
| 14:30.23 | orkid | erm hinder |
| 14:30.45 | *** join/#asterisk botox93 (n=botox93@213.221.82.242) |
| 14:30.46 | Chainsaw | Depends on the speed of the Duron really. Asterisk isn't a supercomputing application that needs tons of CPU. |
| 14:30.48 | *** join/#asterisk VJFROMGT (n=vjfromgt@user-12lcpfg.cable.mindspring.com) |
| 14:31.00 | orkid | oh yeah sharing interrupt with ethernet, i bet that's doing it :P |
| 14:31.03 | VJFROMGT | in which file can i set fake ringback for zap channels? |
| 14:31.26 | Chainsaw | orkid: Quite likely. As they come, NICs are fairly interrupt-intensive. |
| 14:31.37 | Chainsaw | orkid: If you have the lowest PCI slot, try moving the card up one slot. |
| 14:32.13 | orkid | It's not in the lowest slot iirc, but I'll try to find a slot where the interrupt's not shared. Thanks a lot for the help. |
| 14:32.19 | Chainsaw | Any time. |
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| 14:38.51 | [TK]D-Fender | espent: If you're in the dialplan it'll look for "h" whever they are |
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| 14:49.00 | dwery | hi, I'm getting a bunch of zaphfc: bchan rx fifo not enough bytes to receive! (z1=6138, z2=6131, wanted 8 got 7), probably a buffer overrun. |
| 14:49.05 | dwery | anything I coul do? |
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| 14:57.39 | espent | [TK]D-Fender: thats the point - i got the h extensions, but its not called |
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| 14:59.23 | [TK]D-Fender | espent: remember its in the current context... careful where you are in your dialplan. Also... show us. |
| 14:59.31 | [TK]D-Fender | espent: you need to find an error to report |
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| 15:12.09 | espent | [TK]D-Fender: i think maybe you gave me the right hint there |
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| 15:17.01 | ThoMe | 14:35:26 < ThoMe> is it posible sip over secure phone connection? |
| 15:19.13 | ThoMe | i mean, sip over a secure connectino? wihtout vpn? |
| 15:19.33 | merlan8282 | mmm |
| 15:19.41 | merlan8282 | SSL/TLS ? SSH ? |
| 15:19.42 | ThoMe | merlan8282: hm? |
| 15:19.50 | mosty | thome: i don't believe asterisk supports SRTP (yet), see here: http://bugs.digium.com/view.php?id=5413 |
| 15:19.56 | *** join/#asterisk SamaelA (n=samael@innotel.kiev.farlep.net) |
| 15:20.03 | mosty | you might be able to get that patch to work |
| 15:20.12 | SamaelA | Hello |
| 15:20.15 | brutuz | how can you say to stop the loop in the dial plan? |
| 15:20.35 | brutuz | get a leased-line upto the sip provider.. |
| 15:20.39 | ThoMe | mosty: hm. end SRTP iss standard in snom? |
| 15:21.03 | *** join/#asterisk CrazyTux (n=brandon@c-98-196-6-54.hsd1.tx.comcast.net) |
| 15:21.16 | mosty | i believe snom phones support SRTP |
| 15:21.40 | *** join/#asterisk killown (n=ukendt@unaffiliated/killown) |
| 15:21.40 | ThoMe | mosty: ok. |
| 15:21.42 | brutuz | when will Goto stop from Goto? |
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| 15:21.46 | SamaelA | I have one problem with conference. MeetMe creates room, all ok, but voice is so distorted and delay is nearly 4-5 sec. Using g729 |
| 15:21.57 | jasonwoot | don'tGoto? |
| 15:21.57 | ThoMe | mosty: i have a question > dialplan. can you help me also? :-) |
| 15:22.09 | mosty | ThoMe, ask here, maybe someone will help |
| 15:22.15 | ThoMe | mosty: hihi :-) ok |
| 15:22.23 | jjshoe | SamaelA all sip system? |
| 15:22.23 | file | there is a branch for SRTP, don't use the patches |
| 15:22.29 | mosty | SamaelA, do you have a hardware timer or are you using ztdummy? |
| 15:22.48 | SamaelA | ztdummy. and asteriskNOW |
| 15:22.59 | jjshoe | SamaelA buy the sangoma timing device. |
| 15:23.01 | ThoMe | mosty: have a little menu. and: Read(ziel,vm-extension) and GotoIf($["${ziel}" != "0000"]?durchwahl:spy) |
| 15:23.29 | brutuz | jasonwoot: any bright doc supported soln? |
| 15:23.29 | SamaelA | jjshoe: Yes, all sip. Polycom phones |
| 15:23.29 | jjshoe | SamaelA buy the sangoma timing device. |
| 15:23.29 | ThoMe | mosty: but i must press the "#" after the numbers. can i say only the numbers without "#" ? |
| 15:23.29 | mosty | SamaelA, what kernel version, and is this by chance a dell computer? |
| 15:23.29 | [TK]D-Fender | brutuz: huh? |
| 15:23.39 | ThoMe | mosty: wait 2 seconds after the last press... and then gotoIF ? |
| 15:23.49 | SamaelA | mosty: No... Old Cel 433 Calculator ))) |
| 15:23.55 | mosty | ThoMe, you can specify maxdigits in the Read command |
| 15:24.16 | ThoMe | ah ok |
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| 15:24.23 | ThoMe | , 4 ? |
| 15:24.30 | brutuz | [TK]D-Fender: if user didn't give any input use Goto .. but when will it stop? |
| 15:24.45 | SamaelA | jjshoe: And what mixing device astrix is using? Sound card or software? |
| 15:24.55 | [TK]D-Fender | brutuz: What on earth are you talking about? Any input on WHAT? |
| 15:25.07 | [TK]D-Fender | brutuz: pastebin what you're working on. |
| 15:25.09 | [TK]D-Fender | ~pb |
| 15:25.10 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines 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, or , http://bin.cakephp.org/ |
| 15:25.33 | SamaelA | mosty: 2.6.22 kernel |
| 15:25.38 | jjshoe | SamaelA hu? |
| 15:25.40 | jjshoe | SamaelA huh? |
| 15:25.53 | brutuz | [TK]D-Fender: im on it.. |
| 15:25.56 | brutuz | [TK]D-Fender: http://pastebin.com/d67560ea8 |
| 15:26.17 | mosty | SamaelA, is it a dell machine? |
| 15:26.30 | [TK]D-Fender | brutuz: that already looks fine. On timeout it will repeat the menu |
| 15:26.39 | SamaelA | SamaelA: Means i know, that ztdummy is not the best way. But who multiplex channels? |
| 15:26.47 | ThoMe | mosty: but what is the difference ReadExten and Read ? |
| 15:26.52 | SamaelA | mosty: No |
| 15:27.06 | SamaelA | jjshoe: Means i know, that ztdummy is not the best way. But who multiplex channels? |
| 15:27.25 | mosty | ThoMe, well Read is an asterisk dialplan command, and i have never heard of ReadExten |
| 15:27.32 | *** join/#asterisk nfi|ermes (n=ErMeS@217.220.121.62) |
| 15:27.40 | harry__ | apparently rev 43814 added BACKGROUNDSTATUS, but yet: https://gist.github.com/0a4d5e01a494b13cd71b |
| 15:27.48 | ThoMe | mosty: but is it posible set a timeout with read? |
| 15:27.53 | brutuz | [TK]D-Fender: programmatically speaking it will do an endless loop.. unless the use inputted something usefull |
| 15:28.01 | mosty | SamaelA, asterisk does the multiplexing itself with the help of a timing source (in your case, ztdummy) |
| 15:28.02 | harry__ | the first one was played through, the second not. |
| 15:28.06 | harry__ | yet both returns "0" |
| 15:28.14 | ThoMe | ah ok |
| 15:28.22 | [TK]D-Fender | brutuz: well... you also show no options in there. there is nothing for them o dia, and you don't have an invalid handler |
| 15:28.25 | mosty | ThoMe, sure. http://www.voip-info.org/wiki-Asterisk+cmd+Read |
| 15:28.31 | ThoMe | ok :-) thank yo U |
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| 15:29.07 | SamaelA | mosty: Maybe better processor can help me? ))) |
| 15:29.14 | [TK]D-Fender | SamaelA: * uses a Zaptel card for timing, or if not present, ztdummy (or DAHDI equivalent) |
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| 15:29.26 | [TK]D-Fender | SamaelA: What are you running on now? |
| 15:29.39 | SamaelA | [TK]D-Fender: Yes |
| 15:29.50 | *** join/#asterisk Illarane (n=heifer@pdpc/supporter/monthlygold/illarane) |
| 15:29.53 | rbd | hi, for SIP or Local type channels, what's the best way in an extensions.conf dialplan to detect if the caller is still on the line (e.g. I want to execute some steps after a Meetme() command only if the caller didn't hang up during the Meetme() command) ...as it is now I'm just running these commands after and asterisk is saying the channel doesn't exist any more |
| 15:29.56 | brutuz | [TK]D-Fender: http://pastebin.com/d139f0437 |
| 15:29.59 | mosty | SamaelA, what CPU do you have? |
| 15:30.09 | brutuz | [TK]D-Fender: i didn't paste the whole thing.. |
| 15:30.17 | SamaelA | mosty: Celeron 433 ))) |
| 15:30.32 | rbd | SamaelA, lord Jesus |
| 15:30.36 | brutuz | [TK]D-Fender: i don't know if that's relevant to my inquiry |
| 15:30.48 | [TK]D-Fender | brutuz: exten => 3,1,Goto(s,1,supportment) <- order is wrong |
| 15:31.03 | brutuz | [TK]D-Fender: typo error |
| 15:31.11 | SamaelA | mosty: But according to TOP CPU usage nearly 2-3% |
| 15:31.12 | [TK]D-Fender | brutuz: Ok, so you have it loop on invalid, and on timeout. What do you want it to do instead? |
| 15:31.21 | SamaelA | rbd: ))) Old server |
| 15:31.48 | [TK]D-Fender | SamaelA: 2-3% hile this conference is going on? |
| 15:31.50 | Illarane | Got a weird thing happening with out phones. They seem to be syncing to exactly an hour ahead of the actual time (we're in London, and they should be showing 15:30, but instead show 16:30). I'm not entirely sure if this is down to something my predecessor did, or the fact that we're using trixbox to run the phone system, but it's really annoying and changing it on the phone is only temporary. Anyone got any ideas? |
| 15:32.00 | Illarane | The time on the trixbox... box... is correct. |
| 15:32.17 | mosty | Illarane, are the phones autoprovisioned? |
| 15:32.18 | ThoMe | mosty: is this correct: Read(ziel,vm-extension,4, 1) |
| 15:32.20 | [TK]D-Fender | Illarane: Its probably using the new US DST changes |
| 15:32.58 | ThoMe | mosty: ah right, works fine :-) thank you |
| 15:33.02 | [TK]D-Fender | Illarane: and Trixbox is not supported here. |
| 15:33.09 | brutuz | [TK]D-Fender: so if the user didn't input anything.. timeout will be executed/run.. w/c is Goto if the user will not input anything when will it break from the loop? |
| 15:33.16 | SamaelA | [TK]D-Fender: 2-3% CPU usage. Means much system res is free )) |
| 15:33.35 | SamaelA | [TK]D-Fender: While conf |
| 15:33.38 | [TK]D-Fender | brutuz: No, it LOOPS if they enter nothing. thats what "t" is for |
| 15:33.44 | Illarane | [TK]D-Fender: I know nothing about asterisk apart from that it's a pain. ;) |
| 15:33.49 | [TK]D-Fender | SamaelA: rather unusual |
| 15:34.12 | [TK]D-Fender | Illarane: And your probalem has nothing to do with Asterisk |
| 15:34.22 | SamaelA | [TK]D-Fender: I switched to g711. Result the same |
| 15:34.26 | Illarane | mosty: I think so, yes. They're Linksys SPA942s which reboot every couple of hours for no apparent reason, which I assume is to do with the auto-prov. |
| 15:34.42 | brutuz | [TK]D-Fender: ok if they enter nothing it loops.. on the next loop they didn't enter anything.. on the next loop same thing.. when will it break from the loop? |
| 15:34.56 | [TK]D-Fender | brutuz: When they hang up |
| 15:35.00 | Illarane | [TK]D-Fender: If Asterisk sends out the time to its peers, then it does. :) But I don't know if it does, hence asking here. |
| 15:35.10 | [TK]D-Fender | Illarane: It doesn't |
| 15:35.11 | mosty | Illarane, that's most likely where your problem is, but i can't help with trixbox, you should try #trixbox |
| 15:35.27 | brutuz | [TK]D-Fender: if they didn't? what happens is there something like after 5 loops hangup |
| 15:35.27 | Illarane | [TK]D-Fender: Incidentally, if you don't want me to ask in here, some pointers as to the correct place would be nice. :) |
| 15:35.30 | Illarane | mosty: Thanks. |
| 15:35.50 | brutuz | [TK]D-Fender: if they didn't? what happens? Is there something like after 5 loops hangup |
| 15:35.57 | [TK]D-Fender | brutuz: when the CALLER hangs up. |
| 15:36.06 | [TK]D-Fender | \brutIf they don't hang up it will go on forever. |
| 15:36.21 | [TK]D-Fender | brutuz: YOU have to put logic in your loop to limit how many chances they get |
| 15:36.49 | *** join/#asterisk neurosys (n=vinix@173.9.159.182) |
| 15:36.53 | brutuz | [TK]D-Fender: how do you do that? is there a counter that i can increment? |
| 15:37.14 | brutuz | like while [ loopctr < 5 ]; Goto bla... ; |
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| 15:37.37 | *** part/#asterisk Illarane (n=heifer@pdpc/supporter/monthlygold/illarane) |
| 15:38.06 | [TK]D-Fender | brutuz: There is... after you code it all in your dialplan yourself |
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| 15:39.51 | brutuz | [TK]D-Fender: can you give me a simple example? |
| 15:40.41 | [TK]D-Fender | brutuz: http://www.voip-info.org/wiki-Asterisk+tips+IVR+menu |
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| 15:44.09 | SamaelA | Ok.... Thanks ) Try to find better calculator ))) |
| 15:51.21 | WeazelON | Weazel Off |
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| 15:54.08 | SuPrSluG | hello |
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| 15:57.26 | SuPrSluG | is there a way to detect voicemail and go to another number if not answered. I'd like to call several cell phones but not got to their cell phone voicemail. |
| 15:57.47 | jjshoe | SuPrSluG most folks ask the user to press one to accept the call |
| 16:01.12 | *** join/#asterisk questprojects (n=mg@82-71-8-45.dsl.in-addr.zen.co.uk) |
| 16:01.35 | czindy | Hello, I would like to ask for helpof the following: I installed unixodbc / freetds to connect asterisk cdr to mssql. the odbc connection is tested and working to the mssql server, but I got the error from asterisk cli: ERROR[15790]: cdr_odbc.c:133 odbc_log: Unable to retrieve database handle. CDR failed. Please help why asterisk cannot handle this. |
| 16:01.38 | SuPrSluG | jjshoe, thanks i'm trying to get * to ring each cell as many times as it can, but NOT go to vm. Then it will dial the next number and loop 3 times and then it will go to an answering service. |
| 16:01.54 | *** join/#asterisk soylentgreen (n=fgast@missbehave.only640k.net) |
| 16:03.33 | [TK]D-Fender | SuPrSluG: M() + AMD |
| 16:04.46 | *** join/#asterisk Ericounet (n=Ericoune@ACaen-151-1-23-21.w86-215.abo.wanadoo.fr) |
| 16:05.09 | *** part/#asterisk Ericounet (n=Ericoune@ACaen-151-1-23-21.w86-215.abo.wanadoo.fr) |
| 16:05.23 | SuPrSluG | [TK]D-Fender, thanks that sounds like what i need. documented? |
| 16:06.35 | SuPrSluG | [TK]D-Fender, is it reliable? %= |
| 16:11.16 | czindy | Hello, I would like to ask for help of the following: I installed unixodbc / freetds to connect asterisk cdr to mssql. the odbc connection is tested and working to the mssql server, but I got the error from asterisk cli: ERROR[15790]: cdr_odbc.c:133 odbc_log: Unable to retrieve database handle. CDR failed. Please help why asterisk cannot handle this. (debian / asterisk 1.6) Can I check it why asterisk cannot get handler? |
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| 16:23.17 | SuPrSluG | [TK]D-Fender, What's the M() |
| 16:23.32 | [TK]D-Fender | SuPrSluG: "core show application dial" |
| 16:25.41 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
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| 16:31.38 | brutuz | <PROTECTED> |
| 16:31.46 | docid | so, anybody know what the asterisk equlivant signaling is for T1 CAS? |
| 16:32.29 | tzafrir_laptop | docid, depends what you run on top of it |
| 16:33.05 | tzafrir_laptop | CAS is a somewhat equivalent of having 24 (T1) or 30 (E1) separate copper wires |
| 16:33.26 | tzafrir_laptop | (somewhat. Not complete. I know) |
| 16:33.35 | docid | hrmm, well, the UTStarcom box is configured for T1 CAS, AMI, D4, and uses a table called em.dat.... yes, been doing lots of reading on it, just cant figgure out how to configure asterisk to work with it |
| 16:34.17 | tzafrir_laptop | Those parameters should help you set zaptel.conf / system.conf |
| 16:34.40 | tzafrir_laptop | Maybe they just use FXS/FXO signalling on top of that? |
| 16:34.49 | docid | well, i got most of it....im just haveing trouble with the signaling= field |
| 16:35.06 | tzafrir_laptop | waits for someone with experince to offer a better advice |
| 16:35.19 | *** join/#asterisk RoPBX (n=nickserv@200.93.34.175) |
| 16:35.32 | docid | hrmm, didnt think of that......the telco says we have MF signaling... but cant seem to get what feature group MF were useing |
| 16:35.50 | docid | telco not being helpfull |
| 16:41.22 | *** join/#asterisk KermitTheFragger (n=KermitTh@118-197.bbned.dsl.internl.net) |
| 16:42.42 | *** join/#asterisk djin (n=djin@84-104-110-179.cable.quicknet.nl) |
| 16:44.23 | *** part/#asterisk merlin8282 (n=merlin82@AStrasbourg-753-1-23-119.w90-56.abo.wanadoo.fr) |
| 16:47.10 | *** join/#asterisk Imo (n=Imo@brln-4db9f2fd.pool.einsundeins.de) |
| 16:47.14 | Imo | hello |
| 16:47.36 | Imo | i want uninstall asterisk 1.6 and install asterisk 1.4 |
| 16:47.45 | Imo | how can i do this ? |
| 16:48.21 | mort_gib | imo: Did you run into problems?? |
| 16:49.01 | Imo | yes very big problems with asterisk 1.6 |
| 16:50.01 | Imo | mort_gib: you want listen what for problems ? |
| 16:50.03 | Ritzerisk | is it possilbe with asterisk for voicemail to text |
| 16:50.12 | Ritzerisk | like an addon |
| 16:50.31 | mort_gib | Imo: Yeah, I'm going live with a "demo" system with a clients this week |
| 16:51.19 | *** join/#asterisk takashi_85 (n=ahmed@41.196.80.16) |
| 16:51.23 | Imo | mort_gib: my first problem was, when i call a registered sip account, i dont get the incomming call |
| 16:51.40 | *** part/#asterisk takashi_85 (n=ahmed@41.196.80.16) |
| 16:51.56 | *** join/#asterisk SkywaIker (n=pirch@58.147.17.166) |
| 16:52.01 | *** join/#asterisk stevetotaro (n=Steve@pool-72-72-143-197.hrbgpa.dsl-w.verizon.net) |
| 16:52.14 | mort_gib | Imo: Huh?? So incoming calls from a SIP account didn't work for you?? |
| 16:52.37 | Imo | mort_gib: i needed insecure=port,invite |
| 16:53.06 | Imo | and now i can't call stabil |
| 16:53.22 | Imo | all the time broke up the calls after 1 minute |
| 16:53.24 | mort_gib | Imo: So the SIP account didn't register correctly?? |
| 16:53.28 | Imo | sorry for my bad english ;) |
| 16:53.43 | Imo | no my sip account registered |
| 16:53.48 | mort_gib | I have a Soekris 5501/Debian/Asterisk 1.6 running rock solid |
| 16:53.58 | Imo | nsecure=port,invite |
| 16:54.06 | Imo | insecure=port,invite |
| 16:54.10 | mort_gib | Don't worry about your English |
| 16:54.28 | Imo | with this line i can get incomming calls |
| 16:54.59 | tzafrir_laptop | Imo, what version of 1.6, exactly? |
| 16:55.13 | Imo | Connected to Asterisk 1.6.0.6 currently running on server (pid = 4260) |
| 16:56.13 | Imo | sometimes i can call |
| 16:56.28 | Imo | but its so often that the call brokes up after a few minutes |
| 16:57.01 | Imo | and i have installed the asterisk server in a company and the peoples are very angry about me ;) |
| 16:57.30 | Imo | with asterisk 1.4 i dont had problems |
| 16:57.56 | Imo | how i can uninstall the version 1.6 and install asterisk 1.4 ?? |
| 16:58.15 | mmlj4 | while trying to compile wanpipe on suse linux: ‘struct dahdi_chan’ has no member named ‘rxsig’ |
| 16:58.46 | Imo | can anybody help me ? |
| 16:59.02 | Gido-E | Imo isn't that related to your distro? |
| 16:59.46 | apeiron | mort_gib, What kind of volume does your Soekris see? |
| 17:00.28 | mort_gib | aperion: I had 5 calls over it to test it... Not much normally, never more than 3 concurrent calls |
| 17:00.57 | apeiron | mort_gib, Excellent. Thanks for the info. Probably going to pick up one of those, then, or one like it. |
| 17:01.19 | Imo | Gido-E: what du you mean ? |
| 17:01.28 | mort_gib | aperion: I have a Soekris 5501/Debian/Asterisk 1.4/Sangoma A200 system that has 10 phones connected and at least 2 calls at any given time |
| 17:01.59 | mort_gib | aperion: I had 5 participants in MeetMe using g729 and CPU was at 75% |
| 17:02.02 | *** join/#asterisk bob_slacker (n=root@189.27.17.113.dynamic.adsl.gvt.net.br) |
| 17:02.13 | mort_gib | Yelling their silly little heads off :-) |
| 17:02.39 | mort_gib | aperion: I'm running that 5501 off a Sandisk 2GB card :-) |
| 17:04.33 | Imo | Gido-E: i have ubuntu server 8.10 |
| 17:04.51 | *** join/#asterisk StanManCan (n=stan_man@S010600195b3059b4.gv.shawcable.net) |
| 17:05.16 | StanManCan | okay....... did anybody here take my number and am using it to span phone calls to people.. ?: ( |
| 17:05.34 | apeiron | mort_gib, rofl, nice. :D |
| 17:06.28 | *** join/#asterisk RobH (n=RobH@75.101.56.124) |
| 17:06.51 | *** join/#asterisk anonymouz666 (n=anonymou@189.24.56.203) |
| 17:07.12 | StanManCan | About a week after I got help in here I was getting phone calls from people in south carolina saying that they got a call from me and blahblahblah... Got about 40 of them in a weekend. Then it stopped. |
| 17:07.50 | StanManCan | Now it's happening again excepting I've got about 75-80 in the lsat 48 hours saying that they got a message or a phone call about creidt card fraud /or/ accounts being deactivated ect. |
| 17:09.11 | *** join/#asterisk bob_slacker (n=root@189.27.17.113.dynamic.adsl.gvt.net.br) |
| 17:09.27 | kaldemar | StanManCan: do you see such calls in your log? |
| 17:10.04 | [TK]D-Fender | StanManCan: Has anyone traced any of those calls? And indeed, do you have any activity to hint that your system has been compromised? |
| 17:10.30 | Corydon76-dig | StanManCan: I'd say that somebody has cracked one or more of your accounts. Tell me you aren't using all numeric passwords |
| 17:10.54 | Qwell | numeric passwords that also happen to match your username... |
| 17:11.24 | russellb | that's bad, mmmkay |
| 17:11.26 | StanManCan | User name and password is the same, something wrong witht hat ? |
| 17:11.33 | russellb | numeric? |
| 17:11.34 | StanManCan | lol!! kidddding |
| 17:11.57 | russellb | there are a _lot_ of voip brute force attacks going on |
| 17:12.00 | StanManCan | username and passwords are different, my PBX and Voip accounts are safe and fine. I've checked the outgoing call logs on my Voip provider |
| 17:12.03 | russellb | people looking for systems to use for making calls |
| 17:12.03 | [TK]D-Fender | StanManCan: Yes, somethine is very wrong with that. Were all the cool PW's like "12345" and "qwerty" already taken? |
| 17:12.46 | StanManCan | Fender: I've always been careful with my passwords. 8 characcters or longer using letters, numbers, and symbols |
| 17:12.52 | *** join/#asterisk |Krnl| (n=kvirc@190.105.18.156) |
| 17:13.08 | Qwell | on *all* accounts? with guest disabled? on a recent version of Asterisk? |
| 17:13.08 | *** part/#asterisk |Krnl| (n=kvirc@190.105.18.156) |
| 17:13.54 | StanManCan | My PBX is fine. My VOIP provider logs all calls in and out and there's nothing fishy in there |
| 17:14.44 | StanManCan | It's also pay as you go, and with like $7 on my account that wouldn't last long |
| 17:14.45 | StanManCan | lol |
| 17:15.59 | *** join/#asterisk RobH (n=RobH@75.101.56.124) |
| 17:16.20 | StanManCan | Anyways... Nobody in here would of grabbed my number, spoofed it, and used it to spam people / |
| 17:16.43 | *** join/#asterisk CrazyTux (n=brandon@216-110-94-230.static.twtelecom.net) |
| 17:17.00 | pmhaddad | is there a way to reload asterisk gracefully from a bash cli? not the asterisk cli |
| 17:17.13 | pmhaddad | oh nvm |
| 17:17.17 | mosty | asterisk -rx '<put your CLI command here' |
| 17:17.29 | pmhaddad | mosty, yeah, i forgot about -x :P |
| 17:17.56 | kaldemar | but unless i need cluebat, there is no graceful reload. :) |
| 17:18.10 | pmhaddad | graceful restart |
| 17:20.34 | StanManCan | Is there a way to connect a call to another phone number ? |
| 17:21.00 | StanManCan | Somebody calls in and I want them to be connected to another number, but have it not run through my Voip Provider anymore |
| 17:23.51 | mosty | calls in from where? |
| 17:24.19 | StanManCan | anywhere |
| 17:24.59 | StanManCan | phonecall comes in from anybody, if they're from this area code then this happens, the rest of them get forwarded to my cell phone number |
| 17:25.16 | mosty | how does the call arrive at your asterisk machine? |
| 17:25.50 | StanManCan | they will call my old cell phone number which will forward them to one of my phone numbers with my voip provider |
| 17:27.35 | mosty | so, the call comes to you from your VOIP provider? |
| 17:27.58 | StanManCan | I'm confused at your question... |
| 17:28.27 | mosty | i'm trying to figure out the path that the call takes to your server, and where you want the call routed |
| 17:29.10 | StanManCan | I'll explain the situation if it helps better.... |
| 17:30.14 | mosty | please do |
| 17:30.53 | StanManCan | My cellphone is being spammed on and off by numbers in different area codes... I'm going to get a new cellphone number and get my Cell Provider to forward my old number to one of the phone numbers I have with my voip provider..... My asterisk box will then filter the calls based on the area codes... I'll write a dial plan to disconnect certain area codes and forward the rest of them to my NEW cellphone number |
| 17:32.15 | StanManCan | But, I want it to disconnect from my voip provider so that after the calls been forwarded, I'm not usin all my voip-providers minutes |
| 17:32.56 | mosty | if your voip provider supports sip with reinvites, then you might be able to do it |
| 17:33.23 | mosty | otherwise, you might want to look for a "hosted pbx" provider, and just move your dialplan to the voip provider's machine |
| 17:33.47 | *** join/#asterisk didz_ (n=dsad@189.24.56.203) |
| 17:33.48 | StanManCan | What would the difference be ? |
| 17:34.07 | StanManCan | I have no issues running it from home |
| 17:34.15 | didz_ | is it possible to use dynamic_features under meetme? |
| 17:34.19 | jksM | StanManCan, the difference is that you won't be using the minutes... |
| 17:34.41 | StanManCan | oh.. kk |
| 17:35.49 | mosty | StanManCan, just want to check, which outbound route do you want to use for these forwarded calls? |
| 17:40.11 | *** part/#asterisk fcois93 (n=Administ@bagnolet.acropolistelecom.net) |
| 17:49.24 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 17:49.50 | *** join/#asterisk jeffgus (n=jeffgus@green.zimage.com) |
| 17:51.55 | *** join/#asterisk cvnet (n=dahitler@24.156.136.205) |
| 17:55.36 | *** join/#asterisk areay (n=areay@93-97-161-123.zone5.bethere.co.uk) |
| 17:59.19 | areay | im having trouble receiving calls through asterisk... i've enabled sip debugging and i can see stuff happening when i dial the number from a regular phone (i get a busy signal from the phone), i just have no idea what any of it means... here's a pastebin: http://paste.ubuntu.com/129856/ |
| 18:00.52 | mosty | can you pastebin your dialplan (extensions.conf) and sip.conf with the usernames/passwords blanked out? |
| 18:01.36 | *** join/#asterisk bkw_ (n=brian@freeswitch/developer/bkw) |
| 18:01.48 | [TK]D-Fender | areay: Contact: <sip:s@192.168.1.73> <- you are not correctly configured to work behind NAT |
| 18:01.51 | [TK]D-Fender | ~sipnat |
| 18:01.52 | 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 |
| 18:02.11 | [TK]D-Fender | areay: and the inbound call is requesting auth and it doesn't look like they are liking that. |
| 18:02.50 | [TK]D-Fender | areay: Also looks like you used the GUI to create the peer its matching against as opposed to the "do it all yourself" approach you mentioned yesterday |
| 18:03.23 | areay | [TK]D-Fender, yeah i did... the new one this time tho |
| 18:03.59 | [TK]D-Fender | areay: Screw * GUI. ESPECiALLY that ancient version |
| 18:04.11 | areay | [TK]D-Fender, nah i'm using the CURRENT version now |
| 18:04.26 | [TK]D-Fender | areay: slightly better, but BLEH... |
| 18:04.33 | [TK]D-Fender | ~users.conf |
| 18:04.34 | jbot | users.conf is a flaming pile of sh1t that takes the fine control of several perfectly usable * config files and reduces them to the lowest common denominator and makes your system behave like a "toaster grade" PBX system. |
| 18:04.41 | bmoraca | FREEPBX FTW! lol |
| 18:04.49 | areay | lol |
| 18:04.58 | [TK]D-Fender | bmoraca: Currently... yes |
| 18:05.23 | areay | i wondered why it was using users.conf |
| 18:05.36 | [TK]D-Fender | areay: because thats what the GUI was built around. |
| 18:06.04 | areay | [TK]D-Fender, how can i make it use sip.conf and extensions.conf like normal? |
| 18:06.12 | [TK]D-Fender | areay: Stop using the GUI |
| 18:06.26 | areay | there really is no easy way to do this is there |
| 18:06.39 | [TK]D-Fender | areay: I told you yesterday but you wouldn't hear of it |
| 18:07.21 | [TK]D-Fender | areay: Either way you probably need to set the "insecure=port,invite" for your trunk definition nd fix your NAT settings either way |
| 18:07.28 | *** join/#asterisk chrismaster1 (n=chrismas@chello080109200180.3.sc-graz.chello.at) |
| 18:07.51 | areay | [TK]D-Fender, ok kool... then it should work (for the most part), yea? |
| 18:08.04 | [TK]D-Fender | areay: It can be made to work. |
| 18:08.21 | chrismaster1 | what is the best way to check if a peer is online? sip show peers and qualify=yes ? |
| 18:08.24 | [TK]D-Fender | areay: but you have to play more by its rules. and there may be snafu's to deal with |
| 18:08.28 | *** join/#asterisk CapriCoRN^80 (n=int@209.8.41.76) |
| 18:08.36 | [TK]D-Fender | chrismaster1: Place a call to it. |
| 18:08.53 | chrismaster1 | busy |
| 18:08.57 | CapriCoRN^80 | hi all |
| 18:08.59 | areay | [TK]D-Fender, ok i'll give it a go |
| 18:09.17 | CapriCoRN^80 | any body tell me about this error .. app_dial.c:1242 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown) |
| 18:09.38 | mosty | chrismaster1, depends how you define "online", and why you want to check that in the first place |
| 18:10.07 | chrismaster1 | i want to write a tool where everybody sees who is online and who he is calling |
| 18:10.17 | chrismaster1 | like asternic |
| 18:10.20 | chrismaster1 | but not flash |
| 18:11.04 | mosty | then just use qualify=yes and the asterisk manager interface |
| 18:11.17 | [TK]D-Fender | yup |
| 18:11.38 | [TK]D-Fender | CapriCoRN^80: what does SIP debug say? that message by itself means little |
| 18:11.55 | chrismaster1 | mosty: ok, thx |
| 18:12.05 | bmoraca | chrismaster1: don't reinvent the wheel. use either HUDlite or iSymphony |
| 18:12.27 | bmoraca | my preference is for iSymphony, but it's a bit pricy |
| 18:12.40 | bmoraca | cheaper than writing one yourself, though |
| 18:12.49 | mvanbaak | bmoraca: you think ? |
| 18:12.56 | bmoraca | most definitely |
| 18:13.11 | bmoraca | unless you plan on selling it yourself...but it's a crowded market |
| 18:13.16 | mvanbaak | I disagree |
| 18:13.19 | CapriCoRN^80 | [TK]D-Fender: the same i pasted above |
| 18:13.37 | mvanbaak | we put something together in our webbased CRM app. Took roughly 1 hour |
| 18:13.51 | chrismaster1 | oh, thx. i take a look at isymphony, didnt like hudlite |
| 18:13.51 | mvanbaak | 1 hour = 150 euro |
| 18:14.09 | bmoraca | mvanbaak: that's entirely different than a real-time event-driven application. |
| 18:14.27 | mvanbaak | 19:09 < chrismaster1> i want to write a tool where everybody sees who is online and who he is calling |
| 18:14.30 | bmoraca | yes, i could create a simple extension state hook in PHP or ColdFusion very easily. |
| 18:14.34 | mvanbaak | you can fix that in roughly an hour |
| 18:14.40 | [TK]D-Fender | CapriCoRN^80: that is not SIP DEBUG |
| 18:15.18 | cvnet | if i want to change the default port of sip i just add bindport=5555 in sip.conf correct? |
| 18:15.29 | CapriCoRN^80 | [TK]D-Fender: http://pastebin.com/m1fc4c029 |
| 18:15.31 | bmoraca | but a full application which will allow you to see, in real time, that information is much more complex. hence why i9 is able to charge so much for iSymphony and people will pay it. |
| 18:16.26 | bmoraca | i have it deployed at atleast 5 locations...one with 70 CALs. most just use it for the receptionist, though. |
| 18:16.29 | mosty | cvnet, yes- and then do a sip reload |
| 18:17.22 | [TK]D-Fender | CapriCoRN^80: "sip show peer sid" |
| 18:17.23 | chrismaster1 | bmoraca: isymphony looks good, is it stable? |
| 18:18.02 | bmoraca | chrismaster1: i've never had a problem with it. the new version's interface is much improved over previous versions. |
| 18:18.17 | chrismaster1 | bmoraca: eclipse like |
| 18:18.26 | bmoraca | chrismaster1: that's what it's built in. |
| 18:19.40 | chrismaster1 | bmoraca: so i would need a isymphonyServer + client 4 all clients |
| 18:19.44 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 18:20.01 | CapriCoRN^80 | [TK]D-Fender: i be right back. then i will discuss this issue |
| 18:20.11 | bmoraca | chrismaster1: no. one server and CALs for each client. so, if you had 10 clients, one server + 10 client licenses. |
| 18:20.20 | *** join/#asterisk jcoffi1 (n=jcoffi@75.147.155.89) |
| 18:20.54 | bmoraca | chrismaster1: there's a free version too that gives up to 5 concurrent connections |
| 18:21.15 | bmoraca | you don't get all the features, such as queues and chat, but everything else works |
| 18:21.21 | jaytee | iSymphony? CALs? sounds like a linux version of a Microsoft application |
| 18:21.55 | bmoraca | MS isn't the only company that charges per-user fees |
| 18:21.58 | chrismaster1 | bmoraca: full working? only 5 connection limitaiton? |
| 18:22.12 | bmoraca | chrismaster1: with a couple of features turned off, yes |
| 18:22.13 | jaytee | very true |
| 18:22.46 | chrismaster1 | bmoraca: thx |
| 18:22.47 | jaytee | lotta companies charge by number of "users", "seats" or "ports". |
| 18:23.16 | bmoraca | that's all this is...they aren't really CALs, i suppose...you pay for a number of "seats" |
| 18:23.25 | cvnet | I change the default port to 5555 in sip.conf (bindport=5555 ) under general, and now im trying to connect via Zoiper to the server, where shows domain i put IP:5555 and also under advance i changed the sip port to 5555, now mater what i try i can not register |
| 18:23.48 | cvnet | i did reset it as well |
| 18:24.33 | jasonwoot | Does iSymphony support have a pay-per-incident support arrangement? |
| 18:24.58 | cvnet | firewall is off |
| 18:25.00 | bmoraca | i've never had an incident, so I wouldn't know. as far as I'm aware, you can post on their forum or email them for free support |
| 18:25.37 | bmoraca | when i was evaluating their product, i spoke with them on the phone a couple times and they were very helpful |
| 18:26.55 | mosty | cvnet, verify the server is listening on the correct port with netstat, and verify that the client is using the correct port with a packet tracer |
| 18:27.26 | jasonwoot | isymphony is like this _ far away from being a really useful product |
| 18:28.30 | bmoraca | what's it missing? |
| 18:28.42 | jaytee | Bop bopa-a-lu a whop bam boo |
| 18:30.56 | jasonwoot | looks like defining "extension directories" is not shared between users, stays local, prefer it to be global |
| 18:31.24 | bmoraca | i believe you can define extension directories to be local...however, user-defined ones are per-user |
| 18:31.55 | jasonwoot | yeah, I could be missing in in the man page |
| 18:32.52 | cvnet | when i do netstat | more i dont c anything listnhing to port 5555 |
| 18:33.26 | jaytee | Tutti frutti, oh Rudy |
| 18:33.59 | [TK]D-Fender | cvnet: And you're showing us nothing. |
| 18:35.14 | cvnet | one min |
| 18:35.22 | jaytee | Got a girl named Sue, she knows just what to do |
| 18:35.50 | *** join/#asterisk yang (i=yang@CAcert/Assurer/pdpc.supporter.base.yang) |
| 18:37.11 | coppice | Sue Yu? |
| 18:37.17 | cvnet | http://pastebin.com/m59aea59a |
| 18:37.44 | jaytee | "Help me!!!!! I'm channeling Little Richard!!!!" |
| 18:38.12 | *** join/#asterisk jcoffi (n=jcoffi@208.87.0.146) |
| 18:38.32 | coppice | why would someone want to be known as little dick? |
| 18:38.34 | *** join/#asterisk The_Boy_Wonder (n=davidvos@asterisk/batman-developer/dvossel) |
| 18:38.40 | [TK]D-Fender | cvnet: Why am I only seeing TCP, and where are the configs? |
| 18:39.25 | cvnet | one min |
| 18:40.16 | cvnet | http://pastebin.com/m1769a3fa <-- sip.conf |
| 18:40.45 | [TK]D-Fender | waits... |
| 18:40.54 | Kobaz | [Mar 11 14:38:09] WARNING[20871]: pbx_config.c:2358 pbx_load_config: The use of '_.' for an extension is strongly discouraged and can have unexpected behavior. Please use '_X.' instead at line 0 |
| 18:41.00 | Kobaz | how would i turn off that warning |
| 18:41.06 | Kobaz | i really *do* want to use _. |
| 18:41.06 | Qwell | by not using _. |
| 18:41.28 | [TK]D-Fender | "Doctor, doctor, it hurts when I raise my arm like this" |
| 18:41.38 | [TK]D-Fender | <doctor> Then... awww FUKKIT |
| 18:42.01 | [TK]D-Fender | Kobaz: Go into the source and rip it out then. |
| 18:42.06 | Kobaz | well i think it would make sense to have an option to turn it off for uses where you really do want to match anything and everything |
| 18:42.13 | Kobaz | yeah i could... heh |
| 18:42.18 | cvnet | http://pastebin.com/m1769a3fa <-- sip.conf |
| 18:42.25 | mosty | Kobaz, it's just a warning, you can ignore it |
| 18:42.33 | Qwell | Kobaz: asterisk.conf, dontwarn=yes |
| 18:42.36 | Kobaz | yeap. i know... but i dont like seeing warnings |
| 18:42.43 | Kobaz | mmm |
| 18:42.46 | Qwell | That'll be $499.99 |
| 18:42.47 | [TK]D-Fender | cvnGo do a REAL netstat, your last lookup attempt was poort |
| 18:42.49 | Kobaz | but then that's all warnings |
| 18:43.03 | Qwell | Kobaz: it's 2 warnings |
| 18:43.04 | cvnet | waht u mean by real? |
| 18:43.06 | cvnet | i did netstat |
| 18:43.13 | [TK]D-Fender | cvnet: "netstat -an |
| 18:43.13 | Kobaz | ah... what's the other warning? |
| 18:43.20 | *** join/#asterisk JafoJ (i=40506c37@gateway/web/ajax/mibbit.com/x-f043b3748399edb6) |
| 18:43.25 | [TK]D-Fender | Kobaz: ANY OTHER WARNINGS |
| 18:43.32 | Qwell | Kobaz: core show translations |
| 18:43.43 | Kobaz | [TK]D-Fender: he said there's two |
| 18:43.51 | Qwell | yeah, it's just those 2. no idea why |
| 18:44.08 | Kobaz | what about translations? |
| 18:44.12 | Qwell | err, not translations |
| 18:44.15 | Qwell | core show codecs |
| 18:44.15 | cvnet | http://pastebin.com/m19a4d3a4 |
| 18:44.18 | Kobaz | i don't have a translations... i have ah |
| 18:44.39 | *** join/#asterisk jcoffi1 (n=jcoffi@75.147.155.89) |
| 18:44.41 | Kobaz | i have core show transation... but anyway |
| 18:44.44 | Kobaz | what about codecs? |
| 18:44.49 | cvnet | udp 0 0 0.0.0.0:5555 0.0.0.0:* |
| 18:45.03 | Qwell | it lists all codecs Asterisk knows about. It doesn't matter whether Asterisk can use them or not - it still shows them in that list. |
| 18:45.04 | bmoraca | i had a teacher in junior high school whose name was Richard Haire...you can guess what we called him... |
| 18:45.07 | Kobaz | er. translation |
| 18:45.14 | Qwell | so it gives a warning saying "blah blah informational purposes blah blah" |
| 18:45.20 | Kobaz | Qwell: yeah... oh |
| 18:45.43 | Kobaz | i don't see a warning... i see disclaimer |
| 18:45.48 | Kobaz | 1.4.22 |
| 18:46.04 | Qwell | same thing |
| 18:46.06 | *** join/#asterisk Titanous (n=titanous@unaffiliated/titanous) |
| 18:46.14 | Kobaz | ah |
| 18:46.49 | *** join/#asterisk tokozedg (n=rock@89.232.24.53) |
| 18:47.10 | jaytee | Got a girl named Daisy, she almost drives me crazy |
| 18:47.20 | *** join/#asterisk jetlagmk2 (i=jetlag@pool-70-17-58-83.pskn.east.verizon.net) |
| 18:47.31 | Titanous | I just upgraded to 1.4.24-rc1, and I'm getting http://pastebin.com/d546da2fc (a problem with astrealtime) |
| 18:48.27 | cvnet | [TK]D-Fender: did you find any issue? |
| 18:49.14 | [TK]D-Fender | cvnet: sure looks like it is listening |
| 18:49.31 | tokozedg | hi, when i was trying to have two sip gateway for asterisk, there was problem registering second sip number, do i have to set different source port for both sip gateway? and if so how |
| 18:51.56 | Kobaz | hmmmm |
| 18:52.06 | Kobaz | anyone know why a call would get randomly dropped when it's on hold |
| 18:53.27 | Kobaz | http://pastebin.com/m20ca309b |
| 18:54.02 | cvnet | but i cant connect |
| 18:54.32 | *** join/#asterisk suma (n=chinnapa@cpe-76-168-177-23.socal.res.rr.com) |
| 18:55.18 | *** part/#asterisk harry__ (n=h@imperialglamour.com) |
| 18:55.29 | Titanous | I'm gwtting `Unknown column 'lastms' in 'field list'` with astrealtime on 1.4.24-rc1 |
| 18:55.35 | suma | when I originate calls to two persons from asterisk, I want to disconnect from either of the one and connect to a different destination, how is that possible ? |
| 18:57.14 | mosty | huh? |
| 18:57.40 | [TK]D-Fender | cvnet: Check your firewalls & routing, and whree is your softphone relative to *? |
| 18:58.02 | suma | A & B are talking through asterisk ZAP, I want to disconnect either A or B and connect to C |
| 18:58.11 | cvnet | ya it was the firewall |
| 18:58.15 | cvnet | thanks a bunch |
| 18:58.20 | tokozedg | Titanous, create column lastms |
| 18:58.23 | Kobaz | actually the music on hold drops the caller after exactly 28 seconds |
| 18:58.37 | Titanous | tokozedg: what column type? |
| 19:00.02 | tokozedg | Titanous, do you write calls in mysql? |
| 19:00.32 | tokozedg | actually source and dst number ... |
| 19:00.50 | Kobaz | could maybe the music on hold file is corrupted/ |
| 19:00.54 | Kobaz | would that cause a call to dro |
| 19:00.55 | Titanous | tokozedg: I use MySQL for CDR, realtime peers (sip/iax), and voicemail |
| 19:01.32 | tokozedg | so connect mysql and use that database and table, and create a lastms column varchar(255) |
| 19:03.12 | *** join/#asterisk ingenius (n=alektro@69.90.72.173) |
| 19:03.57 | BlargMaN00 | does anyone in here have any experience with using asterisk as Voicemail for Cisco CallManager?? |
| 19:04.39 | Kobaz | callmangler |
| 19:04.47 | BlargMaN00 | yeah, pretty much |
| 19:09.32 | [TK]D-Fender | BlargMaN00: What does "voicemail' have to do with it? |
| 19:10.12 | Kobaz | so anyone have any idea about moh dropping calls? |
| 19:11.19 | cvnet | i have someone from some Arabic country trying to register to my box, i know they have blocked the 5060 but now i changed it to 5555 and also 80 but she still cannot register, is it possible for a isp to block UDP all together? |
| 19:11.39 | mosty | BlargMaN00: no, but the asterisk config would be pretty simple |
| 19:11.54 | apeiron | If an ISP blocked UDP, DNS would be pretty screwed. |
| 19:12.12 | mosty | cvnet, run a packet logger and see what packets are coming from the client's ip address |
| 19:12.53 | Kobaz | well, they might block everything but 53 |
| 19:13.24 | cvnet | mosty not sure what you mean |
| 19:13.37 | *** join/#asterisk aaroneous (n=aaron@cpe-98-14-146-167.nyc.res.rr.com) |
| 19:13.43 | apeiron | wonders if he will ever stop reading mosty as 'moisty' or 'mostly' |
| 19:13.56 | [TK]D-Fender | cvnet: And do you, or do you not get packets? |
| 19:14.15 | [TK]D-Fender | cntBecause its getting annoying seeing "doen't work" with no useful description. |
| 19:14.24 | mosty | cvnet, run something like wireshark (or the command line version, tshark) |
| 19:14.25 | cvnet | from that clients? she can not register, but i dont know if i get pockets or not |
| 19:14.31 | aaroneous | does anyone know which ISDN protocol I want my PRI configured for? |
| 19:14.40 | Kobaz | aaroneous: ask your telco |
| 19:14.50 | aaroneous | Kobaz: they're asking me |
| 19:14.57 | Kobaz | NI2 |
| 19:14.58 | mosty | cvnet, a packet logger can tell you exactly which packets come to/from a particular ip address |
| 19:15.17 | cvnet | mostly: what command do i use for that? |
| 19:15.32 | [TK]D-Fender | cvnet: Don't know if you're getting the packets? that is a lame thing to hear. Open your damn eyes :p |
| 19:15.40 | *** part/#asterisk Titanous (n=titanous@unaffiliated/titanous) |
| 19:15.47 | Kobaz | cvnet: join #your_favorite_operating_system_channel and ask about packet logging |
| 19:16.00 | tokozedg | cvnet, use wireshark |
| 19:16.10 | aaroneous | Kobaz: thanks.. I am going to be using this with a digium card initially and then with a cisco 3640 configured as a SIP<->PRI gateway.. should be cool right? I only ask because google showed various people having trouble with NI2 |
| 19:16.16 | BlargMaN00 | [TK]D-Fender: voicemail has to do with the fact that I am out of Unity licenses, and don't want to spend the money for more, so instead, I am putting all new users on * VM... |
| 19:16.20 | Kobaz | ni2 works fine |
| 19:16.33 | aaroneous | great.. tnx |
| 19:17.00 | bougyman | aaroneous: we've had no trouble with NI2 on * |
| 19:17.06 | bougyman | have dozens of PRIs up with it. |
| 19:17.11 | BlargMaN00 | [TK]D-Fender: i can't go 100% asterisk until the IMAP Storage feature is working correctly, because all my users are spoiled with the whole unity-manage your voicemail in your e-mail box concept... |
| 19:18.01 | bougyman | so when they delete the email it deletes it from the voicemail system? |
| 19:18.08 | BlargMaN00 | yes |
| 19:18.18 | aaroneous | bougyman: cool.. just wanted to make sure |
| 19:18.27 | bougyman | i have a hook for that in mutt, BlargMaN00 |
| 19:18.33 | bougyman | i guess that doesn't help your users :) |
| 19:18.50 | [TK]D-Fender | BlargMaN00: Now what does that have to do with getting CM to talk with *? |
| 19:18.52 | BlargMaN00 | not really... |
| 19:19.00 | Qwell | BlargMaN00: Asterisk can do IMAP voicemail.. |
| 19:19.19 | apeiron | And the * book covers setting it up. |
| 19:19.22 | apeiron | ~book |
| 19:19.23 | jbot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 19:19.23 | areay | [TK]D-Fender, i've got it working and u were right i had to forward the ports to the asterisk server... i've invested in another static ip for it too... im setting up voice menus (yes in the gui again, sorry) and i can't get it to "listen for keypress" on a particular step. it just defaults to donot listen for keypress |
| 19:19.38 | *** join/#asterisk ScarEye (n=scareye@12.27.87.66) |
| 19:19.39 | BlargMaN00 | because all my users are on CM, and I am using IMAP voicemail on *, but it doesn't work 100% yet because i'm using it with exchange 2003 |
| 19:20.09 | bougyman | exchange is not standard imap, is it? |
| 19:20.09 | BlargMaN00 | the only exten that is registered to my * box is my personal exten |
| 19:20.31 | [TK]D-Fender | BlargMaN00: You asked specifically about * & CM. Your problem is exclusively * VM. |
| 19:20.33 | BlargMaN00 | nope, but i did find a way to make it work with a single user and password though... 8)~ |
| 19:20.38 | aaroneous | there shouldn't be any problems setting outbound CPN to anything we want on this PRI right? (or do I need something special other than "Calling Party Number" from the telco?) |
| 19:21.01 | *** join/#asterisk ccitt (n=ccitt@c-98-217-97-45.hsd1.ma.comcast.net) |
| 19:21.04 | [TK]D-Fender | aaroneous: they frown on you setting to "911", etc |
| 19:21.05 | ccitt | hey guys |
| 19:21.16 | Qwell | putnopvut: app_voicemail doesn't work with Exchange? |
| 19:21.26 | aaroneous | [TK]D-Fender: okay but setting it, say, to my cell phone number wont be an issue right? |
| 19:21.29 | [TK]D-Fender | aaroneous: Your telco may also restrict you to DID's that you obtain through them, etc depending |
| 19:21.37 | ccitt | i just have a quick, dumb question about the new asterisk setup i'll have shortly: |
| 19:21.40 | aaroneous | ah.. that's the restriction I am trying to avoid.. |
| 19:21.45 | [TK]D-Fender | aaroneous: or they may ban your setting the number at all |
| 19:21.46 | BlargMaN00 | no, not actually... my problem actually has to do with call routing and 201 forbidden sip notifications... |
| 19:21.50 | putnopvut | Qwell: you mean in IMAP mode? That may be, but I'm almost certain that I've seen reports to the contrary. |
| 19:21.51 | aaroneous | wondering if I had to order it in a special way to avoid that restriction |
| 19:21.59 | [TK]D-Fender | BlargMaN00: then thats another matter. |
| 19:22.04 | Qwell | putnopvut: that's what I thought.. |
| 19:22.20 | bougyman | Qwell: google says it's problematic. |
| 19:22.35 | ccitt | i wil have an asterisk box with a number of sip lines, and be using softphones on workstations as the phones - do i need any sort of channel banks or anything, or can i just do it all with straight ethernet/ip? |
| 19:22.53 | [TK]D-Fender | ccitt: ccOf course not. |
| 19:22.58 | BlargMaN00 | but it deals with voicemail at the same time, because it only happens when voicemail is involved... |
| 19:23.04 | ccitt | of course i wont need the channel banks? |
| 19:23.32 | [TK]D-Fender | ccitt: you said you're runing pure SIP. CB's are to connect analog lines to a T1 card. You said you won't ahve any of that. |
| 19:23.33 | ccitt | or of course i cant use straight ethernet? |
| 19:23.40 | ccitt | awesome |
| 19:23.58 | [TK]D-Fender | ccitt: "Hi I want to listen to MP3's on my computer, do I need a lawnmower?" |
| 19:24.37 | [TK]D-Fender | ccitt: That equipement is for connecting stuff you're telling us you have no intention of using. Answers itself |
| 19:24.39 | BlargMaN00 | the issue is i get 201 Forbidden when this scenario happens: * exten -> CM Exten -> No Answer -> * VM... I think it has something to do with the Caller ID and the way it is being routed... |
| 19:24.40 | ccitt | lmao - i figured id look dumb but oh well - id rather know for sure - secondly, does anyone have the "inside" knowledge of any voip/sip carriers that offer unlimited calling at decent rates, possibly with bulk deals on lines? |
| 19:24.49 | Kobaz | [TK]D-Fender: what are you talking about... of course you need a lawnmower... i have my lawnmower hooked up via usb |
| 19:25.20 | ccitt | for pc-to-phone obviously |
| 19:25.40 | coppice | "Lucent in the carrier space is WAY ahead in the hopelessly filled with bugs category." :-) |
| 19:25.52 | ccitt | hahaha |
| 19:27.10 | BlargMaN00 | putnopvut: BTW... My server is not stressed yet, but I have IMAP VM using Exchange, and it seems to be holding up fairly well, at least on the IMAP side now... I still see some issues every once in a while, but they seem to be slowly going away with each new release... |
| 19:27.44 | putnopvut | BlargMaN00: well, that's a good thing, I suppose. I've never been very happy with Asterisk's IMAP implementation. |
| 19:27.46 | BlargMaN00 | once it works like it's supposed to, then I will be using exclusively asterisk for VM>.. |
| 19:28.19 | *** join/#asterisk dwayne (n=dwayne@c-76-29-245-9.hsd1.al.comcast.net) |
| 19:29.05 | BlargMaN00 | putnopvut: I wouldn't necessarily say that I really like it, but i'm pleased with the fact that it accomplishes what I'm trying to do... I still have issues where it likes to die when making some calls the the app_voicemail and c-client, but like I said, the server isn't stressed ... yet.... |
| 19:29.39 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 19:31.14 | BlargMaN00 | I guess that's one of those things where you really just want to scrap it and start over, but every time you think of doing that, you think to yourself "screw that" |
| 19:34.38 | *** join/#asterisk UQlev (n=kvirc@91.184.221.31) |
| 19:35.11 | putnopvut | BlargMaN00: you don't know how many times I've said that... |
| 19:36.35 | BlargMaN00 | putnopvut: I can only imagine.... |
| 19:37.33 | BlargMaN00 | putnopvut: I say it to myself from time to time... |
| 19:38.56 | putnopvut | BlargMaN00: one thing that may change things is that we're planning on participating in the Google Summer of Code project. One of the projects we have proposed is to re-write the way that storage is accomplished in app_voicemail. If this gets done, then it would help to isolate the IMAP stuff away from the rest of the code and hopefully make it a bit easier to rewrite that stuff. |
| 19:39.36 | mmlj4 | I've got a problem with a sangoma card: http://joeykelly.net/hacks/linux/wanrouter-problem.txt |
| 19:39.51 | *** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose) |
| 19:40.00 | CapriCoRN^80 | [TK]D-Fender: http://pastebin.com/d7415442b |
| 19:40.16 | BlargMaN00 | putnopvut: yeah, it wouldn't hurt to have an app_voicemail, and an app_imap_storage... ya know... |
| 19:40.21 | *** join/#asterisk Deeewayne (n=dwayne@c-76-29-245-9.hsd1.al.comcast.net) |
| 19:40.21 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 19:40.38 | putnopvut | BlargMaN00: that's pretty much the idea behind the project. |
| 19:41.54 | BlargMaN00 | i remember when I first started writing code, I wanted to put everything into one file as messy as possible, so that way only I could debug it... "job security" right?? nope... when you can't debug your own code, you don't get to keep your job long... lol |
| 19:42.57 | *** join/#asterisk lanning (n=lanning@173.8.187.197) |
| 19:43.18 | putnopvut | BlargMaN00: yeah, what sucks is that I inherited that code. I'd be embarrassed to say that I wrote it myself. |
| 19:44.19 | BlargMaN00 | putnopvut: lol... yeah, that does suck... but like you said, the good thing is, you can say you didn't write it to begin with... |
| 19:44.37 | KavanS | [TK]D-Fender: what would you suggest for a least cost dial solution....I have 3 providers....one charges flat fee long distance (except foreign countries), the other charges flat per minute usage, and other has that standard csv rates file |
| 19:45.40 | UQlev | KavanS: it is wise to work with 3-4 at once |
| 19:46.00 | UQlev | KavanS: 1 of all will be down from time to time |
| 19:46.20 | KavanS | UQlev: hrm, okay...I expect to run into the same situation |
| 19:46.25 | CapriCoRN^80 | strange |
| 19:46.31 | KavanS | UQlev: what application/macro/script do you suggest for such? |
| 19:46.31 | UQlev | KavanS: 2 will not be able trace your payments |
| 19:47.04 | UQlev | KavanS: I switche them manually |
| 19:47.13 | KavanS | jesus... |
| 19:47.30 | KavanS | well I want something a little more automatic...choosing the route that's up, as well as the cheapest one |
| 19:47.34 | UQlev | KavanS: keep them all registered but switch your customers |
| 19:47.43 | KavanS | but I understand that I may not get every feature I desire |
| 19:47.54 | KavanS | naw this is for small office... |
| 19:48.08 | KavanS | but I don't want to deal with failure (like everyone in this world right?) |
| 19:48.17 | UQlev | KavanS: it will work perfect for a small office |
| 19:48.46 | CapriCoRN^80 | why my status is unreacable |
| 19:48.49 | UQlev | KavanS: how will you avoid failures with VoIP providers |
| 19:48.50 | CapriCoRN^80 | hmm |
| 19:48.56 | KavanS | UQlev, lol but it will require me doing shit... |
| 19:48.56 | CapriCoRN^80 | be right back |
| 19:49.06 | KavanS | UQlev, going to set it up so it cycles through during dial |
| 19:49.29 | KavanS | UQlev, just like my current PSTN/SIP configuration |
| 19:50.02 | UQlev | KavanS: doubtfully it ill work |
| 19:50.16 | KavanS | UQlev, what experience(s) lead you to believe this? |
| 19:50.51 | UQlev | every call you should wait for 35-40 sec |
| 19:51.00 | BlargMaN00 | putnopvut: well, when you are actually ready to fix it the way it should be, let me know... I will do extensive testing, and even help you debug... |
| 19:51.13 | putnopvut | BlargMaN00: thanks! |
| 19:51.52 | BlargMaN00 | putnopvut: my coding isn't up to the caliber required to write something like that, but I can sure debug code, and test... especially seeing is how that is the only standing in my way of moving from an CM shop to a 100% * shop... |
| 19:52.17 | KavanS | UQlev: most calls are made via the PSTN line...and that's quite reliable |
| 19:52.33 | KavanS | UQlev: I only want to push certain area/country codes via the SIP |
| 19:52.42 | KavanS | currently only SIP is used when PSTN is already in-use |
| 19:52.55 | putnopvut | BlargMaN00: Well, like I said, the Summer of Code could be the opportunity to get it right. |
| 19:53.32 | UQlev | KavanS: look for providers which allow several simultaneous calls at once |
| 19:53.52 | BlargMaN00 | putnopvut: let's hope so... I'm looking forward to it, because I hate CM with a passion... 8)~ |
| 19:54.07 | KavanS | UQlev: we are not even discussing the same subject |
| 19:54.13 | *** join/#asterisk jchonig (n=jch@firewall.honig.net) |
| 19:54.17 | KavanS | lol |
| 19:54.20 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 19:55.02 | jchonig | Any of you fine folks know how to disable MOH on a given channel? SetMusicOnHold(none) where none of type files and directory of /dev/null does not work because /dev/null is not a directory? |
| 19:55.11 | UQlev | KavanS: do we? |
| 19:55.26 | jblack | jchonig: How about giving an empty directory. :) |
| 19:56.02 | jchonig | jblack I can do that, but it seems like a bit of a hack to me. Or an omission |
| 19:56.20 | jblack | So call the directory "nomoh" |
| 19:56.26 | jchonig | Still. |
| 19:57.11 | *** join/#asterisk ReDNeQ (n=ReDNeQ@70.114.226.70) |
| 19:57.20 | jblack | Then you're stuck. |
| 19:57.40 | jchonig | Oh, I'll do it, I just won't like it. ;-) |
| 19:57.52 | jchonig | Got slammed on a con-call, didn't realize * defaulted to MOH |
| 19:58.02 | jchonig | Got a call from school and put them on hold... |
| 19:58.38 | KavanS | jchonig: I sent a salesmen to the "blackhole" today :) it's nelson from the simpsons saying "ha, ha" then a dialtone....disconnects then blacklists them |
| 19:58.38 | jblack | so now you know what '[default]' means. :) |
| 19:58.45 | Chainsaw | KavanS: :D |
| 19:59.21 | jchonig | Just upgraded to 1.4 this weekend, fun stuff like telemarketer torture scripts are lower on my todo list. ;-) |
| 19:59.32 | KavanS | lol |
| 20:00.01 | jblack | you should play some sort of audio while people are on hold, so they know the phone is still off hook. |
| 20:00.52 | jchonig | Well, that doesn't work when you are on a con-call, if one person puts the call on hold, everyone gets to listen to music. They get mad at you |
| 20:01.09 | jblack | gentle white noise is an option. perhaps rip a $3.00 nature sounds cd from target, with a low volume. |
| 20:01.11 | *** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose) |
| 20:01.35 | jchonig | Right now I'm going to disable it on my work line (I'm in a home office), but in the future I may figure out an option to disable it on conference calls |
| 20:01.40 | jblack | Maybe you souldn't put con men on hold. :) |
| 20:01.54 | jchonig | Well, it was a call with marketing.... |
| 20:02.08 | *** join/#asterisk docid (n=eris@69.196.68.142) |
| 20:02.22 | docid | uggg, so hard to stay connected |
| 20:06.17 | *** join/#asterisk flujan (n=flujan@189-039-010-068.static.spo.ctbc.com.br) |
| 20:07.21 | flujan | hello guys, I need to use a asterisk function on my dialplan. |
| 20:07.26 | flujan | The function is new and called AUDIOHOOK_INHERIT |
| 20:07.32 | flujan | how can i set it up on my dialplan? |
| 20:08.51 | *** join/#asterisk horvath (n=horvath@74-51-45-109.telnetcommunications.com) |
| 20:10.11 | horvath | Anyone know where I can find a PDF or powerpoint file of Virtualizing Asterisk - Presented at Digium Asterisk World? The one on scribd wants me to register to download ugh. |
| 20:11.16 | CapriCoRN^80 | [TK]D-Fender: you there ? |
| 20:12.27 | *** join/#asterisk joshaidan (n=joshaida@S01060090f8009fa6.tb.shawcable.net) |
| 20:13.05 | joshaidan | Hey does anyone else get brute force sip REGISTER attacks? |
| 20:13.48 | *** join/#asterisk orkid_ (n=orkid@unaffiliated/orkid) |
| 20:14.15 | mmlj4 | joshaidan: what's your IP? |
| 20:14.39 | joshaidan | mmlj4: why? |
| 20:14.45 | mmlj4 | i'm kiddinfg |
| 20:14.48 | horvath | joelbryan: Thankfully no just your standard ssh brute force attacks |
| 20:14.59 | mmlj4 | BOFH |
| 20:14.59 | orkid_ | Chainsaw, switching PCI slots seemed to have helped. I added another network card though. I notice a problem after a while again (beeps), and move net cards so that they weren't on the same interrupt. So now every device is on its own interrupt, but the beeps are still there :S |
| 20:15.00 | joshaidan | mmlj4: I thought u were, just checking. :) |
| 20:15.10 | NoxIn- | joshaidan: don't see bruteforce attack on my asterisks either |
| 20:15.13 | *** join/#asterisk ghenry (n=ghenry@ghenry.plus.com) |
| 20:15.26 | joshaidan | We've been getting them now and them, seems to be more frequent |
| 20:15.33 | horvath | joshaidan: Setup OSSEC-HIDS to ban ips for 15 min after x number of bad passwords? |
| 20:15.54 | joshaidan | The first attacks were successful, but since then we've tighten things up |
| 20:16.03 | ghenry | Is this like followme with presence? http://www.agile.co.nz/Mobility/Mobility-for-the-SMB/IP-Office-Mobile-Twinning/MenuId/98.aspx |
| 20:16.13 | Chainsaw | orkid: Hm, okay. Just in case there is a firmware bug causing slow interrupt delivery... that BIOS upgrade still sounds like a good idea. |
| 20:16.17 | ghenry | Anyoen have ISDN BRI details for Dubai? |
| 20:16.18 | joshaidan | horvath: thanks, I was going to ask what people do to block them |
| 20:16.25 | ghenry | I'm setting up msidn.conf here |
| 20:16.26 | riddlebox | http://www.agile.co.nz/Mobility/Mobility-for-the-SMB/IP-Office-Mobile-Twinning/MenuId/98.aspx |
| 20:16.30 | riddlebox | oops sorry |
| 20:16.31 | ghenry | will check voip-info.org |
| 20:16.51 | horvath | joshaidan: http://www.ossec.net you will have to make a custom rule but yea ossec is awesome |
| 20:16.56 | ghenry | riddlebox: that's my link ;-) |
| 20:17.00 | orkid_ | Chainsaw, yes ok. Is there anything I should check in BIOS too? Any Latency/ACPI/etc settings that might affect the situation? |
| 20:17.12 | flujan | ping putnopvut |
| 20:17.20 | NoxIn- | joshaidan: are you attacker from a unique IP or multiples ? |
| 20:17.21 | riddlebox | ghenry: yeah went to copy it to check it out cause IP-Office is avaya and thats primarily what we sell |
| 20:17.27 | NoxIn- | attacked* |
| 20:17.27 | putnopvut | flujan: I've got about 5 minutes before I need to go to a meeting... |
| 20:17.30 | Chainsaw | orkid_: I'm assuming you have a linux OS? |
| 20:17.35 | joshaidan | We get them from multiple IPs |
| 20:17.40 | flujan | putnopvut: ok it will be enough |
| 20:17.41 | orkid_ | Chainsaw, yes Hardy 8.04 |
| 20:17.43 | orkid_ | Chainsaw, updated |
| 20:17.49 | flujan | putnopvut: I am trying to use AUDIOHOOK_INHERIT |
| 20:17.54 | joshaidan | They usual exploit us for LD and send telemarketer calls/scams |
| 20:17.56 | putnopvut | flujan: is it working for you? |
| 20:17.57 | Chainsaw | orkid_: Indeed, please make sure you have full ACPI support enabled. |
| 20:17.57 | ghenry | riddlebox: k. It's an old feature which I guess is just done on * via a dialplan I'd write |
| 20:18.03 | riddlebox | ghenry: what was your question? |
| 20:18.10 | Chainsaw | orkid_: If you have an OS selector, it should be on Win2K/XP. |
| 20:18.13 | joshaidan | The worst was one that sent calls to Cuba |
| 20:18.17 | Chainsaw | orkid_: (In the BIOS) |
| 20:18.24 | ghenry | riddlebox: if it's just followme with presence for BLF etc. |
| 20:18.24 | orkid_ | Chainsaw, ok. Also PnP OS should be 'yes' correct? |
| 20:18.32 | horvath | joshaidan: Interesting I wonder if theres some new botnets out in the wild that are just focusing on brute force SIP attcks |
| 20:18.42 | Chainsaw | orkid_: Indeed, that is the correct setting. Any APIC support should be enabled, as should MPS1.4 |
| 20:18.47 | joshaidan | horvath: I wouldn't be surprised |
| 20:18.50 | CapriCoRN^80 | i am very much confused |
| 20:18.55 | Chainsaw | orkid_: (If you have an MP table selector, it should be on 1.4, not 1.1) |
| 20:19.02 | riddlebox | ghenry: yeah the twinning, actually is a bit more than follow me, it also lets you pass the call from the cell phone back to the desk phone |
| 20:19.05 | flujan | putnopvut: no where can i put it on the dialplan? here is a pastie http://pastebin.com/m73323bd |
| 20:19.12 | CapriCoRN^80 | my usera connectioned to * but both user's status are offline |
| 20:19.14 | NoxIn- | joshaidan: care to a few of the attackers IP so I can see if I have some of them in my logs ? |
| 20:19.26 | orkid_ | Chainsaw, hmm, should I take a 'beta' bios? (1012.004) released 3.5 years ago, about a month after the latest 'stable' bios |
| 20:19.27 | jchonig | How do I print the musiconhold classes from the console? |
| 20:19.27 | CapriCoRN^80 | my users are connected to * but both user's status are offline |
| 20:19.39 | ghenry | riddlebox: ok, so a transfer feature too. So a custom dialplan would be needed |
| 20:19.43 | putnopvut | flujan: The place where you have it will work. The problem is that you aren't using the syntax correctly. |
| 20:19.46 | Chainsaw | orkid_: Can I see the Changelog for it please? |
| 20:19.47 | riddlebox | ghenry: yeah |
| 20:19.48 | putnopvut | flujan: what you need to to is: |
| 20:19.52 | joshaidan | NoxIn: sure, the most recent is 68.143.220.226 |
| 20:20.26 | orkid_ | ---------------------- |
| 20:20.26 | orkid_ | Latest beta BIOS. |
| 20:20.26 | orkid_ | [ 1012 ] |
| 20:20.26 | orkid_ | ---------------------- |
| 20:20.26 | orkid_ | Support new CPUs. Please refer to our website at: http://support.asus.com.tw/cpusupport/cpusupport.aspx |
| 20:20.29 | putnopvut | exten => _XXXX,1,Set(AUDIOHOOK_INHERIT(mixmonitor)=yes |
| 20:20.30 | riddlebox | ghenry: you could maybe set it almost as a conf call between the phone and the cell so you could pass it back and forth |
| 20:20.34 | orkid_ | That's not much I know |
| 20:20.38 | putnopvut | exten => _XXXX,1,Set(AUDIOHOOK_INHERIT(mixmonitor)=yes) |
| 20:20.51 | joshaidan | horvath: do you have links to tutorials/info on setting this up to catch those SIP attacks? |
| 20:20.51 | putnopvut | I left off the final closing paren the first time. |
| 20:20.51 | orkid_ | Chainsaw, and nothing in the zip file |
| 20:20.52 | flujan | putnopvut: thanks i will give it a try |
| 20:20.59 | flujan | putnopvut: have a nice meeting bro. |
| 20:21.07 | putnopvut | flujan: it won't be a nice meeting :( |
| 20:21.15 | ghenry | riddlebox: true, that makes sense |
| 20:21.16 | putnopvut | Well, it won't be bad either. It won't be fun though. |
| 20:21.16 | Chainsaw | orkid: New CPU support does not seem harmful. In doing so they may have fixed up embarassing ACPI bugs that they did not feel like publically admitting. |
| 20:21.17 | orkid_ | Chainsaw, the 'beta' is .11 KB bigger. |
| 20:21.20 | flujan | putnopvut: there is no nice meeting lol |
| 20:21.23 | ghenry | riddlebox: seen anyone do it already? |
| 20:21.29 | NoxIn- | well, don't appear on my grep |
| 20:21.36 | orkid_ | I've been running the beta on another comp no problem, maybe i'll try that (if it's not already). anyway, no more rambling from me for now |
| 20:21.37 | orkid_ | bbl |
| 20:22.45 | riddlebox | ghenry: nope, just thought of it since you were wanting to implement the feature |
| 20:22.52 | ghenry | k |
| 20:22.56 | ghenry | thanks riddlebox |
| 20:23.13 | horvath | joshaidan: I don't think theres a tutorial out there but lemme see if I can find the guide about creating new rules |
| 20:23.24 | riddlebox | ghenry: haha they are advertising the oneX its on its way out |
| 20:23.28 | jchonig | Ugh, not only do I need an empty dir, I need an emtpy file in that dir, or a WAV of silence... |
| 20:23.39 | BlargMaN00 | putnopvut: quit |
| 20:23.47 | BlargMaN00 | oops |
| 20:24.09 | ghenry | riddlebox: means nothing to me. |
| 20:24.37 | NoxIn- | joshaidan: you could also use fail2ban |
| 20:24.56 | *** join/#asterisk magronez (n=eusei@unaffiliated/magrao/x-2903) |
| 20:25.04 | joshaidan | NoxIn: thanks |
| 20:26.25 | *** join/#asterisk umpc (n=Justin@unaffiliated/umpc) |
| 20:27.02 | KavanS | who can point me in the direction of a solid least call routing plan? |
| 20:27.10 | KavanS | i.e. lcdial/lcdc/lcdial.sh |
| 20:27.11 | mmlj4 | I've got a problem with a sangoma card, anyone?: http://joeykelly.net/hacks/linux/wanrouter-problem.txt |
| 20:27.19 | docid | call = cost? |
| 20:27.38 | docid | tried sangoma customer service? ive had great results with um |
| 20:27.48 | KavanS | docid: yep lol |
| 20:27.53 | KavanS | least *cost* routing! :) |
| 20:28.05 | horvath | joshaidan: I must be blind but I cant seem to find the page on their wiki |
| 20:28.39 | horvath | joshaidan: fail2ban should work as well but ossec does other things like rootkit checks etc its really quite nice |
| 20:29.10 | mmlj4 | docid: how many questions can you ask before they start charging? |
| 20:29.12 | NoxIn- | mmlj4: when you compile wanrouter you specifie the protocols to compile in |
| 20:29.17 | joshaidan | horvath: cool, I'm going to take a look at both |
| 20:29.21 | mmlj4 | NoxIn-: aye |
| 20:29.27 | *** part/#asterisk jchonig (n=jch@firewall.honig.net) |
| 20:30.25 | NoxIn- | mmlj4: for instance on my case I used the option --protocol=TDM-AFT_TE1 |
| 20:30.25 | mmlj4 | hrm, I can try that |
| 20:30.25 | mmlj4 | ./Setup install --something ? |
| 20:30.25 | NoxIn- | I created a .deb so the complete line I used was ./Setup builddeb --protocol=TDM-AFT_TE1 |
| 20:30.26 | NoxIn- | but should be the same with install |
| 20:30.51 | docid | ummm, dunno, im up to 8 calls and 5 times ssh'd to my box and there was no talk of chargeing |
| 20:31.09 | *** join/#asterisk DavidR2008 (n=chatzill@fw1.safedataisp.net) |
| 20:31.44 | docid | mostly stuff that was broke etc.... just fighting to get some real info from the telco atm |
| 20:32.05 | DavidR2008 | how do you (can you) reload the features.conf without shutting down? |
| 20:33.51 | docid | <PROTECTED> |
| 20:36.16 | Gido-E | DavidR2008 do you know it for voicemail? |
| 20:36.41 | DavidR2008 | Gido-E I don't understand your question |
| 20:37.16 | Gido-E | i needed to know it for voicemail. maybe you would know. |
| 20:37.25 | *** join/#asterisk tobias (n=tobias@cpe-069-134-127-101.nc.res.rr.com) |
| 20:37.47 | DavidR2008 | oh, I understand now, I don't know. I ended up restarting my * server |
| 20:37.58 | *** join/#asterisk Juggie (n=Juggie@CPE001601df17fb-CM001ceac25ada.cpe.net.cable.rogers.com) |
| 20:38.25 | Gido-E | DavidR2008 i think that is the only option. for feature config. |
| 20:38.37 | DavidR2008 | thx |
| 20:38.57 | Gido-E | but i was yust wondering what the minimum is for voicemail.conf changes. |
| 20:39.00 | Juggie | does anyone have a list of north american areacodes in csv? |
| 20:39.59 | DavidR2008 | Juggie: I was able to google some |
| 20:40.04 | kaldemar | DavidR2008: features reload |
| 20:40.32 | DavidR2008 | No such command 'features reload' (type 'help features reload' for other possible commands) |
| 20:41.00 | DavidR2008 | 1.4.23.1 (sorry I forgot to mention) |
| 20:41.14 | *** join/#asterisk orkid_ (n=orkid@unaffiliated/orkid) |
| 20:42.28 | DavidR2008 | If no one knows it's not a big deal. I'm trying to test one touch recording and I can't seem to get it to work. |
| 20:42.52 | orkid_ | Chainsaw, not sure what the bios ver really was because I read it inside the BIOS menu (and after update it was the same there.. but on POST screen it was the updated version).. I thought the beeps were gone (for the first few seconds of the call they were) but then i heard them again .. Hmmm. I didn't have this issue on another motherboard. Could it be the chipset/CPU combo or something? |
| 20:43.14 | DavidR2008 | I copied the example from voip-info.org but it doesn't work for me. Does anyone have any experience with one touch recording? |
| 20:43.42 | mmlj4 | NoxIn-: when I do ./Setup dahdi, it displays "Enabling the TDM Voice Asterisk Support", and remember choosing TDM voice... bah |
| 20:44.07 | kaldemar | DavidR2008: reload res_features.so or module reload res_features.so might do it for 1.4. |
| 20:44.17 | Chainsaw | orkid_: It is possible. Some systems have SMI (System Management Interrupt) activity going on in the background all the time. |
| 20:44.31 | orkid_ | Chainsaw, The current computer is a Duron1600 on asus k8V-X (via chipset), it was before on a P3-1000, gigabyte motheboard, with via chipset |
| 20:44.32 | Chainsaw | orkid_: Which makes them unsuitable for applications like this, where you really want near-realtime response. |
| 20:44.48 | Chainsaw | orkid_: I'm not a fan of VIA chipsets. If you can get something nForce-based it'll probably work better. |
| 20:44.48 | orkid_ | Chainsaw, How can I check if this is the case? |
| 20:45.13 | Chainsaw | orkid_: I can't think of a quick yes/no way to check. |
| 20:45.27 | orkid_ | Chainsaw, yeah you live and you learn. I've got burned with this chipset already because it doesn't do sata2 (or sata DVD burning) |
| 20:46.18 | orkid_ | Chainsaw, well. thanks for your help. I might go back to the old motherboard/chipset and see if that fixes the problem, then I'll know pretty much for certain that it's the hardware |
| 20:46.27 | Chainsaw | orkid_: I mostly remember VIA for the 686B southbridge. |
| 20:46.31 | orkid_ | (or perhaps implementation in linux, who knows though) |
| 20:46.40 | orkid_ | well, the gigabyte has that 686B |
| 20:46.48 | Chainsaw | Awful chipset. |
| 20:47.01 | Chainsaw | It doesn't implement PCI bus parking. You'll have stability problems with SoundBlaster Live! cards as a result. |
| 20:47.24 | orkid_ | should I be seeing interrupts atm on the wcfxo device? |
| 20:47.43 | DavidR2008 | kaldemar: that worked, thanks! |
| 20:47.45 | *** join/#asterisk Toerkeium (i=Toerkeiu@201.216.206.221) |
| 20:47.55 | orkid_ | Btw, the line is " |
| 20:47.55 | orkid_ | <PROTECTED> |
| 20:47.55 | orkid_ | " .. Could disabling APIC help? |
| 20:48.12 | Chainsaw | orkid_: All disabling APIC will do is limit you to 16 IRQs again. |
| 20:48.18 | DavidR2008 | Gido-E: if you're on 1.4 module reload app_voicemail.so will re-parse voicemail.conf |
| 20:48.34 | Chainsaw | orkid_: So sharing becomes more likely again. You might have to reshuffle hardware yet another time to avoid that. |
| 20:49.09 | Chainsaw | Unless you have grave IRQ routing difficulties (which you don't seem to have as you can boot fine), disabling APIC is usually a bad idea. |
| 20:50.02 | orkid_ | Chainsaw, weird that I didn't have this problem on the 686B, but do on the 8237. So disabling APIC won't help you think? Hmm, maybe I'll just try it before I switch back to the other mobo |
| 20:50.08 | orkid_ | bbl |
| 20:50.20 | Chainsaw | orkid_: Yeah, worth a go but it seems unlikely to me. |
| 20:50.25 | docid | anybody got any idea what the trick to getting the dahdi interface to come up in asterisk when signalling=sf_w is set? i get dahdi show channels no such command, but if i change it to em_w or featb or featdmf its there |
| 20:50.32 | docid | whoopz |
| 20:52.54 | *** join/#asterisk bijit (n=benji@190.241.15.48) |
| 20:54.43 | *** join/#asterisk {Sean} (n=sean@adsl-99-49-211-86.dsl.sfldmi.sbcglobal.net) |
| 20:54.56 | mmlj4 | just a sanity check, please... on * 1.4.23.2, what do I need to do to make * compile with dahdi? |
| 20:55.03 | mmlj4 | or support for it, etc. |
| 20:55.24 | mmlj4 | i mean aside from compiling dahdi separately |
| 20:56.32 | *** join/#asterisk orkid_ (n=orkid@unaffiliated/orkid) |
| 20:56.56 | orkid_ | Chainsaw, I can't believe it, it works! |
| 20:57.08 | Chainsaw | orkid_: Neat! |
| 20:57.15 | orkid_ | " 5: 217893 XT-PIC-XT ehci_hcd:usb5, wcfxo, VIA8237 |
| 20:57.15 | orkid_ | " |
| 20:57.28 | Chainsaw | Fair play to you. I hadn't expected that. |
| 20:58.13 | orkid_ | I loaded defaults, turned off ECC (ON BY DEFAULT! i guess it just disables if you don't have ECC RAM), disabled APIC and ACPI, enabled PnP OS again (disabled by default), and it works. |
| 20:58.25 | orkid_ | Could it be something with the APIC implementation? |
| 20:58.46 | Chainsaw | I would look at buggy ACPI firmware more then APIC myself. |
| 20:58.53 | orkid_ | of linux? Anyway, I will turn on ACPI 2 extensions and see if it still works. |
| 20:58.55 | Chainsaw | But I've been proven wrong before, so who knows :) |
| 20:58.56 | orkid_ | brb again |
| 21:01.32 | mmlj4 | ~book |
| 21:01.33 | jbot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 21:07.41 | *** part/#asterisk flujan (n=flujan@189-039-010-068.static.spo.ctbc.com.br) |
| 21:07.50 | DavidR2008 | how are features supposed to work? I can't get my asterisk box to do anything when I press # or * the tone just get passed on the the other side of the call |
| 21:10.53 | docid | mine work fine, ## transfers, etc...now if i could jsut get my t1 configuration information from the stupid telco |
| 21:11.09 | jplank | has anyone every used a viking control module with *? |
| 21:11.35 | *** join/#asterisk voxter (n=voxter@76.77.95.2) |
| 21:12.46 | *** join/#asterisk macli (n=macli@nmc.brc.ubc.ca) |
| 21:13.39 | CapriCoRN^80 | [TK]D-Fender: you there ? |
| 21:16.00 | DavidR2008 | it looks like it had something to do with the Set(DYNAMIC_FEATURES...) command. I removed that and it started working as expected. |
| 21:16.14 | [TK]D-Fender | CapriCoRN^80: yes |
| 21:17.12 | *** join/#asterisk strk (n=strk@ip-123-78.static.adsl.cheapnet.it) |
| 21:17.50 | *** part/#asterisk strk (n=strk@ip-123-78.static.adsl.cheapnet.it) |
| 21:18.08 | CapriCoRN^80 | [TK]D-Fender: http://pastebin.com/d7415442b |
| 21:18.24 | CapriCoRN^80 | my users are unreacable |
| 21:18.30 | *** join/#asterisk orkid_ (n=orkid@unaffiliated/orkid) |
| 21:18.31 | CapriCoRN^80 | strange |
| 21:20.25 | orkid_ | Chainsaw, so ACPI2 can be enabled and it still works ok (I don't see APIC-fastio but XT-PIC-XT in /proc/interrupts), but when I turned on ACPI APIC (pointer?) then I get the beeps. I remember there being some issues with APIC, pointer, RSDT or DSDT or somesuch, and linux; this was a while ago though. unrelated perhaps |
| 21:21.09 | orkid_ | I didn't try APIC w/o ACPI2 extensions though |
| 21:21.16 | *** part/#asterisk Sparky1 (n=Sparky1@12.41.116.4) |
| 21:27.58 | CapriCoRN^80 | [TK]D-Fender: you checked the pastebin ? |
| 21:29.38 | *** join/#asterisk telecos (n=sergio@34.166.219.87.dynamic.jazztel.es) |
| 21:32.58 | [TK]D-Fender | CapriCoRN^80: Check all your networking |
| 21:33.30 | *** join/#asterisk umpc (n=Justin@unaffiliated/umpc) |
| 21:34.50 | CapriCoRN^80 | ok |
| 21:36.02 | *** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose) |
| 21:36.42 | CapriCoRN^80 | [TK]D-Fender: if its networking problem why my user is connecting to the server |
| 21:37.48 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) |
| 21:39.32 | *** join/#asterisk moy (n=chatzill@74.12.124.89) |
| 21:40.45 | *** join/#asterisk jameswf (n=james@unaffiliated/jameswf-home) |
| 21:41.43 | CapriCoRN^80 | [TK]D-Fender: i am little bit confused |
| 21:42.17 | CapriCoRN^80 | [TK]D-Fender: if you can tell me why i am getting request and connecting to server but my status is unreacable |
| 21:49.46 | *** join/#asterisk `Sauron (n=sauron@dsl001-130-033.aus1.dsl.speakeasy.net) |
| 21:50.50 | *** join/#asterisk Der-Tim (n=tkorves@dehhth2srv1.onic.de) |
| 21:50.54 | Der-Tim | hi there |
| 21:52.31 | brunner | does asterisk do a good job of taking advantage of multi-processor environments? |
| 21:52.57 | russellb | it depends on what your system is doing |
| 21:53.09 | russellb | but in general, Asterisk is very heavily multi-threaded, and will use multiple CPUs. |
| 21:53.53 | brunner | in what scenarios would asterisk not take full advantage of multiple CPUs? |
| 21:54.39 | apeiron | erm. That's more a question for your kernel / scheduler / threading library. |
| 21:54.53 | *** join/#asterisk wonderworld (n=ww@ip-62-143-20-187.unitymediagroup.de) |
| 21:57.28 | *** join/#asterisk RoPBX (n=nickserv@200.93.34.175) |
| 21:57.35 | RoPBX | hello all |
| 21:57.49 | wonderworld | would the DoS described here --> http://secunia.com/advisories/34229/ crash asterisk or the whole box? |
| 21:57.53 | RoPBX | please, somebody knows about glare management? |
| 22:00.42 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 22:01.38 | brunner | So in theory, is there any reason why a server like this couldn't handle at least 1000 calls, if there's no transcoding going on? http://tinyurl.com/d7afen |
| 22:02.35 | russellb | 8 quad core CPUs? |
| 22:02.37 | russellb | that's hot. |
| 22:02.40 | brunner | no doubt |
| 22:02.52 | russellb | It's really hard to say until you get to testing, but yeah, I would say that should be fine |
| 22:03.01 | brunner | out of my price range, of course, but I'm still curious about whether there'd be any bottlenecks |
| 22:03.20 | russellb | your bottlenecks would certainly not be CPU |
| 22:03.25 | brunner | lol, yeah |
| 22:03.25 | russellb | it would be network throughput probably. |
| 22:03.30 | brunner | hmm |
| 22:03.35 | russellb | or performance of the NIC(s) |
| 22:03.45 | brunner | well, I'll have to interface via TDM |
| 22:03.56 | brunner | no sip for me, from my telco I'm racking up with =/ |
| 22:04.14 | russellb | 1000 calls worth of T1 cards?! |
| 22:04.15 | brunner | at least not right now. they say they'll offer it before the end of the year |
| 22:04.31 | brunner | russellb: I'm not actually going to buy that thing. I'm only going to try to support 500 calls. |
| 22:04.55 | russellb | So, like, 4 quad span T1 cards perhaps ... |
| 22:05.05 | brunner | but yes, I plan on buying 6 T1 cards |
| 22:05.10 | russellb | nods |
| 22:05.29 | brunner | I thought about getting a SIP gatway, but I think they're just as expensive, right? |
| 22:05.30 | [TK]D-Fender | brunner: What are you doing with these calls? |
| 22:05.34 | russellb | I would actually recommend 2 servers (maybe 3) because of the number of T1 boards. |
| 22:05.44 | brunner | [TK]D-Fender: conferencing |
| 22:05.50 | wonderworld | brunner: model name : Intel(R) Core(TM)2 Duo CPU E6850 @ 3.00GHz -> handles about 20 calls with transcoding with 10% load on 1 cpu |
| 22:05.54 | brunner | russellb: oh yeah? |
| 22:05.56 | russellb | so yes, conferencing takes some CPU, as well |
| 22:06.04 | [TK]D-Fender | brunner: massive overkill. |
| 22:06.15 | brunner | [TK]D-Fender: what is? |
| 22:06.27 | russellb | that huge server? heh, yeah |
| 22:06.29 | brunner | [TK]D-Fender: like I said many, many times, I'm not buying that server I linked to. |
| 22:06.37 | russellb | but it was awesome. |
| 22:06.38 | [TK]D-Fender | brunner: get AudioCodes Mediant SIP gateways, and span 2-3 small servers. |
| 22:06.44 | brunner | russellb: yes |
| 22:07.00 | russellb | nooo, get Digium T1 boards :-D |
| 22:07.04 | brunner | lol |
| 22:07.09 | russellb | (I work for Digium.) |
| 22:07.14 | brunner | yeah, I figured |
| 22:07.22 | brunner | [TK]D-Fender: I'll take a look, thanks |
| 22:07.35 | wonderworld | would the DoS described here --> http://secunia.com/advisories/34229/ crash asterisk or the whole box? |
| 22:07.49 | russellb | if it's an asterisk vulnerability, it's Asterisk |
| 22:10.07 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.static.bezeqint.net) |
| 22:12.34 | RoPBX | russellb, please do you know something about anti-glare? |
| 22:13.18 | brunner | [TK]D-Fender: the Mediant 2000 only support 16 T1's |
| 22:13.20 | russellb | I know nothing :-( |
| 22:13.23 | russellb | about anything. |
| 22:13.43 | RoPBX | oh ok |
| 22:14.03 | outtolunc | is it friday already? <G> |
| 22:19.29 | [TK]D-Fender | brunner: 18:06]<[TK]D-Fender>brunner: get AudioCodes Mediant SIP gateways, and span 2-3 small servers. <- today's magic word is "plural" |
| 22:19.44 | [TK]D-Fender | ;) |
| 22:20.20 | Juggie | does anyone have a CSV/DB of area codes, eg areacode,state/province,country |
| 22:20.33 | brunner | yeah, yeah, I was just hoping to do everything on as few devices as possible |
| 22:20.34 | Juggie | for north america |
| 22:21.09 | [TK]D-Fender | bruAnd did you think you were going to cram all of those T1's into 1 server directly via PCI? |
| 22:22.23 | [TK]D-Fender | brunner: that 1U / 368 channels. Not half bad if you ask me. |
| 22:22.25 | *** join/#asterisk nullable_type (n=kumana@hq.verbx.net) |
| 22:23.00 | nullable_type | Q: What's the best way of upgrading Asterisk from an older version(Without source) to new version (using new source)? Thanks |
| 22:23.37 | brunner | [TK]D-Fender: yep. what's wrong with cramming a bunch of T1's into one box with PCI? |
| 22:24.48 | *** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir) |
| 22:34.12 | tompaw | Hi. |
| 22:35.00 | tompaw | I need some nice cdr browser that will let me generate and view stats, call history etc. * 1.6, could be either based on filed cdrs or mysql, don't care. |
| 22:35.04 | tompaw | Anything you could recommend? |
| 22:35.23 | tompaw | (I don't need billing, account management or anything like that, just pure CDR browser) |
| 22:35.24 | *** join/#asterisk RobH_ (n=RobH@208-106-97-77.dsl.static.sonic.net) |
| 22:38.04 | *** join/#asterisk Bonix (n=Bonix@200-195-41-212.isimples.com.br) |
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| 22:43.06 | KavanS | anyone use least cost routing? |
| 22:43.28 | nullable_type | Is Asterisk 1.6.0.6 stable enough for production. Whats the best version to use for production? |
| 22:45.17 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 22:46.45 | bmoraca | brunner: Sangoma makes a single height 8port T1 card... |
| 22:48.31 | tompaw | KavanS: I used to use it with a2billing. |
| 22:48.51 | KavanS | damn a2billing looks involved |
| 22:49.04 | tompaw | "involved"? |
| 22:49.24 | KavanS | i.e. multiple hours/dependencies of software to configure/mess with to get it working |
| 22:49.26 | KavanS | lol |
| 22:49.31 | KavanS | anyone use LCDIal yet? |
| 22:49.45 | tompaw | KavanS: now we're using voipswitch for that purpose. |
| 22:49.55 | tompaw | a2billing does just... the billing ;) |
| 22:52.11 | KavanS | yeah |
| 22:52.16 | KavanS | no need to mess with billing... |
| 22:52.17 | KavanS | I don't resell |
| 22:58.31 | *** join/#asterisk RobH (n=RobH@dsl017-048-227.sfo4.dsl.speakeasy.net) |
| 22:58.55 | nullable_type | Hey guys I installed 1.6.0.6 over 1.2 without installing, now all screwed up. How do i uninstall everything and then start from scratch? |
| 22:59.42 | jameswf | make uninstall |
| 22:59.58 | *** join/#asterisk jsgoecke (n=jsgoecke@c-67-180-103-93.hsd1.ca.comcast.net) |
| 23:00.20 | nullable_type | will that uninstall both versions? |
| 23:00.55 | jameswf | no.... get 1.2....source and make uninstall as well |
| 23:01.01 | Qwell | 1.2 doesn't have uninstall |
| 23:01.10 | jameswf | well hell |
| 23:01.11 | jameswf | :) |
| 23:02.16 | Juggie | there isnt that much to remove |
| 23:02.17 | *** join/#asterisk SwK (n=SwK@freeswitch/developer/swk) |
| 23:02.24 | Juggie | and uninstalling really doesnt do anything that fancy |
| 23:02.33 | nullable_type | Hey guys whats the best version to go for Production, 1.6 or 1.4 |
| 23:03.21 | Corydon76-dig | nullable_type: depends upon the feature set you need |
| 23:03.59 | nullable_type | pretty much i need is call bridging, http AMI, SIP Reinvite |
| 23:06.59 | nullable_type | Is that all available in 1.4 |
| 23:08.02 | Qwell | nullable_type: doesn't sound like you've done very much research... |
| 23:09.05 | nullable_type | @Qwell >> Yes, sounds like I will have to :) . I only used 1.6.0.6 in dev and wondering if 1.4 is more stable for production |
| 23:12.10 | *** join/#asterisk macli (n=macli@nmc.brc.ubc.ca) |
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| 23:38.18 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194) |
| 23:43.52 | *** join/#asterisk trnzmeta (n=bleh@iinet.guard.com.au) |
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| 23:44.33 | *** part/#asterisk ruben23 (n=IT-ADMIN@124.107.3.178) |
| 23:44.59 | trnzmeta | guys: when someone transfers a phone call whilst I'm on the phone or phone is ringing/busy, it automatically goes to voicemail/engeaged |
| 23:45.15 | trnzmeta | what should I be looking at make sure it rings... (2nd line?) |
| 23:45.27 | [TK]D-Fender | trnzmeta: your phone. |
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| 23:57.09 | Merlin | what does switchvox use for their fax solution? 3rd party software or something open source? |
| 23:58.02 | [TK]D-Fender | Merlin: Yes. |
| 23:59.38 | Merlin | [tk: which? |
| 23:59.47 | Merlin | [tk]d-fender: which though? |