| 00:05.07 | KyleK | the wiki page for Background could use a few more details, if I declare an extension of * and say 12345, can I jump to either while it's playing a message? |
| 00:05.59 | bkw_ | goes back to playing with zRTP |
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| 00:17.00 | orpheee | i need to install other thing for meetme ? |
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| 00:33.40 | [TK]D-Fender | orpheee: Zaptel/DAHDI is REQUIRED and needs to be compiled before * is so that its included in the pre-req's |
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| 00:52.52 | orpheee | [TK]D-Fender> so i need to install Zaptel and DAHDI |
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| 00:56.24 | orpheee | tks for information |
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| 01:14.53 | Tene | The asterisk docs, when talking about imap storage of voicemail, say this: "The directives "authuser" and "authpassword" are not needed when using Kerberos. They are defined to allow Asterisk to authenticate as a single user that has access to all mailboxes as an alternative to Kerberos. ", but I can't find any documentation anywhere at all on actually using kerberos wit hasterisk in any useful way. |
| 01:15.02 | Tene | Am I missing something? |
| 01:23.11 | *** join/#asterisk doolittlework (n=f@196.211.34.2) |
| 01:23.54 | doolittlework | how does one go about dumping cdr records out on server com port? |
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| 01:43.07 | darkavanger | hi |
| 01:43.36 | darkavanger | i need to connect to a sip server what can i use?? |
| 01:46.03 | drmessano | darkavanger: Linux |
| 01:46.17 | *** join/#asterisk egypcio (n=egypcio@unaffiliated/egypcio) |
| 01:47.01 | darkavanger | drmessano: i am using linux i wanted to say wich client application ? :D |
| 01:47.27 | drmessano | I assume youre looking for a softphone...? |
| 01:47.38 | drmessano | Your question was horribly vague |
| 01:47.39 | darkavanger | yess thats it |
| 01:48.13 | drmessano | ~softphone |
| 01:48.14 | infobot | [~softphone] A soft-phone is a program that lets your computer use a compatible audio interface (usually a speaker&mic, or headset), and optionally a webcam, to place VoIP calls with a given protocol (SIP/IAX/H.323/MGCP/etc). For links enter ~zoiper , ~xlite , ~twinkle , ~bria , ~eyebeam , ~ekiga |
| 01:48.22 | darkavanger | in my contry it is 3ham so you can imagine how slow my brain is working |
| 01:48.37 | drmessano | Compared to the usual? Thats impressive |
| 01:49.11 | darkavanger | ~xlite |
| 01:49.12 | infobot | [~xlite] X-Lite is a free SIP soft-phone for Windows, MacOSX, and Linux that can be downloaded from http://www.counterpath.com/ |
| 01:50.03 | darkavanger | ~zoiper |
| 01:50.04 | infobot | [~zoiper] Zoiper (Formerly known as Idefisk) is a free SIP and IAX soft-phone for Windows, MacOSX, and Linux that can be found at http://www.zoiper.com |
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| 01:51.54 | demonist | hello, i have two pcs on a local lan. what can i use to make a call between them using a softphone |
| 01:52.21 | demonist | can i use sip to sip client or will i still need a server such as asterisk |
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| 02:11.24 | Optic | moop |
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| 02:13.32 | hesco | I'm writing my first daemon. On its second iteration through the infinite loop, it dies on "no connection to the (pg) server" error on a query it had no trouble running on the first iteration, and then goes away. the sleep is only 60 seconds, why would I lose my db connection so easily? I earlier included: http://www.perlmonks.org/?node_id=581685 on clinton's advice. |
| 02:14.02 | hesco | sorry folks, wrong channel |
| 02:14.19 | hesco | ignore that |
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| 02:21.02 | TheCoffeMaker | Hi, it's just a little question ... I'm totally new on this ... and have this doubt ... is possible to setup an asterisk PBX with an miti-itx ? |
| 02:22.59 | leifmadsen | TheCoffeMaker: I believe it is |
| 02:23.05 | leifmadsen | mini-itx is just x86 based right? |
| 02:23.15 | TheCoffeMaker | leifmadsen, yes |
| 02:23.18 | leifmadsen | should be fine then |
| 02:23.28 | leifmadsen | I've never done it, but pretty sure I've heard of others doing it |
| 02:23.46 | leifmadsen | I think those are typically via chipsets and such too |
| 02:25.22 | TheCoffeMaker | yes ... I had this doubt coz don't sure about the systems requiremts yet ... but I want to use a mini-itx to create little blackboxes to deploy in small enterprises |
| 02:25.58 | leifmadsen | I don't really see an issue |
| 02:26.06 | leifmadsen | just do some googling, I'm sure others have done it |
| 02:26.48 | TheCoffeMaker | yes ... I will carry on with my researchs and reading everything :) thanks for your help! |
| 02:27.36 | *** part/#asterisk TheCoffeMaker (n=damian@190.245.17.43) |
| 02:28.02 | drmessano | leifmadsen: got 30 seconds for an IMAP + Asterisk question |
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| 02:28.31 | drmessano | (I asked to ask since you normally leave to leave) |
| 02:28.37 | drmessano | .... |
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| 02:30.18 | leifmadsen | drmessano: just ask :) |
| 02:30.30 | leifmadsen | I'm no IMAP pro though |
| 02:30.53 | drmessano | I had some nasty issues with IMAP and 1.6.. using the libraries that I YUM'ed down with CentOS 5.3.. which I think are 2004d |
| 02:31.04 | drmessano | Youve at least tried it, I think.. what worked for you? |
| 02:31.12 | leifmadsen | don't use the package, use the latest source |
| 02:31.15 | leifmadsen | 2007e or something |
| 02:31.20 | leifmadsen | 2004 is oooooold |
| 02:31.24 | drmessano | I suspected that was it.. |
| 02:31.26 | drmessano | Yes, it is |
| 02:31.46 | drmessano | When I compiled it all in, and restarted asterisk, I started getting MWI on phones WITHOUT MAILBOX |
| 02:31.50 | drmessano | MAILBOXES |
| 02:31.54 | drmessano | As in, ALL the phones |
| 02:32.04 | drmessano | So I figured it was something lame |
| 02:32.39 | drmessano | See, you knew more than I was going to ask |
| 02:32.46 | drmessano | Mr Documentation Guy |
| 02:33.24 | drmessano | I have just found LITTLE to NOTHING on IMAP and Asterisk.. and no real world "I did and it broke, do this" blog posts or anything out there |
| 02:33.39 | drmessano | I guess I should contribute something when i get it working |
| 02:33.40 | leifmadsen | because no one uses IMAP |
| 02:33.45 | drmessano | Yeah ^^^^ |
| 02:33.59 | Optic | woop |
| 02:34.16 | leifmadsen | I wrote something small for IMAP, but it isn't public yet since it is so rough |
| 02:34.24 | leifmadsen | poow |
| 02:35.15 | drmessano | Well, I found a nice PP presentation of yours online mentioning CURL and IMAP |
| 02:35.19 | drmessano | Very helpful |
| 02:35.40 | Optic | my asterisk audio works with a sip softphone incoming call, but not with my link2voip did :P |
| 02:36.21 | Optic | maybe link2voip is broken tonight |
| 02:36.26 | leifmadsen | drmessano: yep, that'll basically be how to setup IMAP |
| 02:36.42 | leifmadsen | Optic: or you're having NAT issues |
| 02:37.00 | Optic | server is not behind nat, 5060 and 10000-60000 passed through firewall |
| 02:37.15 | carrar | I use Asterisk and IMA, works great, taste even better |
| 02:37.18 | carrar | IMAP |
| 02:37.21 | leifmadsen | you'd need to check to see if you see the incoming INVITE, etc... |
| 02:37.32 | leifmadsen | carrar: what server? |
| 02:37.35 | leifmadsen | (IMAP server) |
| 02:37.36 | carrar | 1.4 |
| 02:37.38 | Optic | i do, i get the call, rtp packets flow, but there's no audio :( |
| 02:37.38 | carrar | oh |
| 02:37.38 | carrar | UW |
| 02:37.44 | leifmadsen | dovecot? |
| 02:37.47 | carrar | yuppers |
| 02:37.48 | drmessano | Well, not to use the F word in here, leifmadsen, but it inspired me to add IMAP fields for the users maiboxes in FreePBX and the associated voicemail.conf options.. So IMAP may become more of an interest for that subset of users who now have "access" to add it on GUI systems |
| 02:38.05 | drmessano | May help get it out there |
| 02:38.16 | leifmadsen | I'd rather it not get used until it is proven 100% stable :) |
| 02:38.22 | drmessano | heh |
| 02:38.25 | drmessano | Yeah |
| 02:38.29 | carrar | it's stable for the customer I set it up for |
| 02:38.29 | leifmadsen | the worst thing is using something unreliable and giving it to GUI users |
| 02:38.30 | drmessano | YOUR MILEAGE MAY VARY |
| 02:38.46 | carrar | wasn't very hard to setup |
| 02:38.48 | leifmadsen | carrar: yes, I've heard that about a few installations -- it really comes down to what backend server you're using, and what you're using it for |
| 02:39.05 | carrar | They wanted to offload messages off their exchange server |
| 02:39.19 | carrar | guess no one there answers their phone :) |
| 02:39.19 | leifmadsen | I need to figure out dovecot more so that I can set it up and actually have some documentation that would be useful in a production instance |
| 02:39.25 | leifmadsen | heh |
| 02:39.44 | carrar | 1st time I used dovecot too |
| 02:39.47 | carrar | dovecot is nice |
| 02:40.12 | drmessano | Im gonna make it work on Exchange IMAP first.. I want something I can document for those MS SBS users who dont plan to set up UM on SBS 2008 or have SBS 2003 |
| 02:40.15 | carrar | with it's SQL queries |
| 02:40.29 | drmessano | I think its a perfect fit for that folks |
| 02:40.36 | carrar | create your own dovecot queries in postgres for auth |
| 02:40.39 | drmessano | SBS + Asterisk UM |
| 02:40.55 | leifmadsen | wish I had an exchange server to play against |
| 02:41.05 | leifmadsen | not that I want to admin such a server |
| 02:41.11 | jaytee | I do but I don't use it for IMAP |
| 02:41.12 | leifmadsen | lets get that clear right now |
| 02:41.16 | drmessano | Just for the IMAP access |
| 02:41.32 | leifmadsen | I always liked the idea of trying to get IMAP working with Gmail :) |
| 02:41.43 | drmessano | leifmadsen: I'll e-mail you.. Can get you a mailbox to donate for "Science" |
| 02:41.46 | leifmadsen | if only I had the skills to modify the code |
| 02:41.52 | jaytee | works great as an alternative to Comedian Mail though |
| 02:41.53 | drmessano | leifmadsen: Yeah, thats on the list here too |
| 02:41.54 | carrar | and write books |
| 02:41.54 | leifmadsen | drmessano: hmmmm, ok, good call |
| 02:42.00 | carrar | those are excellent skills |
| 02:42.16 | leifmadsen | writing books is for people who can't code :) |
| 02:42.20 | carrar | haha |
| 02:42.32 | leifmadsen | and coding is for people who can't write books |
| 02:42.41 | carrar | thats for sure |
| 02:42.43 | leifmadsen | Corydon76-dig is one of the few who I know who could/can do both |
| 02:42.49 | carrar | Leave no comments beind! |
| 02:42.51 | carrar | behind |
| 02:43.05 | leifmadsen | speaking of behind... I gotta go smack my g/f on the butt |
| 02:43.07 | leifmadsen | brb |
| 02:43.12 | carrar | I take the backpacking theory when writing programs |
| 02:43.13 | leifmadsen | laughs |
| 02:43.16 | carrar | Leave no trace |
| 02:44.08 | jaytee | is that for job security? planning for future consulting services? |
| 02:44.17 | carrar | yes & yes |
| 02:44.47 | carrar | naw, when you are in the groove, writting comments only slows you down |
| 02:45.00 | leifmadsen | until you have to debug it ;) |
| 02:45.04 | carrar | hahah yeah |
| 02:45.06 | jaytee | zing! |
| 02:45.14 | leifmadsen | zap! |
| 02:45.16 | leifmadsen | kerplow! |
| 02:45.22 | leifmadsen | foop! |
| 02:45.24 | jaytee | ok, now what was I trying to do here again? |
| 02:45.32 | carrar | thats when you go back and make comments here and there |
| 02:45.40 | jaytee | I have to comment or I'll forget half of what I've coded. |
| 02:45.48 | carrar | <PROTECTED> |
| 02:45.50 | leifmadsen | I tend to write what I want in the dialplan, then go back and read through what I was trying to do and comment then |
| 02:46.00 | leifmadsen | I don't do dialplan + commenting at the same time (at least rarely) |
| 02:46.16 | carrar | yeah I can't either |
| 02:46.20 | *** part/#asterisk darkavanger (n=darkavan@41.225.122.130) |
| 02:46.43 | carrar | Just in a hurry to complete it and test it |
| 02:46.57 | jaytee | yeah, I comment after I've tested something and I've tweaked it so it's considered "stable" then I comment it verbosely. |
| 02:47.11 | carrar | comment it later once you have something presentable |
| 02:47.52 | carrar | commenting is something you can do once you are too drunk to code |
| 02:47.53 | jaytee | bbiab |
| 02:50.26 | carrar | for the imap db I wrote a init style script to manage the db, works nice. php would be even simpler almost |
| 02:54.04 | drmessano | carrar: Is that like a Kenwood hybrid? |
| 02:54.17 | carrar | heh |
| 02:55.38 | drmessano | Using Asterisk IMAP to store VM in an exchange mailbox seems like the coolest thing since putting a Qmail box in front of Exchange for message sanitizing |
| 02:56.03 | carrar | I don't use IMAP to put voicemails in exange |
| 02:56.46 | carrar | I use IMAP to put the voicemail in a mailbox on a unix box |
| 02:57.23 | carrar | the imap and mail boxes and asterisk are all on the same box |
| 02:57.31 | carrar | superfly easy |
| 02:58.01 | drmessano | Im not talking about YOU |
| 02:58.22 | drmessano | ducks |
| 02:58.30 | carrar | You trying to save money!?? |
| 02:58.33 | carrar | ~savemoney |
| 02:58.34 | infobot | <Gremlin> It's okay, but I was hoping to save some money with a soldering iron and a bunch of sound cards. |
| 02:58.37 | drmessano | ROFL |
| 02:59.06 | drmessano | No, like I said above.. Lots of SBS users out there.. Coupling that with an Asterisk box + IMAP to their existing mailboxes would be awesome |
| 02:59.25 | carrar | Why not just Mail it to their mailbox? |
| 02:59.36 | carrar | why do you need to use imap to stuff it in? |
| 03:00.01 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 03:00.18 | drmessano | Because I HATE duplication of information.. if I mail and delete, no MWI on the phone.. if I mail AND leave it, they have to delete it twice |
| 03:00.32 | drmessano | I like the whole MWI + IMAP sync and having the message be somewhat "live" |
| 03:01.54 | carrar | There is no escape from stupid users |
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| 03:43.44 | KyleK | I might end up writing my own voicemail app to store metadata and .mp3 on gmail via imap and .ulaw stored locally |
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| 04:00.14 | carrar | KyleK, make a shoutcast mp3 voicemail streamer |
| 04:00.19 | carrar | heh |
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| 04:28.57 | sergey | Hi. use Asterisk 1.6.1.0 and app_minivm After MinivmGreet() have core dump, and MinivmGreet(test@test) have greeting but if press * have core dump again. Is it bug? |
| 04:35.33 | carrar | wakarimasen |
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| 04:51.03 | *** join/#asterisk dshap (n=IceChat7@ip68-231-218-208.oc.oc.cox.net) |
| 04:51.42 | dshap | hey everyone, is there someone here who can help me with an Asterisk issue? I'm trying to successfully register my server with my SIP provider |
| 04:51.46 | dshap | and it doesn't appear to be working |
| 04:52.58 | dshap | can anyone read this? just want to make sure i'm successfully logged into the channel |
| 04:54.09 | *** part/#asterisk dshap (n=IceChat7@ip68-231-218-208.oc.oc.cox.net) |
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| 04:55.08 | *** part/#asterisk dshap (n=IceChat7@ip68-231-218-208.oc.oc.cox.net) |
| 04:55.45 | carrar | dshap: c n ou re d th s |
| 04:55.55 | carrar | doh |
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| 05:03.16 | dshap | hey if there is someone here who can help me with an Asterisk issue I would appreciate it very much |
| 05:03.37 | dshap | I'm trying to use voip.ms as a SIP provider and I'm using their sip.conf and extension.conf files that they provide as sample/test configurations |
| 05:03.53 | dshap | and my server is telling me that registration is timining out |
| 05:15.10 | securevoip | dshap: sounds like you have an IP networking issue if reg is timing out |
| 05:15.43 | dshap | my server is behind a consumer-grade belkin router |
| 05:15.52 | dshap | but i set up a DMZ with the local IP of my server |
| 05:16.07 | dshap | and i disabled the firewall |
| 05:16.17 | KyleK | packet sniff the thing :) |
| 05:16.28 | dshap | eh? |
| 05:16.34 | dshap | sry haha i'm a huge n00b with linux |
| 05:16.42 | dshap | i'm basically trying to learn linux and asterisk at the same time |
| 05:16.47 | KyleK | wireshark, look at the sip packets if they go and dont come in |
| 05:16.53 | KyleK | oh |
| 05:16.57 | dshap | oh |
| 05:16.58 | securevoip | Or, grab a FREE DID from http://www.ipcomms.net/html/freedid.html and see if that works... |
| 05:16.59 | dshap | but i have used wireshark |
| 05:17.00 | dshap | randomly |
| 05:17.10 | KyleK | i cant help you too too much gotta go to sleep :-/ |
| 05:17.25 | dshap | i already have a DID though |
| 05:17.55 | carrar | Sounds like you might be behind NAT and your nat stats are timing out on the "consumer-grade belkin router" |
| 05:18.03 | carrar | use qualify |
| 05:18.08 | carrar | see if that makes any difference |
| 05:18.17 | dshap | what do you mean by "use qualify"? |
| 05:18.22 | carrar | look it up |
| 05:18.33 | dshap | if in windows I type "ipconfig" and it says my IP is like 192.168.2.4 |
| 05:18.34 | carrar | it's in your sip.conf |
| 05:18.35 | dshap | a local IP |
| 05:18.38 | dshap | doesn't that mean I'm behind a NAT |
| 05:18.47 | carrar | yes thats a NAT IP |
| 05:18.52 | dshap | ok right |
| 05:18.56 | dshap | so i have nat=yes in sip.conf |
| 05:19.08 | securevoip | set qualify=yes in sip.conf under your ITSP |
| 05:19.13 | carrar | 10/8, 172.16/12 & 192.168/16 are popular NAT IANA IP's |
| 05:20.14 | dshap | gonna try qualify right now |
| 05:20.25 | carrar | securevoip, giving the answer isn't gonna help him look through the sip.conf example and find all the other juicy things |
| 05:20.45 | kaldemar | you need to set externip if your asterisk server is behind a nat. |
| 05:20.48 | carrar | JUICY SIP CONFIGS!! |
| 05:20.50 | kaldemar | ~sipnat |
| 05:20.50 | infobot | [~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 |
| 05:21.02 | *** join/#asterisk SunnyDP (n=scan@bas7-montrealak-1128544871.dsl.bell.ca) |
| 05:21.25 | securevoip | are you testing w/ inbound calls or outbound calls? |
| 05:21.40 | dshap | i will be using both |
| 05:21.52 | dshap | but at this point i just want to get the sample configuration file working |
| 05:21.58 | dshap | the one that my provider gave me |
| 05:22.24 | carrar | read through your sip.conf example file |
| 05:22.36 | securevoip | post your sip.conf to http://pastebin.com/ |
| 05:23.30 | SunnyDP | has anyone of you been succesfull at setting DHCP option 120 (SIP Servers) in Windows Server DHCP ? |
| 05:23.53 | dshap | just pastebinned it under the name "dshap" |
| 05:23.57 | dshap | that's how i have it right now |
| 05:25.18 | carrar | http://tinyurl.com/qjko8n |
| 05:25.45 | kaldemar | dshap: is that really all in your sip.conf? |
| 05:26.14 | dshap | with username & password replaced |
| 05:26.17 | dshap | that's all i've got |
| 05:26.26 | dshap | all i'm trying to do is get my server registered with my SIP provider |
| 05:26.37 | dshap | im not using any analog/softphone stuff |
| 05:26.41 | dshap | just pure VoIP SIP |
| 05:27.31 | kaldemar | you're missing a bunch of needed configuration options. |
| 05:28.01 | dshap | well jeez, that's the file that voip.ms told me to use so they must be wrong =\ |
| 05:28.22 | *** join/#asterisk Subdolus (n=subby@subby.afraid.org) |
| 05:28.23 | dshap | what do i need to add? |
| 05:28.33 | kaldemar | they have only sent you the relevant parts to their service. |
| 05:28.35 | carrar | They probably mean "Add this too your existing sip.conf" |
| 05:28.57 | dshap | damn |
| 05:29.05 | dshap | i added qualify=yes |
| 05:29.16 | dshap | and now i got somethign different before the timeout |
| 05:29.20 | kaldemar | start with the parameters in the quick guide that infobot threw in here. |
| 05:29.31 | dshap | sip_poke_noanswer: Peer 'voipms' is now UNREACHABLE! |
| 05:29.49 | dshap | ok im opening up that link right now |
| 05:29.52 | kaldemar | and then look into asterisk's sample sip.conf and familiarize yourself with all the options in there. |
| 05:32.02 | dshap | at first glance of the sample i have a feeling that there is a TON of stuff in there i'm not gonna understand and or need to get this thing connected to my SIP provider |
| 05:32.05 | kaldemar | start with moving the nat=yes under [general] and add externip or externhost as the guide says. |
| 05:32.10 | dshap | isn't sip.conf like a one-time deal? |
| 05:32.19 | dshap | okay doing that now |
| 05:32.54 | kaldemar | it's a one-time deal if you set _everything_ you need at once. |
| 05:33.04 | dshap | understood |
| 05:33.11 | kaldemar | which is not going to happen for you now. |
| 05:33.58 | dshap | just for clarification |
| 05:34.01 | dshap | i'm a SIP client |
| 05:34.03 | dshap | not a SIP server |
| 05:34.04 | dshap | right? |
| 05:35.07 | securevoip | Asterisk is designed as a B2BUA (http://en.wikipedia.org/wiki/B2BUA) |
| 05:36.50 | dshap | ok yea i read that in the eBook |
| 05:38.00 | dshap | kaldemar, im not sure where externip or externhost are discussed in the "guide" |
| 05:38.41 | dshap | is externip supposed to be the IP address of my router? |
| 05:38.59 | dshap | the one i get if i go to www.whatismyip.com ? |
| 05:42.00 | dshap | sorry |
| 05:42.05 | dshap | i was looking at the wrong link |
| 05:42.07 | dshap | the wiki link |
| 05:42.09 | dshap | im dumb |
| 05:44.24 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-dc73dbee38f639bb) |
| 05:44.53 | Corydon76-dig | -+ |
| 05:45.52 | dshap | ok |
| 05:45.57 | dshap | read that guide/sample |
| 05:46.06 | dshap | my situation i don't think is exactly the same |
| 05:46.14 | dshap | i'm behind a NAT, the people i'm trying to connect to are not |
| 05:46.19 | dshap | and they are not a dynamic host |
| 05:46.24 | dshap | i moved nat=yes up to general |
| 05:46.28 | dshap | and i also added externip |
| 05:47.08 | dshap | im not getting any local callers so i didnt put localip |
| 05:47.14 | dshap | er, localnet |
| 05:47.52 | dshap | it seems as though i wouldn't need to use qualify since the remote host i am connecting to is not behind a NAT |
| 05:48.07 | dshap | either way, registration is still timing out =\ |
| 05:52.20 | [TK]D-Fender | dshap: And nowhere would I assume that you put any of that in the right place... |
| 05:52.25 | [TK]D-Fender | ~sipnat |
| 05:52.26 | infobot | [~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 |
| 05:52.28 | [TK]D-Fender | dshap: ^^^^ |
| 05:54.21 | *** join/#asterisk lanning (n=lanning@173.8.187.197) |
| 05:54.37 | dshap | pretty sure i do have it in the right place |
| 05:54.53 | dshap | nat=yes, externip=myIP,canreinvite=no |
| 05:55.03 | dshap | that's all at the top under [general] |
| 05:55.09 | dshap | then i have [voipms] which is my "A" |
| 05:55.17 | dshap | and under that i have everything else |
| 05:56.43 | [TK]D-Fender | dshap: ... |
| 05:56.45 | [TK]D-Fender | ~pb |
| 05:56.46 | infobot | [~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/ |
| 05:56.47 | [TK]D-Fender | ^^^^^^^^^^ |
| 05:57.40 | SunnyDP | has anyone of you been succesfull at setting DHCP option 120 (SIP Servers) in Windows Server DHCP ? |
| 05:57.45 | dshap | just posted it under the name "dshap" |
| 05:57.47 | dshap | it's up on pastebin |
| 05:58.38 | carrar | sunny, http://tinyurl.com/qjko8n |
| 05:58.43 | [TK]D-Fender | dshap: WHICH ONE?! provide the LINK |
| 05:59.09 | carrar | I'm thinking the 6th one down |
| 05:59.13 | dshap | http://pastebin.com/db90af5c |
| 06:00.07 | [TK]D-Fender | dshap: Location counts. Your REGISTER *ends** the rest of general and those settings you think are part of it are ignored |
| 06:00.26 | [TK]D-Fender | dshap: REGISTER's have to come AFTER everything else under pgeneral] |
| 06:01.15 | carrar | Synny, option 120 needs to be type "ip-address" |
| 06:02.01 | SunnyDP | carrar: array or not? have you been succesful at settimg up under windows server ? |
| 06:02.34 | carrar | You can use "array of ip-address" |
| 06:02.37 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
| 06:02.38 | *** join/#asterisk trentcreek (n=kvirc@200.94.231.94) |
| 06:02.41 | carrar | if you have more then 1 IP |
| 06:03.05 | SunnyDP | carrar: ahhh ok, i get it, did not know what array was... |
| 06:03.06 | carrar | however make sure array's of IP is support for that field |
| 06:03.18 | SunnyDP | ok great |
| 06:03.24 | carrar | start with "ip-address" |
| 06:03.29 | carrar | once that works |
| 06:03.38 | carrar | try a array if you have more then 1 IP |
| 06:03.49 | dshap | D-Fender: just moved my register statement to be the last line under [general], it is still tming ou |
| 06:04.03 | dshap | timing out* |
| 06:05.04 | [TK]D-Fender | dshap: And I don't see new configs, no call debug, and what do you have forwarded to your server? |
| 06:05.48 | dshap | basically i signed up for this SIP trunking origination/termination service at www.voip.ms |
| 06:05.53 | dshap | i ordered a DID |
| 06:06.04 | dshap | when I call the DID, the call is supposed to be forwarded to my server via SIP |
| 06:06.19 | [TK]D-Fender | dshap: not what I asked. what PORTS are forward to your server? |
| 06:06.24 | carrar | hahah |
| 06:06.27 | dshap | sorry |
| 06:06.27 | dshap | haha |
| 06:06.31 | dshap | okay |
| 06:06.46 | dshap | in linux i typed "ifconfig" |
| 06:06.52 | dshap | to get my server's local IP |
| 06:06.57 | dshap | which is 192.168.2.12 |
| 06:07.01 | dshap | then i went to my router setup page |
| 06:07.09 | dshap | and set up a DMZ for 192.168.2.12 |
| 06:07.14 | dshap | which i believe forwards ALL ports |
| 06:07.35 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 06:07.38 | dshap | i was trying to eliminate that as a possible issue for why i have not been able to get this working |
| 06:07.40 | [TK]D-Fender | dshap: Overkill, but OK. Now show me your NEW configs, and some SIP debug |
| 06:07.59 | dshap | new configs on the way, but how do i pull up the SIP debug? |
| 06:08.11 | [TK]D-Fender | dshap: Configs first then I'll grill for the rest |
| 06:08.21 | carrar | and grill he will!! |
| 06:09.14 | [TK]D-Fender | has 3 cooking modes : raw, rare, and burnt to a cinder |
| 06:10.03 | carrar | Then Sushi is a great fit |
| 06:10.03 | dshap | i am prepared to be grilled. my new config is the one i just sent with the register line moved - i am generating another pastebin right now |
| 06:11.03 | dshap | new sip.conf: http://pastebin.com/db917ddd |
| 06:12.32 | [TK]D-Fender | dshap: Also if you read the guide you'd know your [voipms] section has to be "nat=no" |
| 06:12.55 | dshap | *sigh*, i swear to you i read the guide but i just missed that =\ |
| 06:13.03 | dshap | ill update that now |
| 06:13.10 | [TK]D-Fender | dshap: Fix this, and enable SIP DEBUG at * CLI and show me your registration attempt |
| 06:14.38 | dshap | ok now with SIP DEBUG there is a lot of stuff on the screen |
| 06:14.42 | dshap | should i just paste it all to pastebin? |
| 06:14.48 | [TK]D-Fender | YES |
| 06:16.26 | dshap | SIP debug: http://pastebin.com/d11d99244 |
| 06:16.33 | dshap | that has my username on it but whatever, i dont care |
| 06:16.38 | dshap | i just want to get this thing working! |
| 06:16.42 | dshap | thank you very much for your help, by the way |
| 06:16.44 | dshap | i appreciate it |
| 06:16.55 | [TK]D-Fender | dshap: pastebi your exact sip.conf masking ONLY passwords |
| 06:17.08 | dshap | coming right up |
| 06:17.16 | dshap | ill put *** for the passwords |
| 06:17.38 | [TK]D-Fender | dshap: fine |
| 06:18.53 | *** join/#asterisk ctp (n=ctp@brsg-d9beed54.pool.mediaWays.net) |
| 06:19.14 | dshap | http://pastebin.com/d590164a0 |
| 06:19.22 | dshap | only passwords masked, everything else untouched ^^ |
| 06:19.44 | [TK]D-Fender | dshap: AH... |
| 06:19.56 | [TK]D-Fender | dshap: You didn't specify your LOCALNET <---------- |
| 06:20.09 | [TK]D-Fender | dshap: You clearly have real trouble reading directions |
| 06:20.11 | dshap | i read in the guide that this is only for local calls |
| 06:20.15 | dshap | no no no |
| 06:20.15 | dshap | i read it |
| 06:20.19 | dshap | i guess i just misunderstood it |
| 06:20.24 | dshap | i thought that because i'm getting a call from voip.ms |
| 06:20.27 | dshap | that it would NOT be a local call |
| 06:20.34 | [TK]D-Fender | dshap: How the &#^$ is * supposed to know what COUNTS as local if you don't tell it? |
| 06:20.49 | [TK]D-Fender | dshap: things taht are NOT local get the externip. |
| 06:21.09 | dshap | i guess i just thought that if i left it out then it would just consider everything at not local =\ |
| 06:21.10 | [TK]D-Fender | dshap: Since you didn't define your local subnet range it isn't SENDING the WAN IP for your contac: |
| 06:21.12 | dshap | ok im adding localnetright now |
| 06:21.29 | [TK]D-Fender | dshap: Contact: <sip:s@192.168.2.12> <--------- |
| 06:22.35 | *** join/#asterisk MrNaz (n=mrnaz@203.214.68.222) |
| 06:22.37 | dshap | should i add |
| 06:22.42 | dshap | localnet=192.168.2.12 |
| 06:22.46 | dshap | what about the /24 do i need that? |
| 06:22.52 | [TK]D-Fender | YES |
| 06:23.16 | [TK]D-Fender | and its "localnet = 192.168.2.0/24" |
| 06:23.23 | [TK]D-Fender | not .12 |
| 06:23.41 | dshap | ugh |
| 06:23.48 | dshap | got it |
| 06:23.53 | dshap | that makes more sense |
| 06:24.03 | [TK]D-Fender | reaches for his ClueBat (tm) |
| 06:27.42 | dshap | still timing out. new config @ http://pastebin.com/d54edf741 and new SIP debug @ http://pastebin.com/d20b49521 |
| 06:28.37 | carrar | nothing is getting in |
| 06:28.59 | carrar | possibly nothing getting out |
| 06:29.04 | carrar | check your network |
| 06:29.15 | *** join/#asterisk xrmx__ (n=rm@host119-200-dynamic.180-80-r.retail.telecomitalia.it) |
| 06:29.25 | dshap | what should i do to check it |
| 06:29.27 | [TK]D-Fender | dshap: Check the firewall on you * server |
| 06:30.14 | dshap | it is enabled! |
| 06:30.15 | dshap | damnnnnnn |
| 06:30.22 | [TK]D-Fender | reaches for his ClueBat (tm) |
| 06:30.32 | carrar | reaches for TK's ClueBat also |
| 06:30.32 | dshap | 1 sec |
| 06:30.36 | [TK]D-Fender | bludgeons the McFuck out of dshap |
| 06:30.48 | carrar | waits in line |
| 06:31.16 | [TK]D-Fender | hands carrar a STRAW to finish off the pulp he's left with |
| 06:31.26 | carrar | haha |
| 06:31.53 | [TK]D-Fender | 2 words : chunky fucking salsa. |
| 06:32.26 | dshap | ok im using CentOS 5.3/KDE which i'm extremely new to...i went to Administration --> Security Level and Firewall --> Firewall Options --> I clicked disable |
| 06:32.38 | [TK]D-Fender | dshap: "iptables --flush" |
| 06:32.39 | dshap | what about SELinux...should I make that Disabled as well? |
| 06:32.55 | dshap | right now SELinux is "Enforcing" |
| 06:32.56 | [TK]D-Fender | FFS we'd better not be dealing with that TOO |
| 06:33.03 | [TK]D-Fender | ARGHHHHHHHHHHHHHHHHHHHHHHHHHHH |
| 06:33.33 | [TK]D-Fender | ok, I'm done for the night. |
| 06:33.40 | dshap | ahhh |
| 06:33.42 | dshap | i disabled everything |
| 06:33.44 | dshap | SELinux |
| 06:33.45 | dshap | Firewall |
| 06:33.47 | [TK]D-Fender | dshap: Your configs are better now, fix your box. |
| 06:33.49 | dshap | i did "iptables --flush" |
| 06:33.57 | [TK]D-Fender | is off |
| 06:34.02 | dshap | what the hell could be wrong with it? the only thing i did was install asterisk |
| 06:34.31 | dshap | sigh |
| 06:34.52 | carrar | haha |
| 06:35.10 | drmessano | yum remove iptables |
| 06:35.18 | dshap | what does that do? |
| 06:35.32 | drmessano | It buys goats to eat your lawn |
| 06:35.36 | drmessano | Come on now |
| 06:36.04 | dshap | i just dont know what "iptables" is |
| 06:36.16 | dshap | obviously yum remove uninstalls it |
| 06:36.21 | drmessano | iptables is the firewall thats fucking your shit up |
| 06:36.26 | dshap | well fuck |
| 06:36.28 | dshap | ok |
| 06:36.35 | drmessano | Kick that bitch out, shes fucking your brother |
| 06:36.44 | drmessano | Then move on |
| 06:36.53 | trentcreek | I got IP tables on my box |
| 06:37.04 | dshap | and asterisk still works? |
| 06:37.13 | dshap | (iptables in the process of being kicked out |
| 06:37.14 | trentcreek | yes |
| 06:37.15 | dshap | ) |
| 06:37.38 | dshap | to be honest i just want to get my box up and running so i can start to learn the cool stuff |
| 06:37.43 | dshap | ill worry about security later on |
| 06:37.45 | drmessano | Then help him config it, trentcreek |
| 06:37.50 | dshap | no no |
| 06:37.52 | dshap | im kickin the bitch out |
| 06:37.52 | dshap | haha |
| 06:37.58 | dshap | she's DONEZO |
| 06:38.16 | drmessano | Change the locks, keep her stereo.. fo shizzle |
| 06:38.35 | trentcreek | iptables -P INPUT ACCEPT |
| 06:38.53 | trentcreek | iptables -P OUTPUT ACCEPT |
| 06:39.06 | trentcreek | iptables -P FORWARD ACCEPT |
| 06:39.15 | trentcreek | iptables -F |
| 06:39.18 | drmessano | So keep iptables, just make it wide open? |
| 06:39.27 | trentcreek | iptables -X |
| 06:39.30 | trentcreek | hehe |
| 06:39.32 | drmessano | NO POINT.. remove that shit |
| 06:40.18 | drmessano | Thats like buying a $500 security door and putting a door stopper on it to keep it open all the time |
| 06:40.38 | trentcreek | hey..good idea |
| 06:41.04 | carrar | open root policy |
| 06:41.32 | dshap | ok |
| 06:41.36 | dshap | is there anythign i need to do |
| 06:41.39 | dshap | to refresh or whatever |
| 06:41.41 | dshap | after removing iptables |
| 06:41.46 | dshap | or can i just go right back into asterisk |
| 06:41.49 | dshap | and fire it up |
| 06:42.16 | drmessano | is it 4:20 again? |
| 06:42.32 | drmessano | dshap is firin one up |
| 06:42.35 | dshap | REGISTRATION TIMED OUT |
| 06:42.36 | dshap | :( |
| 06:42.41 | dshap | fuckkkk this shit |
| 06:42.45 | dshap | iptables gone |
| 06:42.46 | drmessano | Is asterisk running |
| 06:42.48 | dshap | no firewall |
| 06:42.54 | dshap | YES |
| 06:42.59 | dshap | i got "Asteisk Ready" |
| 06:43.11 | dshap | followed by *CLI> |
| 06:43.25 | dshap | and then after 10 seconds or so i start getting the timeout messages |
| 06:43.44 | drmessano | This is where I would get drawn and quartered for recommending a nice GUI based installed |
| 06:43.47 | carrar | cause A) nothing is getting out of your network or B) nothing is getting in |
| 06:43.48 | drmessano | Like AsteriskNOW |
| 06:43.52 | drmessano | ducks |
| 06:44.35 | carrar | You should update SSH on your Asterisk box also |
| 06:44.47 | carrar | 4.3 is old |
| 06:44.59 | dshap | 'yum update ssh' ? |
| 06:46.21 | dshap | how might i verify that i have a network problem by using something other than asterisk? |
| 06:46.58 | drmessano | openssh-server.i386 4.3p2-29.el5 |
| 06:47.03 | drmessano | Thats current |
| 06:48.58 | dshap | what do you guys use asterisk for? if you don't mind me asking |
| 06:49.08 | trentcreek | washing dishes |
| 06:49.17 | trentcreek | making cheap calls |
| 06:49.46 | drmessano | making calls |
| 06:49.58 | trentcreek | set up one server and hand out ATAs to everyone for free calls |
| 06:50.11 | dshap | ATA? |
| 06:50.20 | carrar | dshap |
| 06:50.26 | carrar | I can't reach your SIP port |
| 06:50.30 | carrar | You have a network issue |
| 06:50.40 | carrar | PORT STATE SERVICE |
| 06:50.41 | carrar | 5060/udp closed sip |
| 06:50.43 | drmessano | Asterisk gives me an inexpensive PBX solution I can run on modest hardware, and also allows me access to VoIP service providers who offer much cheaper phone service |
| 06:50.56 | trentcreek | omfg there is no tilde on this KB |
| 06:50.59 | dshap | so maybe the DMZ isn't working |
| 06:51.06 | *** join/#asterisk grEvenX (n=even@apb9hb.ip.ssc.net) |
| 06:51.11 | carrar | will it IS working for TCP |
| 06:51.11 | drmessano | DMZ sucks |
| 06:51.18 | carrar | but not for UDP appearently |
| 06:51.20 | drmessano | Its a half ass solution |
| 06:51.21 | dshap | hm |
| 06:51.24 | dshap | ok |
| 06:51.27 | dshap | 1 seclet mesee |
| 06:51.34 | drmessano | DMZ on SOHO routers is often "kinda DMZ" |
| 06:51.38 | drmessano | Open some ports |
| 06:51.44 | drmessano | 5060, 10000-20000 |
| 06:51.49 | drmessano | We'll wait |
| 06:51.57 | dshap | would i put 5060 for both inbound port and private port? |
| 06:52.06 | drmessano | yup |
| 06:52.08 | trentcreek | bot: ata |
| 06:52.11 | drmessano | What kind of router is this? |
| 06:52.11 | carrar | for now |
| 06:52.13 | dshap | belkin |
| 06:52.14 | carrar | open everything |
| 06:52.14 | trentcreek | bot ATA |
| 06:52.17 | trentcreek | jbot |
| 06:52.17 | drmessano | Oh my god |
| 06:52.21 | drmessano | A belkin? |
| 06:52.22 | carrar | all UDP |
| 06:52.32 | dshap | yea why |
| 06:52.32 | drmessano | All yours, carrar |
| 06:52.35 | carrar | s/open/forward/ |
| 06:52.38 | carrar | haha |
| 06:52.38 | dshap | haha |
| 06:52.47 | dshap | i havent had issues with port forwarding before |
| 06:52.49 | drmessano | I hate Belkin |
| 06:52.52 | drmessano | s/belkin/shit/ |
| 06:53.07 | carrar | 5060 UDP |
| 06:53.09 | trentcreek | ~ATA |
| 06:53.10 | infobot | well, ata is Analogue Terminal Adapter which provides an FXS and/or FXO and ethernet, see http://www.voip-info.org/wiki/view/ATA |
| 06:53.18 | carrar | not TCP |
| 06:53.24 | dshap | just forwarded 5060 UDP |
| 06:53.25 | dshap | check it |
| 06:53.28 | dshap | however u were checking it |
| 06:53.47 | carrar | nope |
| 06:54.05 | trentcreek | may want to install fail2ban |
| 06:54.22 | trentcreek | yum -y install fail2ban |
| 06:54.40 | trentcreek | ~fail2ban |
| 06:54.41 | infobot | it has been said that fail2ban is a program to ban people using iptables based on information in logs: http://www.voip-info.org/wiki/view/Fail2Ban+(with+iptables)+And+Asterisk |
| 06:55.02 | dshap | y would i need to ban people? |
| 06:55.04 | trentcreek | oops..need IP tables |
| 06:55.24 | trentcreek | people that keep trying to log into your system with scripts |
| 06:55.37 | drmessano | He hasnt even gotten his firewall working.. Please fuck off with the fail2ban crap |
| 06:55.44 | dshap | carrar, you're tellin me my router is fucked |
| 06:55.48 | dshap | is that correct? |
| 06:56.01 | carrar | I'm just saying I can't reach your UDP/5060 |
| 06:56.02 | dshap | you are somehow able to check if port 5060 (required for SIP) is open on my router |
| 06:56.07 | dshap | and it's showing that it is closed? |
| 06:56.20 | carrar | nmap -sU -p 5060 68.231.218.208 |
| 06:56.26 | carrar | PORT STATE SERVICE |
| 06:56.27 | carrar | 5060/udp closed sip |
| 06:56.39 | carrar | one that works should display: |
| 06:56.40 | carrar | PORT STATE SERVICE |
| 06:56.40 | carrar | 5060/udp open|filtered sip |
| 06:57.01 | carrar | for instance |
| 06:57.02 | carrar | nmap -sU -p 5060 sip.us3b.voip.ms |
| 06:57.07 | carrar | PORT STATE SERVICE |
| 06:57.07 | carrar | 5060/udp open|filtered sip |
| 06:57.26 | carrar | You however are unreachable |
| 06:57.34 | dshap | so that is definitely my problem |
| 06:57.37 | carrar | yes |
| 06:57.54 | dshap | could it be a linux issue? |
| 06:57.57 | dshap | on my machine |
| 06:58.09 | carrar | doublefully is you have done: iptables -F |
| 06:58.17 | dshap | i have the router set to forward port 5060 UDP --> 192.168.2.12 |
| 06:58.29 | dshap | i dont have iptables |
| 06:59.06 | dshap | i just changed the local IP on the port forwarding |
| 06:59.10 | dshap | to forward to another computer on my network |
| 06:59.18 | dshap | try to reach UDP 5060 1 more time please? |
| 06:59.19 | carrar | Just for fun |
| 06:59.24 | carrar | add this to the top of your sip.conf |
| 06:59.24 | carrar | bindport=5060 |
| 06:59.24 | carrar | bindaddr=0.0.0.0 |
| 06:59.34 | dshap | above [general] ? |
| 06:59.41 | carrar | just below |
| 06:59.51 | carrar | and restart * |
| 06:59.58 | carrar | "restart now" |
| 07:00.07 | dshap | it's not currently running |
| 07:00.12 | dshap | hasn't been for a while |
| 07:01.00 | dshap | but ok i just did that |
| 07:01.03 | dshap | now starting asterisk again |
| 07:01.07 | carrar | there you go |
| 07:01.10 | carrar | thats it |
| 07:01.14 | carrar | PORT STATE SERVICE |
| 07:01.14 | dshap | ? |
| 07:01.14 | carrar | 5060/udp open|filtered sip |
| 07:01.25 | *** join/#asterisk grndslm (n=grndslm@96.19.110.120) |
| 07:01.38 | dshap | haven't gotten the timeout message yet... |
| 07:02.05 | dshap | what asterisk command do i type to see if i am successfully registered? |
| 07:02.37 | carrar | sip show reg? |
| 07:02.46 | dshap | it worked! |
| 07:02.51 | dshap | it's showing up on my control panel website for voip.ms |
| 07:02.52 | dshap | my IP |
| 07:02.55 | dshap | it registered |
| 07:02.56 | dshap | omg |
| 07:03.03 | dshap | thank you so much dude |
| 07:03.08 | dshap | what was the problem? |
| 07:03.12 | carrar | omg you FAILED to read the instructions :) |
| 07:03.21 | dshap | WHERE in the instructions did i fail to read? |
| 07:03.34 | carrar | Instead of deleting everything in the example sip.conf |
| 07:03.46 | carrar | try only removing things you know you don't want |
| 07:04.09 | carrar | start by removing eveything that is commented out |
| 07:04.12 | dshap | my stupid SIP provider made it seem to me like i just needed their file |
| 07:04.14 | dshap | which was obviously wrong |
| 07:04.44 | dshap | ok wow |
| 07:04.47 | dshap | i have a lot of reading to do |
| 07:04.51 | dshap | so now that i'm registered |
| 07:04.53 | dshap | i can call my DID |
| 07:05.03 | dshap | and it will execute the dialplan in my extensions.conf? |
| 07:05.03 | carrar | well |
| 07:05.08 | carrar | probably not |
| 07:05.13 | dshap | y not |
| 07:05.17 | carrar | since you don't have any SIP phones registered |
| 07:05.45 | dshap | why do i need a SIP phone? |
| 07:05.47 | carrar | unless you just answer it and play a file |
| 07:06.03 | carrar | Where is the call gonna go? |
| 07:06.08 | carrar | once it hits your * box |
| 07:06.11 | dshap | what if i want to answer it, play a file, and then transfer the call to another number on the PSTN? |
| 07:06.22 | carrar | sure |
| 07:06.24 | dshap | (via my SIP provider) |
| 07:06.24 | carrar | can do that |
| 07:06.28 | carrar | send it back out |
| 07:06.31 | dshap | does that require further editing of the sip.conf file? |
| 07:06.38 | trentcreek | dshap: That is what google voice is for |
| 07:06.44 | carrar | hopefully not |
| 07:06.47 | dshap | hah |
| 07:06.53 | dshap | trent i'm fully aware of what google voice does |
| 07:06.56 | dshap | i have a different plan :-p |
| 07:07.47 | dshap | my goal for now is to have many different audio files stored on my server |
| 07:08.01 | trentcreek | http://en.wikipedia.org/wiki/index.html?curid=20887118 |
| 07:08.02 | dshap | when someone calls the server, i want it to look up their phone number in a MySQL database |
| 07:08.08 | dshap | and then play the appropriate file |
| 07:08.11 | carrar | no |
| 07:08.17 | carrar | don't use MySQL / Oracle |
| 07:08.21 | dshap | why not |
| 07:08.23 | carrar | use PostgreSQL |
| 07:08.40 | dshap | i read that you can use PHP to interact with AGI |
| 07:08.44 | carrar | yes |
| 07:08.49 | carrar | db doesn't matter |
| 07:08.55 | dshap | and i already have experience doing PHP/MySQL development |
| 07:08.56 | dshap | for web apps |
| 07:08.57 | carrar | but PostgreSQL is better |
| 07:09.02 | dshap | how come |
| 07:09.11 | carrar | PHP with PostgreSQL is the same |
| 07:09.11 | Nugget | oh man, that's the understatement of the century. |
| 07:09.19 | dshap | haha yea? |
| 07:09.29 | dshap | well i could probably learn the PostgreSQL functions in PHP |
| 07:09.36 | carrar | you should |
| 07:09.39 | dshap | is it free? |
| 07:09.41 | Nugget | agrees |
| 07:09.41 | carrar | yes |
| 07:09.45 | Nugget | it's free-er than mysql. |
| 07:09.48 | carrar | yes yes |
| 07:09.56 | dshap | does it have like an easy-to-use GUI for managing tables and stuff? |
| 07:10.01 | carrar | yes |
| 07:10.10 | carrar | PgAdminIII |
| 07:10.12 | carrar | nice gui |
| 07:10.14 | dshap | is it the same relational database setup like tables/columns |
| 07:10.16 | *** join/#asterisk SebastianS (n=schu@dsl-static-111.212-5-200.telecom.sk) |
| 07:10.17 | carrar | for the CLI Impaired |
| 07:10.24 | dshap | and similar SQL syntax for interacting? |
| 07:10.27 | carrar | yes |
| 07:10.31 | carrar | SQL is SQL |
| 07:10.37 | carrar | only better |
| 07:10.38 | dshap | ok well you guys seem like you know what you are talking about |
| 07:10.41 | carrar | on POstgreSQL |
| 07:10.44 | dshap | i will definitely look into PostgreSQL |
| 07:11.00 | Nugget | postgresql also has much better documentation |
| 07:11.12 | drmessano | carrar has me convinved PGSQL is the way to go |
| 07:11.17 | drmessano | convinced |
| 07:11.20 | drmessano | Damn carrar |
| 07:11.22 | carrar | heh |
| 07:11.32 | dshap | let's say i want someone to call my asterisk server |
| 07:11.33 | carrar | I've used both MySQL and POstgreSQL |
| 07:11.40 | dshap | and then i want to connect them to another number on the PSTN |
| 07:11.48 | dshap | will that call be routed through my asterisk server at all times? |
| 07:11.56 | dshap | such that i will be incurring both origination and termination costs? |
| 07:11.58 | drmessano | MySQL is nice.. just pisses me off how bad it sucks.. otherwise, its nice |
| 07:12.10 | carrar | dshap, thats up to your sip carrier |
| 07:12.18 | carrar | how many SIP Channels they let you have |
| 07:12.24 | carrar | if there is a limit |
| 07:12.24 | dshap | unlimited they say |
| 07:12.35 | carrar | then you probably pay a per min charge? |
| 07:12.35 | dshap | but there is a different rate for origination & termination |
| 07:12.35 | *** join/#asterisk NetEcho (n=NetEcho@unaffiliated/netecho) |
| 07:12.37 | dshap | yes |
| 07:12.41 | dshap | $1/month for the DID |
| 07:12.46 | dshap | and $0.01 per minute |
| 07:12.47 | carrar | You're golden then |
| 07:12.50 | dshap | with a 6-seond billing increment |
| 07:12.54 | dshap | seemed like the best deal i could find |
| 07:13.14 | dshap | right but if i'm doing termination & origination |
| 07:13.19 | dshap | it would be $0.02 per minute |
| 07:13.30 | carrar | yeah |
| 07:13.32 | dshap | since they are calling the server and the server is calling the person they are talking to |
| 07:13.49 | dshap | is there no way for my asterisk server to set up a completely separate connection between 2 PSTN lines |
| 07:13.49 | carrar | well inbound "should" be free |
| 07:13.56 | carrar | unless it's a TOLLFREE DID |
| 07:13.57 | NetEcho | hey guys, question you may be able to answer, I saw an old Systm episode where they set up asterisk using a SIPTURA box to convert the PSTN line to voip, but I just found out cisco bought SIPTURA, so I was wondering where I might be able to get an affordable SIP adapter to convert my PSTN line to VoIP |
| 07:14.03 | dshap | nah it's def not free |
| 07:14.06 | dshap | and it's not a tollfree DID |
| 07:14.58 | carrar | "affordable SIP adapter" is relative |
| 07:15.16 | dshap | hey carrar and everyone else - thanks for your help |
| 07:15.17 | carrar | AudioCodes makes nice FXS/FXO devices |
| 07:15.20 | dshap | i'm glad i got this up and running |
| 07:15.22 | NetEcho | well not like digium's $1000+ adapters |
| 07:15.22 | dshap | im gonna get to bed |
| 07:15.39 | carrar | TDM400? |
| 07:15.44 | carrar | they aren't $1k |
| 07:15.50 | carrar | they work nice |
| 07:16.01 | dshap | thanks again |
| 07:16.02 | dshap | bye |
| 07:16.04 | carrar | np |
| 07:16.06 | *** part/#asterisk dshap (n=IceChat7@ip68-231-218-208.oc.oc.cox.net) |
| 07:16.14 | carrar | ok maybe that was a problem ;) |
| 07:17.21 | NetEcho | ideally I'd need a setup that takes my PSTN line and pipes it into the asterisk system and to 1 or 2 standard non-voip phones, that I can route calls to and then the rest of the phones would probably be Cisco voip phones over a router |
| 07:18.00 | carrar | So a TDM400 with 1 FXO and 2 FXS ports |
| 07:18.09 | NetEcho | the normal non-voip phones are my cordless ones, unless there are cordless voip phones now |
| 07:18.16 | carrar | oh |
| 07:18.19 | carrar | none standard voip |
| 07:18.23 | carrar | but still voip |
| 07:18.26 | carrar | not analog |
| 07:18.34 | carrar | So a TDM400 with 1 FXO |
| 07:19.33 | NetEcho | can only find the TDM410 |
| 07:19.46 | NetEcho | FXO is for the PSTN line right? |
| 07:19.57 | carrar | yes |
| 07:20.10 | NetEcho | k |
| 07:20.50 | carrar | think of O as connects to the central Office |
| 07:21.02 | carrar | the local CO |
| 07:21.24 | NetEcho | what I eventually plan to have set up is to make it so the phone only rings in rooms that people are in and add in silent mode that silences phones when you don't want them to ring and if all phones are set to silent it automaticly goes to voicemail |
| 07:21.27 | carrar | FXS, connects to a Station aka analog phone in your house |
| 07:22.01 | carrar | FXS provides dialtone |
| 07:22.07 | carrar | FXO accepts dialtone |
| 07:22.18 | NetEcho | ah |
| 07:22.51 | NetEcho | yea it would seem they don't sell the TDM400 anymore |
| 07:23.13 | carrar | lies |
| 07:23.25 | carrar | http://www.digium.com/en/products/analog/ |
| 07:23.30 | carrar | they just call them something different |
| 07:23.36 | carrar | 410 |
| 07:23.46 | NetEcho | $500 and up wow |
| 07:24.17 | *** join/#asterisk botox93 (n=botox93@213.221.82.242) |
| 07:24.36 | NetEcho | thats more expensive than the computer lol |
| 07:26.10 | NetEcho | hrm I just stumbled upon openvox |
| 07:26.14 | *** join/#asterisk |Cybex| (n=John@80.100.126.176) |
| 07:26.42 | NetEcho | fully compatible with Digium hardware aparently |
| 07:27.29 | NetEcho | hrm |
| 07:27.49 | NetEcho | couldn't asterisk be set up to use a computer modem ? |
| 07:28.51 | carrar | X100P if you want to do it cheap and have issues |
| 07:28.58 | drmessano | Modems are not voice quality hardware |
| 07:29.11 | carrar | find copies for like $7 on ebay |
| 07:29.25 | drmessano | So no one would bother beyond an effort made years ago thats still biting us in the ass |
| 07:29.47 | carrar | spend the money and do it right |
| 07:30.53 | NetEcho | I'd probably go with OpenVox A400P |
| 07:30.56 | carrar | here are some |
| 07:30.56 | carrar | http://www.voipsupply.com/atas/fxo-fxs/ |
| 07:31.07 | NetEcho | not the cheapest but not insanely expensive either |
| 07:31.17 | NetEcho | this is just a pet project of mine |
| 07:31.40 | *** join/#asterisk hastinapur (n=gomel77@mail3.sis.com.by) |
| 07:32.01 | carrar | stay away from grandstream anything |
| 07:32.22 | drmessano | Openvox isnt bad for a hobby card |
| 07:32.26 | NetEcho | well the A400P is apparently similar to the TDM400 |
| 07:32.32 | NetEcho | just a lot cheaper |
| 07:32.37 | drmessano | You at least get the feel of how a TDM card works |
| 07:32.40 | NetEcho | yea |
| 07:32.47 | drmessano | I wouldnt run a business on one |
| 07:32.55 | NetEcho | nah this is all for home use |
| 07:33.03 | *** join/#asterisk cjk_ (n=cjk@vodsl-8547.vo.lu) |
| 07:33.08 | carrar | How much is the A400P |
| 07:33.10 | NetEcho | once I get a feel for how everything works I can upgrade |
| 07:33.23 | NetEcho | on sale for $199.95 right now |
| 07:33.32 | drmessano | $170 incl. Shipping with 4 cards on ebay |
| 07:33.39 | drmessano | They're all over ebay |
| 07:33.40 | NetEcho | 4 cards holy crap |
| 07:33.40 | carrar | twince as much as the audiocodes 1FXO1FXS |
| 07:33.51 | drmessano | 1, 2, 3, and 4 |
| 07:33.55 | NetEcho | 2FXS 2 FXO |
| 07:34.03 | drmessano | 1 is like $90 |
| 07:34.26 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) |
| 07:34.28 | carrar | You want A400P01 |
| 07:34.33 | drmessano | carrar: Thats fully loaded for $170 incl shipping |
| 07:34.51 | NetEcho | the one I'm looking for allows me to change out the FXO and FXS modules with TDM modules later on |
| 07:34.55 | carrar | $99 |
| 07:35.00 | carrar | http://www.voiplink.com/OpenVox_A400P01_1_FXO_p/openvox-a400p01.htm |
| 07:35.13 | NetEcho | wait sorry |
| 07:35.19 | *** join/#asterisk sulex (n=sulex@pdpc/supporter/professional/sulex) |
| 07:35.37 | drmessano | http://cgi.ebay.com/Openvox-A400P-1FXS-FXO-Digium-Asterisk-Trixbox-TDM400_W0QQitemZ180315562192QQcmdZViewItemQQptZLH_DefaultDomain_0?hash=item29fba520d0&_trksid=p3286.c0.m14&_trkparms=72:1234|66:2|65:12|39:1|240:1318|301:1|293:1|294:50 |
| 07:35.39 | NetEcho | Digium's X100M/S100M <-- those any decent |
| 07:35.40 | *** join/#asterisk gr0mit (n=tim@extrt.txrx.org.uk) |
| 07:35.50 | drmessano | Thats just a MODULE |
| 07:35.54 | NetEcho | yea |
| 07:35.56 | carrar | woah $89 |
| 07:35.57 | NetEcho | are they good modules? |
| 07:35.57 | drmessano | Look at the link I posted |
| 07:36.00 | carrar | how about $79 |
| 07:36.09 | NetEcho | cause the card is swapable with them |
| 07:36.30 | carrar | I've recommend all I am gonna recommend |
| 07:36.32 | carrar | good night |
| 07:36.36 | NetEcho | ttyl |
| 07:36.41 | carrar | passes the tourch to drm |
| 07:37.06 | NetEcho | I got a perfect little mini-itx system that would handle this just nicely |
| 07:37.57 | NetEcho | brb |
| 07:47.29 | NetEcho | I gotta run, I'll do some research later |
| 07:47.35 | *** part/#asterisk NetEcho (n=NetEcho@unaffiliated/netecho) |
| 07:55.53 | *** join/#asterisk \void\ (n=void@fer227.internetdsl.tpnet.pl) |
| 07:56.46 | *** join/#asterisk scardinal (n=supreme@0905ds1-rdo.0.fullrate.dk) |
| 07:57.38 | *** join/#asterisk war9407 (i=war@liquidswords.org) |
| 07:58.52 | *** join/#asterisk joako (n=joako@opensuse/member/joak0) |
| 07:59.20 | joako | Anyone use egika softphone? I just tried and even though asterisk says answered I just hear continuous ringing |
| 08:17.22 | KyleK | ekiga works for me |
| 08:25.26 | *** join/#asterisk joobie (n=joobie@210-84-22-83.dyn.iinet.net.au) |
| 09:08.16 | *** join/#asterisk shareenergy (n=go@87.74.7.50) |
| 09:09.14 | shareenergy | hello guys anyone knows why mixmonitor does not save on the folder if i dont use exten => _X.,1,Answer() ? |
| 09:10.42 | shareenergy | exten => _X.,1,Answer() |
| 09:10.42 | shareenergy | exten => _X.,1,Dial(SIP/${EXTEN}|30) |
| 09:10.42 | shareenergy | exten => _X.,2,MixMonitor(in-${EXTEN}--${STRFTIME(,,%F-%T)}-${CALLERID(num)}.wav) |
| 09:10.42 | shareenergy | exten => _X.,3,Congestion |
| 09:10.42 | shareenergy | exten => _X.,102,Busy |
| 09:10.49 | shareenergy | this works without any problem |
| 09:17.30 | kaldemar | that definitely won't work without any problems, since you have two lines as priority 1. |
| 09:18.00 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 09:19.22 | kaldemar | put the MixMonitor before the dial and it'll work. |
| 09:25.09 | shareenergy | kaldemar I tried that |
| 09:25.12 | shareenergy | no result |
| 09:25.46 | shareenergy | how can i check if the call was answerd and then mixmonitor? |
| 09:26.50 | kaldemar | look at the CLI prints. but anyway, put the MixMonitor first. |
| 09:26.59 | plundra | I'm pretty new to pgAdmin, but the Explain tab, what's it for? :-) I would assume my explain-query would give me some nicely formatted stuff there. |
| 09:27.12 | plundra | bah, wrong channel :-P |
| 09:32.56 | shareenergy | kaldemar the problem is that I recording agents |
| 09:34.12 | *** join/#asterisk |Cybex| (n=John@80.100.126.176) |
| 09:35.15 | *** join/#asterisk botox93 (n=botox93@213.221.82.242) |
| 09:35.58 | shareenergy | kaldemar if i use answer |
| 09:36.04 | *** join/#asterisk botox93 (n=botox93@213.221.82.242) |
| 09:36.11 | shareenergy | my queues go crazy on the queuemetrix |
| 09:48.46 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
| 09:51.41 | *** join/#asterisk \void\ (n=void@fer227.internetdsl.tpnet.pl) |
| 10:03.14 | *** join/#asterisk |Cybex| (n=John@80.100.126.176) |
| 10:06.01 | *** join/#asterisk miloux (n=KVIrc@milu.rit.se) |
| 10:06.35 | miloux | can someone elaborate what hangupcause AST_CAUSE_INTERWORKING 127 means? |
| 10:06.47 | *** join/#asterisk MrNaz (n=mrnaz@203.214.68.222) |
| 10:10.15 | *** join/#asterisk |Cybex| (n=John@80.100.126.176) |
| 10:12.25 | *** join/#asterisk KermitTheFragger (n=KermitTh@118-197.bbned.dsl.internl.net) |
| 10:30.36 | *** join/#asterisk Takapa (i=vegard@junior.svanberg.no) |
| 10:39.06 | *** join/#asterisk nighty^ (n=nighty@x122091.ppp.asahi-net.or.jp) |
| 10:39.13 | kaldemar | miloux: that the call has ended and the remote end wasn't probably capable of sending an appropriate cause code. |
| 10:43.20 | *** join/#asterisk eject_ck (n=eject@85.223.182.86) |
| 10:44.16 | eject_ck | I'm writing extension I need use it for XXX or XXXX. what I need use instead last X in XXXX ? |
| 10:44.20 | eject_ck | exten => _0XXX,1,Dial(SIP/111,,WtrD(${EXTEN})) |
| 10:44.20 | eject_ck | exten => _0XXX,2,Hangup() |
| 10:44.55 | eject_ck | to be equal to |
| 10:44.55 | eject_ck | exten => _0XXX,1,Dial(SIP/111,,WtrD(${EXTEN})) |
| 10:44.55 | eject_ck | exten => _0XXX,2,Hangup() |
| 10:44.55 | eject_ck | exten => _0XXXX,1,Dial(SIP/111,,WtrD(${EXTEN})) |
| 10:44.55 | eject_ck | exten => _0XXXX,2,Hangup() |
| 10:45.37 | eject_ck | I mean write extension using one line without additional extension |
| 10:48.54 | *** join/#asterisk DarKnesS_WolF (n=wolf@unaffiliated/sherif) |
| 10:50.33 | kaldemar | eject_ck: there are only "one or more" or "zero or more" pattern characters. "." and "!" respectively. |
| 10:52.46 | miloux | kaldemar: thanks! |
| 10:53.14 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 10:53.24 | *** join/#asterisk jeff (i=jeff@unaffiliated/jeff) |
| 10:55.59 | eject_ck | kaldemar: how can I remove first zero form ${EXTEN} - e.g. I need use 0466 or 05433 and dial 466 and 5433 |
| 10:56.56 | kaldemar | eject_ck: ${EXTEN:1} will remove the first digit, whatever it is. |
| 10:57.14 | eject_ck | thanks! |
| 10:57.29 | eject_ck | so finally it's |
| 10:57.29 | eject_ck | exten => _0XXX!,1,Dial(SIP/111,,WtrD(${EXTEN:1})) |
| 10:57.29 | eject_ck | exten => _0XXX!,2,Hangup() |
| 10:57.32 | eject_ck | right / |
| 10:57.37 | eject_ck | ? |
| 10:59.03 | eject_ck | yes, thanks! |
| 11:02.54 | *** join/#asterisk HeMan (n=jimmy@ssh.southpole.se) |
| 11:03.44 | HeMan | Hi! is it possible to set up asterisk so anyone can call in, not just other registered sip phones? |
| 11:05.50 | kaldemar | eject_ck: yes, but keep in mind that it also matches everything that is longer than 4. |
| 11:06.06 | kaldemar | HeMan: yes it is. |
| 11:07.20 | \void\ | I am developing 3G gateway, I've build system according to sip.fontventa.com instructions, when I am making a videocall from mobile, asterisk hangs up on h324m_loopback() or h324m_gw_answer() for aprox. 10-15 seconds, and then hangs up (no video, no audio, no errors), anyone know this issue? |
| 11:07.54 | *** join/#asterisk ttl- (n=patrick@d5153AE82.access.telenet.be) |
| 11:08.11 | HeMan | kaldemar: I get "Failed to authenticate user " when I try, what do I miss? |
| 11:08.57 | \void\ | asterisk hangs* for aprox 10-15sec, and then hangs up |
| 11:10.57 | HeMan | what are those types of calls called? |
| 11:12.57 | *** join/#asterisk mikkel (n=mikkel@84.238.113.66) |
| 11:13.43 | kaldemar | HeMan: guest calls. allowguest=yes in sip.conf. |
| 11:14.46 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 11:17.34 | *** join/#asterisk pif (n=ldm@zenon.apartia.fr) |
| 11:20.20 | HeMan | kaldemar: an it should be in [general], right? |
| 11:22.12 | eject_ck | kaldemar: i need use . instead ! ?\ |
| 11:22.31 | eject_ck | where I can read doc about wildcard mask ? |
| 11:27.04 | \void\ | eject_ck, here http://www.voip-info.org/wiki/view/Asterisk+Dialplan+Patterns |
| 11:27.27 | HeMan | kaldemar: no difference with if I have yes or no to allowguest in [general] in sip.conf |
| 11:29.48 | *** join/#asterisk coppice (n=chatzill@204.196.17.210.dyn.pacific.net.hk) |
| 11:34.17 | kaldemar | eject_ck: if you use 0XXX., it only matches to numbers that are 4 or more digits long: http://www.voip-info.org/wiki/view/Asterisk+Dialplan+Patterns |
| 11:34.25 | *** join/#asterisk DarkRift (n=dark@65.92.169.102) |
| 11:34.35 | *** join/#asterisk ingenius (n=alektro@186.136.12.198) |
| 11:34.57 | kaldemar | HeMan: are you calling with a user that matches a device? show a cli output of a call with sip debug enabled. |
| 11:35.59 | kaldemar | HeMan: if you want to prevent all authentication, use insecure=port,invite. |
| 11:36.22 | HeMan | kaldemar: no, I gave it a name that I'm sure is not used |
| 11:36.40 | HeMan | kaldemar: could there be security implications to do that? |
| 11:37.09 | kaldemar | no calls are authenticated after that. you do the math. :P |
| 11:41.50 | HeMan | kaldemar: worked! |
| 11:42.06 | *** part/#asterisk eject_ck (n=eject@85.223.182.86) |
| 11:42.11 | HeMan | kaldemar: now I have to figure out why it got in the context it got... |
| 11:44.04 | HeMan | ...and turn it off directly since it got in the context where it could call out without authentication... |
| 12:01.21 | *** join/#asterisk kamanashisroy (n=kamanash@119.30.36.8) |
| 12:04.56 | *** join/#asterisk esaym (n=user@cpe-24-174-186-34.satx.res.rr.com) |
| 12:08.19 | *** join/#asterisk SuPrSluG (n=SuPrSluG@firewall-a.buf.ny.i-evolve.net) |
| 12:09.21 | HeMan | Can I make all guest calls to get into a specific context? |
| 12:10.31 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 12:20.23 | SuPrSluG | you can force any call into any context you want |
| 12:20.48 | *** join/#asterisk wonderworld (n=ww@ip-62-143-16-28.unitymediagroup.de) |
| 12:21.06 | kaldemar | HeMan: define a context under [general] |
| 12:22.23 | *** join/#asterisk leifmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 12:22.23 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 12:23.52 | [TK]D-Fender | NA NA NA NA NA! |
| 12:23.57 | [TK]D-Fender | LEIF IS LEIF! |
| 12:26.35 | creativx | life is life |
| 12:26.47 | *** join/#asterisk j_kroon (n=blenda@dsl-244-51-73.telkomadsl.co.za) |
| 12:29.41 | leifmadsen | Life is Leif |
| 12:29.49 | leifmadsen | and vice-versa |
| 12:36.26 | *** join/#asterisk j_kroon (n=blenda@41.240.146.113) |
| 12:38.10 | *** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
| 12:40.43 | *** join/#asterisk j_kroon (n=blenda@dsl-240-146-113.telkomadsl.co.za) |
| 12:41.17 | j_kroon | hi guys, i seem to have run into a file descriptor leak when using Monitor() |
| 12:42.25 | j_kroon | how would i go about trying to track it down? |
| 12:42.40 | *** join/#asterisk tcseke (n=chatzill@217.20.134.239) |
| 12:43.28 | *** join/#asterisk |Cybex| (n=John@80.100.126.176) |
| 12:44.44 | leifmadsen | j_kroon: sounds like something you might use valgrind to find |
| 12:45.31 | *** join/#asterisk ariel_ (i=3fd6eca9@gateway/web/ajax/mibbit.com/x-a8a8bec4b580180b) |
| 12:46.13 | *** join/#asterisk qdk (n=qdk@81.7.168.130) |
| 12:48.44 | *** join/#asterisk j_kroon (n=jkroon@dsl-244-51-73.telkomadsl.co.za) |
| 12:49.12 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
| 12:52.48 | *** join/#asterisk ccesario (n=ccesario@189.19.9.100) |
| 12:55.08 | jaytee | valgrind :-) , sounds like one of those bulk bargain coffees. "Our Valgrind brand is only $3.99 a pound but has the same rich robust flavor of gourmet coffees costing twice as much" |
| 12:56.35 | coppice | "Our Valgrind brand is only $3.99 a pound but has the same rich robust flavor of gourmet coffees costing twice as much per kilo" |
| 12:57.08 | j_kroon | ok well, different question ... how do i trouble-shoot a file descriptor leak in asterisk? |
| 12:57.32 | j_kroon | i'm getting lots of dangling file descriptors to files in /var/spool/asterisk/montior |
| 13:02.03 | *** join/#asterisk Aiatek (n=Asterisk@75.112.88.200.m.sta.codetel.net.do) |
| 13:02.26 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:03.55 | KyleK | j_kroon: turn debugging up really high? |
| 13:04.19 | KyleK | (no idea what the monitor directory is for) |
| 13:07.39 | *** join/#asterisk Subdolus (n=subby@subby.afraid.org) |
| 13:09.17 | *** join/#asterisk p4tr0p1 (n=jam@201.34.141.194) |
| 13:10.04 | j_kroon | KyleK, for call recording. |
| 13:10.09 | rhassing | KyleK, the monitor directory is the directory where the files are placed for monitored calls (core show application Monitor ) |
| 13:10.29 | j_kroon | and no, i'd wager I'd need to run asterisk inside of gdb or something ... and on that particular box it's not really an option, so I need to reproduce elsewhere. |
| 13:15.00 | *** join/#asterisk propellerhead (n=yogurt2u@host44.190-136-118.telecom.net.ar) |
| 13:15.49 | *** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim) |
| 13:20.39 | j_kroon | rhassing, that's the easy part yes. but in order to function the app opens files in that folder, it's never closing them. |
| 13:20.43 | j_kroon | i'm trying to figure out why. |
| 13:20.49 | *** join/#asterisk magronez (n=eusei@unaffiliated/magrao/x-2903) |
| 13:21.53 | *** join/#asterisk neurosys (n=vinix@173-9-159-182-miami.txt.hfc.comcastbusiness.net) |
| 13:22.39 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 13:36.53 | *** join/#asterisk [intra]lanman (n=intralan@freeswitch/developer/intralanman) |
| 13:41.22 | *** join/#asterisk putnopvut (n=putnopvu@asterisk/master-of-queues/mmichelson) |
| 13:41.27 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 13:44.04 | j_kroon | rhassing, ok, it seems it's not the Monitor() app, it's file convert. |
| 13:44.10 | j_kroon | it doesn't close it's input file. |
| 13:44.40 | rhassing | j_kroon, why do you want to convert the file? |
| 13:46.40 | *** join/#asterisk cesar_CR (n=cesar@201.195.239.11) |
| 13:47.23 | *** join/#asterisk Dr-Linux|home (n=Nothing@119.63.130.34) |
| 13:48.26 | Dr-Linux|home | I'm using asterisk 1.4.25, problem when i dial from outside pstn DHHDI channel execute the AGI and suddendly agi crashes, |
| 13:48.45 | Dr-Linux|home | however if i dial the same extensions directly from softphone local extensions, all works fine |
| 13:49.28 | Dr-Linux|home | maybe due to : DAHDI/4-1 |
| 13:50.31 | j_kroon | can anybody please confirm the following bug: when issueing file convert via the cli asterisk does NOT close the input file? |
| 13:50.35 | j_kroon | Dr-Linux|home, AGI vs DeadAGI perhaps? |
| 13:51.11 | Dr-Linux|home | j_kroon: where DeadAGI invovled? |
| 13:51.22 | j_kroon | rhassing, the reason for conversion is manyfold, mainly the recording happens in one format and the client wants to convert to mp3 for archiving purposes (not the best, but whatever) |
| 13:51.35 | *** join/#asterisk theron (n=theron@216.51.246.211) |
| 13:51.36 | *** part/#asterisk hastinapur (n=gomel77@mail3.sis.com.by) |
| 13:51.52 | j_kroon | Dr-Linux|home, DeadAGI should be used if the channel is not yet answered. I've never used AGI myself ... so take it with a pinch of salt. |
| 13:52.00 | j_kroon | either way, it shouldn't cause a crash imho. |
| 13:52.09 | Optic | files ticket with link2voip support |
| 13:52.11 | Dr-Linux|home | j_kroon: it was just working fine when it was Zap/4-1 but it is not working now when it is DAHDI/4-1 |
| 13:52.14 | Optic | not sure what's wrong with my * |
| 13:52.38 | j_kroon | rhassing, so the idea with the script is to first convert to .g729 for playing back if the client "retrieves" a call from a phone, and it's the input file for this that's leaking. |
| 13:53.15 | Dr-Linux|home | what's the difference between zapata.conf and chan_dahdi.conf? |
| 13:53.20 | j_kroon | Dr-Linux|home, i've seen some strange differences between Zap/ and DAHDI/ could be another one. try adding an explicit Answer() before invoking AGI() and retest. just to make sure. |
| 13:53.39 | j_kroon | Dr-Linux|home, Zap got renamed to dahdi for version 2 of zaptel. |
| 13:53.51 | j_kroon | same thing though, but obviously there are differences. and some of them are obscure. |
| 13:53.54 | Dr-Linux|home | j_kroon: already Answer() is there |
| 13:54.21 | Dr-Linux|home | yeah but any difference in parameter in .conf file? |
| 13:54.46 | Dr-Linux|home | because i simply copy data from zapata.conf and put in chan_dahdi.conf |
| 13:55.06 | Dr-Linux|home | do i need some difference parameters in chan_dahdi.conf? |
| 13:55.49 | *** join/#asterisk fiesch (n=fiesch@dslb-188-098-227-142.pools.arcor-ip.net) |
| 13:55.56 | *** join/#asterisk KavanS (n=KavanS@static-71-117-242-28.ptldor.dsl-w.verizon.net) |
| 13:56.20 | *** join/#asterisk Silicium_ (n=Silicium@2001:bf0:c080:200:0:0:0:23) |
| 13:56.22 | Silicium_ | hi there |
| 13:56.41 | j_kroon | Dr-Linux|home, not that i can recall. |
| 13:57.02 | Silicium_ | what means _X and s in the extensionField in extension.conf? |
| 13:57.35 | *** join/#asterisk intralanman (n=intralan@va-67-76-163-226.sta.embarqhsd.net) |
| 13:57.56 | [TK]D-Fender | Silicium_: ... |
| 13:57.58 | [TK]D-Fender | ~book |
| 13:57.58 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 13:58.00 | [TK]D-Fender | ^^^^^^^ |
| 13:58.18 | [TK]D-Fender | Silicium_: This is Dialplan 101 and something you NEED to master. Dialplan is 95% of *. |
| 13:58.21 | Silicium_ | yea i actually own this book |
| 13:58.29 | Silicium_ | but it is to difficult to find. |
| 13:58.38 | *** join/#asterisk jasonwoot (n=some@bookit-dev.com) |
| 13:58.58 | [TK]D-Fender | Silicium_: PDF is right thre, as is HTML |
| 14:00.31 | fiesch | heya. what's the state of the"page" function in the current asterisk build... can i assume that it "works" if the auto-answer header is supported? Been assigned to a project to add mobile forklifts on the whole company campus to the pbx via omni-wlan antennas and sth like Siemens S75 (optimal), Polycom 8030 or a UTStarcom Wifi |
| 14:00.32 | *** join/#asterisk theron (n=theron@216.51.246.211) |
| 14:01.20 | [TK]D-Fender | fiesch: Page has always worked. |
| 14:01.22 | fiesch | communication should be like with all forklifts having two way radios on the same channel and can dial out and to specifix extensions within the administration building |
| 14:01.30 | [TK]D-Fender | fieWhat your DEVICE does is ITS problem. |
| 14:01.41 | [TK]D-Fender | fiesch: What your DEVICE does is ITS problem. |
| 14:01.56 | fiesch | [TK]D-Fender: well yes, i should possibly have been a bit clearer |
| 14:02.25 | [TK]D-Fender | fiesch: Apples & oranges. Page is not a suspect here. |
| 14:02.40 | Silicium_ | thanks for help... |
| 14:02.42 | fiesch | what's the general level of interoperability as seen by the community.. like number confirmed working / number available |
| 14:03.25 | fiesch | I'll have some over for testing in the following weeks but as time is like always not on my side ... |
| 14:03.47 | *** part/#asterisk Silicium_ (n=Silicium@2001:bf0:c080:200:0:0:0:23) |
| 14:04.49 | [TK]D-Fender | fiesch: huh?! |
| 14:05.06 | [TK]D-Fender | fiesch: "number working"? you aren't making any sense |
| 14:05.25 | fiesch | [TK]D-Fender: nevermind. |
| 14:05.32 | Optic | bwok! |
| 14:06.25 | fiesch | number working spelled out reads "Number of SIP hardware end devices confirmed working with the standard Asterisk Page command" |
| 14:06.51 | *** join/#asterisk ingenius (n=alektro@host176.190-230-72.telecom.net.ar) |
| 14:06.58 | [TK]D-Fender | fiesch: All of them. |
| 14:07.28 | [TK]D-Fender | fiesch: Page doesn't do ANYTHING. Its just a combined Dial command. |
| 14:08.01 | [TK]D-Fender | fiesch: there is no magic in "Page". Its dial + MeetMe, and every bit as dumbed down as that description sounds. |
| 14:08.13 | fiesch | [TK]D-Fender: great, that's what i wanted to hear. I'll need to whip up sip support details for the phones then. |
| 14:08.20 | [TK]D-Fender | fiesch: You could do the same thing spawning Originates into a MettMe yourself |
| 14:08.34 | jaytee | dumbed down is being polite. wicked frakkin retahded is more like it |
| 14:08.40 | [TK]D-Fender | fiesch: and Every SIP phone has its own header / AA format. |
| 14:08.53 | eppigy | HOLLER BACK YOUNGIN |
| 14:09.05 | jaytee | hehe |
| 14:09.08 | [TK]D-Fender | jaytee: No, I save that for OTHER commands, but I've made a promise not to rag on it so much ;) |
| 14:09.46 | [TK]D-Fender | eppigy: "I'm not sure what a Holla-Back girl is but I want her dead." - Brian on 'Family Guy' |
| 14:10.34 | fiesch | *g* |
| 14:10.34 | eppigy | haha |
| 14:10.34 | fiesch | well I'm off thangs for the info |
| 14:10.34 | fiesch | aarrgh |
| 14:10.34 | fiesch | g=k |
| 14:10.35 | fiesch | hate those |
| 14:11.04 | *** join/#asterisk securevoip (n=securevo@173-15-197-73-BusName-Richmond.hfc.comcastbusiness.net) |
| 14:14.16 | *** join/#asterisk [acer]lanman (n=intralan@va-67-76-163-226.sta.embarqhsd.net) |
| 14:20.04 | timeshell_atwork | Hey, does asterisk and dahdi work under 64 bit? |
| 14:21.00 | [TK]D-Fender | timeshell_atwork: http://www.google.ca/search?hl=en&q=DAHDI+64bit&btnG=Google+Search&meta=&aq=f&oq= |
| 14:21.11 | [TK]D-Fender | timeshell_atwork: JFG.... awww fukkit |
| 14:21.30 | *** join/#asterisk telnettech (n=telnette@cpe-71-74-91-116.insight.res.rr.com) |
| 14:21.33 | timeshell_atwork | lol [TK]D-Fender Wouldn't a yes or no be easier? |
| 14:22.16 | timeshell_atwork | Yah I coulda googled it. |
| 14:22.32 | timeshell_atwork | But that would be anti-social of me. |
| 14:22.34 | timeshell_atwork | :D |
| 14:26.10 | *** join/#asterisk ck_28 (n=CK@212.98.141.199) |
| 14:26.14 | ck_28 | hi |
| 14:26.15 | [TK]D-Fender | timeshell_atwork: Yup.. putting the "fun" back into "dysfunctional" |
| 14:26.23 | timeshell_atwork | heh |
| 14:26.24 | ck_28 | i am using a call file to generate a call |
| 14:26.37 | ck_28 | how can i get the status of the call |
| 14:26.41 | timeshell_atwork | Ok then let me rephrase. Do asterisk and dahdi work WELL under 64 bit? Are they stable? |
| 14:27.15 | ck_28 | how to know if the channel available or busy or unavailable ? |
| 14:27.38 | ck_28 | i used HANGUPCAUSE is ${HANGUPCAUSE} and DIALSTATUS is ${DIALSTATUS}) |
| 14:33.17 | [TK]D-Fender | ck_28: DIALSTATUS sure seems to say it... |
| 14:33.55 | j_kroon | https://issues.asterisk.org/view.php?id=15181 - not closing the descriptors causes asterisk to eventually die with out of file descriptors! |
| 14:34.26 | ck_28 | [TK]D-Fender i am using a call file |
| 14:34.44 | j_kroon | timeshell_atwork, yes. |
| 14:34.56 | ck_28 | [TK]D-Fender what i am doing is moving the file to /var/spool/asterisk/outgoing |
| 14:35.27 | timeshell_atwork | j_kroon Thank you |
| 14:35.30 | j_kroon | russellb, you the same russel who closed my bug report? |
| 14:35.39 | ck_28 | [TK]D-Fender not dialing it direcktly |
| 14:35.52 | [TK]D-Fender | ck_28: whatever that means... |
| 14:36.34 | ck_28 | [TK]D-Fender http://www.voip-info.org/wiki/view/Asterisk+auto-dial+out |
| 14:36.38 | *** join/#asterisk Jacke (i=jacke@valhalla.bofh.org) |
| 14:36.45 | [TK]D-Fender | ck_28: Yes i know all about call files... |
| 14:36.54 | ck_28 | sorry i know that you know |
| 14:37.06 | [TK]D-Fender | ck_28: "dialing directly" as though that term necessarily meant something specific or special |
| 14:37.27 | *** join/#asterisk SlipperyChicken (n=andrew@LONDON14-1168107385.sdsl.bell.ca) |
| 14:38.00 | ck_28 | how can i get the call status for the call |
| 14:38.01 | [TK]D-Fender | ck_28: See if you're looking to process things yourself for the "Channel:" then you should be choosing a channel type that LETS you. |
| 14:38.29 | [TK]D-Fender | ck_28: Go read over *'s list of channel types a few dozen times ro until your eyes bleed :) |
| 14:38.44 | [TK]D-Fender | ck_28: One should hopefully stand out to you befor that happens. |
| 14:39.11 | lost_soul | wondering, if I enable followme, and turn off the soft phone on my pc the followme doesn't seem to be used. For whatever reason asterisk just hangs the call up. Is this normal behavior or are my settings improperly configured |
| 14:39.14 | *** join/#asterisk moy (n=moy@74.12.124.89) |
| 14:39.20 | ck_28 | [TK]D-Fender i will make what i am asking for more clear |
| 14:39.28 | j_kroon | lost_soul, freepbx? |
| 14:39.34 | ck_28 | [TK]D-Fender at the call file Context: fax-tx |
| 14:39.41 | lost_soul | asterisk running on openbsd |
| 14:39.52 | lost_soul | version 1.4 something |
| 14:39.59 | ck_28 | at [fax-tx] i add exten => s,n,Noop(HANGUPCAUSE is ${HANGUPCAUSE} and DIALSTATUS is ${DIALSTATUS}) |
| 14:40.20 | ck_28 | [TK]D-Fender i cant get nothing for the value ${DIALSTATUS} |
| 14:40.49 | [TK]D-Fender | ck_28: pastebin the REST of that dialplan. |
| 14:40.51 | [TK]D-Fender | ~pb |
| 14:40.52 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste , or , http://bin.cakephp.org/ |
| 14:41.00 | ck_28 | [TK]D-Fender because asterisk manager test if channel available it will continue to context |
| 14:41.06 | [TK]D-Fender | ck_28: because things aren't occuring in the order you think they are. |
| 14:41.40 | j_kroon | lost_soul, follow me is not an asterisk feature by itself but implemented in the dialplan. so how did you configure the follow me? |
| 14:42.25 | [TK]D-Fender | j_kroon: followme.conf <- sort of is... |
| 14:42.32 | lost_soul | j_kroon: exten => _1XX,n(lbl_incoming_calls_3),Followme(${EXTEN}|a) |
| 14:42.35 | j_kroon | o.O |
| 14:42.40 | lost_soul | is what I used to call it in extensions.conf |
| 14:42.44 | j_kroon | feels stupid now |
| 14:42.54 | lost_soul | then setup the followme.conf |
| 14:43.17 | *** join/#asterisk Whitor (n=Whitor@rrcs-24-97-4-146.nys.biz.rr.com) |
| 14:43.17 | ck_28 | [TK]D-Fender kindly check it http://pastebin.com/d30a51e3c |
| 14:43.27 | lost_soul | basically all I wanted it to do was if the softphone on pc was off cuz I want to shut that pc down... have it ring the house phone |
| 14:43.55 | [TK]D-Fender | Follwme is a complete waste as you could alsways have done it in pure dialplan... |
| 14:43.55 | ck_28 | [TK]D-Fender i add my call file and extension.conf + sip.conf |
| 14:44.07 | [TK]D-Fender | lost_soul: that IS a complete waste |
| 14:44.12 | KyleK | ~followme |
| 14:44.18 | [TK]D-Fender | lost_soul: Just dial them back to back |
| 14:44.19 | KyleK | aww come on |
| 14:45.14 | lost_soul | [TK]D-Fender: so have a dial command after the local ext and before the voicemail? |
| 14:46.17 | [TK]D-Fender | ck_28: http://pastebin.com/m18ec27f5 |
| 14:46.18 | *** join/#asterisk Deeewayne (n=dwayne@75.76.254.162) |
| 14:46.18 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 14:46.26 | [TK]D-Fender | lost_soul: Yes |
| 14:46.59 | lost_soul | [TK]D-Fender: will give that a shot, thanks |
| 14:47.52 | *** join/#asterisk intralanman (n=intralan@va-67-76-163-226.sta.embarqhsd.net) |
| 14:48.06 | *** join/#asterisk |Cybex| (n=John@80.100.126.176) |
| 14:49.22 | *** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler) |
| 14:51.44 | [TK]D-Fender | [intra]lanman: [i][n][t][r][a][l][a][n][m][a][n] |
| 14:52.42 | [intra]lanman | ^[T][K][D-F]ender$ ? |
| 14:52.51 | Dr-Linux|home | I can't recieve callerid, when i plug phone line to a box where DAHDI running |
| 14:53.09 | *** join/#asterisk Defraz (n=T0tal@fw-poky.fuzecore.com) |
| 14:53.12 | Dr-Linux|home | but if i plug same cable in to a box that running zap .. i can get callerid |
| 14:53.41 | *** join/#asterisk seanmh (n=johndoe@198.59.129.24) |
| 14:53.41 | Dr-Linux|home | on both boxes zapata.conf and chan_dahdi.conf are identical to each other |
| 14:53.44 | Dr-Linux|home | any advice? |
| 14:54.30 | [TK]D-Fender | Dr-Linux|home: ... show us everything because right now we see nothing and trust even less :) |
| 14:54.54 | Aiatek | ~pastebin |
| 14:54.55 | infobot | [~pastebin] 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 |
| 14:55.07 | Dr-Linux|home | [TK]D-Fender: you mean, show you chan_dahdi.conf? |
| 14:55.11 | *** part/#asterisk securevoip (n=securevo@173-15-197-73-BusName-Richmond.hfc.comcastbusiness.net) |
| 14:55.20 | [TK]D-Fender | Dr-Linux|home: Obviously |
| 14:55.22 | Aiatek | both |
| 14:55.27 | [TK]D-Fender | Dr-Linux|home: and failed call with debug, etc |
| 14:55.47 | [TK]D-Fender | Dr-Linux|home: you've been at this for years. Do yuo really need to ask what to show? |
| 14:55.58 | [TK]D-Fender | Dr-Linux|home: Seriously Shah...... |
| 14:56.05 | [TK]D-Fender | TERRIBLE |
| 14:56.08 | ck_28 | [TK]D-Fender did you see the link i paste http://pastebin.com/m18ec27f5 |
| 14:56.23 | Dr-Linux|home | Aiatek: both ..mmm both files are just same |
| 14:56.27 | [TK]D-Fender | ck_28: tahts MY link. READ IT |
| 14:56.50 | [TK]D-Fender | Dr-Linux|home: Don't give us a story, give us your configs and a failed call with debug |
| 14:56.52 | Aiatek | thats what you say |
| 14:57.09 | Aiatek | show what you have in both side |
| 14:57.47 | *** join/#asterisk Chuggs (n=Chuggs@s142-179-186-158.ab.hsia.telus.net) |
| 14:58.08 | Dr-Linux|home | [TK]D-Fender: not sure how to debug DAHDI calls |
| 14:58.22 | ck_28 | [TK]D-Fender what can i do to detect |
| 14:58.46 | ck_28 | before answers |
| 14:58.58 | *** join/#asterisk Victor_Yure (n=victor@unaffiliated/victoryure/x-837844) |
| 14:58.59 | ck_28 | is there any lines i can add ? |
| 14:59.36 | ck_28 | [TK]D-Fender or any other solution |
| 14:59.40 | [TK]D-Fender | ck_28: Go read over *'s list of channel types a few dozen times or until your eyes bleed :) <-------- |
| 14:59.52 | [TK]D-Fender | ck_28: See if you're looking to process things yourself for the "Channel:" then you should be choosing a channel type that LETS you. |
| 15:00.16 | ck_28 | :P |
| 15:00.46 | Dr-Linux|home | [TK]D-Fender: see there: http://pastebin.ca/1431033 |
| 15:01.54 | [TK]D-Fender | Dr-Linux|home: Core debug and NoOp the damn callerid. |
| 15:02.01 | [TK]D-Fender | Dr-Linux|home: I don't see you NO getting it there |
| 15:02.11 | [TK]D-Fender | Dr-Linux|home: And no, there is no way in hell I trust your AGI :) |
| 15:02.16 | Dr-Linux|home | [TK]D-Fender: NoOP doesn't show |
| 15:02.25 | [TK]D-Fender | Dr-Linux|home: SHOW ME. |
| 15:02.30 | Dr-Linux|home | okey |
| 15:02.34 | Dr-Linux|home | hang on |
| 15:03.19 | *** join/#asterisk mv2 (n=maverick@83.240.229.38) |
| 15:03.31 | mv2 | <PROTECTED> |
| 15:06.48 | Dr-Linux|home | [TK]D-Fender: see there: http://pastebin.ca/1431041 |
| 15:06.57 | Dr-Linux|home | no caller id |
| 15:07.19 | [TK]D-Fender | Dr-Linux|home: I don't see the CODE to go with that either |
| 15:07.47 | mv2 | <PROTECTED> |
| 15:08.38 | Dr-Linux|home | [TK]D-Fender: want to see dialplan, what i have? |
| 15:08.46 | *** join/#asterisk grEvenX (n=even@cB78D5AC1.dhcp.bluecom.no) |
| 15:09.40 | Dr-Linux|home | [TK]D-Fender: There is nothing to do with AGI, that is just example |
| 15:09.45 | Dr-Linux|home | caller id should be first thing |
| 15:09.48 | KyleK | mv2: I dont think IAX2 does video |
| 15:09.52 | Dr-Linux|home | see here the dialplan: http://pastebin.ca/1431042 |
| 15:10.09 | Dr-Linux|home | but that doesn't matter much in my case |
| 15:10.38 | Dr-Linux|home | [TK]D-Fender: tell me one thing, do you think anything wrong at chan_dahdi.conf? |
| 15:11.48 | [TK]D-Fender | Dr-Linux|home: What ver of *? |
| 15:12.05 | Dr-Linux|home | 1.4.25 |
| 15:12.18 | [TK]D-Fender | Dr-Linux|home: well exten => 8800,n,NoOp(${CALLERID}) <--- not a proper way to get the CID |
| 15:12.32 | [TK]D-Fender | Dr-Linux|home: there is a FUNCTION for this since 1.2 and the vars are dead crap. |
| 15:13.01 | Dr-Linux|home | yes i know, but still i grabs the callerid |
| 15:13.04 | [TK]D-Fender | Dr-Linux|home: Please do this the RIGHT way |
| 15:13.14 | Dr-Linux|home | okey |
| 15:13.19 | [TK]D-Fender | Dr-Linux|home: I am not going to trust half-assed deprecated bits and pieces |
| 15:14.15 | Aiatek | which one is the proper way to get the caller ID? |
| 15:14.39 | mv2 | Kylek: ok thanks |
| 15:15.50 | [TK]D-Fender | Aiatek: "core show function CALLERid" |
| 15:15.55 | [TK]D-Fender | Aiatek: "core show function CALLERID" |
| 15:15.56 | *** part/#asterisk ck_28 (n=CK@212.98.141.199) |
| 15:16.09 | [TK]D-Fender | guesses ck_28's eyes have bled out... |
| 15:16.50 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 15:20.31 | *** join/#asterisk hfb (n=hfb@pool-96-247-109-136.lsanca.dsl-w.verizon.net) |
| 15:20.32 | Aiatek | ok thanks |
| 15:20.35 | *** join/#asterisk gr0mit (n=tim@extrt.txrx.org.uk) |
| 15:21.02 | *** join/#asterisk fred-tmft (n=fred-tea@69.244.180.112) |
| 15:21.46 | *** part/#asterisk fred-tmft (n=fred-tea@69.244.180.112) |
| 15:23.05 | *** join/#asterisk SebastianS (n=schu@adsl-dyn58.78-98-36.t-com.sk) |
| 15:23.20 | *** join/#asterisk andrebarbosa (n=andrebar@212.13.49.67) |
| 15:25.15 | eppigy | Katty: caught up on sleep :D |
| 15:27.18 | Katty | :>>> |
| 15:27.27 | eppigy | yesh |
| 15:27.47 | *** join/#asterisk jtodd (i=qmj61rjt@ns.fox-den.com) |
| 15:27.47 | *** mode/#asterisk [+o jtodd] by ChanServ |
| 15:28.18 | *** join/#asterisk ming_zym (n=ming_zym@f0-0-tep-rtr1.corp.cnb.yahoo.com) |
| 15:30.35 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 15:31.14 | phix | hi cunts |
| 15:31.18 | phix | how are you? |
| 15:31.23 | phix | geat? |
| 15:35.37 | Katty | scowls. |
| 15:36.27 | eppigy | wow |
| 15:36.44 | eppigy | I thought I was looking at a different channel for a sec |
| 15:38.16 | [TK]D-Fender | Dr-Linux|home: Well? |
| 15:38.58 | KavanS | in one sentence, why would one use 1.6 vs. 1.4...what sizeable additions have been made? |
| 15:41.07 | [TK]D-Fender | KavanS: Go read the CHANGES docs included with the tarball |
| 15:41.29 | KavanS | so you wouldn't upgrade? |
| 15:42.23 | [TK]D-Fender | KavanS: Any more words you'd like to put in my mouth while you're at it? :) |
| 15:42.25 | Dr-Linux|home | [TK]D-Fender: sorry that server is disconnected |
| 15:42.50 | KavanS | [TK]D-Fender, ok, so 1.4 is the way to go then, roger that |
| 15:43.05 | [TK]D-Fender | KavanS: That would indeed be "more". |
| 15:43.21 | [TK]D-Fender | reaches for his ClueBat (tm) |
| 15:43.53 | *** join/#asterisk marv[work] (n=timr@router.asteriasgi.com) |
| 15:43.55 | *** join/#asterisk intralanman (n=intralan@va-67-76-163-226.sta.embarqhsd.net) |
| 15:44.37 | *** join/#asterisk BCS-Satori (n=BCS-Sato@75-148-21-113-WashingtonDC.hfc.comcastbusiness.net) |
| 15:45.22 | *** join/#asterisk Torrieri (n=Torrieri@nelug/crew/torrieri) |
| 15:45.37 | BCS-Satori | My VoIP carrier decided to charge extra for CNAME's a few months back on each DID. Is there a way or service that can lookup CNAME's based on the caller ID number and display them on the phones without having to go through the carrier? |
| 15:46.17 | *** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim) |
| 15:46.18 | carrar | BCS-Satori, yes there is |
| 15:46.44 | ctooley | BCS-Satori, Yeah, you can do an SS7 Dip. But, I'm guessing that you're probably going to be paying less for the CNAM service on your DID's than those trunks cost. |
| 15:48.18 | BCS-Satori | ctooley: so there is no open free alternative? Its $3.00 a month per DID but at 100 DID's it gets a little crazy for something that use to be free. |
| 15:48.47 | carrar | Get a provider who doesn't nail you in the ass for CNAM |
| 15:49.24 | BCS-Satori | carrar: I can not agree more |
| 15:49.28 | carrar | They should be passing CNAM |
| 15:49.33 | carrar | if they are any good |
| 15:49.45 | carrar | passing to you |
| 15:49.45 | BCS-Satori | carrar: they use to for the past year, and decided April 1st to make it a pay for service per DID |
| 15:50.12 | carrar | so port your numbers to someone else who does |
| 15:50.21 | *** join/#asterisk s14ck (n=s14ck@ccscliente154.ifxnetworks.net.ve) |
| 15:50.36 | BCS-Satori | carrar: Probably what we will be looking at doing. |
| 15:54.46 | *** join/#asterisk da__d00d (n=da__d00d@dsl-vlan422-66-18-194-227.nucleus.com) |
| 16:10.37 | *** join/#asterisk gr0mit (n=tim@extrt.txrx.org.uk) |
| 16:17.41 | *** join/#asterisk pfn (n=pfnguyen@hanhuy.com) |
| 16:18.14 | *** join/#asterisk ck_28 (n=CK@212.98.141.199) |
| 16:18.42 | ck_28 | [TK]D-Fender how can i install Ms sql odbc driver and use with asterisk |
| 16:19.07 | *** join/#asterisk jameswf (n=james@unaffiliated/jameswf-home) |
| 16:19.19 | [TK]D-Fender | ck_28: #odbc |
| 16:19.41 | *** join/#asterisk jmacz (n=jmacz@190.144.75.22) |
| 16:20.53 | *** join/#asterisk SirThomas (n=tomc@mail.kendeco.com) |
| 16:22.55 | *** join/#asterisk timholum (n=timholum@64-91-67-5.stat.centurytel.net) |
| 16:30.03 | rhassing | I passed my dCAP exam :-) |
| 16:30.19 | *** join/#asterisk BlargMaN00 (n=blarg@12.234.16.130) |
| 16:30.32 | rhassing | I'm so happy, I just wanted to share this moment :) |
| 16:35.34 | *** join/#asterisk SirThomas (n=tomc@mail.kendeco.com) |
| 16:38.58 | jaytee | congratulations |
| 16:39.23 | *** join/#asterisk subl (i=sublime@xmission.xmission.com) |
| 16:48.55 | *** join/#asterisk eppigy (n=Dave@216-139-245-58.aus.us.siteprotect.com) |
| 16:50.28 | *** join/#asterisk neurosys (n=vinix@173-9-159-182-miami.txt.hfc.comcastbusiness.net) |
| 16:51.32 | jameswf | is infobot talkin to himself |
| 16:51.49 | eppigy | haha |
| 16:51.51 | eppigy | no i am dave |
| 16:52.13 | coppice | no, dave is a TV station in London |
| 16:52.17 | *** join/#asterisk telnettech (n=telnette@cpe-71-74-91-116.insight.res.rr.com) |
| 16:52.18 | neurosys | [TK]D-Fender: Correct me if im wrong, but i have 26 rooms in this resort, if i buy a Mediatrix 4124 a 4108, for an extra 100 or so i can grab 2 4116's and leave room for more phones if need be? make sense? |
| 16:52.29 | jameswf | ~infobot |
| 16:52.30 | infobot | well, infobot is [infobot], or infobot, or likes abuse |
| 16:52.49 | jameswf | ~dave |
| 16:52.49 | infobot | dave is probably the renowned inventor of the FR-clapper |
| 16:52.56 | neurosys | heh |
| 16:53.52 | telnettech | neurosys: you are correct. Just depends on what you want |
| 16:54.02 | telnettech | either way would work |
| 16:54.26 | neurosys | telnettech: thanks :) |
| 16:54.27 | Nugget | telnet is eeeeeeevil! |
| 16:54.46 | telnettech | good day Nugget |
| 16:54.52 | eppigy | botsnack |
| 16:54.58 | eppigy | ~botsnack |
| 16:54.58 | infobot | :), eppigy |
| 16:55.57 | *** join/#asterisk Victor_Yure (n=victor@unaffiliated/victoryure/x-837844) |
| 16:58.15 | *** join/#asterisk MrNaz (n=mrnaz@ppp118-208-203-152.lns10.mel6.internode.on.net) |
| 16:58.18 | *** part/#asterisk seste (n=rseste@mail.daitanlabs.com) |
| 17:06.51 | [TK]D-Fender | neurosys: Whatever combo is most cost-effective do you |
| 17:07.32 | *** join/#asterisk hardwire (n=hardwire@216-67-98-253.static.acsalaska.net) |
| 17:07.38 | hardwire | stupid nickserv |
| 17:08.20 | hardwire | anybody in the US found a trunk or DISA system that allows a call to go through to a mobile client without charging them for minutes - and instead charging you? |
| 17:10.08 | hardwire | basically making zero-rated calls to mobile customers on a variety of mobile networks |
| 17:10.24 | [TK]D-Fender | hardwire: doesn't work that way. you are calling out to multiple destination carriers. THESE people want to gouge their customers so no.... not really viable |
| 17:10.36 | Katty | man i feel fat. |
| 17:10.50 | hardwire | [TK]D-Fender: I'll take your word on it. |
| 17:11.12 | hardwire | I didn't think there would be a clearing house that is dealing with all mobile subscribers. |
| 17:11.19 | hardwire | that's a lot of work |
| 17:11.19 | [TK]D-Fender | hardwire: And it'd be per carrier as well |
| 17:12.03 | hardwire | [TK]D-Fender: indeed. If I want to make zero-rated calls I will need a means to trunk in to a mobile network and pass a mobile ANI for all concurrent channels. |
| 17:12.06 | hardwire | thinks more. |
| 17:12.11 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 17:12.30 | hardwire | given that those people have free inbound for mobile-mobile |
| 17:14.11 | *** join/#asterisk voxter (n=voxter@76.77.91.251) |
| 17:20.05 | hardwire | foxy voxter |
| 17:23.42 | *** join/#asterisk jtodd (i=vawopl0x@ns.fox-den.com) |
| 17:23.42 | *** mode/#asterisk [+o jtodd] by ChanServ |
| 17:23.58 | *** join/#asterisk Shazaum (n=Shazaum@unaffiliated/shazaum) |
| 17:24.08 | Shazaum | hi |
| 17:25.02 | *** part/#asterisk Shazaum (n=Shazaum@unaffiliated/shazaum) |
| 17:26.59 | hardwire | bye |
| 17:27.27 | eppigy | Katty: girl dont be silly |
| 17:32.43 | *** join/#asterisk PanicMan (i=Learner@122.102.33.80) |
| 17:33.15 | *** join/#asterisk ming_zym (n=ming_zym@220.181.34.178) |
| 17:35.55 | ck_28 | [TK]D-Fender i telneted to 127.0.0.1 5038 |
| 17:36.16 | PanicMan | hello to all, I'm new in Asterisk, Want some help regarding Asterisk+H323+SS7 |
| 17:36.17 | ck_28 | when i type Action: Login |
| 17:36.20 | phix | hi |
| 17:36.20 | ck_28 | nothin appears |
| 17:36.38 | PanicMan | may i proceed ! |
| 17:37.20 | PanicMan | any helper care to help :( |
| 17:40.13 | PanicMan | hello |
| 17:40.25 | PanicMan | anybody played with asterisk with SS7 ? |
| 17:41.10 | *** join/#asterisk cyford-tech (n=allen@12.22.184.2) |
| 17:42.40 | *** join/#asterisk chandoo (n=chandoo@ool-4353b978.dyn.optonline.net) |
| 17:45.09 | *** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim) |
| 17:46.53 | watchy | hardwire: callout and fake your cid to a cell on the carrier |
| 17:47.45 | watchy | from what i understand that will work |
| 17:52.04 | *** join/#asterisk grEvenX (n=even@cB78D5AC1.dhcp.bluecom.no) |
| 17:56.18 | *** part/#asterisk ming_zym (n=ming_zym@220.181.34.178) |
| 18:27.03 | *** join/#asterisk lanning (n=lanning@nat/yahoo/x-336505e1821e8f6d) |
| 18:27.34 | hardwire | watchy: you'd think they would know what trunk/means the call came in on. |
| 18:27.46 | *** part/#asterisk guax (n=guaxinim@unaffiliated/guaxinim) |
| 18:32.41 | *** join/#asterisk theHub (n=theHub@69.177.93.21) |
| 18:34.20 | *** join/#asterisk seb- (n=seb@li30-51.members.linode.com) |
| 18:34.47 | seb- | [TK]D-Fender: got access to a Windows machine? if yes do you mind calling my server and testing chat w/ me? i think i'm good to go |
| 18:35.13 | seb- | [TK]D-Fender: Also, possible to create a custom Ekiga that has config for a specific server embedded you can hand out to people? |
| 18:38.37 | [TK]D-Fender | SebWhy would i require a Windows machine? :) |
| 18:39.01 | seb- | [TK]D-Fender: i agree...they are lame but my students will undoubted have 100% windows |
| 18:39.02 | [TK]D-Fender | seb-: I know that Zoiper allows you to roll-out preconfigured clients. |
| 18:39.38 | seb- | [TK]D-Fender: zoiper...but you got my heart set on ekiga |
| 18:39.41 | seb- | :) |
| 18:39.45 | [TK]D-Fender | seb-: Mind you anyone dumb enough not to be able to follow filling in the 4 blanks in the account entry screen shold be dragged out and shot anyway :) |
| 18:40.32 | seb- | [TK]D-Fender: i just tried on a windows machine...didn't work...maybe a firewall issue....anyhoo can i pm you my password..i need to test my headset (mic + headphones) i bought as you suggested |
| 18:41.11 | seb- | [TK]D-Fender: shoudl only take a sec |
| 18:43.17 | [TK]D-Fender | seb-: @work, can't work on this now |
| 18:43.36 | seb- | [TK]D-Fender: ok..maybe another time |
| 18:43.41 | seb- | [TK]D-Fender: how are things w/ you? |
| 18:45.11 | seb- | [TK]D-Fender: is there a time on a weekday we can test the chat? (I can test on weekends as this only works when *i'm* at work) |
| 18:49.08 | Optic | i figured out what was wrong with my * |
| 18:49.15 | Optic | i had a server connecting that I had forgotten all about :( |
| 18:49.23 | Optic | so I turned it off, and now all is well |
| 18:52.36 | *** join/#asterisk SebastianS (n=schu@adsl-dyn123.91-127-211.t-com.sk) |
| 19:01.34 | therealcircut | ok all |
| 19:01.39 | therealcircut | so i got a great one for ya |
| 19:01.50 | therealcircut | we have this overhead paging device that works on a WCTDM400P |
| 19:02.14 | therealcircut | u call its extension, it rings once, picks up and plays whatever over the intercom |
| 19:02.30 | therealcircut | well we installed a TDM800P card with 2 FXS modules, and now the intercom has stopped working |
| 19:02.50 | therealcircut | we measured the voltage coming from the TDM400, and it was 49.1 base |
| 19:02.57 | *** part/#asterisk Optic (n=dfraser@miso.capybara.org) |
| 19:03.03 | therealcircut | then we measured the voltage coming from the TDP800P, it was 45.1 |
| 19:03.05 | therealcircut | erm |
| 19:03.06 | therealcircut | 45.5 |
| 19:03.35 | marv[work] | is there any way from the dialplan to call multiple devices, play something to them before they're bridged, then bridge one of them? I tried using app_dial and the M flag, but it still hangs up on the others as soon as one person answers |
| 19:03.46 | therealcircut | also the polarity on the 800P was opposite from the 400P, I was able to flip |
| 19:04.15 | therealcircut | the 800P polarity to be the same as the 400P, but im still seeing only 45.5v on the 800p |
| 19:04.31 | therealcircut | ive tried tinkering with the 'fxovoltage' setting, but the voltage remains at 45.5 |
| 19:04.37 | therealcircut | even after a restart / reload |
| 19:05.01 | therealcircut | anyone have any ideas on how to bump up the voltage on the 800P card? |
| 19:05.24 | *** part/#asterisk seb- (n=seb@li30-51.members.linode.com) |
| 19:07.40 | *** join/#asterisk troy|work (n=troy@142.166.111.20) |
| 19:08.35 | therealcircut | good one right? |
| 19:09.14 | therealcircut | marv[work]: any output in your asterisk console? |
| 19:10.03 | cyford-tech | hi, i have a question, are there any privacy laws against outgoing calls? |
| 19:10.34 | cyford-tech | in america |
| 19:10.57 | cyford-tech | opps i mean call snooping on outgoing calls |
| 19:12.34 | *** join/#asterisk spck (n=spck@unioncab.com) |
| 19:13.39 | marv[work] | therealcircut: looks like if I instead dial local channels and have them dial the real channels and then pass the M option on that second dial, it does what I wanted it to do. (iow the local's don't pass the answer back to the originating channel until the macro is finished running) |
| 19:13.44 | marv[work] | but that's kind of nasty |
| 19:15.27 | *** join/#asterisk lesouvage (n=lesouvag@82.73.69.76) |
| 19:16.51 | spck | what's the diff between trixbox and asterisk? |
| 19:16.52 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 19:17.16 | therealcircut | spck: trixbox is a distribution designed specifically to run an asterisk pbx |
| 19:17.39 | therealcircut | has some neat features, and probably is a good starting distro for people not comfortable with linux / asterisk |
| 19:18.33 | *** join/#asterisk vasundhar (n=vasundha@122.169.149.93) |
| 19:18.49 | *** join/#asterisk jameswf (n=james@unaffiliated/jameswf-home) |
| 19:19.01 | [TK]D-Fender | Nope... a BAD start using more dated * versions, a forked GUI that the main branch doesn't want to her about, etc |
| 19:19.22 | therealcircut | oh i forgot, [TK]D-Fender knows all |
| 19:19.48 | [TK]D-Fender | And charges a reasonable rate for selective recollection ;) |
| 19:20.37 | jameswf | ~[TK]D-Fender |
| 19:20.38 | infobot | [TK]D-Fender is the Zen Master of the blatantly obvious, and #asterisk 's resident Google-proxy. |
| 19:21.12 | lesouvage | I run 1.4.24.1 and just found out that channel variable ${DIALSTATUS} isn't available anymore. I did some Googling and I read about the function DEVSTATE that should be a replacement in svn but this function isn't available in 1.4.24.1. Should the channel variable still be there in this version? |
| 19:21.23 | therealcircut | whatever helps you sleep at night |
| 19:21.38 | [TK]D-Fender | lesouvage: PARDON? |
| 19:21.44 | therealcircut | lesouvage: i think u need to install the bristuff package to ge that |
| 19:21.46 | eppigy | haha |
| 19:21.55 | eppigy | man what |
| 19:22.14 | jameswf | therealcircut: The three wise men help me sleep at night |
| 19:22.18 | [TK]D-Fender | therealcircut: Can I have some of that crack you're on? thats some good stuff ;) |
| 19:22.18 | eppigy | < therealcircut> has some neat features, and probably is a good starting distro for people not comfortable with linux / asterisk |
| 19:22.21 | eppigy | read: lazy |
| 19:22.50 | jameswf | ~lazy |
| 19:22.51 | infobot | Hard work may pay off later, but LAZINESS pays off now! Work hard at hardly working! |
| 19:23.11 | therealcircut | jameswf: sleeping with 3 men helps you sleep at night? |
| 19:23.21 | eppigy | lol |
| 19:23.42 | eppigy | to each his own |
| 19:23.46 | therealcircut | [TK]D-Fender: heres something i bet u dont know |
| 19:23.57 | therealcircut | will a 2002 ZX9r engine fit on a ZX6r frame |
| 19:24.02 | therealcircut | with no modifications |
| 19:24.02 | jameswf | therealcircut: I was talking about Jim , Jack and Johnny but that could work too wanna come over |
| 19:24.05 | marv[work] | as for as I know, app_dial still sets DIALSTATUS |
| 19:24.11 | therealcircut | jameswf: zing |
| 19:24.45 | eppigy | therealcircut: if we were in #greasemonkey |
| 19:24.52 | therealcircut | :) |
| 19:24.52 | eppigy | that might be relevant |
| 19:25.35 | lesouvage | AFAIK ${DIALSTATUS} used to be one of the normal channelveriables. see http://www.voip-info.org/wiki/view/Asterisk+variables#PredefinedChannelVariables. But the variable stops having soe value at all. |
| 19:26.08 | eppigy | I find it hard to believe it has been depricated |
| 19:27.25 | [TK]D-Fender | lesouvage: Bet you believe everything you read on Wikipedia too ;) |
| 19:27.36 | marv[work] | DIALSTATUS is still set by app_dial in 1.6.1.0 |
| 19:28.03 | [TK]D-Fender | lesouvage: And they can't remove DIALSTATUS in a 1.4 release. Thats be a major change and its in MAINTENENCE mode <- |
| 19:28.19 | [TK]D-Fender | lesouvage: So umm... WTF are you doing? :) |
| 19:28.43 | *** join/#asterisk ayeso (n=chatzill@ext-52.sagetelecom.net) |
| 19:29.18 | marv[work] | also the DEVICE_STATE function does something different |
| 19:29.20 | ayeso | is there a way in the dialplan to do something after a caller has hung up? I want to call the voice mail application then do something after the caller has disconnected. |
| 19:29.41 | *** join/#asterisk ingenius (n=alektro@host176.190-230-72.telecom.net.ar) |
| 19:29.42 | marv[work] | ayeso: the h extension |
| 19:29.42 | [TK]D-Fender | marv[work]: COMPLETE different :) |
| 19:29.59 | ayeso | marv[work]: Ill check it out, thanks for the info |
| 19:30.20 | [TK]D-Fender | ayeso: Who is going to tak to Voicemail if they've hung up? |
| 19:30.34 | [TK]D-Fender | talk* |
| 19:30.38 | marv[work] | [TK]D-Fender: i think he means do something after they hung up from the voicemail |
| 19:30.39 | *** join/#asterisk seanmh (n=johndoe@c-69-254-131-168.hsd1.nm.comcast.net) |
| 19:31.24 | therealcircut | chuck norris thats who |
| 19:31.31 | ayeso | [TK]D-Fender: I want to check if the called party has a new voicemail after the voicemail app was called. if not, then i need to send an email to the called party with the souce ani of the caller. |
| 19:31.32 | [TK]D-Fender | marv[work]: If thats the case there is an exit script hook anyway. |
| 19:32.03 | [TK]D-Fender | ayeso: there are scripting hooks in voicemail.conf go read the samples |
| 19:32.12 | *** join/#asterisk propellerhead (n=yogurt2u@host130.190-226-46.telecom.net.ar) |
| 19:32.31 | therealcircut | anyone know what the fxovoltage parameter for the wctdm24xx.ko module does? |
| 19:32.45 | ayeso | [TK]D-Fender: cant say i'v seen anything like that in there, ill take a look. |
| 19:35.27 | lesouvage | [TK]D-Fender: I will pastebin the asterisk code and the cli output when making a call. It shows that ${DIALSTATUS} doesn't have a value. |
| 19:35.40 | *** join/#asterisk Torrieri (n=Torrieri@nelug/crew/torrieri) |
| 19:37.33 | [TK]D-Fender | lesouvage: Forget code, show me your dialplan and its execution. |
| 19:37.41 | [TK]D-Fender | lesouvage: Proof is in the pudding... |
| 19:38.01 | [TK]D-Fender | lesouvage: Don't need anything more than stark reality. |
| 19:38.10 | lesouvage | [TK]D-Fender: I'm working on it. |
| 19:38.19 | [TK]D-Fender | :) |
| 19:38.34 | *** join/#asterisk M1s3ry (n=M1s3ry@boromir.api-digital.com) |
| 19:41.53 | lesouvage | [TK]D-Fender: see http://www.pastebin.ca/1431316 |
| 19:43.13 | *** join/#asterisk jtodd (i=ubt1z1mz@ns.fox-den.com) |
| 19:43.13 | *** mode/#asterisk [+o jtodd] by ChanServ |
| 19:44.31 | lesouvage | [TK]D-Fender: see is not a command it is more a short version of "could you please check my pastebin and see if it makes any sense":-) |
| 19:45.57 | ajohnson | I'm having an issue with ODBC voicemail storage. When Asterisk goes to insert a voicemail message into the database, I get an error: http://pastebin.com/de55f5d8 |
| 19:46.16 | ajohnson | ODBC realtime is working for my queues, sip peers, voicemail users, etc |
| 19:46.28 | *** join/#asterisk chendy (n=chatzill@58.61.40.229) |
| 19:47.07 | ajohnson | Happening on 1.6.0, .1, and .2-beta2 |
| 19:47.08 | beek | ajohnson: Does the user you're connecting as have SELECT,UPDATE,DELETE,INSERT rights on that table? |
| 19:47.17 | ajohnson | Yes, root |
| 19:47.22 | ajohnson | (during testing) |
| 19:47.41 | *** join/#asterisk chendy (n=chatzill@58.61.40.229) |
| 19:47.47 | ajohnson | but I'm also using it for CDR, SIP realtime, and those also involve inserting/updating |
| 19:48.33 | beek | Any errors appearing in your MySQL logs? |
| 19:49.01 | ajohnson | Hard to tell |
| 19:49.19 | ajohnson | I have query logging turned on |
| 19:49.36 | ajohnson | but the binary data that gets logged screws up my terminal when I use tail to follow it |
| 19:49.49 | wdoekes | non-null columns in your voicemail table, not mentioned in the insert? |
| 19:49.53 | ajohnson | the insert statement gets issued, and binary data is attached, but it does not get inserted |
| 19:50.25 | ajohnson | only non-nul field is msgnum |
| 19:51.03 | ajohnson | all fields appear to be populated when I look at the query log |
| 19:51.04 | wdoekes | use less or cat -v to view the binary log |
| 19:51.36 | [TK]D-Fender | lesouvage: .. OK, WTF are you doing here? |
| 19:51.49 | ajohnson | thanks, I must not have binary logging enabled |
| 19:51.54 | wdoekes | or enable text logging of all queries (I do not know what it does with binary data though) |
| 19:52.05 | [TK]D-Fender | lesouvage: You aren't even ISSUING a dial, how the hell are you supposed to get a DIALSTATUS? |
| 19:52.17 | ajohnson | I have text logging of all queries enabled |
| 19:52.22 | ajohnson | ls |
| 19:52.27 | ajohnson | wrong window :) |
| 19:52.52 | wdoekes | so when you issue the insert yourself, all is well? |
| 19:52.54 | ajohnson | ahh cat -v is much better, didn't screw up my terminal |
| 19:53.20 | ajohnson | it contains a BLOB, so I'm not sure how to handle that when doing an insert (other than leave it null) |
| 19:54.06 | wdoekes | me neither.. no experience with blobs.. sorry :) |
| 19:55.24 | lesouvage | [TK]D-Fender: I tried it with inbound and outbound dialing. |
| 19:55.34 | [TK]D-Fender | lesouvage: You aren't even DIALING. |
| 19:57.43 | lesouvage | [TK]D-Fender: I will run the same on a 1.4.18.1 box. |
| 19:57.50 | [TK]D-Fender | lesouvage: YOU HAVE NO DIAL!!!!!!!!! |
| 19:57.52 | ajohnson | muahha |
| 19:58.00 | [TK]D-Fender | lesouvage: I don't care what you run it on. |
| 19:58.10 | ajohnson | wdoekes: Thank you |
| 19:58.13 | [TK]D-Fender | lesouvage: DIALSTATUS is set on EXITING a dial |
| 19:58.26 | ajohnson | there is missing information in the documentation |
| 19:58.32 | [TK]D-Fender | lesouvage: You can't get a result without taking an action. |
| 19:58.47 | ajohnson | the insert included with the sources as documentation to create the table is wrong, it is missing a field |
| 19:59.08 | wdoekes | :) |
| 19:59.23 | ajohnson | I'm just going to verify and then see if I can submit something on bugs |
| 19:59.45 | watchy | hardwire: a company i do business with does it |
| 20:00.07 | ajohnson | woot, that was it |
| 20:04.27 | *** join/#asterisk jcims (n=chatzill@oh-69-34-176-18.sta.embarqhsd.net) |
| 20:04.42 | jcims | hey folks, are there any good sites for hosted pbx reviews? |
| 20:05.18 | jcims | i'm looking at coredial and jive, prices vary a bit but the features are both the same. i'm pretty sure both use asterisk |
| 20:07.12 | [TK]D-Fender | jcims: Does it matter what they use? In going with a hosted solution you generally have no say in anything |
| 20:07.28 | *** join/#asterisk lost_soul (n=shawn@cpe-67-241-67-197.twcny.res.rr.com) |
| 20:07.41 | jcims | [TK]D-Fender: no, not really...just sometimes lends to a common feature set |
| 20:08.04 | [TK]D-Fender | jcims: Suppose tahts all fine & dandy... |
| 20:10.38 | *** join/#asterisk pewsh (n=pjf@obey.org) |
| 20:11.45 | therealcircut | woot |
| 20:11.47 | therealcircut | works |
| 20:11.49 | therealcircut | later boys |
| 20:14.08 | ajohnson | Where is the info for submitting a core dump? |
| 20:14.37 | Qwell | ajohnson: issues.asterisk.org |
| 20:14.57 | Nugget | Qwell has "issues" |
| 20:15.02 | vasundhar | \q |
| 20:15.08 | vasundhar | \quit |
| 20:15.16 | *** join/#asterisk seanmh (n=johndoe@c-69-254-131-168.hsd1.nm.comcast.net) |
| 20:15.24 | *** part/#asterisk subl (i=sublime@xmission.xmission.com) |
| 20:17.06 | ajohnson | hrm |
| 20:17.22 | ajohnson | having difficulty finding it, though I know I've seen it before |
| 20:17.35 | [TK]D-Fender | Nugget: No, Qwell helps on the bug tracker.. he SOLVES problems :) |
| 20:18.12 | Nugget | It can be both! |
| 20:18.50 | ajohnson | when in doubt, google |
| 20:19.23 | Qwell | when in doubt, visit the URL people tell you to? O.o |
| 20:20.04 | ajohnson | Qwell: Which would be nice, if the information was on the page given |
| 20:20.07 | ajohnson | or at least easy to find |
| 20:21.59 | neurosys | oh this is gonna get good ;) |
| 20:22.13 | *** join/#asterisk ingenius (n=alektro@host176.190-230-72.telecom.net.ar) |
| 20:28.38 | [TK]D-Fender | checkout time, BBIAB |
| 20:31.14 | hardwire | watchy: testing that today. |
| 20:32.34 | *** join/#asterisk telnettech (n=telnette@cpe-71-74-91-116.insight.res.rr.com) |
| 20:33.24 | *** join/#asterisk Whitor (n=Whitor@cpe-74-76-185-31.nycap.res.rr.com) |
| 20:39.02 | *** join/#asterisk nsgn (n=nsgn@cpe-24-27-55-224.austin.res.rr.com) |
| 20:40.03 | *** join/#asterisk nicob (n=nibou@chez.nicolas.bouthors.org) |
| 20:40.23 | nicob | hello, I have some trouble setting up my digium with asterisk anyone can help ? |
| 20:41.19 | *** join/#asterisk SlipperyChicken (n=andrew@LONDON14-1168107385.sdsl.bell.ca) |
| 20:43.22 | *** join/#asterisk porche (n=kursad@88.233.134.236) |
| 20:43.27 | porche | hiya |
| 20:43.37 | porche | I have a question about faxing with asterisk |
| 20:43.49 | porche | which solution is the most stable? |
| 20:44.43 | *** join/#asterisk DarkLogik (n=darklogi@76.73.51.195) |
| 20:49.45 | coppice | the stability of the FAX solutions mostly comes down to the stability of the timing you are achieving in your particular Asterisk installation |
| 20:52.30 | porche | hmms |
| 20:52.44 | porche | so there is no stable solution? |
| 20:53.06 | asteriskmonkey | go with hylafax :) it works nice with iaxmodem |
| 20:53.11 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194) |
| 20:54.37 | Nugget | I need to write an irssi plug-in that stick an "s" in porche's nick. |
| 20:54.53 | *** join/#asterisk LND (n=LND@user-514d70b0.l2.c2.dsl.pol.co.uk) |
| 20:55.12 | coppice | if your timing is stable you'll have stable results. there are many many installations handling hundreds of thousands of FAXes a day using iaxmodem+hylafax and using app-fax |
| 20:55.24 | nsgn | can anyone offer any helpful advice for a first timer trying to configure one polycom phone with asteriskNOW? |
| 20:56.09 | [TK]D-Fender | nsgn: Go download the Admin guide & firmware appropriate to your phone and hop to it... |
| 20:56.21 | [TK]D-Fender | nsgn: http://downloads.polycom.com/voice/voip/sip_sw_releases_matrix.html |
| 20:56.22 | *** join/#asterisk seb- (n=seb@li30-51.members.linode.com) |
| 20:56.34 | nsgn | [TK]D-Fender: clicking, thanks |
| 20:57.01 | seb- | anyone need to test their sip phone? i need to test my * server....i'll pm you my password if you want to call |
| 20:57.50 | [TK]D-Fender | seb-: I'm home & can test |
| 20:57.57 | seb- | [TK]D-Fender: you rock! |
| 20:58.20 | porche | coppice, so you point to iaxmode + hylafax |
| 20:58.32 | porche | there is a commercial one from asterisk, ever used? |
| 20:58.35 | porche | sorry |
| 20:58.39 | porche | from digium |
| 20:59.28 | coppice | there don't seem to be many reports about that one. there are quite a few reports from people having problems setting it up, but not many from people who have figured it out |
| 21:00.56 | seb- | [TK]D-Fender: ok..pm'd you the info |
| 21:02.58 | porche | got it |
| 21:03.09 | porche | so it's relatively new, to jump in |
| 21:05.01 | coppice | it doesn't offer anything more than the free options. I thought they would have released V.34 FAXing, but what they released is quite basic |
| 21:05.17 | nicob | my b410 recieves calls fine but says "everyone is busy" when trying to place an outgoing call. Any quick fix for an asteisk newby ? |
| 21:06.56 | *** join/#asterisk frantic667 (n=toffifee@dsbg-4db5c854.pool.einsundeins.de) |
| 21:07.32 | [TK]D-Fender | nicob: pastebin your failed call |
| 21:07.34 | [TK]D-Fender | ~pb |
| 21:07.34 | infobot | [~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/ |
| 21:09.19 | porche | i see |
| 21:09.31 | nicob | [TK]D-Fender: http://pastebin.com/d407f628e |
| 21:10.29 | nicob | (I have setup an IAX fallback right now so the call goes out to another server and out, so users don't complain) |
| 21:10.56 | nicob | line 68 shows my "symptom" |
| 21:10.57 | [TK]D-Fender | nicob: What is your zap interface? |
| 21:11.19 | [TK]D-Fender | nicob: And what is it connected to? |
| 21:11.32 | nicob | errr |
| 21:12.31 | nicob | the digium B410 card is connected to ISDN if that answers the question |
| 21:12.35 | nicob | (sorry I'm new to all this) |
| 21:13.02 | [TK]D-Fender | nicob: Yeah, I'm multi-tasking and it just went over me... You sure a number starting with 00 is legal? |
| 21:13.39 | nicob | 0 is to identify the "outside" route defined in/through FreePBX |
| 21:13.55 | nicob | 06 is the start of my mobile number |
| 21:14.12 | nicob | I'm sure surprised BOTH zeros show in this log |
| 21:14.33 | KyleK | ZAP/g0/00620716234|300| |
| 21:14.42 | KyleK | the zappyy thingy is dialing that |
| 21:14.55 | KyleK | maybe the dialplan needs to eat a zero? |
| 21:15.00 | [TK]D-Fender | nicob: well its dialing 00 to the PSTN |
| 21:15.03 | nicob | fixing that right now |
| 21:15.09 | [TK]D-Fender | nicob: it isn't stripping a 0 off the front |
| 21:16.01 | nicob | same thing |
| 21:16.13 | lesouvage | [TK]D-Fender: I updated my expertise about ${DIALSTATUS} . I seem to have mixed things up leading to asking a not that smart question. |
| 21:16.22 | nicob | changed the config to be 0|0X. |
| 21:16.53 | nicob | and the log still says "Everyone is busy/congested at this time" |
| 21:17.35 | nicob | KyleK: what's the |300| suposed to mean here ? |
| 21:18.02 | eppigy | time limit on dial |
| 21:18.10 | [TK]D-Fender | lesouvage: :) |
| 21:19.12 | nicob | any other clue to hit me with ? |
| 21:20.55 | KyleK | can you try making a call with a specific channel? either the hardware doesn't like you, or maybe its trouble with the grouping |
| 21:21.47 | eppigy | nicob: what is the output of zap show status? |
| 21:21.58 | *** part/#asterisk seb- (n=seb@li30-51.members.linode.com) |
| 21:22.03 | eppigy | clearly the channels in g0 are not up |
| 21:24.07 | nicob | eppigy: only talks about "ZTDUMMY". Smells bad. |
| 21:24.21 | eppigy | yeah bro |
| 21:24.27 | eppigy | that is not good |
| 21:26.08 | nsgn | i'm having one heck of a time just testing out a single polycom phone. i'm trying to use it's web panel to manually point it at asterisk and seem to be failing. input? |
| 21:26.34 | eppigy | i just do it manually |
| 21:26.50 | eppigy | plus I mean what does sip.cfg contain? |
| 21:26.57 | eppigy | it may be set in there |
| 21:27.00 | nicob | but I understood (maybe wrongly) that I didn't need Zap with this B410 and only needed misdn ? |
| 21:27.11 | [TK]D-Fender | eppigy: H'es nowhere near the point of touching provisioning yet |
| 21:27.18 | eppigy | o |
| 21:27.37 | eppigy | bottom line grab the keypad |
| 21:27.43 | eppigy | and set provisioning type |
| 21:27.45 | eppigy | and ip address |
| 21:27.50 | eppigy | OFF TO THE RACES |
| 21:27.51 | [TK]D-Fender | eppigy: He could be and I pointed him that way, but has 0 experience and seems to be trying the web interface as he said |
| 21:27.58 | eppigy | o |
| 21:27.59 | eppigy | :[ |
| 21:28.08 | eppigy | nicob: well brother |
| 21:28.14 | eppigy | if you are calling ZAP channels |
| 21:28.16 | eppigy | you need zap |
| 21:28.20 | eppigy | you know what I mean |
| 21:28.25 | eppigy | YA FEEL ME |
| 21:28.40 | KyleK | nicob: so these incoming calls that work, where do they come from? |
| 21:28.45 | [TK]D-Fender | eppigy: AT&T's "reach out and touch someone" hasn't gone tactile yet ;) |
| 21:28.52 | eppigy | yet |
| 21:28.52 | nicob | KyleK: that's a good question ! |
| 21:28.55 | nicob | checking |
| 21:28.55 | eppigy | being the operative |
| 21:29.00 | eppigy | YET |
| 21:29.30 | [TK]D-Fender | eppigy: I miss the good 'ole days when you could throw 10,000 volts down the line to fry that motherfucker pissing you off ;) |
| 21:29.37 | eppigy | lol |
| 21:29.58 | eppigy | now you have ot drive rto their house |
| 21:30.01 | eppigy | and it is all very messy |
| 21:30.33 | *** join/#asterisk wubbla (n=wubbla@213-33-22-213.adsl.highway.telekom.at) |
| 21:30.37 | wubbla | hi there! |
| 21:30.40 | eppigy | hey girl |
| 21:30.43 | eppigy | what u doin |
| 21:31.22 | wubbla | is not a girl... |
| 21:31.27 | wubbla | ;-) |
| 21:31.30 | eppigy | damn girl dont be liek that |
| 21:31.57 | nicob | so incoming calls DO travel through mISDN. I updated my conf to route outbounds the same way |
| 21:31.59 | KyleK | yea there are no girls on the internet |
| 21:32.07 | nicob | but outgoing calls still don't go out |
| 21:32.15 | KyleK | so whats the error now? |
| 21:32.39 | KyleK | Kirk to enterprise, pastebin us up |
| 21:32.50 | eppigy | COPY IN CHANNEL AT WILL |
| 21:33.56 | [TK]D-Fender | eppigy: My sensei gave me another surprise taht he planned for ANOTHER kyu test for this Sunday, making it the FOURTH in about 2 months time |
| 21:34.12 | nicob | yeah yeah |
| 21:34.21 | nicob | trin' to figure WHAT to paste |
| 21:34.32 | KyleK | kyu test? |
| 21:35.01 | KyleK | nicob: the line that says "i hate you nicob -- not love, mISDN" |
| 21:35.08 | nicob | :) |
| 21:35.36 | nicob | http://pastebin.com/de5d71a2 |
| 21:35.54 | nicob | hates the part saying "port down" |
| 21:37.39 | nicob | hey ! |
| 21:37.56 | nicob | "misdn port up 1" did the trick |
| 21:38.18 | nicob | now it goes out |
| 21:38.27 | nicob | but after a few seconds the port(s) go down again |
| 21:39.09 | nicob | so it's a misdn conf problem right ? |
| 21:39.23 | wubbla | is anyone here familiar with SNOM telephones? |
| 21:39.26 | KyleK | hopefully |
| 21:39.33 | hardwire | wubbla: enough to make a killing |
| 21:39.40 | hardwire | literally.. they have sharp edges. |
| 21:39.44 | hardwire | you could whack somebody. |
| 21:40.08 | wubbla | hardwire: okay then... |
| 21:40.18 | *** part/#asterisk porche (n=kursad@88.233.134.236) |
| 21:40.27 | wubbla | hardwire: is there any working solution to the "pickup-issue"? |
| 21:40.27 | hardwire | wonders if you have a question :0 |
| 21:40.28 | wubbla | ;) |
| 21:40.40 | hardwire | as in you pick up the phone.. and it answers? |
| 21:40.56 | Deeewayne | pickup lines |
| 21:41.05 | hardwire | like "hey baby.. whats your sign"? |
| 21:41.11 | Deeewayne | yes! |
| 21:41.27 | hardwire | wubbla: I'm being serious.. really.. |
| 21:41.35 | hardwire | what's the issue cause maybe I call it the "hangs up on me issue" |
| 21:42.04 | wubbla | hardwire: no i mean pickup in the context of extension monitoring and BLF... |
| 21:42.26 | *** join/#asterisk Greek-Boy (n=greek@41.222.89.77) |
| 21:42.32 | hardwire | dunno.. how would any other phone handle that? |
| 21:42.39 | hardwire | mostly through asterisk itself right? |
| 21:42.44 | hardwire | it would send hints as needed, etc.. |
| 21:44.39 | *** join/#asterisk qdk (n=qdk@0x573d8dd1.bynqu1.dynamic.dsl.tele.dk) |
| 21:44.57 | smth | exten=>s,n,gotoif["${qwe}" = " ${asd}"]? 2:4 Does it work? |
| 21:45.31 | *** part/#asterisk pewsh (n=pjf@obey.org) |
| 21:47.01 | nicob | so how do I tell those stupid ports to *not* go down ? |
| 21:47.11 | nicob | isdn port viagra ? |
| 21:49.49 | [TK]D-Fender | [17:39]<hardwire>literally.. they have sharp edges. <- I work with far sharper ones :) |
| 21:50.52 | [TK]D-Fender | smth: Looks plenty broken to me. |
| 21:53.10 | *** join/#asterisk michaely (n=Mike@207.114.199.107) |
| 21:53.18 | *** part/#asterisk michaely (n=Mike@207.114.199.107) |
| 21:55.01 | smth | [TK]D-Fender] , how can I make it to be conditional by comparing two variable |
| 21:57.22 | nsgn | eppigy: [TK]D-Fender: i've gotten the phone talking to asterisk manually now, and am ultimately trying to test an outgoing call on my new digium PCI 8xFXO card. i'm already painfully confused. dialing 9 gives me a fresh dialtone, but dialing a number gives me all circuits busy |
| 21:57.54 | [TK]D-Fender | smth: You have added whitespace in there that KILLS it |
| 21:58.03 | telnettech | i know this is probably the wrong channel but why would the amportal from freepbx not start when server is rebotted? Why would external incoming and outgoing calls not work thru created trunks but phone to phone internal calls work fine? why would i have to do amportal restart manually before the inbound and outgoing calls work? |
| 21:58.06 | M1s3ry | sngn, check to see what the CLI says when you hear the circuits busy.... more info can be found there. |
| 21:58.26 | [TK]D-Fender | telnettech: You're right... this IS the wrong channel :) |
| 21:58.43 | telnettech | thanks TK |
| 21:58.51 | M1s3ry | @nsgn ^^^ (I've come to find that mac's apparently don't like tab completing with xchat) |
| 21:59.06 | nicob | l1watcher_timeout seems to answer my question |
| 21:59.08 | [TK]D-Fender | telnettech: And check your DISTRO as to why your startup process isn't doing what you like. That isn't even FreePBX's prolem |
| 21:59.21 | nsgn | M1s3ry: hah, its ok. how would i check the cli? sorry for the likely stupid questions, i'm quite new |
| 21:59.23 | nicob | thanks to all for the quick and effective help, pointing to my stupid mistakes |
| 21:59.32 | nsgn | M1s3ry: i've got a keyboard/monitor on the box right now for ease |
| 21:59.32 | telnettech | you are saying it maybe a CentOS problem? |
| 21:59.43 | M1s3ry | nsgn, "asterisk -rvvvvvv" |
| 22:00.06 | bmoraca | telnettech: it's a configuration problem, related to your init scripts. |
| 22:00.24 | telnettech | ok i will check....thanks |
| 22:01.20 | M1s3ry | nsgn, if you can, try to pastbein what you see when the call fails, you'll get better answers from here if we can see what's happening. |
| 22:01.28 | nsgn | M1s3ry: wow, ok, it scrolls by more crap than i can even read in about half a second |
| 22:01.39 | M1s3ry | that happens |
| 22:01.57 | M1s3ry | especially if you have multiple channels open |
| 22:02.00 | nsgn | M1s3ry: FYI, i'm on asteriskNOW and can't really figure out how to even tell if my digium card is even being properly detected or configured |
| 22:02.12 | nsgn | M1s3ry: this is a brand new install, one phone, one POTS line plugged in |
| 22:02.30 | wubbla | is there any special development channel available for asterisk or should i just ask here? |
| 22:03.02 | M1s3ry | wubbla, #asterisk-dev I believe |
| 22:03.20 | M1s3ry | yup |
| 22:03.26 | wubbla | M1s3ry: thanks! |
| 22:03.42 | M1s3ry | nsgn, what version of asterisknow? |
| 22:03.52 | nsgn | M1s3ry: latest. installed today. 1.5 32bit |
| 22:04.59 | M1s3ry | exit out of the CLI and run dahdi_scan to make sure the server sees the card, or dahdi_tool would suffice as well. |
| 22:05.10 | nsgn | M1s3ry: thanks, doing. one sec |
| 22:05.24 | *** join/#asterisk [intra]lanman (n=intralan@freeswitch/developer/intralanman) |
| 22:05.35 | nsgn | "Wildcard TDM800P with VPMADT032" |
| 22:05.42 | nsgn | showing all 8 FXO ports |
| 22:05.59 | nsgn | identified as "Board 1" |
| 22:06.05 | M1s3ry | then at least we know your server sees the card, have you configured it? |
| 22:06.48 | nsgn | M1s3ry: i've got outbound routes and trunks but i FreePBX isnt really giving me further config than that. i feel i'm missing something :) |
| 22:07.29 | M1s3ry | to be honest I don't understand the FreePBX GUI nor do I want to. |
| 22:07.44 | M1s3ry | but that's just me. |
| 22:07.48 | nsgn | M1s3ry: that's ok. anything you can teach me in concept or directions you can point me is much appreciated |
| 22:08.20 | M1s3ry | go back into the CLI , and do "dahdi show channels" see if you have any channels configured. |
| 22:08.40 | nsgn | M1s3ry: ah, channels. now this is coming back to me. i played with * about 5 years ago |
| 22:09.06 | nsgn | M1s3ry: ahha! no channels |
| 22:09.27 | M1s3ry | sounds like an issue. :) |
| 22:09.28 | nsgn | and apparently no way to configure channels on freepbx? |
| 22:09.31 | nsgn | i cant find one |
| 22:09.50 | telnettech | nsgn: here is the link to freepbx documents....this may help |
| 22:09.50 | M1s3ry | I'm sure there is, but I couldn't help with that. |
| 22:09.52 | telnettech | http://www.freepbx.org/support/documentation/getting-started |
| 22:09.55 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 22:10.02 | telnettech | or try the freepbx channel |
| 22:11.02 | M1s3ry | telnettech, docs ftw! |
| 22:11.06 | telnettech | reading the docs are good for your heart...less stress |
| 22:11.36 | M1s3ry | hits up a few redbulls.... |
| 22:11.42 | M1s3ry | I'm ready to read now! |
| 22:13.55 | *** join/#asterisk Alborracho (n=chatzill@190.25.135.1) |
| 22:26.18 | nsgn | where is zapata-auto.conf located? |
| 22:26.37 | Chuggs | Anyone doing a Municipal * setup in Alberta with a Telus PRI? |
| 22:29.40 | nsgn | anyone know why when i run try to run genzaptelconf i just get "command not found"? |
| 22:30.12 | [TK]D-Fender | nsgn: Maybe you don't have Zaptel installed? |
| 22:30.18 | *** join/#asterisk lasko (n=lasko@70.102.15.210) |
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| 22:30.30 | *** part/#asterisk lasko (n=lasko@70.102.15.210) |
| 22:30.52 | nsgn | [TK]D-Fender: does it not come with asteriskNOW? |
| 22:31.15 | *** join/#asterisk BadHAL (n=nn@cpe-72-179-194-139.stx.res.rr.com) |
| 22:31.31 | *** join/#asterisk [acer]lanman (n=intralan@173-137-68-3.pools.spcsdns.net) |
| 22:32.14 | [TK]D-Fender | nsgn: probably depends on what VERSION |
| 22:32.22 | nsgn | 1.5 32bit. latest |
| 22:32.46 | [TK]D-Fender | nsgn: Odds are NO and that it comes with DAHDI which is the new name for ZAPTEL |
| 22:33.10 | nsgn | ah, ok. am i reading out-of-date docs when it tells me to run genzaptelconf? is there a newer command, perhaps, reflecting the new name? |
| 22:33.59 | carrar | Are there any uptdate docs? :) |
| 22:34.04 | carrar | besides core show application |
| 22:34.42 | carrar | and the readmes |
| 22:34.46 | carrar | changelogs |
| 22:35.10 | [TK]D-Fender | and stuff |
| 22:35.13 | [TK]D-Fender | and things |
| 22:35.16 | carrar | heh |
| 22:35.18 | [TK]D-Fender | Especially things... |
| 22:35.28 | carrar | and stuff is just cool |
| 22:35.29 | [TK]D-Fender | nsgn: Again, good odds :) |
| 22:35.30 | nsgn | carrar: i'm reading the POTS doc on the freepbx site |
| 22:35.38 | nsgn | clicked right from the docs page onto it |
| 22:35.48 | nsgn | http://www.freepbx.org/support/documentation/administration-guide/interfacing-to-a-pstn |
| 22:36.06 | carrar | freepbx? Never heard of it |
| 22:36.09 | [TK]D-Fender | nsgn: Yes, and keep in mind the DATES for those docs and the fact that FreePBX is a THIRD PARTY APP |
| 22:36.33 | nsgn | sure. which is why i'm asking if the command they're giving me is out of date or if my system simply isnt configured properly |
| 22:37.07 | *** join/#asterisk [acer]lanman (n=intralan@173-137-68-3.pools.spcsdns.net) |
| 22:37.27 | *** join/#asterisk Alborracho (n=chatzill@190.25.135.1) |
| 22:37.47 | Alborracho | how can i see this kind of debug in cli ? ast_log(LOG_WARNING, "I don't know about this dtmf mode: %s\n",mode); |
| 22:38.04 | Alborracho | or in /var/log/asterisk/full (messages) |
| 22:38.18 | [TK]D-Fender | Alborracho: why are trying to find ways to see errors you don't have? |
| 22:39.34 | Alborracho | because i need to find a solution to my dtmf problem and debugging the chan_sip is the only thing left for me |
| 22:40.09 | [TK]D-Fender | Alborracho: So far I see you looking for backup for problems noone has confirmed you have. I don't see any debug... |
| 22:41.02 | Alborracho | the thing is in the telco they have another provider like me but they use "apex" platform |
| 22:41.16 | Alborracho | and the can use dtmf in g729 with no problem |
| 22:41.33 | Alborracho | so they keep telling me that my platform is the problem |
| 22:42.58 | [TK]D-Fender | Alborracho: And I still don't see debug for your failures, nor configs to match |
| 22:43.02 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 22:43.41 | Alborracho | i just need based arguments to prove them im right |
| 22:43.56 | Alborracho | and the only arguments i have is the trace |
| 22:44.00 | Alborracho | and debugs |
| 22:44.14 | [TK]D-Fender | Alborracho: You've got them.... we don't |
| 22:45.39 | *** join/#asterisk lesouvage (n=lesouvag@82.73.69.76) |
| 22:46.22 | Alborracho | i dont understand... |
| 22:48.54 | nsgn | alright, my reading tells me that genzaptelconf should be current, and that my issue is that i've got no channels defined in zapata.conf. i'm having a difficult time at this point determining why this is the case |
| 22:49.01 | [TK]D-Fender | Alborracho: i don't see your debug anywhere. |
| 22:49.08 | [TK]D-Fender | Alborracho: wHAT PART OF THIS IS NOT CLEAR? |
| 22:49.40 | [TK]D-Fender | nsgn: It isn't |
| 22:50.07 | nsgn | [TK]D-Fender: which isnt? |
| 22:50.21 | [TK]D-Fender | nsgn: It isn't CURRENT |
| 22:51.27 | nsgn | i know zapata.conf is what is currently being sought by FreePBX because it says on the "Add Trunk" page that a Zap Identifier can either be a group or a channel defined in zapata.conf |
| 22:51.50 | *** join/#asterisk jmacz (n=jmacz@190.144.75.22) |
| 22:51.58 | nsgn | my issue is that i have no zapata.conf file, and the only utility i can find that is supposed to create it, genzaptelconf, doesnt seem to exist |
| 22:53.09 | [TK]D-Fender | Go ask in FreePBX how to better build these with their DAHDI replacements |
| 22:53.21 | [TK]D-Fender | #freepbx |
| 22:53.53 | nsgn | [TK]D-Fender: alright, i'll hit them up with it. thanks |
| 22:55.26 | *** join/#asterisk ctp (n=ctp@brsg-d9beed54.pool.mediaWays.net) |
| 22:55.29 | *** part/#asterisk M1s3ry (n=M1s3ry@boromir.api-digital.com) |
| 23:02.28 | nsgn | [TK]D-Fender: they don't seem to be on top of that. it seems to be that, on the command line level, i simply need to configure my TDM800p properly |
| 23:03.58 | *** join/#asterisk war9407 (i=war@liquidswords.org) |
| 23:20.42 | nsgn | could someone kindly help me configure FXO ports to be active channels? i've been googling and asking specific questions for nearly two hours now |
| 23:20.49 | nsgn | and have gotten essentially nowhere |
| 23:26.30 | *** join/#asterisk djin (n=djin@84-104-110-179.cable.quicknet.nl) |
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| 23:35.43 | nsgn | argh, for the love of god, i've got everything except getting my digium's FXO ports to be added as channels |
| 23:35.51 | nsgn | "dahdi show channels" still shows me blank |
| 23:38.52 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 23:50.19 | nsgn | ..anybody? |
| 23:57.32 | nsgn | my pissue is a blank chan_dhadi.conf file |
| 23:57.34 | nsgn | *issue |
| 23:57.39 | nsgn | though pissue isn't far from it |
| 23:58.23 | nsgn | why is this file blank? |
| 23:58.29 | nsgn | what am i supposed to populate it with? |
| 23:58.38 | nsgn | and is there a script/utility to populate it for basic setups? |
| 23:58.47 | jaytee | have you tried running dahdi_cfg -vvvv |