| 00:03.05 | p3nguin | Are you going to be able to do that soon? |
| 00:03.39 | Atriks | I noticed you |
| 00:03.43 | p3nguin | erm |
| 00:03.46 | p3nguin | Why would you do that? |
| 00:03.59 | Atriks | pm |
| 00:04.04 | p3nguin | Why would you do that? |
| 00:04.17 | Atriks | It's for you |
| 00:04.28 | p3nguin | *sigh* |
| 00:04.38 | p3nguin | That's not how we do things here. |
| 00:04.45 | p3nguin | Private support will cost you. |
| 00:04.58 | Atriks | urf |
| 00:05.30 | p3nguin | There shouldn't be any sensitive information in a sip debug. |
| 00:05.30 | Atriks | http://pastebin.com/cGTERyW6 |
| 00:05.39 | Atriks | just my name yes |
| 00:05.43 | Atriks | never mind |
| 00:07.17 | p3nguin | It appears that your ITSP requires you to be registered before they will accept calls from you. |
| 00:07.27 | p3nguin | Did you create a register statement in sip.conf? |
| 00:07.37 | Atriks | I don't think so |
| 00:07.42 | p3nguin | You'll have to do that. |
| 00:07.51 | Atriks | how to ? |
| 00:08.02 | p3nguin | It goes in the general section. Above the authentication section if it exists, before any peer entries. |
| 00:08.24 | Atriks | oh |
| 00:08.33 | Atriks | you mean register => user:pass@host ? |
| 00:08.34 | p3nguin | register => user[:secret[:authuser]]@host[:port][/extension] |
| 00:08.37 | p3nguin | exactly |
| 00:08.43 | Atriks | already done |
| 00:08.52 | p3nguin | What does "sip show registry" say? |
| 00:09.14 | Atriks | freephonie.net:5060 N 0953534363 1785 Registered |
| 00:09.26 | p3nguin | Registered. |
| 00:09.29 | p3nguin | hmm |
| 00:09.36 | p3nguin | So why would the sip debug indicate that you aren't registered? |
| 00:09.48 | Atriks | I don't know :x |
| 00:10.17 | p3nguin | Any chance you mistyped the user or secret in the peer entry? |
| 00:10.20 | Atriks | during sip reload : |
| 00:10.22 | Atriks | WARNING[7813]: chan_sip.c:18305 handle_response_register: Got 423 Interval too brief for service 0953534363@freephonie.net, minimum is 1800 seconds |
| 00:10.45 | p3nguin | 30 minutes! |
| 00:12.02 | Atriks | <PROTECTED> |
| 00:12.02 | Atriks | [Aug 21 11:11:47] NOTICE[7813]: chan_sip.c:21594 handle_request_subscribe: Received SIP subscribe for peer without mailbox: Hatrix |
| 00:12.30 | p3nguin | The subscribe thing is normal when you didn't create a mailbox for that user. You can safely ignore it for now. |
| 00:12.45 | Atriks | But then when I try to call |
| 00:12.46 | Atriks | http://pastebin.com/MSZ7fkJB |
| 00:13.55 | p3nguin | I'm curious. Show me your entire peer entry for the ITSP and your phone. MASK your passwords; don't want to see passwords. |
| 00:14.48 | Atriks | k |
| 00:15.29 | Atriks | http://pastebin.com/6x4kEFqS |
| 00:16.37 | p3nguin | I think your ITSP might want more information from you for calling. Let me give you my config example. |
| 00:17.04 | p3nguin | http://pastebin.com/tER2jGnY |
| 00:17.51 | Atriks | my phone is at the end of the file |
| 00:18.03 | Atriks | hum ok |
| 00:18.04 | Atriks | as you |
| 00:18.04 | Atriks | :) |
| 00:18.06 | p3nguin | Some ITSPs need an insecure setting, some need the trustrpid/sendrpid settings. |
| 00:18.34 | p3nguin | I see you don't have username or defaultuser in your peer for the ITSP. |
| 00:18.51 | p3nguin | Which asterisk version do you use? |
| 00:19.08 | Atriks | do you have a command to know it ? |
| 00:19.15 | p3nguin | core show version |
| 00:19.41 | Atriks | Asterisk 1.6.2.9-2+squeeze3 |
| 00:20.05 | p3nguin | username should work in that one. Later versions would need defaultuser. |
| 00:20.14 | p3nguin | Add your username to the freephonie peer. |
| 00:20.28 | Atriks | replace fromuser to username ? |
| 00:20.53 | p3nguin | If you don't need fromuser, yes. If you need fromuser, you'll want to keep it and add username. |
| 00:21.09 | Atriks | I keep it |
| 00:21.21 | p3nguin | I don't use fromuser in my configs. |
| 00:21.54 | Atriks | Calling |
| 00:21.58 | Atriks | seems to work |
| 00:22.01 | Atriks | No |
| 00:22.02 | Atriks | :x |
| 00:22.10 | Atriks | service unavailable, but later |
| 00:22.13 | p3nguin | Save changes, run sip reload |
| 00:22.18 | Atriks | already done |
| 00:23.57 | *** join/#asterisk postconf (~postconf@msfree.com) |
| 00:24.59 | *** join/#asterisk phix (~threat@123-243-44-131.static.tpgi.com.au) |
| 00:25.03 | *** join/#asterisk drynish (~drynish@modemcable016.169-21-96.mc.videotron.ca) |
| 00:25.19 | drynish | Who is using debian here and is compiling asterisk on it? |
| 00:25.22 | drynish | Is it feasible? |
| 00:25.35 | Atriks | yes |
| 00:25.35 | *** join/#asterisk razu (~razu@2001:ad0:1:1:202:b3ff:fe36:921c) |
| 00:25.38 | drynish | Ok |
| 00:25.45 | drynish | what's the secret? |
| 00:25.50 | p3nguin | Feasible to install Asterisk on a Debian system? What kind of weird question is that? |
| 00:25.57 | drynish | I'm sorry :) |
| 00:26.02 | Atriks | aptitude install asterisk |
| 00:26.14 | drynish | But whenever I'm not using the asterisk package |
| 00:26.26 | drynish | it fails to works... I get segfault all the time |
| 00:26.57 | postconf | you might try emailing the package maintainer, ask for a Makefile... |
| 00:28.28 | drynish | it's so broken in so many packages |
| 00:28.39 | drynish | dunno why |
| 00:28.57 | drynish | I will try to make it work |
| 00:29.11 | p3nguin | What steps are you taking to do it? |
| 00:29.43 | drynish | svn download libpri |
| 00:29.53 | p3nguin | You're using a PRI? |
| 00:30.16 | drynish | no |
| 00:30.22 | p3nguin | Why do you need libpri? |
| 00:30.22 | drynish | But a dahdi interface card |
| 00:30.30 | p3nguin | oh |
| 00:30.37 | p3nguin | You're going to run analog phones? |
| 00:30.44 | drynish | Yes |
| 00:30.48 | p3nguin | I see |
| 00:31.18 | drynish | no... just linking to a landline |
| 00:31.22 | drynish | and also using an ATA |
| 00:31.25 | drynish | to my home phone |
| 00:31.30 | p3nguin | Is something wrong with libpri and dahdi from the repos? |
| 00:31.58 | drynish | It's compiling well |
| 00:32.06 | drynish | and when I use debian ones, it works well |
| 00:35.15 | drynish | oups sorry |
| 00:35.23 | drynish | libpri and dahdi seems to compile correctly |
| 00:35.28 | drynish | but when I use them, segfault |
| 00:35.34 | drynish | I will just try to debug it |
| 00:36.40 | postconf | what does strace say? |
| 00:38.23 | drynish | forgot to try it |
| 00:38.24 | drynish | let me check |
| 00:40.35 | *** join/#asterisk LemensTS (~matthew@adsl-70-238-144-220.dsl.stlsmo.sbcglobal.net) |
| 00:40.41 | LemensTS | http://support.dell.com/support/edocs/systems/dime521/en/SM_EN/specs.htm and http://www.tigerdirect.com/applications/searchtools/item-details.asp?EdpNo=6611056&SRCCODE=WEBLET03ORDER&cm_mmc=Email-_-WebletMain-_-WEBLET03ORDER-_-Deals is that motherboard and that processor compatible? |
| 00:42.41 | Atriks | search for socket |
| 00:42.51 | Atriks | AMD Athlon⢠64 X2 dual-core processor |
| 00:43.08 | Atriks | it should |
| 00:50.59 | LemensTS | Atriks: Thanks, thats what I ordered I can't get it to work for anything. Done dozens of Intel machines, think this is first AMD...lol |
| 00:51.34 | Atriks | Rho |
| 00:51.35 | Atriks | AMD rox |
| 00:51.38 | Atriks | Better than intel |
| 00:51.48 | p3nguin | I'd rather use Intel for servers. |
| 00:51.55 | Atriks | servers ok |
| 00:51.59 | p3nguin | They have better cache. |
| 00:52.20 | p3nguin | Back in the day, I only ran AMD for desktop machines. |
| 00:53.14 | *** join/#asterisk Guest8383 (~Geek@unaffiliated/cain) |
| 00:56.55 | *** join/#asterisk BuenGenio (~BuenGenio@059148208218.ctinets.com) |
| 00:57.33 | *** part/#asterisk LemensTS (~matthew@adsl-70-238-144-220.dsl.stlsmo.sbcglobal.net) |
| 01:00.38 | *** join/#asterisk drynish (~drynish@modemcable016.169-21-96.mc.videotron.ca) |
| 01:01.09 | drynish | My strace is saying that /dev/dahdi/channel was just before the segfault |
| 01:02.12 | Atriks | so p3nguin it still doesn't work :s |
| 01:02.15 | *** join/#asterisk jayson_r (~jayson@cpe-071-076-046-081.sc.res.rr.com) |
| 01:02.24 | p3nguin | Did you already paste another sip debug? |
| 01:02.39 | Atriks | one is suffisant |
| 01:02.45 | Atriks | I didn't change something |
| 01:02.51 | p3nguin | You were supposed to. |
| 01:02.59 | Atriks | Ok |
| 01:03.01 | Atriks | I'll do |
| 01:04.58 | Atriks | http://pastebin.com/cbc2wUrb p3nguin |
| 01:06.38 | p3nguin | I don't see a call in that paste. |
| 01:06.56 | Atriks | yes |
| 01:07.01 | Atriks | that's a connection |
| 01:07.09 | p3nguin | There's no CALL in that paste. |
| 01:07.15 | p3nguin | I wanted to see a call. |
| 01:07.17 | Atriks | It will |
| 01:07.20 | Atriks | wait |
| 01:07.57 | Atriks | http://pastebin.com/sqVgNJq0 here |
| 01:08.29 | p3nguin | Now I see one. |
| 01:11.21 | p3nguin | Did you ever try adding sendrpid=yes to your freephonie peer? |
| 01:17.34 | Atriks | I'll try |
| 01:17.55 | p3nguin | freephonie.org does not say they need that in the configuration, but I'd try it. |
| 01:18.44 | Atriks | calling |
| 01:18.48 | Atriks | unavailable :( |
| 01:19.47 | Atriks | <--- SIP read from UDP:212.27.52.5:5060 ---> |
| 01:19.47 | Atriks | <PROTECTED> |
| 01:20.03 | ChannelZ | * got a packet from that IP on port 5060 |
| 01:20.14 | Atriks | someone is using my asterisk ? |
| 01:20.28 | ChannelZ | Not necessarily |
| 01:20.33 | p3nguin | freephonie.net has address 212.27.52.5 |
| 01:20.41 | ChannelZ | depends on the rest of the packet and what * did with it |
| 01:20.50 | Atriks | oh ok |
| 01:20.59 | Atriks | I receive packet from it every 2 seconds |
| 01:21.08 | Atriks | but calls don't work |
| 01:21.15 | ChannelZ | Makes sense |
| 01:21.23 | ChannelZ | At least it's persistent. |
| 01:22.11 | p3nguin | My only remaining thought is that you are not sending calls to an extension that they accept. |
| 01:22.59 | p3nguin | They don't say what format they want. |
| 01:25.41 | Atriks | :( |
| 01:25.54 | Atriks | trinkets 6/12 |
| 01:25.57 | Atriks | time 9:38 |
| 01:26.33 | p3nguin | Try calling a number with full country code and area code. |
| 01:27.06 | Atriks | with ident numer ? |
| 01:27.09 | Atriks | like +33 ? |
| 01:27.18 | *** join/#asterisk benklop (~ben@c-67-176-102-107.hsd1.co.comcast.net) |
| 01:27.21 | p3nguin | If 33 is the country code, that's what I mean. |
| 01:27.42 | p3nguin | If I wanted to call you, I'd call 00 33 .... |
| 01:27.52 | benklop | is google voice /gtalk integration broken again? |
| 01:27.57 | p3nguin | yes |
| 01:27.58 | Atriks | call failed : not found |
| 01:28.13 | p3nguin | Try with 00 33 ... |
| 01:28.24 | benklop | it's either that or a firewall issue on my side.. |
| 01:28.31 | Atriks | service unavailable |
| 01:28.34 | benklop | penguin: was that directed at me? |
| 01:28.42 | p3nguin | benklop: yes |
| 01:28.45 | Atriks | But I've no sound with xlite |
| 01:28.50 | ChannelZ | infobot broken gtalk is <reply>see https://issues.asterisk.org/jira/browse/ASTERISK-18301 |
| 01:28.50 | infobot | ChannelZ: okay |
| 01:28.56 | benklop | okay. thanks. |
| 01:29.05 | Atriks | It says something I think, but no sound |
| 01:29.07 | p3nguin | ~broken gtalk |
| 01:29.12 | *** join/#asterisk freeedrich| (~eeePC@hansaserver.de) |
| 01:29.26 | ChannelZ | I think the stupid ~ trigger messes with factoids. |
| 01:29.35 | p3nguin | infobot: broken gtalk |
| 01:29.39 | benklop | is there a place I can look this stuff first so as not to bother you guys next time? google doesnt seem to be relevant enough date waise |
| 01:30.14 | p3nguin | ~broken |
| 01:30.14 | infobot | broken is, like, mailto:nothing@machine.cx -> http://machine.cx/debian/ or screen shots are at http://nivda.machine.cx or that's sid for you. |
| 01:30.19 | p3nguin | ~gtalk |
| 01:30.24 | ChannelZ | ~broken-gtalk |
| 01:30.25 | ChannelZ | There |
| 01:30.41 | ChannelZ | oh you dirty infowhore |
| 01:30.43 | p3nguin | Sometimes I just don't understand that silly thing. |
| 01:31.47 | Atriks | why isn't it wooorkiiing :( |
| 01:32.03 | ChannelZ | Maybe he runs on the same server as JIRA and his disk is full.. cuz we're having a private conversation and he's being a boob too. |
| 01:32.46 | ChannelZ | ~brokengtalk |
| 01:32.53 | ChannelZ | kicks infobot |
| 01:33.09 | ChannelZ | He knows it |
| 01:33.10 | p3nguin | ~channelz |
| 01:33.10 | infobot | methinks channelz is something else |
| 01:33.14 | p3nguin | haha |
| 01:33.40 | ChannelZ | infobot literal brokengtalk |
| 01:33.40 | infobot | "brokengtalk" is "<reply>see https://issues.asterisk.org/jira/browse/ASTERISK-18301" |
| 01:33.42 | p3nguin | I wonder if it picked up someone saying, "channelz: you're something else." |
| 01:33.45 | ChannelZ | see? dummy |
| 01:34.05 | ChannelZ | That's how it normally works, though I don't know if whomever runs infobot has autolearn turned on |
| 01:35.18 | ChannelZ | I run an old original infobot named regurg on another network. |
| 01:35.38 | ChannelZ | I'm not exactly sure which flavor this one is |
| 01:36.12 | Atriks | I'll sleep |
| 01:36.14 | Atriks | I'm tired |
| 01:36.59 | ChannelZ | ~brokengtalk |
| 01:36.59 | infobot | brokengtalk is probably see https://issues.asterisk.org/jira/browse/ASTERISK-18301 for more info |
| 01:37.04 | p3nguin | w00t |
| 01:37.47 | ChannelZ | He wasn't liking the <reply> for whatever reason. Maybe the lack of space after. |
| 01:37.48 | ChannelZ | shrugs |
| 01:38.05 | p3nguin | I've used it with and without the space after reply. |
| 01:38.30 | p3nguin | infobot: brokengtalk |
| 01:38.30 | infobot | hmm... brokengtalk is see https://issues.asterisk.org/jira/browse/ASTERISK-18301 for more info |
| 01:38.51 | p3nguin | no, brokengtalk is <reply>see https://issues.asterisk.org/jira/browse/ASTERISK-18301 for more info |
| 01:38.59 | p3nguin | infobot: no, brokengtalk is <reply>see https://issues.asterisk.org/jira/browse/ASTERISK-18301 for more info |
| 01:38.59 | infobot | okay, p3nguin |
| 01:39.03 | p3nguin | infobot: brokengtalk |
| 01:39.07 | ChannelZ | see |
| 01:39.14 | ChannelZ | BRAIN DAMAGED |
| 01:39.16 | p3nguin | ~brokengtalk |
| 01:39.25 | p3nguin | infobot: no, brokengtalk is <reply> see https://issues.asterisk.org/jira/browse/ASTERISK-18301 for more info |
| 01:39.25 | infobot | p3nguin: okay |
| 01:39.31 | p3nguin | ~brokengtalk |
| 01:39.51 | p3nguin | infobot: brokengtalk is see https://issues.asterisk.org/jira/browse/ASTERISK-18301 for more info |
| 01:39.51 | infobot | ...but brokengtalk is already something else... |
| 01:40.00 | p3nguin | infobot: no, brokengtalk is see https://issues.asterisk.org/jira/browse/ASTERISK-18301 for more info |
| 01:40.00 | infobot | okay, p3nguin |
| 01:40.03 | p3nguin | ~brokengtalk |
| 01:40.03 | infobot | from memory, brokengtalk is see https://issues.asterisk.org/jira/browse/ASTERISK-18301 for more info |
| 01:40.10 | p3nguin | Messed. Up. |
| 01:40.37 | ChannelZ | ~barfo |
| 01:40.37 | infobot | i think i gonna be sick |
| 01:40.53 | ChannelZ | It wants a space between <reply> and the factoid |
| 01:41.11 | ChannelZ | but this will be funner anyway.... |
| 01:41.40 | ChannelZ | infobot no brokengtalk is GTalk is a little busted at the moment... see https://issues.asterisk.org/jira/browse/ASTERISK-18301 for more info |
| 01:41.40 | infobot | ChannelZ: okay |
| 01:41.45 | p3nguin | infobot: poop is <reply>it is just poop! |
| 01:41.45 | infobot | ACTION poops on is <reply>it is just poop! |
| 01:41.55 | p3nguin | erm |
| 01:42.09 | p3nguin | infobot: poopy is <reply>it is just poopy! |
| 01:42.09 | infobot | okay, p3nguin |
| 01:42.15 | p3nguin | infobot: poopy |
| 01:42.15 | infobot | it is just poopy! |
| 01:42.15 | ChannelZ | infobot poop p3nguin |
| 01:42.16 | infobot | ACTION poops on p3nguin |
| 01:42.31 | p3nguin | The lack of space works there. |
| 01:42.35 | p3nguin | infobot: forget poopy |
| 01:42.36 | infobot | p3nguin: i forgot poopy |
| 01:43.06 | p3nguin | I think it's just having another bad day. |
| 01:43.32 | ChannelZ | Well if it's anything like old classic infobot, he has many many quirks |
| 01:43.37 | ChannelZ | Perl gone wrong |
| 01:44.09 | p3nguin | I have an rbot with similar issues. |
| 01:44.25 | *** join/#asterisk Cain (~Geek@unaffiliated/cain) |
| 01:44.43 | *** part/#asterisk postconf (~postconf@msfree.com) |
| 01:45.07 | benklop | well, thank you for the excellent entertainment after the excellent fix. :-P |
| 01:45.07 | Atriks | good night, 03:45 here |
| 01:45.13 | Atriks | see you later ! |
| 01:48.00 | ChannelZ | adios |
| 01:48.50 | ChannelZ | I actually have my own version of infobot half rewritten in PHP. I didn't like most of the infobot derivitaves/clones that are already out there, it messed with my bot's personality too much. |
| 02:01.43 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 02:11.46 | *** join/#asterisk jwiggins (~James@gateway/tor-sasl/jwiggins) |
| 02:17.03 | Nugget | in php? what was your goal, to somehow make an implementation in a language worse than perl? |
| 02:17.56 | drmessano | PHP: Because mIRC doesn't run well in *nix |
| 02:18.02 | Nugget | heh |
| 02:19.49 | *** join/#asterisk kaushal (~kaushal@14.99.152.194) |
| 02:20.04 | *** join/#asterisk godmachine-x6 (~godmachin@unaffiliated/godmachine-x6) |
| 02:24.49 | kaushal | Hi |
| 02:27.27 | ChannelZ | I'm not sure there is a language worse than perl |
| 02:27.37 | Nugget | php |
| 02:28.01 | Nugget | maybe also INTERCAL or something |
| 02:30.07 | ChannelZ | PHP make sense. Perl winds up looking like hieroglyphics most the time |
| 02:30.45 | *** join/#asterisk Clear4ciD (~Clear4ciD@c-76-100-161-152.hsd1.md.comcast.net) |
| 02:30.53 | kaushal | I have this script http://fpaste.org/MZ7U/ when i run this script as sh -xv outbound.sh run, all call does not happen concurrently instead when the first number answers then the second gets answered and so on and so forth. |
| 02:31.04 | kaushal | Any clue ? |
| 02:32.37 | p3nguin | When I wrote the script, it would call an unlimited amount of times, and there is no way for it to care if there is an answer or not. |
| 02:33.04 | *** join/#asterisk gogasca (~gogasca@c-71-202-75-108.hsd1.ca.comcast.net) |
| 02:35.01 | kaushal | the only change i did was line number 17 --> asterisk -rx "channel originate DAHDI/g0/$num Application MP3Player /home/kaushal/obd-demo.mp3" |
| 02:35.32 | p3nguin | Then I have to assume that you're wrong and the script will still generate an unlimited number of calls. |
| 02:36.04 | kaushal | instead of asterisk -rx "originate Local/$num@auto-outbound-spf extension s@auto-outbound" |
| 02:36.23 | p3nguin | I use it to spawn usually like 20 calls at a time, starting just two seconds apart. |
| 02:36.36 | kaushal | p3nguin: ok |
| 02:36.37 | p3nguin | There is no way for the script to know or care about an answer. |
| 02:39.04 | kaushal | p3nguin: shall i pastebin the verbose debug of that script ? |
| 02:39.24 | p3nguin | It isn't necessary. I wrote the thing myself, so I know what it does. |
| 02:41.29 | kaushal | I have tried all options to make it work, the issue is the same |
| 02:41.45 | kaushal | has it to do with line no 17 ? |
| 02:41.55 | p3nguin | There are no options. You put in a list of phone numbers, and run the script during the appropriate hours, and it calls the numbers. |
| 02:42.26 | p3nguin | Is the asterisk cli command to originate calls actually channel originate ... ? |
| 02:42.33 | p3nguin | It wasn't when I wrote it. |
| 02:42.42 | kaushal | yes |
| 02:42.55 | p3nguin | It was and possibly still is just originate ... |
| 02:43.44 | p3nguin | I guess channel originate is the new way, but omitting channel still works. |
| 02:44.28 | kaushal | p3nguin: i tried omitting channel also now |
| 02:44.42 | kaushal | still the same |
| 02:45.17 | *** join/#asterisk Clear4ciD (~Clear4ciD@c-76-100-161-152.hsd1.md.comcast.net) |
| 02:45.43 | p3nguin | I would be more interested to see what happens on the asterisk cli when you run the script. |
| 02:45.58 | kaushal | ok |
| 02:46.41 | kaushal | p3nguin: meaning to inoke the script from CLI > ? |
| 02:46.45 | kaushal | invoke* |
| 02:46.47 | p3nguin | nope |
| 02:46.56 | p3nguin | I mean core set verbose 4, then run the script. |
| 02:47.06 | kaushal | ok |
| 02:47.27 | kaushal | so i would have to have two consoles right ? |
| 02:47.43 | p3nguin | Or learn how to use tmux or screen. |
| 02:47.56 | kaushal | yes i use screen |
| 02:48.06 | p3nguin | Then you need only one console. |
| 02:48.10 | kaushal | so let me set CLI in screen |
| 02:48.16 | kaushal | and then run the script |
| 02:48.24 | kaushal | p3nguin: please give me a moment |
| 02:52.41 | kaushal | p3nguin: http://fpaste.org/ZnLB/ |
| 02:54.28 | p3nguin | I have no idea what you think that shows me. |
| 02:55.34 | ChannelZ | I see boobies! |
| 02:55.40 | kaushal | p3nguin: let me also pastebin the verbose debug mode of the script itself |
| 02:56.07 | p3nguin | The script has no output. |
| 02:57.12 | kaushal | http://fpaste.org/ko0Y/ |
| 03:03.21 | kaushal | please suggest further |
| 03:03.40 | p3nguin | There is nothing else to suggest. I wrote the script and it does what it was written to do. |
| 03:03.52 | p3nguin | I know it does because I use the damn thing myself. |
| 03:04.04 | p3nguin | I don't know what more you want. |
| 03:04.18 | p3nguin | If you don't like what my script does, go write your own. |
| 03:05.05 | kaushal | p3nguin: is there a way if you can accomodate this line originate DAHDI/g0/$num Application MP3Player obd-demo.mp3 in your environment just to ensure if it works for you ? |
| 03:05.16 | p3nguin | Nope. |
| 03:05.20 | kaushal | i will fix it |
| 03:05.55 | kaushal | thats the only difference |
| 03:06.06 | kaushal | I am really failing to understand |
| 03:06.35 | p3nguin | The command being run should have nothing to do with the script. The script just runs asterisk -rx ... over and over and over and over until it find - in the file, then it waits 60 seconds. |
| 03:07.04 | p3nguin | It does not listen for answers, it does not accept feedback. |
| 03:07.13 | p3nguin | That's all it does. |
| 03:07.47 | p3nguin | Using the test command with it will show you exactly what it does. |
| 03:09.25 | kaushal | p3nguin: I completely agree with you |
| 03:09.31 | kaushal | i ran the test too |
| 03:09.45 | kaushal | but its really weird |
| 03:10.01 | kaushal | it doesnot work as expected |
| 03:10.46 | kaushal | otherwise i would not have complained at all in the first instance itself |
| 03:10.51 | kaushal | believe me |
| 03:11.25 | kaushal | I have already shared the details |
| 03:12.01 | p3nguin | And I've told you that the script does not accept feedback, so I guess you're fucked. |
| 03:12.06 | p3nguin | It works for me, and that's what I wrote it for. |
| 03:13.37 | kaushal | anyways thanks |
| 03:13.37 | p3nguin | When you run the test, does it show you that it is calling all the numbers in the list? |
| 03:13.47 | kaushal | yes |
| 03:13.51 | kaushal | it shows |
| 03:14.02 | p3nguin | How many numbers are in your list? |
| 03:14.25 | kaushal | three numbers only to ensure |
| 03:14.31 | kaushal | later i will populate more |
| 03:15.10 | kaushal | all the three numbers are at my desk |
| 03:15.15 | p3nguin | When you run the test, it will output three originate commands. |
| 03:15.39 | p3nguin | Have you tried running the originate commands that are output from the test in your asterisk CLI? |
| 03:15.57 | kaushal | let me test it again |
| 03:16.07 | p3nguin | one, then the second, then the third, in sequence |
| 03:16.15 | p3nguin | That's what the shell script does. |
| 03:16.51 | p3nguin | originate does not wait for the channel to answer before returning to the prompt. |
| 03:17.01 | p3nguin | You can run it over and over and over and over. |
| 03:17.11 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 03:17.20 | kaushal | it works perfectly fine when i ran originate DAHDI/g0/xxxxxxxxxx Application MP3Player /home/kaushal/obd-demo.mp3 |
| 03:17.37 | kaushal | at CLI > |
| 03:17.40 | p3nguin | Now run all three. |
| 03:17.46 | kaushal | ok |
| 03:20.30 | *** join/#asterisk radic (~radic@dslb-178-002-209-094.pools.arcor-ip.net) |
| 03:21.50 | kaushal | p3nguin: i am suspecting the issue here |
| 03:22.00 | kaushal | when i ran all the three at once |
| 03:22.33 | kaushal | it didnot ring the second and third at all |
| 03:22.44 | kaushal | only the first |
| 03:23.29 | kaushal | is it some config issue ? |
| 03:23.47 | kaushal | but whereas when i call individually it works as expected |
| 03:23.49 | p3nguin | What did the verbose output indicate? |
| 03:24.25 | p3nguin | My only limitation is my bandwidth. That's why I usually limit my calls to 20 per batch. |
| 03:26.06 | kaushal | p3nguin: http://fpaste.org/vkwY/ |
| 03:28.06 | p3nguin | That information means nothing to me. |
| 03:28.21 | kaushal | p3nguin: let me set verbose mode to 10 |
| 03:28.30 | p3nguin | It won't change the output. |
| 03:28.49 | kaushal | i am running out of ideas now |
| 03:28.49 | p3nguin | You could change it to 15000 and it would be exactly the same. |
| 03:28.54 | kaushal | :/ |
| 03:30.43 | p3nguin | If you have more than one phone on the system, can two people make outbound calls at the same time? |
| 03:36.19 | kaushal | nope |
| 03:38.56 | kaushal | p3nguin: trying to understand further |
| 03:39.04 | kaushal | please give me a moment |
| 03:39.05 | kaushal | brb |
| 03:47.16 | p3nguin | That must be pretty shitty to have over 200 PRI channels and can only make one call at a time. |
| 04:03.40 | ChannelZ | That he can spam with? Sounds like it's perfect the way it is. |
| 04:09.38 | kaldemar | cause 17 means busy... |
| 04:11.30 | *** join/#asterisk salz212 (~chatzilla@182.178.160.108) |
| 04:18.43 | kaushal | back after further research |
| 04:18.47 | kaushal | when i do |
| 04:18.48 | kaushal | originate DAHDI/g0/xxxxxxxxxx Application MP3Player /home/kaushal/obd-demo.mp3 |
| 04:18.49 | kaushal | originate DAHDI/g1/xxxxxxxxxx Application MP3Player /home/kaushal/obd-demo.mp3 |
| 04:18.58 | kaushal | it still does not work |
| 04:19.14 | kaushal | since g0 and g1 are different call groups |
| 04:20.28 | kaushal | what i have observed here is that there is some sort of time interval |
| 04:21.01 | kaushal | Do i need to revisit the configs under /etc/asterisk ? |
| 04:21.09 | p3nguin | Only being able to make one call at a time must be painful. |
| 04:21.32 | kaushal | yes |
| 04:21.40 | kaushal | p3nguin: i agree with you |
| 04:22.55 | kaushal | p3nguin: the script works perfect |
| 04:23.13 | kaushal | the issue here is at the CLI > prompt |
| 04:24.24 | kaushal | so nailing down the issue |
| 04:24.36 | kaushal | so at a time only one number is being called |
| 04:24.46 | kaushal | thats the observation |
| 04:25.03 | kaushal | inspite of using different call groups |
| 04:25.11 | ChannelZ | Why are you not using call files |
| 04:25.37 | kaushal | ChannelZ: ok |
| 04:25.50 | kaushal | let me look at it |
| 04:25.52 | ChannelZ | Doing asterisk -rx with the channel originate like that is not going to return until the channel answers (or times out) |
| 04:26.17 | p3nguin | It does for me. |
| 04:26.27 | p3nguin | I can throw out 100 originates if I want. |
| 04:26.29 | ChannelZ | Doesn't here |
| 04:26.37 | kaushal | ChannelZ: yeah |
| 04:26.45 | p3nguin | That's why I made the script like I did. |
| 04:26.51 | kaushal | ChannelZ: for p3nguin it works |
| 04:27.12 | kaushal | the only difference is i am using E1 and in US it is T1 |
| 04:27.19 | ChannelZ | I haven't been paying attention much or seen the script to be honest |
| 04:27.22 | p3nguin | I'm not using analog, though. |
| 04:27.34 | *** join/#asterisk atan (atan3@unaffiliated/atan) |
| 04:27.45 | kaushal | ChannelZ: do you wish to see the script ? |
| 04:27.51 | ChannelZ | not really |
| 04:27.55 | p3nguin | I limit my calls to 20 for reasons of bandwidth, but if I wanted to spawn a crapload more, I could. |
| 04:28.00 | ChannelZ | doing it from a shell pretty much behaved as I said |
| 04:28.24 | kaushal | ChannelZ: Any clue about CLI > ? |
| 04:28.43 | ChannelZ | I'm not sure what you even mean |
| 04:29.02 | kaushal | I am pretty sure if it works at CLI > then asterisk -rx should work too |
| 04:29.35 | WIMPy | Good morning! |
| 04:30.00 | kaushal | *CLI> originate DAHDI/g0/xxxxxxxxxx Application MP3Player /home/kaushal/obd-demo.mp3 |
| 04:30.03 | kaushal | *CLI> originate DAHDI/g1/xxxxxxxxxx Application MP3Player /home/kaushal/obd-demo.mp3 |
| 04:30.10 | kaushal | WIMPy: good morning |
| 04:30.32 | kaushal | so when i run that only one number is being called |
| 04:31.24 | kaushal | meaning if i paste both two lines at once |
| 04:31.55 | kaushal | Am i understanding it correctly ? |
| 04:32.31 | ChannelZ | I have no idea |
| 04:33.17 | kaushal | WIMPy: Any clue ? |
| 04:33.18 | p3nguin | I just confirmed it. I now have two phones here ringing at the same time from two asterisk -rx originates. |
| 04:33.30 | kaushal | ok |
| 04:33.39 | kaushal | is it T1 or E1 PRI ? |
| 04:33.42 | WIMPy | Where are you now? |
| 04:34.03 | ChannelZ | As separate processes perhaps but at least here the first asterisk -rx ... doesn't return to the shell prompt until the channel answers |
| 04:34.10 | p3nguin | And I didn't answer the first one... I just ran asterisk -rx 'originate ...' ; asterisk -rx 'originate ...' |
| 04:34.33 | kaushal | WIMPy: are you referring to me ? |
| 04:34.37 | WIMPy | yes |
| 04:34.49 | kaushal | not sure i understand that question |
| 04:35.05 | kaushal | are you talking about error 101 ? |
| 04:36.00 | WIMPy | Let me read... |
| 04:36.07 | ChannelZ | p3nguin: what are you doing with the originate? Running an application, or specifying an extension? |
| 04:36.39 | p3nguin | This test I used application, but in the script, I originate against a local channel to an extension. |
| 04:38.36 | WIMPy | kaushal: You reallt should use AMI so you can get feedback. |
| 04:39.52 | p3nguin | He's saying that he can only make one outgoing call at a time, even from multiple phones. He has well over 200 PRI channels, so I would think he could make as many calls as he wants. |
| 04:40.07 | kaushal | p3nguin: http://fpaste.org/1v3c/ |
| 04:41.03 | WIMPy | Yes, but that was only using the CLI, I guess. |
| 04:41.04 | kaushal | p3nguin: yes i can make multiple calls since i have 200 channels |
| 04:41.20 | p3nguin | You told me that you can't. |
| 04:41.36 | kaushal | sorry |
| 04:41.55 | kaushal | i mean i should be able to make multiple calls |
| 04:42.06 | kaushal | since i have more than 1 channel |
| 04:42.24 | p3nguin | Then you can originate more than one call, too. |
| 04:42.29 | kaushal | but at the moment only one call being made |
| 04:42.46 | kaldemar | and what do you see in pri debug when you try to make two calls? what does the dial line look like? what does your chan_dahdi.conf look like? |
| 04:42.46 | kaushal | surprising |
| 04:42.54 | WIMPy | kaushal: Use AMI. It's the only way forward. |
| 04:43.03 | ChannelZ | In your script, why don't you just do & at the end of the asterisk -rx line to spawn it in the background? |
| 04:43.20 | kaldemar | i saw in an earlier paste that you got a hangup from PRI with cause 17, which means busy. don't dismiss that. |
| 04:43.38 | kaushal | kaldemar: sure and let me pastebin the configs |
| 04:43.41 | p3nguin | Mine returns to the prompt, so I never had a reason to put & on it. |
| 04:43.51 | WIMPy | if you use asterisk -rx or .call files, you won't know how many channels you have available. |
| 04:44.00 | p3nguin | He has plenty. |
| 04:44.23 | ChannelZ | Won't * queue with the call files its self? |
| 04:44.25 | WIMPy | But I guess he wants to use all of them. |
| 04:44.39 | p3nguin | 8 ports, E1 |
| 04:44.55 | WIMPy | ChannelZ: That might actually work. Just check how many are left. |
| 04:46.13 | kaushal | kaldemar: http://fpaste.org/kGDq/ |
| 04:46.50 | p3nguin | I don't need any feedback, so I'm satisfied with throwing a batch of asterisk -rx at my asterisk. |
| 04:47.41 | kaushal | kaldemar: let me know if you need more information about configs |
| 04:47.49 | p3nguin | But if I knew how to write stuff for AMI, I probably would have used it. |
| 04:48.14 | kaushal | WIMPy: AMI -> Asterisk Management interface ? |
| 04:48.17 | WIMPy | Well, thinking about it, Asterisk shouldn't care if it it able to dial out when processing .call file. You just could use the retry. So that's not that effective. |
| 04:48.30 | WIMPy | yes |
| 04:48.30 | ChannelZ | WIMPy: what I meant was, if you put a call file in the spool and the channel you set it for is unavailable (let's just say there's only 1) does the call file immediately fail? |
| 04:48.52 | WIMPy | What else could it do? |
| 04:49.04 | ChannelZ | With retry, it'll wait and try again. |
| 04:49.06 | p3nguin | If you have no retry, it should be removed. |
| 04:49.14 | ChannelZ | I think anyway, I don't really use them but thought that was the point |
| 04:49.22 | WIMPy | yes |
| 04:49.31 | ChannelZ | There's an archive function that will put the call file in a new directory and tell you what happened to it even. |
| 04:52.05 | kaldemar | kaushal: i made three questions and you answered one. |
| 04:52.54 | kaushal | kaldemar: please give me a moment |
| 04:56.07 | kaushal | kaldemar: http://fpaste.org/BRUa/ |
| 05:00.45 | kaldemar | kaushal: no pri debug there. |
| 05:01.05 | kaushal | its set to verbosity of 4 |
| 05:01.31 | p3nguin | That has very little to do with pri debug. |
| 05:01.48 | kaushal | ok |
| 05:01.57 | kaushal | got it |
| 05:02.07 | kaushal | let me enable it and then pastebin it |
| 05:02.13 | kaushal | please give me a moment |
| 05:05.39 | kaldemar | not very little but nothing. pri intense debug span ... |
| 05:06.13 | kaushal | sure |
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| 05:20.27 | p3nguin | It takes so long. |
| 05:20.59 | kaushal | back again |
| 05:21.00 | kaushal | http://sprunge.us/TZPB |
| 05:23.01 | kaushal | brb after sometime |
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| 05:30.25 | WIMPy | Looks ok |
| 05:33.44 | *** join/#asterisk Atriks (~Hatrix@2a01:e34:ee51:4670:61c4:c8e5:69b7:a868) |
| 05:48.34 | kaldemar | the debug does not reflect the described situation though. |
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| 05:53.44 | singler | is where problems with asterisk's jira? |
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| 06:03.23 | Atriks | wake up |
| 06:03.28 | Atriks | Hi |
| 06:03.33 | singler | hi |
| 06:04.03 | Atriks | I'm a problem with my asterisk server. When I try to make a call with x-lite, it says "service unavailable" after few seconds |
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| 06:10.01 | Atriks | Everybody's sleeping ? |
| 06:10.17 | WIMPy | I wish I would |
| 06:11.05 | kaldemar | Atriks: what do you see in CLI when you make a call? |
| 06:11.19 | Atriks | Wait |
| 06:11.23 | Atriks | I've a debug somewhere |
| 06:11.43 | Atriks | http://pastebin.com/sqVgNJq0 here |
| 06:11.44 | *** join/#asterisk kleszcz (tick@80.54.23.253) |
| 06:12.40 | Atriks | I was told that it's a problem with the connection to the itsp |
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| 06:15.01 | kaldemar | Atriks: SIP/2.0 403 not registered received from the ITSP. |
| 06:15.05 | *** join/#asterisk felimwhiteley (~quassel@46.7.101.58) |
| 06:15.26 | Atriks | Yeah, but I register in sip.conf |
| 06:16.00 | *** join/#asterisk felimwhiteley (~quassel@46.7.101.58) |
| 06:16.19 | kaldemar | Atriks: seems like they require you to register before being able to make calls or there is something wrong with your configuration. pastebin the dial line from extensions.conf and the the used peer from sip.conf if there is one. |
| 06:16.46 | kaldemar | so registration should not be the issue. probably the credentials are not what they should be. |
| 06:18.18 | Atriks | http://pastebin.com/jJNkasya |
| 06:18.29 | Atriks | I've not many in extension.conf |
| 06:18.53 | Atriks | I've a register => in general in sip.conf |
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| 06:23.50 | kaldemar | Atriks: add defaultuser parameter to the peer definition. |
| 06:24.21 | Atriks | to hatrix ? |
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| 06:24.37 | Atriks | hum |
| 06:24.45 | Atriks | No, freephonie, defaultuser=Hatrix ? |
| 06:25.20 | kaldemar | to freephonie. and set it to what ever you use to register to them with. |
| 06:26.02 | Atriks | Have I to write my Asterisk username or itsp username? |
| 06:26.57 | kaldemar | itsp username naturally, since your dialing to the itsp with it. |
| 06:27.53 | Atriks | <PROTECTED> |
| 06:27.53 | Atriks | <PROTECTED> |
| 06:28.39 | kaldemar | yes, because you have a name as the host. |
| 06:28.57 | Atriks | and WARNING[7813]: chan_sip.c:18305 handle_response_register: Got 423 Interval too brief for service 0953534363@freephonie.net, minimum is 1800 seconds |
| 06:28.58 | Atriks | <PROTECTED> |
| 06:29.47 | kaldemar | you're registering too often to them. |
| 06:30.27 | kaldemar | see expiry parameters to change that. |
| 06:30.46 | Atriks | where ? .-. |
| 06:31.22 | kaldemar | in sip.conf |
| 06:31.57 | Atriks | yes, but where ? |
| 06:32.06 | kaldemar | what may cause the ITSP to not accept your registration which may lead to the "403 not registered" |
| 06:32.30 | Atriks | ;maxexpiry=3600 ; Maximum allowed time of incoming registrations ? |
| 06:32.41 | kaldemar | Atriks: under [general]. sample sip.conf has those. |
| 06:32.42 | Atriks | or min ? |
| 06:32.53 | Atriks | it's in general yes |
| 06:33.55 | kaldemar | a registration from you to the ITSP is outgoing |
| 06:34.14 | Atriks | I don't understand... |
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| 06:44.27 | kaldemar | Atriks: you're sending a register message that gets responded with interval too brief. meaning that the ITSP thinks that the expiry time in your register message is too small. make it larger with a conf parameter that changes the value for an outgoing registration. that would be defaultexpiry as the sample says. |
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| 06:45.22 | Atriks | yes kaldemar |
| 06:45.25 | Atriks | by default 120 |
| 06:45.28 | Atriks | what should y put ? |
| 06:45.57 | kaldemar | your ITSP knows. |
| 06:46.14 | kaldemar | ask them or try larger values until they accept. |
| 06:46.15 | Atriks | I'm not mi iTSP |
| 06:46.35 | kaldemar | me neither. |
| 06:48.14 | Atriks | 500 not enough |
| 06:48.45 | Atriks | Oh |
| 06:48.47 | Atriks | It works !!!! |
| 06:48.49 | Atriks | I love you kaldemar |
| 06:49.17 | Atriks | Hum |
| 06:49.37 | Atriks | It works only for normal phones |
| 06:49.40 | Atriks | not mobiles |
| 06:49.41 | Atriks | what |
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| 08:11.23 | kaushal | hi again |
| 08:11.31 | kaushal | kaldemar: back |
| 08:11.37 | kaushal | Any clue ? |
| 08:13.53 | kaldemar | kaushal: your debug does not match with earlier output. |
| 08:14.05 | kaushal | ok |
| 08:14.35 | kaushal | kaldemar: let me describe it here |
| 08:14.50 | kaushal | so i have set debug for pri span 1-8 |
| 08:14.56 | kaushal | and then redirected to a file |
| 08:15.11 | kaushal | and then initiated the outbound campaign |
| 08:15.12 | kaldemar | how about enabling it for a single span that is used for dialing only? |
| 08:15.27 | kaushal | ok |
| 08:15.55 | kaushal | kaldemar: so how would i know which span is used for dialling ? |
| 08:16.05 | kaushal | not sure actually |
| 08:16.33 | kaushal | so i have g0-g7 call groups and span1-7 |
| 08:16.58 | WIMPy | You should have span 1-8. |
| 08:17.14 | kaushal | yeah typo |
| 08:17.15 | WIMPy | But what are you trying to do right now, exactely? |
| 08:17.21 | kaushal | g0-g7 call groups and span1-8 |
| 08:17.25 | kaushal | WIMPy: ok |
| 08:17.41 | *** join/#asterisk cerberus_za (~coert@41-134-110-162.dsl.mweb.co.za) |
| 08:17.44 | WIMPy | group 0=span 1, group 1=span 2, etc. |
| 08:18.03 | WIMPy | With your configuration that is. |
| 08:18.39 | kaushal | WIMPy: ok |
| 08:18.55 | *** join/#asterisk james_zhu (~Administr@183.16.215.134) |
| 08:19.00 | kaldemar | however, earlier output suggested that you received a hangup with cause 17 from the PRI but the last debug didn't have such a DISCONNECT. |
| 08:19.08 | kaushal | WIMPy: so when i run this |
| 08:19.09 | kaushal | *CLI> originate DAHDI/g0/xxxxxxxxxx Application MP3Player /home/kaushal/obd-demo.mp3 |
| 08:19.12 | kaushal | *CLI> originate DAHDI/g1/xxxxxxxxxx Application MP3Player /home/kaushal/obd-demo.mp3 |
| 08:19.29 | kaushal | i just see only one call being made at a time instead of multiple calls |
| 08:20.10 | kaushal | the numbers are ofcourse different |
| 08:20.32 | WIMPy | Yes, channel originate locks something. |
| 08:20.34 | kaushal | kaldemar: ok |
| 08:20.55 | kaushal | kaldemar: i would re run the campaign and pastebin the output again |
| 08:21.24 | kaushal | WIMPy: i have in total 8 PRI lines which comprises 240 channels |
| 08:21.44 | WIMPy | Yes, I know |
| 08:21.47 | *** join/#asterisk ollii (~risker@2001:470:1f15:1384:62eb:69ff:fe31:b4) |
| 08:21.51 | kaushal | WIMPy: How do i overcome this locking issue |
| 08:22.14 | WIMPy | Don't do it from the *CLI. Use .call files or better use AMI. |
| 08:22.22 | kaushal | or as suggested by you do i need to use AMI for this purpose ? |
| 08:22.25 | kaushal | WIMPy: ok |
| 08:22.46 | kaushal | WIMPy: do i need to have apache web server running on this asterisk box ? |
| 08:23.02 | *** join/#asterisk jacc0 (~jacc0@D522448D.static.ziggozakelijk.nl) |
| 08:23.06 | jacc0 | hi all!! |
| 08:23.09 | WIMPy | No. Where do you see a link to Apache? |
| 08:23.11 | jacc0 | gooooooood mornign |
| 08:23.32 | kaushal | WIMPy: when i look at /etc/asterisk/manager.conf file |
| 08:23.37 | ollii | good morning |
| 08:23.43 | kaushal | it says https or http |
| 08:24.10 | WIMPy | kaushal: You don't need the http(s) stuff. |
| 08:25.08 | kaushal | i enabled port 5038 |
| 08:25.30 | WIMPy | And now you have to get coding. |
| 08:25.50 | kaushal | ok |
| 08:26.34 | WIMPy | The good thing is that you will receive events that way about the calls you started. So you know when a call has ended. |
| 08:26.47 | WIMPy | And why, if you're interested. |
| 08:27.01 | kaushal | WIMPy: http://fpaste.org/Lqww/ |
| 08:27.13 | WIMPy | But the important bit is to know how many calls are active at any time. |
| 08:27.24 | kaushal | WIMPy: Any wiki page which mentions about configuring AMI ? |
| 08:27.45 | jacc0 | 20+ production envirements still running smooth :) |
| 08:27.54 | WIMPy | You don't need 'webenabled'. |
| 08:28.37 | WIMPy | kaushal: You need to set up a user and then you need to write an application that connects to AMI and talks to Asterisk that way. |
| 08:28.45 | kaushal | ok |
| 08:28.49 | WIMPy | It's pretty powerfull. |
| 08:28.54 | kaushal | oh ok |
| 08:29.06 | WIMPy | But it involves programming. |
| 08:29.11 | kaushal | How is different from AGI or .call files ? |
| 08:29.25 | kaushal | WIMPy: still learning and trying to understand |
| 08:29.49 | irroot | morning folks im in and about again |
| 08:30.01 | WIMPy | AGIs are called from the dialplan. I.e. from Asterisk to your programm on an active call. |
| 08:30.02 | jacc0 | AGI can do a lot more then only originating a call |
| 08:30.13 | WIMPy | Call files will make Asterisk set up a call. |
| 08:30.24 | ollii | agis can be written in c,php,python, bash...what you need ;) |
| 08:30.34 | ollii | *depends on your needs |
| 08:30.44 | WIMPy | And AGI works both ways. You can tell Asterisk what to do and you get information form Asterisk about what it's doing. |
| 08:32.03 | kaushal | ok |
| 08:35.36 | *** join/#asterisk mandla (~mandla@168.167.180.161) |
| 08:36.03 | kaldemar | kaushal: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Manager+Interface+(AMI) |
| 08:46.26 | *** join/#asterisk dwmw2___ (~ctrlproxy@twosheds.infradead.org) |
| 08:48.02 | mandla | hello, can anyone hook me up with a working chan_dahdi.conf file, please |
| 08:49.47 | kaldemar | mandla: http://svn.digium.com/svn/asterisk/tags/1.8.5.0/configs/chan_dahdi.conf.sample |
| 08:50.10 | mandla | kaldemar, thanx man. |
| 08:54.30 | kaushal | kaldemar: Thanks |
| 08:57.18 | Atriks | Back. I can't make calls to mobile phones via SIP due of my internet provider. Is it a way to make asterisk reconized as a phone plugged in the box or somethign like that ? |
| 08:59.02 | kaldemar | what is the reason for not being able to make calls to mobiles? |
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| 09:05.27 | dwmw2___ | hm, I edit utils/smsq.c and run 'make' in the top level, and it isn't rebuilt |
| 09:08.31 | Atriks | kaldemar, My internet provider block it |
| 09:08.38 | Atriks | from SIP, I mean |
| 09:09.07 | Atriks | For some security and fraud problems |
| 09:09.07 | kaldemar | internet provider or ITSP? |
| 09:09.16 | Atriks | It's the same |
| 09:09.35 | kaldemar | there's nothing you can do about it. change your provider. |
| 09:09.40 | Atriks | Fu |
| 09:09.45 | Atriks | It will be the same |
| 09:09.57 | Atriks | Every Internet provider are like that in France |
| 09:10.12 | kaldemar | dwmw2___: do a make menuselect and see if it is enabled under Utilities. |
| 09:10.29 | kaldemar | fu? |
| 09:11.11 | dwmw2___ | it should be; I just did a rebuild of the Fedora SRPM (since I'm running the Fedora package) and then went to fix the bugs |
| 09:11.13 | dwmw2___ | checks |
| 09:12.05 | dwmw2___ | hm, it's turned off. |
| 09:12.23 | dwmw2___ | oh, I think the package does multiple builds, to do things like voicemail-imap, voicemail-odbc etc. |
| 09:12.26 | dwmw2___ | thanks :) |
| 09:14.39 | dwmw2___ | gr |
| 09:15.21 | dwmw2___ | I made smsq name its queue file with '.call' at the end, and Asterisk *still* doesn't notice it until I manually rename it |
| 09:15.54 | dwmw2___ | smsq was making a file named 'smsq.motx.0.1314004450-18200.1' which didn't get noticed until I renamed it to 'smsq.motx.0.1314004450-18200.1.call' manually |
| 09:16.14 | dwmw2___ | so I changed smsq to make 'smsq.motx.0.1314004450-18200.1.call' and now it didn't get acted upon until I manually renamed it to 'asd.call' |
| 09:16.30 | dwmw2___ | maybe it's the *rename* that is the trigger? smsq links it into place and then removes the original manually |
| 09:20.44 | kaldemar | do the created files have their last modified date in the future for some reason? |
| 09:21.36 | dwmw2___ | no |
| 09:21.43 | dwmw2___ | I'm staring at the inotify code in pbx_spool.c now |
| 09:22.02 | dwmw2___ | I suspect if I make smsq.c *move* the file into place instead of hard-linking it, it'll be fine. |
| 09:22.30 | dwmw2___ | not sure what inotify event you get on a link. IN_CREATE perhaps ? |
| 09:23.06 | dwmw2___ | I think pbx_spool.c waits for an IN_CLOSE_WRITE notification on a created file, before it eats it |
| 09:23.12 | dwmw2___ | oh, that's horrid |
| 09:23.38 | dwmw2___ | [root@obelisk outgoing]# echo -n >> smsq.motx.0.1314004602-18213.1.call |
| 09:23.39 | dwmw2___ | haha |
| 09:23.41 | dwmw2___ | that made it work |
| 09:27.18 | dwmw2___ | hm, can inotify really not tell the difference between a hardlink and a creat() ? |
| 09:30.27 | dwmw2___ | this is wrong. Using inotify like this is cute, but surely it should have a timeout? If the file is non-zero size and not touched for a few seconds after it's created, then it was *fine* |
| 09:30.50 | dwmw2___ | and if the file *is* written after it was created, asterisk should surely bitch and refuse to run it? The rules state that you should make it first and move it into place! |
| 09:31.00 | dwmw2___ | or did that change? |
| 09:35.18 | dwmw2___ | adds the analysis to https://bugzilla.redhat.com/show_bug.cgi?id=732374 |
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| 09:51.13 | *** join/#asterisk Polysics (~luca@host210-142-static.228-95-b.business.telecomitalia.it) |
| 09:51.15 | Polysics | hello |
| 09:51.23 | Polysics | i ran into an interesting problem |
| 09:51.33 | Polysics | MoH blocks the call flow? |
| 09:51.51 | Polysics | i though i could use moh to keep people waiting while i do stuff |
| 09:52.19 | mandla | hello guys, i cant make outgoing calls work. |
| 09:52.38 | mandla | Im on Asterisk 1.7 with Xorcom Atribank |
| 09:52.43 | mandla | Im on Asterisk 1.7 with Xorcom Astribank |
| 09:53.10 | mandla | I dont know if the problem is with my chan_dahdi.conf file. |
| 09:53.45 | WIMPy | mandla: There is no Asterisk 1.7. Are you still on to the same story you were on to a month ago? |
| 09:54.36 | Polysics | did 1.7 ever exist? |
| 09:55.21 | Polysics | am i right in sayng MusicOnHold blocks execution? |
| 09:55.47 | WIMPy | Polysics: Of an AGI or what? |
| 09:56.21 | Polysics | WIMPy, yes, i call MusicOnHold from AGI, the script stays there and only continues when caller hangs up and MoH stops |
| 09:57.02 | WIMPy | Yes, while an Application executes, the AGI waits. |
| 09:57.13 | *** join/#asterisk asterisk-Tester (~RAMYT@210.5.215.39) |
| 09:57.53 | Polysics | the idea was to put the caller on hold, then use Originate to dial the various possible callees |
| 09:58.07 | Polysics | since each of them can accept or reject the call |
| 09:58.16 | Polysics | and Bridge the first available one |
| 09:58.58 | mandla | WIMPy, same story my man. 1.6. |
| 10:00.15 | kaldemar | Polysics: sounds like a queue with ringall strategy. |
| 10:00.19 | mandla | WIMPy, the project is been on hold, was tackling other open source projects. |
| 10:00.54 | Polysics | kaldemar, each potential callee has to be first verified for available credit, as credits are pooled between caller group |
| 10:01.12 | Polysics | and thus a receiver that was OK before could not be now, due to credit decreasing |
| 10:01.20 | Polysics | but that is not the point |
| 10:01.27 | mandla | WIMPy, Now this other pig head was hired in my co. to continue where i left off, he totally screwed everything up. |
| 10:01.39 | Polysics | the point is "how can i play audio/park an user somewhere while i do my stuff?" |
| 10:01.48 | kaldemar | Polysics: you set local channels as members and do what you want in dialplan. |
| 10:02.03 | mandla | WIMPy, where is Irroot?? |
| 10:02.18 | Polysics | kaldemar, isn't that the same thing as what i am doing using AGI? |
| 10:02.23 | WIMPy | Polysics: Use AMI. |
| 10:02.30 | Polysics | don't i still have the problem of keeping the person waiting? |
| 10:02.39 | WIMPy | mandla: Still here regularly. |
| 10:02.56 | kaldemar | Polysics: sure, but you'd tackle the moh problem with a queue. |
| 10:03.33 | Polysics | so you say i should build a single local channel queue and call that? intriguing :-D |
| 10:03.45 | Polysics | it's interesting that there isn't any other method though |
| 10:04.00 | WIMPy | Polysics: I told you one :-) |
| 10:04.59 | Polysics | WIMPy, AMI with which command? |
| 10:05.14 | Polysics | what would the call flow be? |
| 10:05.54 | kaldemar | Polysics: a queue with local channels as members. and in those extensions you do the required checks. |
| 10:06.10 | dwmw2___ | kaldemar: ok, all fixed: https://bugzilla.redhat.com/show_bug.cgi?id=732374#c5 |
| 10:06.50 | WIMPy | Polysics: You can let the caller run in to MOH, while trying to dial out independently. On success you bridge them. |
| 10:06.52 | Polysics | one local channel for each member? |
| 10:07.34 | kaldemar | Polysics: you can't have more than one. :) |
| 10:09.23 | Polysics | WIMPy, that is where i stop. how do i dial out if the user is in MoH, since AGI stops ? |
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| 10:16.24 | kaldemar | Polysics: have you noticed app StartMusicOnHold? it does not not block dialplan execution. |
| 10:16.43 | Polysics | kaldemar, no, i did not .-( |
| 10:17.04 | kaldemar | StopMusicOnHold will stop it for you. |
| 10:17.11 | *** part/#asterisk mirko_brankovic (~mirko_bra@212.200.146.253) |
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| 10:17.53 | Polysics | thanks |
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| 10:21.21 | Polysics | hmm, no, though, this strategy won't work |
| 10:21.34 | Polysics | no dialplan blocking also means the call is hung up right away |
| 10:21.47 | Polysics | fire an UserEvent and handle with AMI is probably the only way |
| 10:27.35 | Polysics | ok, that works |
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| 11:15.27 | dwmw2___ | PRI Span: 2 q931.c:8707 post_handle_q931_message: Call 32779 enters state 12 (Disconnect Indication). Hold state: Idle |
| 11:15.37 | dwmw2___ | yet ast_check_hangup() in app_sms doesn't return true. |
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| 11:27.34 | dwmw2___ | aha, inbanddisconnect=yes in chan_dahdi causes it. |
| 11:27.49 | dwmw2___ | is there a way for app_sms to detect that the call has been disconnected, despite that setting? |
| 11:30.21 | WIMPy | You should never use inband on a PRI. |
| 11:30.28 | WIMPy | or BRI |
| 11:31.09 | dwmw2___ | really? |
| 11:31.36 | dwmw2___ | what's the equivalent of 'earlyb3' in mISDN then? Where you get a DISCONNECT from the exchange but the with audio that tells you the actual error? |
| 11:33.03 | WIMPy | Earlyb3 is for bedia before the call connects. There is no explicit handling of late media. Depending on versions and combinations of cahnnels, either the disconnect is delayed or it doesn't work. |
| 11:33.19 | dwmw2___ | ok |
| 11:33.37 | WIMPy | And I'm missing that, actually. |
| 11:33.38 | dwmw2___ | so I can set inbanddisconnect=no and still get the earlyb3? |
| 11:34.11 | WIMPy | They are not related. |
| 11:34.17 | dwmw2___ | ok, thanks. |
| 11:34.35 | dwmw2___ | I was confused by the help text in the sample chan_dahdi.conf |
| 11:35.23 | WIMPy | Which one? |
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| 11:35.31 | dwmw2___ | ; Allow inband audio (progress) when a call is DISCONNECTed by the far end of a PRI |
| 11:35.31 | dwmw2___ | ; |
| 11:35.45 | tzafrir_laptop | mandla, asterisk 1.7? 1.8? |
| 11:36.03 | WIMPy | Hmm. Strange |
| 11:36.12 | dwmw2___ | strange that I was confused? :) |
| 11:36.21 | WIMPy | No |
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| 11:37.29 | mandla | tzafrir_laptop, http://pastebin.com/fg6mmHVU |
| 11:37.33 | WIMPy | Well, yes. |
| 11:37.43 | mandla | 1.6 |
| 11:38.12 | WIMPy | dwmw2___: Early meadi can be handled in the dialplan e.g. with playback(bla,noanswer) |
| 11:38.28 | WIMPy | Late media can't. |
| 11:38.56 | dwmw2___ | WIMPy: my telco doesn't let me *send* early media |
| 11:39.26 | dwmw2___ | I'm only interested in receiving it, when the early media may well be a recorded message telling me the reason the call hasn't been connected, and sometimes giving an alternative phone number to call. |
| 11:41.05 | mandla | tzafrir_laptop, Asterisk 1.6.2.11 |
| 11:45.54 | nuken | hi guys |
| 11:46.38 | nuken | does anybody know any feature that can be like a phonebook in my analog phones ? |
| 11:47.40 | nuken | for example, I program any number in a nuber one in phone keyboard |
| 11:48.27 | nuken | and when the user call number 1 will be automatic call to that number |
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| 11:52.39 | WIMPy | dwmw2___: That should happen after disconnect. |
| 11:56.31 | kaldemar | nuken: make single digit extensions that dial the wanted numbers. |
| 11:57.41 | nuken | huum ok |
| 11:58.15 | nuken | but, can I do this for one group of extension for example ? |
| 11:58.54 | nuken | for example, the extension '1' will be avaliable just for extentions of 100 to 110 |
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| 12:04.38 | kaldemar | nuken: yes. just make a context in extensions.conf for each phone and then include other contexts in those. |
| 12:09.15 | IsUp | i am able to do Echo test with my PBX, no audio or any problems |
| 12:09.27 | IsUp | but when i use my provider to Dial, i am getting one way audio |
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| 12:12.46 | kaldemar | IsUp: using SIP and you're behind a NAT? |
| 12:13.11 | IsUp | yes i am behind NAT |
| 12:13.22 | IsUp | my PBX has public ip and i am connecting it with softphone |
| 12:13.24 | IsUp | Echo works fine |
| 12:15.00 | kaldemar | enable sip debug for a call and pastebin it. |
| 12:15.53 | IsUp | br |
| 12:15.55 | IsUp | brb |
| 12:18.51 | eduzimrs | hi i got this "exten => _[7-9]XXXXXXX,n,GotoIf($[${GROUP_COUNT(GSM-OUTBOUND-LOCAL)} > "4" ]?gsmlocal:e1local)" for example if group count=3 it should go to the laber "e1local" right??? |
| 12:19.43 | eduzimrs | something smaller than 4 is treated as FALSE right? |
| 12:19.51 | kaldemar | eduzimrs: change "4" to 4. |
| 12:20.25 | eduzimrs | hum, i think thats the problem why is not working. |
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| 12:22.16 | eduzimrs | kaldemar: works , tks |
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| 12:23.04 | eduzimrs | kaldemar: could u explain me why? the " " treat that as a string ? am i right? |
| 12:23.51 | donkeh | Hi all, having a little problem with the AMD application hanging up calls in 1.8 - anyone that can assist/advise ? |
| 12:27.35 | kaldemar | eduzimrs: yes, it will be treated as a string when quoted. |
| 12:30.23 | dwmw2 | java.io.IOException: No space left on device |
| 12:30.31 | dwmw2 | hm, jira seems unhappy when I try to file a new ticket |
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| 12:48.50 | fireman_biff | Hi, what would cause this error: "Got a UA, but i'm in state 7" ? |
| 12:49.03 | fireman_biff | asterisk 1.4.22 |
| 12:51.37 | psilikon | anyone in here have experience with Astmanproxy? Every time I send an 'originate' command * and Astmanproxy crashes. I would love to see some samples of how other people used Astmanproxy. |
| 12:53.17 | dwmw2 | app_v110.c makes my brain hurt |
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| 12:54.04 | treborsux | yo |
| 12:54.23 | treborsux | well i made outgoings ok and made phone calls from soft phones |
| 12:54.31 | treborsux | watching tutorials on incoming today |
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| 13:23.08 | brad_mssw | I'm running 1.8.6.0-rc1 (would run 1.8.5, but digiums ubuntu 10.04 binaries didn't support PRI). Having a voicemail issue where the _calculated_ message duration is _way_ off. For instance, a 25s message is calculated as 8s. This is causing issues with 'minsecs' in voicemail.conf. Anyone else experiencing this? |
| 13:23.52 | ollii | 1.8.5.0 + libpri 1.4.12 is working fine from source |
| 13:23.57 | ollii | on ubuntu server 10.04 |
| 13:24.42 | brad_mssw | the issue isn't with source, it is as per this thread: http://www.spinics.net/lists/asterisk/msg144090.html |
| 13:25.41 | brad_mssw | I did find a similar issue reported to what I'm experiencing https://issues.asterisk.org/jira/browse/ASTERISK-16981 |
| 13:27.04 | brad_mssw | but can't confirm this is the same issue ... |
| 13:27.20 | brad_mssw | (since that was so long ago that it was reported) |
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| 14:22.03 | pabelanger | brad_mssw: if you do as the thread suggest, confirming PRI suppor is enabled, I'll rebuild the package today |
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| 14:32.51 | mjordan | brad_mssw: with respect to the VM duration issue, what is your format parameter in voicemail.conf? |
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| 14:33.44 | brad_mssw | mjordan: format = wav49|gsm|wav |
| 14:34.25 | mjordan | thanks, I'll take a look with 1.8 and see if I can find anything |
| 14:34.29 | Katty | ohai |
| 14:34.33 | brad_mssw | pabelanger: yes, the 1.8.6.0-rc1 _definitely_ works with PRI, and 1.8.5.0 definitely did NOT work with PRI (I'm not the original reporter, just happened to stumble across that trying to get the PRI to work) |
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| 14:35.17 | brad_mssw | pabelanger: none of the 'pri' functions existed in the 1.8.5 build |
| 14:35.20 | pabelanger | brad_mssw: thanks, I'll rebuild 1.8.5.0 now |
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| 14:53.18 | *** part/#asterisk asterisk-Tester (~RAMYT@210.5.215.39) |
| 14:57.54 | ruben23 | hi there any suggestion guys i have this error on my asterisk- flooding my asterisk console ------> http://i52.tinypic.com/20128oi.jpg |
| 14:59.48 | ruben23 | any idea guys..? |
| 15:00.19 | irroot | hey hey |
| 15:00.44 | irroot | just got linux 3.0.3 + dahdi 2.5.0 + wanpipe 3.5.20 to play nice |
| 15:01.06 | Tim_Toady | ruben23: seems thers a prob with some wav sound file it tries to read |
| 15:01.33 | Tim_Toady | some voice mail or some sound message |
| 15:03.39 | ruben23 | Tim_Toady: ok i will look into it, thanks for the idea. |
| 15:04.11 | Tim_Toady | it can also be music on hold, thats what usually is in wav format |
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| 15:06.56 | x1user | Hi, i have agi debug, core debug and core verbose to maximum, and i got no errors, but i still have no sound?? |
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| 15:13.02 | mjordan | brad_mssw: doesn't look like its just a format issue. Would you mind posting the [general] section of your voicemail.conf as an attachment on ASTERISK-16981? |
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| 15:18.07 | brad_mssw | mjordan: I can, but that's overkill since it is mostly default ... this is the exact command used when transforming it: http://pastebin.com/napSWLCt ... only additional is appending [default] with some voice mail boxes |
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| 15:18.20 | brad_mssw | mjordan: ok, I changed it to use a different e-mail |
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| 15:26.57 | brad_mssw | mjordan: just attached it to the ticket |
| 15:28.32 | mjordan | thanks - just to double check, you're running Ubuntu 10.04. Is it a 32 or 64 bit system? |
| 15:29.53 | brad_mssw | mjordan: 64bit |
| 15:30.02 | mjordan | thanks |
| 15:39.58 | *** part/#asterisk ollii (~risker@2001:470:1f15:1384:62eb:69ff:fe31:b4) |
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| 15:52.29 | pabelanger | brad_mssw: rebuild 1.8.5.0 for lucid, it should have PRI support now |
| 15:53.27 | brad_mssw | pabelanger: great, thanks ... did it get reversioned, or just replace the existing .deb? |
| 15:56.34 | pabelanger | brad_mssw: bumped; asterisk-1.8.5.0-1digium2 |
| 15:56.58 | pabelanger | you'll have to drop lucid-proposed from your apt source.list to use it |
| 16:05.33 | brad_mssw | pabelanger: yep, thanks. |
| 16:05.45 | brad_mssw | I'll try it tonight outside of business hours ;) |
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| 16:07.53 | pigpen | would anyone have an idea why my psql cdr field statement is bing doubled such as: |
| 16:07.54 | pigpen | INSERT INTO cdr ("calldate","clid","src","dst","dcontext","channel","dstchannel","lastapp","lastdata","duration","billsec","disposition","amaflags","accountcode","uniqueid","calldate","clid","src","dst","dcontext","channel","dstchannel","lastapp","lastdata","duration","billsec","disposition","amaflags","accountcode","uniqueid") |
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| 17:03.43 | *** join/#asterisk sereal-work (~sereal@unaffiliated/sereal) |
| 17:04.22 | sereal-work | Does anyone know if there is a known issue with asterisk 1.4 where you can't register two DIDs from the same fromdomain? |
| 17:04.35 | sereal-work | and same sip proxy |
| 17:05.11 | *** join/#asterisk sequencer (~something@196.218.255.29) |
| 17:05.13 | sequencer | Hi all |
| 17:06.22 | *** join/#asterisk Ryushin (proxy@cl-412.phx-01.us.sixxs.net) |
| 17:06.29 | sequencer | where do i set the timout for recieving a 200 OK response in asterisk ? |
| 17:09.01 | Ryushin | Just upgrade to the latest version of asterisk. Trying to compile asterisk-addons-1.6.2.3 for mp3 support, but I'm getting a compile error in format_mp3.c. |
| 17:09.20 | Ryushin | Has 1.6.2.3. compile cleanly for anyone else? |
| 17:09.58 | citywok | Ryushin: yes, it has |
| 17:11.06 | Ryushin | Then I guess something is hosed on my end. This should be fun. :( |
| 17:11.54 | sequencer | does anyone know where to set the timeout for recieving a 200 OK for SIP ? |
| 17:17.58 | Ryushin | citywok: Whate version of asterisk are you running? I'm trying to compile with with 1.8.5 |
| 17:18.21 | Qwell | Ryushin: pastebin the errors |
| 17:18.21 | citywok | you said 1.6.2.3... |
| 17:18.27 | Qwell | wait, wat |
| 17:18.31 | citywok | but i'm using 1.6.2.3 & 1.6.2.20 in production |
| 17:18.36 | Qwell | Why are you installing addons with 1.8? |
| 17:18.40 | citywok | if you are trying to use 1.8.5 you don't need addons at all |
| 17:18.54 | Ryushin | That how do I add mp3 support? |
| 17:19.03 | citywok | nothce the 1.6? |
| 17:19.06 | Ryushin | Is it included native to 1.8? |
| 17:19.06 | *** part/#asterisk sequencer (~something@196.218.255.29) |
| 17:19.08 | citywok | s/nothce/notice/ |
| 17:19.18 | Ryushin | Yes. |
| 17:19.24 | citywok | yea. b/c they go together |
| 17:19.24 | Ryushin | But there was no other newer addons. |
| 17:19.33 | citywok | b/c addons are built in in 1.8 |
| 17:19.39 | Ryushin | Okay, I'll look in the menu options then in 1.8 for mp3. |
| 17:19.50 | Ryushin | Thanks for pointing me in the right direction. |
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| 17:53.35 | anonymouz666 | in JIRA, where I can attack the debug info ? |
| 17:53.39 | anonymouz666 | attach |
| 17:53.40 | anonymouz666 | sorry |
| 17:53.42 | anonymouz666 | hehe |
| 17:53.51 | asilva | Does anyone know if DCAP is over asterisk 1.8? |
| 17:57.44 | psilikon | Anyone ever stumble across an AMI proxy that is well documented? |
| 17:58.02 | CJ0NeS | where can i get some hardware embbeded to use with asterisk? |
| 17:58.10 | CJ0NeS | any suggestion? |
| 17:58.23 | KavanS | CJ0NeS, I believe digium.com has some hardware on their website, and they are the authors of asterisk :) |
| 17:59.04 | *** part/#asterisk sereal-work (~sereal@unaffiliated/sereal) |
| 17:59.09 | CJ0NeS | yes... i saw digium products... but i want some more options... |
| 18:00.21 | kaldemar | CJ0NeS: what exactly are you looking for? interface cards? a pbx box? |
| 18:00.40 | CJ0NeS | kaldemar, asterisk server embbeded |
| 18:01.15 | kaldemar | elaborate please |
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| 18:03.02 | serafie | anonymouz666: More Actions -> Attach Files |
| 18:03.35 | CJ0NeS | i'm looking to asterisk server with hardware embbeded... like Linksys NSLU2 |
| 18:04.25 | CJ0NeS | or something like that... |
| 18:06.59 | *** join/#asterisk xnfinite (~xnfinite@164.145.223.87.dynamic.jazztel.es) |
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| 18:15.13 | anonymouz666 | serafie: already did thanks. |
| 18:30.13 | *** join/#asterisk treborsux (~treborsux@75-144-117-117-Jacksonville.hfc.comcastbusiness.net) |
| 18:30.33 | treborsux | why would a poe switch power a 560 but not a 501 polycom? |
| 18:31.17 | treborsux | http://www.ebay.com/itm/Linksys-Cisco-SRW208P-8-Port-Gigabit-Switch-PoE-/220770464400?pt=COMP_EN_Hubs&hash=item3366f1f290 |
| 18:32.32 | Qwell | because 501s were PoE? |
| 18:33.00 | Qwell | weren't* |
| 18:33.54 | carrar | heh |
| 18:34.04 | treborsux | what?? |
| 18:34.39 | treborsux | that would be tough since there is not place to plug in power |
| 18:34.50 | Qwell | they require a special cable |
| 18:36.12 | *** join/#asterisk nny (~SM@cpe-174-107-223-014.sc.res.rr.com) |
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| 18:37.17 | Qwell | http://www.voiplink.com/Polycom_PoE_Cable_for_501_and_301_IP_Phones_p/polycom-poe-cable.htm |
| 18:37.51 | treborsux | what the hell |
| 18:37.59 | treborsux | i screwed up i thought these were poe |
| 18:38.07 | treborsux | i got bad advice |
| 18:38.07 | nny | for ODBC Set: Set(dispositionstatus=${ODBC_CALLRESULTINSERT(${LeadID},${CAMPAIGNID},${ARG1},${DMEET})}) , wouldn't the four variables just be VAL1 in func_odbc.conf? |
| 18:38.09 | treborsux | pooo |
| 18:38.12 | Qwell | You screwed up, you thought they were made in the last 5 years. |
| 18:38.19 | Qwell | There are much better, cheaper, replacements. |
| 18:38.33 | Qwell | (which, surprise surprise, are PoE) |
| 18:38.43 | treborsux | with the cord I can get those |
| 18:38.51 | treborsux | damnit now i have to send these back |
| 18:39.20 | atheos | add that there are two PoE "standards". early Cisco PoE products were reverse polarity |
| 18:40.57 | treborsux | so these always had to be pluged into the wall |
| 18:41.05 | treborsux | damn vendor told me they were poe |
| 18:41.20 | nny | Qwell sorry to bother can you sanity check Set(dispositionstatus=${ODBC_CALLRESULTINSERT(${LeadID},${CAMPAIGNID},${ARG1},${DMEET})}) ? I assume the entire end string of variables is just VAL1 ? |
| 18:41.53 | treborsux | What is poe and as cheap |
| 18:41.53 | nny | or should I make each variable Set as VAL1 VAL2 etc? |
| 18:42.55 | treborsux | is there an injector? Can I make one? |
| 18:44.20 | atheos | treborsux sure, 48V - positive to pin 1+2, neg to 3+6 |
| 18:45.26 | jaytee | ~itsplist-us |
| 18:45.26 | infobot | Here are some popular ITSPs (USA) starting with the more respected ones: http://www.teliax.com , http://www.voicepulse.com/connect/ , http://www.broadvoice.com , http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net , http://voip.ms and http://flowroute.com |
| 18:45.29 | treborsux | i have 6 of the cords |
| 18:45.38 | treborsux | i just pluged one in |
| 18:45.48 | treborsux | and it works powered with the switch |
| 18:46.21 | treborsux | question is the cord have an integrated circut or just direction |
| 18:46.38 | treborsux | anyone made a cord so poe switch will power 501? |
| 18:47.04 | treborsux | I bought these cords for use where i dont have poe because they come with power transformers |
| 18:47.46 | treborsux | what is the adapter doing to allow it to work with the poe switch? |
| 18:49.39 | atheos | <PROTECTED> |
| 18:49.45 | atheos | <PROTECTED> |
| 18:50.40 | treborsux | on the 301 can i make a cable that allows the poe switch to work like this cable does |
| 18:51.09 | treborsux | i know i can make one with a transformer |
| 18:51.35 | treborsux | but can i make one that works like the real one and allows 802.3af to work like it is now |
| 18:52.09 | *** join/#asterisk KNERD (~KNERD@99.72.119.220) |
| 18:53.35 | treborsux | http://www.8774e4voip.com/PhotoGallery.asp?ProductCode=Polycom+PoE+Cable+-+301%2F501 |
| 18:53.39 | treborsux | i want to make this |
| 18:54.45 | atheos | treborsux - all that does, is inject power to two of your twisted pairs. You can build this, easy. |
| 18:55.16 | treborsux | kewl |
| 18:55.25 | KNERD | Why does the gtalk plug in keep screwing up? It's yet AGAIN back to fail |
| 18:55.26 | treborsux | ill take it apart than |
| 18:55.36 | atheos | might have been a hazard, but I used a single Cisco power supply to power 10 phones where I used to work. I just used a punchdown block to distribute the power |
| 18:56.06 | atheos | no need to take it apart treboursux - 48V - positive to pin 1+2, neg to 3+6 |
| 18:56.37 | atheos | but, use a meter to verify your cable |
| 18:56.47 | *** join/#asterisk d_preston215 (~chatzilla@static-76-161-250-54.t1.cavtel.net) |
| 18:56.50 | d_preston215 | What is everyone's feeling on trixbox at this point? |
| 18:57.45 | *** join/#asterisk sunfone (~jeff@c-67-184-36-158.hsd1.il.comcast.net) |
| 18:58.39 | KNERD | ask in the tribox channel |
| 18:58.45 | pabelanger | d_preston215: I heard their community was pretty much dead |
| 18:58.53 | pabelanger | not sure though, never use it |
| 18:59.11 | d_preston215 | I wanted to ask the general asterisk community as a whole. |
| 18:59.39 | Gugge | my fealing is that i dont like a webinterface limiting my options :) |
| 19:00.20 | *** join/#asterisk nmjnb (~nmjnb@c-567e72d5.026-18-73746f23.cust.bredbandsbolaget.se) |
| 19:01.49 | d_preston215 | I'm just researching options for moving away from TB (2.8 being the broken ring-strategy mess that it is). |
| 19:02.10 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 19:02.33 | d_preston215 | I was all of about to call TB dead as well, but apparantly they want to start up some kind of development again: |
| 19:02.35 | d_preston215 | http://www.fonality.com/trixbox/forums/community-edition/open-discussion/trixbox-community |
| 19:03.03 | treborsux | if that is all i had to do the 501 would already work |
| 19:03.50 | treborsux | the 501 cant be just stick it in those lines or it would work right now |
| 19:04.22 | treborsux | blue and brown are 12v |
| 19:04.23 | atheos | treborsux - there aren't too many variables when it comes to PoE. You've got polarity, or you have wattage requirements. |
| 19:04.37 | treborsux | right but 501 is not poe |
| 19:04.57 | treborsux | when you use the cable it changes 48 to 12 |
| 19:06.01 | treborsux | I can use a transformer on the brown and blue at 12 v but i cant use the poe switch i have with them |
| 19:06.12 | treborsux | the adapter steps down the power |
| 19:06.37 | *** join/#asterisk navaismo (~navaismo@189.249.55.244) |
| 19:07.07 | treborsux | these 501s are useless to me poop |
| 19:07.22 | atheos | treborsux - I guess I'm unclear on your objective, if you're phone is not PoE. |
| 19:07.24 | nmjnb | anyone have any idea why I can't connect to my asterisknow 1.7.1 server? I installed it on a virtual host, assigned an unused public IP but can't access it anyway |
| 19:08.03 | nmjnb | any system service needed to connect through a browser? |
| 19:08.38 | atheos | treborsux - either way, you have two unused pair of cable in ethernet. If you inject 12V in your ethernet run, you can certainly power the phone if you wire it appropriately. |
| 19:08.42 | treborsux | freakind cable costs more than the phones |
| 19:08.49 | atheos | haha, yup. |
| 19:08.50 | nmjnb | I can ping the server, and it can ping my local computer back |
| 19:09.50 | navaismo | @nmjnb mayb the iptables is blocking you |
| 19:09.54 | nmjnb | treborsux: there are units to connect between the switch and phone to get PoE if that's of any interest. |
| 19:10.19 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 19:10.30 | nmjnb | navaismo: I thought iptables was blank in a fresh install |
| 19:10.50 | treborsux | yes i have 6 of them |
| 19:10.57 | treborsux | but 30 phones |
| 19:11.20 | nmjnb | treborsux: lol.. get yourself some PoE switches then.. :P |
| 19:11.32 | treborsux | i have poe switches!!! |
| 19:11.34 | nmjnb | navaismo: iptables is blank, as I guessed |
| 19:11.47 | navaismo | iptables -L what shows? |
| 19:11.52 | navaismo | ok |
| 19:12.02 | treborsux | The problem is that i have only 6 of these cable because they are also the cable needed to use them with wall jack not poe |
| 19:12.22 | treborsux | u have to use the cable if you want to poe though 2 |
| 19:12.24 | navaismo | with telnet you can connect to 80 port? |
| 19:12.24 | Nugget | telnet is eeeeeeevil! |
| 19:12.27 | treborsux | and i did not know whtis |
| 19:13.16 | treborsux | so this jack ass on ebay sold me 30 ip501 and cords just for the ones i need to plug in the wall and said that was fine. But i should of done my research so i am the jack ass |
| 19:13.24 | p3nguin | You need a special Ethernet cable to run PoE? |
| 19:13.28 | treborsux | yes |
| 19:13.36 | jaytee | iptables is not typically "blank" in a fresh install...at least on CentOS. It's set to block any incoming traffic and allow any established/related connections initiated from the inside network and masquerade. you have to allow traffic inbound for any ports you need 2 way traffic for. |
| 19:13.38 | treborsux | because the phone is 12v not 48 |
| 19:13.38 | KavanS | I don't think you do |
| 19:13.46 | KavanS | oh... |
| 19:13.50 | atheos | p3nguin they aren't not spec PoE. 12v bastard implementation of PoE |
| 19:13.51 | treborsux | the cable steps it down |
| 19:13.56 | treborsux | i just opened one |
| 19:14.05 | treborsux | and it only works poe with the cable |
| 19:14.18 | nmjnb | jaytee: it's an install of Asterisknow, and the iptables are empty |
| 19:14.25 | navaismo | nmjnb try with service iptables stop and try again |
| 19:14.39 | *** join/#asterisk linusXtorvalds (~e_dot_zil@pool-98-118-168-221.bflony.fios.verizon.net) |
| 19:15.03 | linusXtorvalds | hello all |
| 19:15.29 | navaismo | hi |
| 19:15.38 | *** join/#asterisk tzafrir_laptop (~tzafrir@bzq-218-155-147.cablep.bezeqint.net) |
| 19:17.32 | nmjnb | navaismo: still can't connect |
| 19:17.47 | nmjnb | should I connect to some port or just http? |
| 19:18.02 | *** join/#asterisk linusXtorvalds (~e_dot_zil@pool-98-118-168-221.bflony.fios.verizon.net) |
| 19:18.13 | linusXtorvalds | wuz up |
| 19:18.19 | navaismo | as i know only http |
| 19:18.33 | navaismo | the apache is running? |
| 19:18.38 | nmjnb | the guides I saw didn't mention any ports |
| 19:18.55 | nmjnb | I would guess, since it should all be automatic in the asterisknow, but I can check |
| 19:19.32 | *** join/#asterisk fenlander (~fenlander@82.152.81.57) |
| 19:19.51 | KNERD | Why does the gtalk plug in keep screwing up? It's yet AGAIN back to fail |
| 19:20.31 | Gugge | KNERD: because google changes things, and they dont support the plugin |
| 19:20.59 | KNERD | free switch has not have these problems at all |
| 19:21.09 | atheos | KNERD - I haven't seen it stable for more than a day on incoming. outgoing is solid, find a DID to forward incoming calls to. |
| 19:21.49 | KNERD | i am starting to see why people are flooding to freeswitch |
| 19:22.18 | KNERD | outgoing is screwed..outgoing does not function |
| 19:22.19 | Gugge | competition is nice |
| 19:22.34 | KNERD | i mean incoming is fine |
| 19:22.35 | *** join/#asterisk hobodave (~hobodave@pdpc/supporter/professional/hobodave) |
| 19:23.05 | KNERD | they just patched it 2 days ago |
| 19:23.07 | atheos | KNERD - strange, my outing calls are flawless. incoming is all I've had a problem with. |
| 19:23.25 | nmjnb | navaismo: it seems httpd isn't running, or I'm not seeing it. |
| 19:23.57 | KNERD | https://issues.asterisk.org/jira/browse/ASTERISK-18301 |
| 19:25.26 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 19:25.56 | nmjnb | navaismo: problem solved, I checked netstat -a to get a hint of what it was listening to, and found some radan-http, googled it and apparantly it's port 8088 to connect to the server.. |
| 19:26.40 | navaismo | ok |
| 19:28.00 | linusXtorvalds | does asterisk like rtpmap: 101 or something else? |
| 19:31.00 | *** join/#asterisk angryuser_laptop (~angryuser@90-156-167-83.reverse.alphalink.fr) |
| 19:34.40 | sunfone | I would like to commission a company to build a custom phone... anyone have that experience? Any known manufacturers that white-label or otherwise allow for custom (presumably high volume order) engineering? |
| 19:35.02 | *** join/#asterisk jkroon (~jkroon@197.168.172.250) |
| 19:37.42 | treborsux | now i am trying to figure out why they sold me these phones without inline cord because they cant be used in anyway without them |
| 19:43.20 | *** join/#asterisk treborsux (~treborsux@75-144-117-117-Jacksonville.hfc.comcastbusiness.net) |
| 19:44.23 | linusXtorvalds | *bored* |
| 19:44.31 | sunfone | meh |
| 19:46.26 | carrar | drink! |
| 19:46.34 | linusXtorvalds | *at work* |
| 19:46.46 | carrar | even more reason too |
| 19:46.58 | carrar | vodka |
| 19:47.42 | carrar | sides, if you can use peer pressure get your coworkers to drink, now you're talking some fun |
| 19:48.08 | carrar | peer pressure, peer pressure, come on |
| 19:48.11 | carrar | everyone is doing it |
| 19:48.42 | carrar | streaking through the hallway around cubicals at noon |
| 19:48.48 | carrar | It' |
| 19:48.54 | carrar | It's what it's all about! |
| 19:50.09 | *** join/#asterisk kerx (~kerx@76-240-161-60.lightspeed.irvnca.sbcglobal.net) |
| 19:51.30 | *** join/#asterisk mintee (~mintee@2001:470:7:a41::2) |
| 19:51.39 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 19:51.43 | linusXtorvalds | someone ban carrar |
| 19:52.07 | mintee | so it there an easier way to combine the in and out files from a Monitor() than using a secondary script? |
| 19:52.39 | dijib | does anybody in here have any experience with PBX in a flash? or freepbx? |
| 19:52.47 | carrar | heh |
| 19:53.18 | navaismo | @mintee Mixmonitor |
| 19:53.20 | sunfone | dijib: /join #freepbx |
| 19:53.28 | dijib | k |
| 19:55.21 | mintee | Mixmonitor(filename.extn) |
| 19:55.25 | mintee | that's it? |
| 19:55.46 | mintee | does it create a wav? |
| 19:56.06 | navaismo | yes |
| 19:56.31 | treborsux | what is a model higher than 501 but not 550 that has poe |
| 19:56.48 | navaismo | https://wiki.asterisk.org/wiki/display/AST/Application_MixMonitor |
| 19:58.06 | sunfone | treborsux: how many lines? I went to the 450 after the 501. |
| 19:58.14 | sunfone | It has "normal" POE |
| 19:59.09 | treborsux | I need a lot of 21 501 power cords |
| 19:59.51 | p3nguin | mintee: It'll create whatever supported format you tell it to use. wav, WAV, gsm, etc. |
| 19:59.56 | *** join/#asterisk chrisjunkie (~Chris@2001:4428:22d:2:208:2ff:fe7e:6312) |
| 20:01.18 | mintee | p3nguin: great! works good |
| 20:01.20 | *** join/#asterisk dwmw2_gone__ (~ctrlproxy@twosheds.infradead.org) |
| 20:01.21 | mintee | thanks y'all |
| 20:03.56 | *** join/#asterisk rdahlin_1_ (~rdahlin_1@78-73-19-238-no168.tbcn.telia.com) |
| 20:04.40 | *** join/#asterisk leed (~quassel@75-150-13-105-Atlanta.hfc.comcastbusiness.net) |
| 20:10.29 | *** join/#asterisk hobodave (~hobodave@pdpc/supporter/professional/hobodave) |
| 20:15.07 | *** join/#asterisk linuxgecko (~gecko@99-182-113-98.lightspeed.clmboh.sbcglobal.net) |
| 20:20.08 | *** join/#asterisk sequencer (~something@196.218.255.29) |
| 20:20.11 | sequencer | Hi all |
| 20:20.33 | sequencer | Where do i set the callerID settings per each extension ? :s |
| 20:20.43 | p3nguin | obviously in the extension. |
| 20:20.48 | sequencer | right.. |
| 20:20.55 | p3nguin | Extensions are in extensions.conf. |
| 20:20.57 | sequencer | just cant figure the correct line to do it :s |
| 20:21.03 | WIMPy | Extensions have CallerIDs? |
| 20:21.07 | nny | is there a way to dynamically create a meetme room by specifying the room number? |
| 20:21.11 | navaismo | in the sip.conf for devices |
| 20:21.20 | p3nguin | Every extension that you want to set the caller ID, you have to set the caller ID. |
| 20:21.21 | nny | thought d but it created a room number different than what i told it to |
| 20:21.39 | sequencer | p3nguin right, whats the structure for that ? :s |
| 20:21.46 | sequencer | callerid = DID ? |
| 20:21.57 | p3nguin | Set(CALLERID(num)=3215551212) |
| 20:22.24 | sequencer | is that in the dialplan itself, or where i set the extensions ? :s |
| 20:23.01 | p3nguin | extensions are the dial plan. |
| 20:23.11 | sequencer | hmm.. |
| 20:23.18 | navaismo | if you want to set the CID for SIP devices is in sip.conf |
| 20:23.19 | p3nguin | You're not making a lot of sense right now. |
| 20:23.29 | p3nguin | He asked to set caller id per extension. |
| 20:23.45 | p3nguin | That's done in extensions.conf where the extensions are. |
| 20:23.57 | sequencer | can i do this in my outgoing settings: |
| 20:24.10 | Kobaz | you know what he's talking about... in sip.conf you can set the callerid |
| 20:24.12 | sequencer | Set(CALLERID(num)=321555${EXTEN}) ? |
| 20:24.24 | p3nguin | You could, but don't. |
| 20:25.13 | p3nguin | Since you won't be dialing 4-digit phone numbers, and you don't want to set the caller id to the number you're calling, don't. |
| 20:25.52 | nny | odd exten => _551XX,1,Meetme(${EXTEN},sdqcaA) |
| 20:26.00 | nny | <PROTECTED> |
| 20:26.01 | nny | wtf? |
| 20:26.13 | nny | why wouldn't it create room 55101? |
| 20:26.20 | p3nguin | Will every call going out need to have the same CID number? |
| 20:26.29 | Kobaz | nny it's just an internal identifier |
| 20:26.45 | nny | Kobaz: oh so the meetme room number is still 55101? |
| 20:26.52 | Kobaz | yes |
| 20:26.54 | *** join/#asterisk ipc9 (~any@173-162-245-206-NewEngland.hfc.comcastbusiness.net) |
| 20:27.05 | nny | oh.. heh. that's not confusing in the least o_0 |
| 20:27.13 | Kobaz | it's because you're using the dynamic create option |
| 20:27.24 | Kobaz | yeah that message should be changed, the user doesn't need to know the internal id |
| 20:30.32 | sequencer | p3nguin how can i set it to my own extension ? |
| 20:30.49 | p3nguin | Are you calling out to the PSTN? |
| 20:31.01 | sequencer | for instrance, if am on ext 1234 i want my caller id to be 333-333-1234 |
| 20:31.10 | sequencer | am calling outside to a SIP provider |
| 20:31.28 | sequencer | that allows the caller id to be changed |
| 20:31.42 | p3nguin | You could use the callerid parameter in the sip peer entry. |
| 20:31.49 | sequencer | exacthly |
| 20:31.54 | sequencer | thats what am referring to |
| 20:32.06 | *** join/#asterisk azv4 (~azv4@host-66-202-7-218.pit.choiceone.net) |
| 20:32.10 | sequencer | i just dont know whats the callerid line should be |
| 20:32.15 | p3nguin | callerid=Null <3333331234> |
| 20:32.21 | sequencer | alrighty! |
| 20:32.26 | sequencer | lets try that :) |
| 20:32.48 | p3nguin | Or you can set it in the EXTENSION like you originally asked. |
| 20:33.10 | p3nguin | Both ways will yield caller id of your choice. |
| 20:33.18 | navaismo | "I knew it!" |
| 20:34.18 | sequencer | great |
| 20:34.18 | sequencer | how do i restart when convenient ? |
| 20:34.21 | p3nguin | core restart when convenient |
| 20:34.27 | sequencer | core ! |
| 20:34.30 | sequencer | thats what i missed |
| 20:34.32 | sequencer | thanks man! |
| 20:34.45 | p3nguin | I'm surprised it isn't aliased so you can leave off the core part. |
| 20:34.53 | p3nguin | restart when convenient should have done the same thing. |
| 20:35.44 | *** join/#asterisk hobodave (~hobodave@pdpc/supporter/professional/hobodave) |
| 20:37.33 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 20:38.56 | sequencer | i think so |
| 20:39.20 | sequencer | the call is using the sip trunk's default CId |
| 20:39.30 | sequencer | and ignoring my own extension |
| 20:39.40 | p3nguin | It's not an extension. |
| 20:39.44 | sequencer | let me try to remove the trunk's caller id.. :s |
| 20:39.44 | p3nguin | It's caller id. |
| 20:39.57 | sequencer | and ignoring my own extension's caller id * |
| 20:40.03 | p3nguin | still wrong. |
| 20:40.10 | p3nguin | It's your own PHONE's caller id. |
| 20:40.20 | sequencer | hmm.. |
| 20:40.26 | p3nguin | Phones are not extensions. |
| 20:40.30 | sequencer | a phone is a device |
| 20:40.31 | p3nguin | Extensions are not devices. |
| 20:41.18 | sequencer | yep so basically no matter what the device is, as long the extension is set it should display the extension's caller id ? ;) |
| 20:41.33 | p3nguin | no |
| 20:41.35 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
| 20:41.37 | p3nguin | You're not making sense again. |
| 20:41.42 | sequencer | or am tottaly missing terminology again |
| 20:41.55 | p3nguin | Extensions are the dialing rules found in extensions.conf. |
| 20:42.06 | p3nguin | They start with "exten =>" or "same =>" |
| 20:42.09 | sequencer | ok. |
| 20:42.15 | p3nguin | That is all. |
| 20:42.25 | sequencer | so its a word terminology confusion |
| 20:42.26 | sequencer | ;) |
| 20:42.39 | p3nguin | Phones are phones or devices or peers. |
| 20:43.36 | p3nguin | To set caller ID in an extension, you'll use Set(CALLERID(num)=somenumber) in an extension. Extensions are in extensions.conf. |
| 20:44.11 | p3nguin | Do it before the Dial(). |
| 20:44.31 | p3nguin | You can set it to whatever you want. Your ITSP may or may not accept it and pass it along. |
| 20:45.17 | p3nguin | The caller ID you set before the Dial() in an extension has shit-all to do with your phones. |
| 20:46.32 | p3nguin | If the extension used to Dial() your phone happens to be a 10-digit number that can be dialed from the PSTN, you can use your extension number as caller ID. |
| 20:47.12 | p3nguin | That's how I would do it if I had a DID for every phone on my system. |
| 20:51.52 | sequencer | but the thing is, i have a 4-digit extensions |
| 20:52.06 | p3nguin | But 4-digit phone numbers cannot be dialed from the PSTN. |
| 20:52.21 | sequencer | right |
| 20:52.37 | p3nguin | So you'll have to have 10-digit numbers, with the last four matching the extension used to dial your phone. |
| 20:52.41 | sequencer | 4 digit numbers goes to local phones |
| 20:52.49 | sequencer | exactly |
| 20:52.51 | p3nguin | This is very basic stuff. |
| 20:52.54 | p3nguin | Asterisk 101. |
| 20:53.10 | sequencer | now if i dont want to set a caller id explicitly for each extension |
| 20:53.18 | sequencer | what woul be the logical solution? |
| 20:53.24 | p3nguin | Set it for each phone, then. |
| 20:53.26 | sequencer | i have 300 DID |
| 20:53.39 | sequencer | i cant through 300 lines for them |
| 20:53.49 | p3nguin | You'll either set it per extension or per phone. You don't have much other choice. |
| 20:54.15 | p3nguin | It doesn't make sense to have 300 extensions just for each phone to set its own caller id. |
| 20:54.18 | sequencer | cant i set it where i put the common numbers in the callerId and then add the 4-digit numbers? |
| 20:54.24 | p3nguin | So that's out. |
| 20:54.36 | p3nguin | Sure, you can do that. |
| 20:54.44 | sequencer | great.. |
| 20:54.49 | sequencer | how and where? |
| 20:54.54 | p3nguin | extensions.conf |
| 20:55.07 | sequencer | if i used Set(CALLERID(num)=somenumber) |
| 20:55.12 | p3nguin | what are hte first 6 digits? |
| 20:55.24 | sequencer | it would be Set(CALLERID(num)=333333) |
| 20:55.39 | p3nguin | exten => _1NXXNXXXXXX,1,Set(CALLERID(num)=333333${CALLERID(num)}) |
| 20:55.40 | sequencer | then how do i append another 4 digitd? |
| 20:55.45 | sequencer | alrighty |
| 20:55.53 | p3nguin | Set the callerid value PER PHONE to the 4-digit extension number. |
| 20:55.53 | sequencer | this makes much sense |
| 20:56.25 | nny | is there an extension for failed? like n? |
| 20:56.33 | p3nguin | n isn't an extension. |
| 20:56.37 | nny | er h |
| 20:56.38 | nny | sorry ha |
| 20:56.46 | p3nguin | h is hangup, i is invalid |
| 20:56.55 | nny | [Aug 22 15:53:13] NOTICE[26299]: pbx_spool.c:352 attempt_thread: Queued call to SIP/backup2/14132561400 expired without completion after 0 attempts |
| 20:57.01 | nny | where would that fall into ? |
| 20:57.24 | p3nguin | Like if you are using WaitExten() or BackGround() and someone presses a number which you do not have an extension for, it will fall onto i. |
| 20:57.44 | *** join/#asterisk angryuser_laptop (~angryuser@90-156-167-83.reverse.alphalink.fr) |
| 20:57.57 | p3nguin | That's probably not what you need for your queue situation. |
| 20:58.03 | jaytee | people often overlook the t extension |
| 20:58.12 | p3nguin | t for timeout |
| 20:58.24 | p3nguin | T for a different kind of timeout |
| 20:58.36 | jaytee | yep, and handy if you use WaitExten() |
| 20:58.42 | nny | what does http://pastebin.com/aRnUYX9E fall into? |
| 21:00.48 | navaismo | exten => failed,1,.... |
| 21:01.00 | nny | navaismo: will try thanks |
| 21:01.38 | sequencer | p3nguin i guess this thing worked |
| 21:01.46 | sequencer | now how can i add the name to the caller ? |
| 21:01.52 | p3nguin | You can't. |
| 21:02.08 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 21:02.08 | p3nguin | CNAM lookup is performed on the receiving end. |
| 21:02.25 | sequencer | i used to have it on my other asterisk |
| 21:02.35 | p3nguin | You can send it, but it won't matter. |
| 21:02.42 | p3nguin | Why, you ask? |
| 21:02.42 | sequencer | i need to send it |
| 21:02.47 | p3nguin | CNAM lookup is performed on the receiving end. |
| 21:03.07 | sequencer | if it's sent then SIP provider will forward it |
| 21:03.15 | p3nguin | THEY CAN'T |
| 21:03.17 | p3nguin | CNAM lookup is performed on the receiving end. |
| 21:03.25 | p3nguin | Do you speak English? |
| 21:03.33 | p3nguin | If you want to send it anyway, use: CALLERID(all)=Your Name <yournumber> |
| 21:03.43 | sequencer | hmm.. |
| 21:03.45 | p3nguin | CNAM is not something that is sent. |
| 21:03.48 | Kobaz | you can send it but it won't do anything |
| 21:03.51 | p3nguin | It is looked up by the recipient. |
| 21:04.00 | p3nguin | That's what I already said multiple times. |
| 21:04.01 | Kobaz | on the pstn anyway |
| 21:04.16 | Kobaz | if it's completely on your own system, you can pass the name |
| 21:04.31 | WIMPy | on some PSTN maybe. |
| 21:04.35 | sequencer | i think it already does between the phones |
| 21:04.47 | sequencer | my SIP provider allows me to do anything |
| 21:04.57 | p3nguin | Between the phones, locally, you can set it in the callerid parameter in sip.conf |
| 21:04.58 | sequencer | setting up my own CID and CNAM |
| 21:05.10 | p3nguin | Sure you can send it, but like you've already been told, it won't matter. |
| 21:05.18 | Kobaz | sequencer: try it... you won't get the name on the other end |
| 21:05.30 | sequencer | ill try it, no harm done :) |
| 21:05.40 | p3nguin | (1603.17) <p3nguin> CNAM lookup is performed on the receiving end. |
| 21:05.42 | p3nguin | (1603.33) <p3nguin> If you want to send it anyway, use: CALLERID(all)=Your Name <yournumber> |
| 21:06.06 | p3nguin | Or set it in the callerid setting like I told you earlier. |
| 21:06.22 | sequencer | i can use a variable as well, right ? |
| 21:06.26 | p3nguin | callerid=Your Name <yournumber> |
| 21:06.28 | p3nguin | If you wanted. |
| 21:06.37 | sequencer | like exten => _1NXXNXXXXXX,1,Set(CALLERID(num)=${name} 333333${CALLERID(num)}) |
| 21:06.45 | p3nguin | nope |
| 21:06.59 | p3nguin | You can't put name in CALLERID(num). num means NUMBER. |
| 21:07.10 | sequencer | oh |
| 21:07.11 | p3nguin | You can use CALLERID(name) or CALLERID(all) |
| 21:07.17 | sequencer | like exten => _1NXXNXXXXXX,1,Set(CALLERID(all)=${name} 333333${CALLERID(num)}) |
| 21:07.44 | sequencer | so it would be.. |
| 21:07.48 | sequencer | like exten => _1NXXNXXXXXX,1,Set(CALLERID(all)=${fullname} 333333${CALLERID(num)}) |
| 21:07.54 | p3nguin | If ${name} expandes correctly that should work. But even if you send it out, it will never make it to the receiving end of the call over the PSTN. |
| 21:08.10 | sequencer | ${fullname} is defined in the peer |
| 21:08.19 | sequencer | with the person name |
| 21:08.26 | p3nguin | But you have to use the format like I showed you. |
| 21:08.29 | p3nguin | CALLERID(all)=Your Name <yournumber> |
| 21:08.32 | Kobaz | you're wasting your time |
| 21:08.33 | p3nguin | Notice the <> |
| 21:10.10 | sequencer | like exten => _1NXXNXXXXXX,1,Set(CALLERID(all)=${fullname} <333333${CALLERID(num)}>) |
| 21:10.14 | sequencer | would this be right? |
| 21:10.31 | p3nguin | Looks okay to me. |
| 21:10.35 | sequencer | alrighty.. |
| 21:10.40 | sequencer | waiting for convenience.. |
| 21:11.26 | *** join/#asterisk rsmiley (~rsmiley@vpn.fortrust.biz) |
| 21:11.34 | rsmiley | Hello. |
| 21:11.37 | p3nguin | If you were going to bother with a setvar in each phone's sip entry, you could have just defined the entire CALLERID(all) there. |
| 21:12.04 | p3nguin | setvar=CIDout=Your Name <yournumber> |
| 21:12.20 | p3nguin | exten => _1NXXNXXXXXX,1,Set(CALLERID(all)=${CIDout}) |
| 21:13.48 | Katty | hi |
| 21:14.03 | Katty | today i learned you can bake mason jars. |
| 21:14.07 | Katty | and then won't asplode. |
| 21:14.18 | Katty | which is FABULIOUS if you need to ship cake. |
| 21:14.21 | ChannelZ | With the lids? |
| 21:14.26 | Katty | no, no lids |
| 21:14.28 | Katty | lids after the baking |
| 21:14.34 | ChannelZ | was gonna say... |
| 21:14.36 | Katty | they also recommend not shipping, frosted. |
| 21:14.40 | Katty | just plain ole cake |
| 21:14.51 | p3nguin | Frosted cakes get soggy. |
| 21:14.53 | Katty | yes. |
| 21:14.58 | p3nguin | and eww |
| 21:15.08 | Katty | expanding on ideas.... |
| 21:15.16 | Katty | you can use them as tupperware |
| 21:15.20 | ChannelZ | Now you need one of those vacuum sealers |
| 21:15.23 | Katty | cause you can nukerwave them, and they won't stain |
| 21:15.31 | Katty | which is great for porton control |
| 21:15.31 | ChannelZ | You could make your own little packets of frosting that could be cut open and squeezed on |
| 21:15.37 | Katty | YES |
| 21:15.39 | sequencer | nice topic |
| 21:15.40 | Katty | that'd be hottt |
| 21:15.46 | sequencer | any idead fo cupcakes ? :S |
| 21:15.47 | sequencer | :D |
| 21:15.49 | rsmiley | I have a question. Lets say that I want to set up my sip.cfg. Does it require all the dialplans? |
| 21:15.53 | Katty | http://www.seriouseats.com/recipes/images/10.11.10cscakejartop.jpg <- cake in a jar |
| 21:16.01 | Katty | sequencer: cupcakes are ...cakes |
| 21:16.06 | Chainsaw | rsmiley: Your dialplan goes in extension.conf, not sip.conf |
| 21:16.07 | Katty | sequencer: they're just little cakes, with frosting |
| 21:16.11 | sequencer | no no |
| 21:16.12 | Katty | sequencer: muffins are cakes, without frosting |
| 21:16.12 | ChannelZ | nice |
| 21:16.15 | sequencer | they arent just cakes |
| 21:16.21 | Katty | everything is just cake |
| 21:16.23 | sequencer | theyre my favs ;) |
| 21:16.28 | Qwell | Katty: I am not cake. |
| 21:16.38 | Katty | Qwell: you re what you eat. |
| 21:16.40 | p3nguin | I think rsmiley might be talking about digit map and sip.cfg for some phone. |
| 21:16.41 | Katty | Qwell: you are cake. |
| 21:16.52 | rsmiley | I am confused... |
| 21:16.58 | Katty | blueberry muffin, in a jar. |
| 21:17.18 | Katty | ohoh, and i saw meatloaf in muffin tins |
| 21:17.22 | Katty | with 'mashed potato' frosting on top |
| 21:17.25 | nny | navaismo: hmm failed just killed it without calling the extension |
| 21:17.28 | Katty | it was /adorable/ |
| 21:17.52 | p3nguin | I haven't heard of the "failed" extension before. Sounds completely made up. |
| 21:17.56 | Katty | poor design for baking meatloaf. you should have all the edges exposed for proper crisping |
| 21:18.00 | nny | p3nguin: ha -_-... |
| 21:18.01 | Katty | and for grease run off |
| 21:18.04 | rsmiley | I have about 40 sip cisco 7960's, and I want to make it easy. To set up their tftp stuff what do I need to do? |
| 21:18.16 | nny | p3nguin: so is there ANY way to trap and perform something if http://pastebin.com/xfZaFiWp happens?? |
| 21:18.19 | p3nguin | rsmiley: install tftpd |
| 21:18.23 | rsmiley | check |
| 21:18.40 | rsmiley | and dns, and the phone network is on its own vlan. |
| 21:18.41 | p3nguin | rsmiley: Get SIPDefault.cfg and a sample for SIP<MAC>.cfg |
| 21:18.57 | rsmiley | also check. |
| 21:20.08 | p3nguin | rsmiley: Use your dhcpd to send the IP address (or host name) of the tftp server to the phone when they boot up. If it's the same as the dhcpd, you can skip that part. |
| 21:20.18 | sequencer | anyways.. Katty |
| 21:20.24 | sequencer | how woul we do it ? ;) |
| 21:20.27 | sequencer | would*\ |
| 21:20.33 | Katty | do what |
| 21:20.43 | sequencer | just put em in the oven ? |
| 21:20.46 | Katty | yep |
| 21:20.48 | Katty | it's glass |
| 21:20.53 | Katty | like a casserole dish or a lasagna pan |
| 21:20.57 | Katty | it's just shaped different |
| 21:21.04 | sequencer | hmm.. |
| 21:21.04 | rsmiley | yeah, that I did in dhcp.conf and that all works. |
| 21:21.17 | Katty | obviously adjust baking times |
| 21:21.21 | sequencer | sounds interesting |
| 21:21.22 | Katty | the smaller the pan, the quicker it cooks |
| 21:21.30 | p3nguin | katty: IGA sells breads in Mason jars. |
| 21:21.30 | sequencer | absoloutly |
| 21:21.31 | p3nguin | sweet breads |
| 21:21.46 | Katty | oh yes, and since you can buy them in different sizes, it's a great portion control thing |
| 21:21.48 | p3nguin | rsmiley: And the problem was... what? |
| 21:21.48 | nny | anyone know how to perform actions after a call fails? |
| 21:21.56 | Katty | 16 oz for soups, 8oz for main dishes |
| 21:21.57 | p3nguin | dialplan! |
| 21:21.59 | nny | or do I need to make a workaround? |
| 21:22.05 | Katty | who am i kidding, 16 oz for everything including dessert! nomnom |
| 21:22.10 | p3nguin | heh |
| 21:22.17 | p3nguin | 32 oz! |
| 21:22.21 | Katty | YES |
| 21:22.23 | rsmiley | I was under the impresson that I needed to do some special magic with the sip.cfg... |
| 21:22.30 | nny | p3nguin: failed: used if an auto-dial out call fails (that had context, priority and extension specified) |
| 21:22.34 | p3nguin | rsmiley: There's no sip.cfg |
| 21:22.40 | nny | p3nguin: from: http://www.voip-info.org/wiki/view/Asterisk+standard+extensions |
| 21:23.13 | p3nguin | nny: Cool, so it's not made up. |
| 21:23.38 | Katty | those clear mason jars would be adorable for halloween too |
| 21:23.53 | Katty | i'm making 'worm pie' which is actually just no bake cheese cake on top of crushed oreos (dirt) and gummie worms |
| 21:23.55 | nny | p3nguin: yeah, just doesn't seem to be working.. urgh |
| 21:24.09 | p3nguin | rsmiley: And if I said SIPDefault.cfg, I meant SIPDefault.cnf. Your cfg stuff confused me. |
| 21:24.41 | p3nguin | 'Cause there is no cfg for 7940/7960 with SIP. |
| 21:25.04 | rsmiley | so all I have to have is the genaric conf for the phones and point it at its sip<mac>.cnf with its numberid and secret? |
| 21:25.31 | p3nguin | The phone will automatically look for SIP<MAC>.cnf on the tftpd. |
| 21:25.47 | p3nguin | It will look for SIPDefault.cnf and then SIP<MAC>.cnf |
| 21:27.02 | rsmiley | the tutorial that I read adds a file for the displayname and address for the lines. is that correct? |
| 21:27.12 | p3nguin | SIPDefault.cnf will contain all the settings common to all phones. SIP<MAC>.cnf will have the settings that are for each individual phone. |
| 21:27.30 | navaismo | @nny we use failed or you can use hangupcause or reason variable within gotoif |
| 21:27.43 | p3nguin | rsmiley: Let me find some examples for you. |
| 21:28.00 | rsmiley | thanks, im a little lost. |
| 21:28.58 | nny | navaismo: yeah it should* work, but after the failure it just exits and ignores my exten => failed,1, etc lines |
| 21:30.10 | navaismo | what asterisk version? |
| 21:30.22 | p3nguin | rsmiley: I can't find one online quickly, so I'll post mine. |
| 21:30.53 | nny | navaismo: 1.8.3.2 |
| 21:32.32 | navaismo | its a normal call or call file? |
| 21:32.40 | navaismo | Im using 1.6.2.20 |
| 21:33.15 | p3nguin | rsmiley: http://pastebin.com/JgVEaf2t |
| 21:33.47 | nny | navaismo: call file |
| 21:33.51 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-rofclrlxaqggjpsa) |
| 21:33.59 | Katty | HERE FILE |
| 21:34.01 | Katty | HERE FILE FILE FILE |
| 21:34.18 | navaismo | stupid question: do you make a dialplan reload? |
| 21:34.25 | p3nguin | dialplan reload |
| 21:34.43 | nny | navaismo: hmm think i see it now, one sec |
| 21:35.04 | nny | p3nguin: amd caught it, thought it was hitting a different context |
| 21:35.08 | nny | er navaismo ^^ |
| 21:38.15 | nny | navaismo: pebkac :S |
| 21:38.49 | p3nguin | rsmiley: Do you also need a sample of SIP<MAC>.cnf? |
| 21:38.49 | rsmiley | @p3nguin http://pastebin.com/nuNBxxLr are the configs I have been working on. |
| 21:38.51 | carrar | hugs FILE one last time before untieing it and letting it run back to KATTY |
| 21:39.03 | Katty | :> |
| 21:39.24 | p3nguin | rsmiley: It's all wrong. You said you have Cisco 7960. This file is for a Polycom. |
| 21:39.34 | rsmiley | that might be an issue... |
| 21:39.36 | Katty | time to go get dog food |
| 21:39.44 | Katty | and then i have a party to host |
| 21:39.48 | Katty | so, i'll see you crazy kids tmw |
| 21:39.49 | navaismo | i dont undestand? |
| 21:40.01 | p3nguin | This is the first time I have ever seen someone trying to use a Polycom config on a Cisco 7960. |
| 21:40.04 | *** join/#asterisk devmikey (~irc@96.46.249.230) |
| 21:40.19 | p3nguin | rsmiley: I just pasted a proper SIPDefault.cnf for you. Use it. |
| 21:40.46 | rsmiley | its been a long day... |
| 21:40.51 | p3nguin | rsmiley: And here is the SIP<MAC>.cnf: http://pastebin.com/Y2nSPFbT |
| 21:41.49 | rsmiley | thanks. |
| 21:41.52 | *** join/#asterisk LittleFool (~LittleFoo@over-dozed.com) |
| 21:42.14 | p3nguin | If you don't understand what a setting is, just ask. |
| 21:44.20 | LittleFool | How do i activate the mysql addon to record cdr? |
| 21:44.53 | rsmiley | so for my deployment I can make a script to kick out the sip<mac>.cnf and tweek the sipdefault.cnf and then I have to have it add the numbers to the dailplan and the extensions confs. then its plug and play. |
| 21:46.08 | p3nguin | Sure. For only a few phones, I'd do it by hand, but for 40, I'd probably use bash, sed, awk, perl, etc. |
| 21:46.36 | p3nguin | SIPDefault.cnf is the global file, so do it by hand. |
| 21:46.47 | rsmiley | bash and echo with an input file is what im thinking... |
| 21:47.42 | p3nguin | I only have settings for two line keys in the SIP<MAC>.cnf. For more keys, just add more sections. |
| 21:48.53 | rsmiley | I dont think we have any phones with more then one line, such a think might complicate my bash-fu. |
| 21:48.55 | *** join/#asterisk IsUp (5db65305@gateway/web/freenode/ip.93.182.83.5) |
| 21:49.12 | p3nguin | Some people will use all 2 or 6 keys for the same SIP account. |
| 21:49.19 | p3nguin | 7940/7960 |
| 21:49.32 | p3nguin | I feel like one is enough for one account. |
| 21:51.11 | rsmiley | me too |
| 21:55.26 | *** join/#asterisk beek (~klinebl@pdpc/supporter/bronze/beek) |
| 21:56.18 | *** join/#asterisk lanning (~lanning@173-8-187-197-SFBA.hfc.comcastbusiness.net) |
| 22:00.18 | *** part/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
| 22:09.12 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 22:20.38 | rsmiley | quit() |
| 22:20.42 | rsmiley | quit |
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| 22:22.21 | *** join/#asterisk ChrisInSydney (~Chris@60-242-81-231.tpgi.com.au) |
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| 22:27.37 | carrar | Is KATTY having another ZORK party again? |
| 22:28.44 | Maliuta | hehe. ZORK was the first book I ever bought |
| 22:29.04 | Maliuta | I think I still have it, if not the one of my ex's sons has it |
| 22:30.33 | sunfone | Book? I thought it was the original MUD. |
| 22:31.33 | Maliuta | were there MUD's in 1983? |
| 22:32.25 | *** join/#asterisk hobodave_ (~hobodave@pdpc/supporter/professional/hobodave) |
| 22:32.37 | sunfone | They weren't called that then :) |
| 22:32.53 | Maliuta | http://en.wikipedia.org/wiki/Zork_books would say "yes" |
| 22:33.40 | sunfone | Cool. I wonder which one came first... the book or the game? |
| 22:34.33 | Maliuta | "The first version of Zork was written in 19771979 on a DEC PDP-10" |
| 22:34.33 | sunfone | http://en.wikipedia.org/wiki/Zork |
| 22:34.43 | sunfone | right :) Same page |
| 22:34.43 | Maliuta | source was http://en.wikipedia.org/wiki/Zork |
| 22:34.54 | sunfone | So the game came first then? |
| 22:34.57 | Maliuta | just more reason for me to get a PDP 10 |
| 22:35.01 | Maliuta | jah |
| 22:35.02 | sunfone | Ha! |
| 22:35.11 | Maliuta | they books didn't start 'til '83 |
| 22:35.14 | sunfone | I wrote a C compiler for a PDP 11 in school |
| 22:35.32 | sunfone | kind of dating myself I guess |
| 22:35.37 | Maliuta | I want a PDP 10, there are still a couple hooked up to the 'net |
| 22:36.05 | sunfone | We ran BSD on a PDP 11 |
| 22:36.39 | Maliuta | I'm afraid I am little post that era, still love the kit and practices though |
| 22:36.48 | Maliuta | I was 5 in '83 |
| 22:37.16 | sunfone | :) |
| 22:37.23 | Maliuta | reminds me I need to track down a copy of the H2G2 game |
| 22:38.02 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 22:38.38 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
| 22:38.58 | Maliuta | oh. It seem that the same guys responsible for zork did the H2G2 game |
| 22:39.47 | sunfone | Friend of mine just BOUGHT the Zork series... runs it via wine on Ubuntu |
| 22:39.51 | *** join/#asterisk IsUp (5db65305@gateway/web/freenode/ip.93.182.83.5) |
| 22:40.24 | sunfone | Some company is wrapping emulators around all these old games... so you don't have to worry about highmem, etc. |
| 22:43.45 | Maliuta | Yeah, I'm going to grab some emulator stuff and get me H2G2 (apparently it's classed as "abondonware") |
| 22:45.31 | sunfone | I used to be of the opinion that because of the 'net these games would never die, but the truth is once everyone who used to play them passes on, they too will be forgotton :( |
| 22:45.38 | sunfone | forgotten |
| 22:45.49 | sunfone | forgottun? |
| 22:47.17 | KavanS | g00gle? |
| 22:51.02 | *** join/#asterisk devmikey (~irc@96.46.249.230) |
| 22:51.11 | Maliuta | are you sure that the library on congress isn't archiving them |
| 22:52.31 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
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| 23:03.16 | *** join/#asterisk drynish (~drynish@modemcable016.169-21-96.mc.videotron.ca) |
| 23:03.20 | drynish | Hello guys |
| 23:03.32 | drynish | What is the use of answeronpolarityswitch |
| 23:04.31 | drynish | Oh I just notice the AMD use |
| 23:08.16 | ChannelZ | It's useful (maybe) on analog lines which don't otherwise have call progress |
| 23:08.49 | ChannelZ | but it depends on what your telco actually does. |
| 23:09.26 | drynish | Oh |
| 23:09.34 | drynish | Can I use it to detect if someone answered a call |
| 23:09.46 | nny | do all lines for a specific channel contain the channel name? |
| 23:09.51 | ChannelZ | That's the idea |
| 23:10.10 | drynish | Ok I will tell you the idea: I got so many issues with echo, that I decided to make the asterisk just as a answering machine |
| 23:10.13 | ChannelZ | nny: all "lines" of what? |
| 23:10.19 | nny | trying to use Notepad ++ to separate 30 calls in a log file, trying to find a common denominator that I can use to cut each channel section out |
| 23:10.24 | drynish | Just use my fxo card next to my phones in my house |
| 23:10.33 | nny | ChannelZ: sorry log entry lines |
| 23:10.38 | drynish | and make it answer a few rings later |
| 23:10.48 | drynish | It works well however, it always answer on each call! |
| 23:10.50 | drynish | ;) |
| 23:10.53 | drynish | that is not the idea :) |
| 23:11.04 | drynish | I just want it to answer when my line is not answered |
| 23:11.50 | nny | ChannelZ: nm it appears *most do* but not all |
| 23:11.53 | drynish | by other phones |
| 23:12.01 | ChannelZ | nny: the channel name should generally be the same unless transfers happened or whatever. Depends on what you're tracing I guess. |
| 23:12.16 | *** join/#asterisk Cesare (~Adium@creati59.lnk.telstra.net) |
| 23:12.44 | ChannelZ | drynish: make your dialplan do a Wait for however many seconds you want to delay for, and then have it Answer and dump them into voicemail or whatever. |
| 23:14.05 | navaismo | nny do you need the call flow in full.log ? |
| 23:14.10 | nny | someday i will find a log parser that can search asterisk logs for channels etc easily. Not saying they don't exist, just don't know of any |
| 23:15.38 | navaismo | I always use the full.log for search the call flow just search the number between brackets [] and then grep that number. Hope helps |
| 23:23.18 | nny | navaismo: ahh perfect, thanks |
| 23:24.25 | ChannelZ | splat |
| 23:29.31 | *** join/#asterisk Atriks (~Hatrix@2a01:e34:ee51:4670:61c4:c8e5:69b7:a868) |
| 23:29.31 | *** join/#asterisk cnu (cnu@the.ultimate.lamer.la) |
| 23:29.31 | *** join/#asterisk SunTsu (miyamoto@unaffiliated/suntsu) |
| 23:33.17 | *** join/#asterisk psykon (~psilikon@147-112.96-97.tampabay.res.rr.com) |
| 23:34.02 | *** join/#asterisk cnu (cnu@the.ultimate.lamer.la) |
| 23:34.45 | psykon | Should I expect 'PITCH_SHIFT' to work with an origination action from the manager interface? |
| 23:34.59 | *** part/#asterisk chrisjunkie (~Chris@2001:4428:22d:2:208:2ff:fe7e:6312) |
| 23:35.36 | ChannelZ | hmm are you able to specify what channel it's supposed to operate on? otherwise I don't see how |
| 23:36.41 | psykon | You can change the voice of both the caller and the called party by substituting rx or tx with both. |
| 23:36.48 | psykon | Set(PITCH_SHIFT(rx)=.7) |
| 23:38.58 | ChannelZ | right.. but where are you trying to do it from - you said 'wtih an origination', you mean like a 'channel originate' that calls an exten and as part of that exten you call the PITCH_SHIFT function? |
| 23:39.21 | ChannelZ | (or not 'channel originate', I guess the AMI command is just 'Originate') |
| 23:41.43 | psykon | I was originating to a local channel. Something like Channel: Local/16095551212@context. Then in the context the first priority was the pitch_shift command. |
| 23:43.58 | ChannelZ | Hmm. |
| 23:44.30 | ChannelZ | I assume you're asking because it's not working :) Not sure. I wonder if the channel has to be answer()ed first for it to do anything |
| 23:46.51 | *** join/#asterisk IPNixon (~IPNixon@unaffiliated/ipnixon) |
| 23:47.22 | psykon | ChannelZ, Yep, it isn't working. Actually crashing *. |
| 23:48.01 | IPNixon | hey all, i have an x100p in a newly built asterisk box. i'm having a problem with disconnect supervision, though...any calls coming in from the POTS line aren't being dropped after the other person hangs up...anything i can do? |
| 23:50.35 | ChannelZ | Oh that's fun. And I guess it wouldn't have to be answered first normally since otherwise you couldn't (easily) call the function. I just tested it anyway though not from AMI |
| 23:52.10 | psykon | ChannelZ, cool. It would be nice if you could add it as an option during meetme confs. |
| 23:52.50 | ChannelZ | Didn't crash on a 'channel originate' from the console... |
| 23:53.16 | drynish | ChannelZ, that's what I did, however it answers even if I answered the call |
| 23:53.22 | psykon | ChannelZ, how did you set it from the console? |
| 23:53.44 | drynish | so the answeronpolarityswitch should do the job |
| 23:53.51 | ChannelZ | drynish: show me your dialplan.. either something else is going on or your ATA is answering it or something before Asterisk gets to it |
| 23:54.08 | *** part/#asterisk nny (~SM@cpe-174-107-223-014.sc.res.rr.com) |
| 23:54.16 | ChannelZ | psykon: channel originate DAHDI/4/mycellphone extension 210@internal |
| 23:54.18 | drynish | my dialplan is exactly what you told me |
| 23:54.48 | psykon | ChannelZ, How did you set PITCH_SHIFT from the console? |
| 23:54.51 | ChannelZ | I didn't tell you anything specific. Pastebin or show the call coming in on the console with verbose on 3 |
| 23:55.02 | ChannelZ | psykon: it's part of the 210 extension in extensions.conf |
| 23:55.22 | psykon | ChannelZ, oh right. Ok that is how I am doing it. |
| 23:55.28 | ChannelZ | psykon: pastebin what AMI commands you are sending and I can test that |
| 23:55.36 | psykon | k |
| 23:55.48 | ChannelZ | I don't use AMI much to know the syntax off the top of my head :) |
| 23:55.48 | drynish | http://pastebin.com/DfTT2deB |
| 23:56.12 | drynish | Oh no I know, sooner you told me something but I had to take care of the kids |
| 23:56.42 | ChannelZ | drynish: and you reloaded the dialplan? Show the console output for a call. |
| 23:56.48 | *** join/#asterisk FainaUkraina (~Gene@cm61-10-82-188.hkcable.com.hk) |
| 23:56.58 | drynish | I can't test it right now |
| 23:57.05 | drynish | so I'll have to wait for tomorrow |
| 23:57.09 | drynish | My card is a x100p |
| 23:57.17 | drynish | so it's just an analog card |
| 23:57.55 | drynish | cannot do anything else than waiting for the answeronreversepolarity |
| 23:57.56 | brad_mssw | pabelanger: no go |
| 23:58.14 | psykon | ChannelZ, http://pastie.org/2414136 |
| 23:58.14 | brad_mssw | pabelanger: just downgraded to the current 1.8.5.0 package and pri still doesn't work on ubuntu 10.04 lts amd64 |
| 23:58.31 | pabelanger | brad_mssw: *CLI> core show version |
| 23:58.56 | brad_mssw | pabelanger: Asterisk 1.8.5.0-1digium2~lucid built by pbuilder @ nighthawk on a x86_64 running Linux on 2011-08-22 15:28:44 UTC |
| 23:59.18 | ChannelZ | psykon: sec |
