| 00:00.06 | ZX81 | the bug I'm working on in IAX2 is just based on crap network connectivity with no calls |
| 00:00.18 | pcrane | hmm... |
| 00:00.27 | ZX81 | might want to swap their internal trunks over to SIP just in case - they're all on the same VPN |
| 00:00.32 | ZX81 | so no fw problems |
| 18:41.26 | *** join/#asterisk jbot (i=ibot@pdpc/supporter/active/TimRiker/bot/apt) |
| 18:41.26 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- CentOS 5.2 Zaptel troubles:type ~centos52bug -=- Asterisk 1.6.0-beta9 (2008/05/14) Asterisk 1.4.21.1 (2008/06/30) Asterisk 1.2.29 (2008/06/03), *-Addons 1.4.6 (2008/02/21) 1.6.0-beta3 (2008/04/02), Zaptel 1.4.11 (2008/05/28), Libpri 1.4.4 (2008/...) -=- #asterisknow or #asterisk-gui for AsteriskNOW -=- #switchvox for Switchvox (switchvox.com) -=- #freepbx for FreePBX |
| 18:44.14 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 18:44.46 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 18:45.37 | *** join/#asterisk angeldavid (n=angeldav@nelug/coreteam/pepo) |
| 18:52.01 | youseenothing | who is a good business grade voip provider in texas? |
| 18:52.07 | youseenothing | any suggestions? |
| 18:52.39 | willy23123 | any good manuals on asterisk? |
| 18:52.49 | *** join/#asterisk freezey (i=hidden-u@gw.mypublisher.com) |
| 18:53.00 | Assimilate | The source |
| 18:53.42 | freezey | just setup an asterisk call center simulation but i am wondering if their is a way to make it so that each agent has to login to phone... so if they move their seat they can login and the extension will follow with them? |
| 18:54.46 | beek | ~book |
| 18:54.47 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 18:54.55 | beek | willy23123: ^^^^^^^^ |
| 18:55.17 | Assimilate | freezey, I use agent callback login, but I think thats old.. |
| 18:56.13 | freezey | Assimilate: is that so they can login on the phone itself? cause i know nortel has something where the agent has to login into the phone before calls start working |
| 18:56.20 | [TK]D-Fender | freezey: use local channels, or agentcallbacklogin, etc |
| 18:57.23 | freezey | [TK]D-Fender: but would that present the login on the say polycom phone itself? |
| 18:57.43 | freezey | [TK]D-Fender: where the agent would actually have to enter in userid and pw |
| 18:57.47 | [TK]D-Fender | freezey: You mean an indication taht someone is loggedin on it? |
| 18:57.57 | *** join/#asterisk [hC] (n=hardcore@209.82.105.158) |
| 18:58.17 | [hC] | Anyone familiar with common problems that would cause Polycom phones to hang on "running sip.ld" indefinitely? |
| 18:58.36 | [TK]D-Fender | freezey: you can use the MicroBrowser for that if you want, but logging in/out is best done through the dialplan. Idications can be done via Presence, or the MicroBrowser. |
| 18:58.40 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 18:58.46 | [TK]D-Fender | [hC]: Corrupt configs. |
| 18:59.06 | [TK]D-Fender | [hC]: Or of course a corrupt sip.ld |
| 18:59.29 | freezey | [TK]D-Fender: i mean like when somebody sits down at the certain phone they would have to enter in their ID and PW for the extension to be pulled from asterisk.. |
| 18:59.44 | freezey | but enter it in physically on the phone for their information to be pulled |
| 18:59.59 | [hC] | [TK]D-Fender: I bet its a corrupt sip.ld, we had a power outage during an upgrade to 3.0.3... Ill have to factory reset it i suppose. It gets stuck trying to upload its boot log at a 0 byte file. |
| 19:00.00 | freezey | so it makes spaces swappable they can move anywhere and login to any phone |
| 19:00.00 | [TK]D-Fender | freezey: Fine, have them call in to log in. |
| 19:00.18 | freezey | [TK]D-Fender: and how would that be accomplished? |
| 19:00.19 | [TK]D-Fender | [hC]: do a downgrade, then re-up it |
| 19:00.44 | [TK]D-Fender | freezey: Make an exten to log them in by whatever means you want to manage your queue members like. |
| 19:02.36 | *** join/#asterisk chandoo (n=chandra@ool-4353b9e5.dyn.optonline.net) |
| 19:03.06 | [hC] | [TK]D-Fender: ah good call. |
| 19:03.21 | [hC] | [TK]D-Fender: i thought i could hold 4,6,8,* at bootup or something as well? |
| 19:03.46 | [TK]D-Fender | [hC]: that could fush the configs, but I'm not sure about SIP image. |
| 19:03.58 | [TK]D-Fender | [hC]: At best that might force a complete new download |
| 19:03.58 | [hC] | [TK]D-Fender: hmm.. right yeah. |
| 19:04.02 | freezey | [TK]D-Fender: so the person would sit at the phone and call in to this other extension and they would enter their exntesion number and pw and that would log them into the phone? |
| 19:04.08 | [TK]D-Fender | [hC]: So that MIGHT do it, but I've never had to personally. |
| 19:04.20 | [TK]D-Fender | freezey: Yes, something like that. |
| 19:10.07 | *** join/#asterisk gr0mit (n=tim@extrt.txrx.org.uk) |
| 19:12.00 | [hC] | [TK]D-Fender: very strange. its like the FTP server chokes when the phone tries to upload its boot log. the phone never sends the data. (put older bootrom and sip firmware on there) |
| 19:12.13 | [TK]D-Fender | [hC]: check your perms |
| 19:12.29 | [TK]D-Fender | [hC]: and don't forget that your hard reset might hit FTP creds ass well |
| 19:13.01 | [hC] | [TK]D-Fender: thats all good. tcpdumping i see the phone do a STOR <mac>-boot.log and the server says ok send data, then it just hangs. perms are good. I'll have to muck about some more |
| 19:16.32 | *** join/#asterisk Ebola (n=Ebola@unaffiliated/ebola) |
| 19:20.47 | tzanger | FME is good from DC-2GHz and is designed for mobile antennas |
| 19:21.07 | tzanger | TNC is threaded BNC ok |
| 19:21.22 | tzanger | and TNC is good for 11GHz |
| 19:26.45 | [hC] | [TK]D-Fender: sigh.. this polycom is borked. it goes to update the bootrom, and then tries to upload its logfile, but hangs trying to upload its boot log (ftp server is fine, manually ftping works). i wonder if its bricked or something. |
| 19:27.24 | *** join/#asterisk lanning (n=lanning@ip67-152-85-190.z85-152-67.customer.algx.net) |
| 19:27.50 | [TK]D-Fender | [hC]: Down/up the BR, that should force a complete format and reDL ofboth images |
| 19:28.04 | [hC] | [TK]D-Fender: during bootup? |
| 19:28.23 | [hC] | [TK]D-Fender: oh. sorry. i see what you mean. thats what im doing. Its trying to download the downgraded firmware. |
| 19:28.46 | freezey | [TK]D-Fender: one of the bigger issues is that somebody is using elastix to get asterisk up and running and the way that the gui creates the conf files etc etc does it all weird and lays it out just odd |
| 19:29.05 | [hC] | [TK]D-Fender: "Updating BootROM" - it starts to download the bootrom, but in the middle of that tries to upload its boot log, It gets as far as sending the STOR command to the FTP server and the whole process hangs from there. |
| 19:29.29 | [TK]D-Fender | [hC]: I first suggested jsut fixing the SIP. Now I'm suggesting using the BR to force BOTH to refresh |
| 19:29.36 | [hC] | [TK]D-Fender: i did do both. |
| 19:30.08 | [TK]D-Fender | freezey: If they're running FreePBX, then it owns their sorry asses. Good luck with that. |
| 19:31.22 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.151) |
| 19:31.53 | [TK]D-Fender | freezey: And asking in here on their behalf for that queue question was a complete waste of time. |
| 19:35.17 | [hC] | if i start asterisk with -vvvvgc is there any way to 'background' it without killing it entirely and reloading it with safe_asterisk? |
| 19:35.32 | *** join/#asterisk angryuser (n=fdgfdg@df01t2-212-194-246-58.d4.club-internet.fr) |
| 19:35.36 | angryuser | hello |
| 19:35.41 | [TK]D-Fender | [hC]: Thats a #linux question for sure... |
| 19:35.55 | angryuser | fender, need a little of you help |
| 19:35.59 | [hC] | [TK]D-Fender: I mean, is there a way from the asterisk cli to detach from the process and background it? |
| 19:36.26 | [TK]D-Fender | [hC]: Same answer. |
| 19:36.26 | *** join/#asterisk mvanbaak_ (n=michiel@vanbaak.xs4all.nl) |
| 19:36.27 | freezey | [TK]D-Fender: yeah thats what i figured with freePBX you can only do so much correct? |
| 19:36.29 | angryuser | my sip show registry show nothing at all.... 1.4.19.1 |
| 19:36.30 | [TK]D-Fender | [hC]: nothing I can think of. |
| 19:36.32 | *** part/#asterisk gbr_ (n=gbr@200.103.96.98) |
| 19:36.37 | angryuser | bug ? |
| 19:36.38 | *** join/#asterisk gbr_ (n=gbr@200.103.96.98) |
| 19:36.39 | [hC] | [TK]D-Fender: thanks. |
| 19:36.47 | [TK]D-Fender | angryuser: Then you aren't registered to anything |
| 19:36.58 | angryuser | i got the lines set |
| 19:37.12 | *** join/#asterisk gbr_ (n=gbr@200.103.96.98) |
| 19:37.13 | [TK]D-Fender | angryuser: set your verbose to 10 and retry |
| 19:37.19 | rwaite | yay, my asterisk phone system is officially gold |
| 19:37.27 | [TK]D-Fender | angryuser: Still nothing = no registration. |
| 19:37.30 | rwaite | no probs yet and it only took me 6 weeks :) |
| 19:37.39 | [TK]D-Fender | rwaite: Sell while the market is high. |
| 19:37.54 | rwaite | hah. 20 million? |
| 19:38.03 | rwaite | once... twice |
| 19:38.05 | tzanger | rwaite: you must have some really good MOH |
| 19:38.41 | [TK]D-Fender | rwaite: Sold. |
| 19:38.53 | [TK]D-Fender | rwaite: I'll be shipping your the box of confetti shortly :p |
| 19:39.02 | angryuser | i got register => id:pass@provider.com/id Nothing ;( |
| 19:39.11 | *** join/#asterisk SQLDarkly (n=administ@199-117-163-66.dia.static.qwest.net) |
| 19:39.13 | *** join/#asterisk pa (n=pa@unaffiliated/pa) |
| 19:39.23 | [TK]D-Fender | dumps a ream of paper into his industrial shredder |
| 19:39.23 | rwaite | lol |
| 19:39.26 | SQLDarkly | Does anyone use USB Headsets with IP Softphones? |
| 19:39.27 | angryuser | under [general] of course |
| 19:39.45 | rwaite | i'll even include my 15 yr old test phone, free of charge! |
| 19:39.51 | [TK]D-Fender | angryuser: then you've done something wrong, or it simply FAILS, so nothing to see. Go watch your SIP DEBUG |
| 19:39.58 | [TK]D-Fender | SQLDarkly: Likely. |
| 19:39.59 | angryuser | ok |
| 19:40.11 | SQLDarkly | Looking for an external usb solution so the audio can be split after the DSP |
| 19:40.34 | [TK]D-Fender | SQLDarkly: to what end? |
| 19:40.48 | rwaite | i am considering trying to setup some way to have the asterisk debug output scroll on my x11 desktop |
| 19:41.03 | rwaite | since i always have a term open with it anyway |
| 19:41.17 | SQLDarkly | Trainer has asked to be able to plug in a headset with an agent for coaching. This of course is not possible simply by plugging two headsets in as there is no way to control the feed |
| 19:41.24 | CVirus | http://rafb.net/p/WEsOio37.html .. why isn't the Goto application executed ? |
| 19:41.25 | [TK]D-Fender | rwaite: "asterisk -rvvvvvvv" <- |
| 19:42.19 | [TK]D-Fender | CVirus: when/why do you think it should be? |
| 19:42.31 | rwaite | yeah but how to get it to go onto the root window? |
| 19:42.43 | [TK]D-Fender | rwaite: huh? |
| 19:42.48 | angryuser | ok have 3 packet till it's destroys channel any ideas ? http://www.pastebin.ca/1071694 |
| 19:43.05 | CVirus | [TK]D-Fender: I think it should go back to Answer() after I insert any digit .. right ? |
| 19:43.17 | [TK]D-Fender | CVirus: No. |
| 19:43.33 | [TK]D-Fender | CVirus: your Goto is in "s", not "_X" |
| 19:43.33 | rwaite | there are programs that will tail the system logs on your X11 root window consistently, but i dont know how to make them. it'd be nice to have the same for asterisk |
| 19:43.49 | [TK]D-Fender | rwaite: Oh, you mean like a "background?" |
| 19:43.58 | SQLDarkly | I was thinking about some sort of device similarly used to split the signal from some of m-audios products. Curious if anyone had come across a request like this and was able to solve it |
| 19:43.59 | rwaite | yeah |
| 19:44.17 | [TK]D-Fender | rwaite: Oh... not a question for "here" :) |
| 19:44.29 | rwaite | lol i know :) |
| 19:44.35 | freezey | [TK]D-Fender: so if theirs a module out their to do this and its not built into freepbx then its not able to be used.. on the otherhand using entire asterisk CLI anything is possible? |
| 19:44.46 | [TK]D-Fender | SQLDarkly: Easier jsut to ChanSpy on your agent call rather than hack something ugle and spend money doing it. |
| 19:44.49 | [TK]D-Fender | ugly* |
| 19:44.59 | SQLDarkly | no doubt :) the bastards |
| 19:45.00 | CVirus | [TK]D-Fender: I changed it to exten => _X,n,Goto(incoming,s,1) |
| 19:45.11 | CVirus | [TK]D-Fender: and yet it doesn't go back to Answer() ... it hangs up the call |
| 19:45.40 | [TK]D-Fender | freezey: No such thing as a "module". Yu can do this in *. FreePBX builds your configs for you and as such does things ITS way. If you don't like it, tough luck. |
| 19:46.00 | [TK]D-Fender | CVirus: pastebin the new code and CLI output of the failed attenot at verbose 10. |
| 19:46.06 | [TK]D-Fender | attempt* |
| 19:46.19 | *** join/#asterisk j0 (n=dan@S01060016b6b541d2.va.shawcable.net) |
| 19:46.26 | angryuser | fender can you help me out on this one, i passed like 2 hrs befor asking, i dont know how to read sip debug:http://www.pastebin.ca/1071694 |
| 19:46.37 | [TK]D-Fender | angryuser: those are QUALIFY packets in your peer and does not have anything to do with a "register" |
| 19:46.49 | j0 | my WaitExten seems to wait for at least 2 digits before continuing.. i have digittimeout set to 1, but i'd only like it to take the first digit and continue |
| 19:46.51 | freezey | [TK]D-Fender: yeah that was the problem its using freepbx and thats whats limiting to certain other asterisk functions that could be used |
| 19:46.55 | angryuser | so there is no at all traffec |
| 19:46.59 | angryuser | traffic |
| 19:47.01 | j0 | strange thing is, it used to work, but i think an upgrade several months back broke it |
| 19:47.11 | [TK]D-Fender | j0: PASTEBIN is your friend |
| 19:47.24 | CVirus | [TK]D-Fender: the call is hungup from the side of the other server ... IAX terminates the call as soon as the dialplan is finished ... got it now .. thanks |
| 19:47.33 | [TK]D-Fender | j0: waiting for 2 digits has nothing to do with digittimeout |
| 19:47.54 | angryuser | and i have 0 line under 'sip show registry' ... |
| 19:48.21 | [TK]D-Fender | freezey: Don't ask for muffins out of a cookie-cutter system. |
| 19:48.33 | [TK]D-Fender | angryuser: And you are going around in circles. |
| 19:49.12 | angryuser | yes |
| 19:49.36 | angryuser | i am pretty sure it is not a context/syntax error |
| 19:49.52 | j0 | here's what's happening when I dial double digits into my dialplan. http://pastebin.com/d7cf9a16 |
| 19:49.52 | [TK]D-Fender | angryuser: Now should be the poitn where you realize you didn't show us your CONFIGS so we can tell you whats broken. |
| 19:50.05 | [TK]D-Fender | angryuser: No, it couldn't POSSIBLY be a config error on your part! |
| 19:50.16 | angryuser | <PROTECTED> |
| 19:50.21 | angryuser | ;) ok 1 min |
| 19:50.57 | [TK]D-Fender | j0: So they dialed 87... and don't pastebin some tinly little segment of your dialplan. pastebin the ENTIRE damn context |
| 19:51.07 | j0 | there's about a 5 second delay before it continues in my dial plan if i press 8 |
| 19:53.07 | j0 | [TK]D-Fender: sorry. |
| 19:54.06 | j0 | ah.. i got it while sifting through my plan.. i included a different context with extensions longer than 1 digit. |
| 19:54.44 | [TK]D-Fender | j0: Congratulations. |
| 19:54.56 | seanbright | so if you get a hangup with cause 99, what exactly does that mean? |
| 19:54.57 | [TK]D-Fender | j0: You can now pick a plushie from the middle shelf :) |
| 19:55.09 | seanbright | i don't know what AST_CAUSE_IE_NONEXIST means |
| 19:55.25 | Juggie | i can tell you IE = Information Element :) |
| 19:55.40 | *** join/#asterisk CanWood (n=chatzill@24.108.64.80) |
| 19:56.35 | seanbright | hmmm |
| 19:57.07 | Juggie | its a pri thing |
| 19:57.13 | angryuser | http://www.pastebin.ca/1071705 Fender ? |
| 19:57.45 | seanbright | well our PRI isn't showing any alarms |
| 19:57.50 | seanbright | but all of our outbounds get that |
| 19:57.59 | seanbright | and i hear the standard congestion stuff |
| 19:58.03 | [TK]D-Fender | ~sipregister |
| 19:58.04 | jbot | [~sipregister] SIP Registration is to tell your provider as to what IP address & EXTEN to send INCOMING calls to. Some ITSP's let you use a fixed IP or host rather that register. Registration is NOT normally needed to PLACE a call, as those are typically auth'd independently. Others accept UN-AUTH'D calls once you are registered (saves on negotiation BW) |
| 19:58.07 | [TK]D-Fender | angryuser: ^^^^ |
| 19:58.30 | bootc | seanbright: "This cause indicates that the information element(s)/parameter(s) were discarded. However, the information element is not required to be present in the message in order for the equipment sending the cause to process the message. " |
| 19:58.30 | angryuser | my server is inside my local net |
| 19:58.42 | angryuser | i need to register * inside my local net |
| 19:58.47 | bootc | as to what that means exactly I have no idea |
| 19:59.00 | Juggie | seanbright, does incoming work ok? |
| 19:59.12 | seanbright | Juggie: testing that now |
| 19:59.53 | [TK]D-Fender | angryuser: And your credentials are wrong for placing that call |
| 20:00.03 | [TK]D-Fender | angryuser: NOTHING to do with registering |
| 20:00.07 | Juggie | seanbright, pastebin console and also do you have your switch type configured properly. |
| 20:00.28 | angryuser | they are not wrong, i created them, 901/901 how hard they can be wrong really ? |
| 20:01.06 | [TK]D-Fender | angryuser: they clearly don't match. |
| 20:01.20 | [TK]D-Fender | angryuser: Where you screwed up I don't see enough to know yet. |
| 20:01.35 | angryuser | when my third party software is configured like sip client it works, hm |
| 20:02.02 | angryuser | let's say i have other registration lines to other providers |
| 20:02.22 | angryuser | why i dont see 0 lines when i type 'sip show registry' ? |
| 20:02.33 | angryuser | why i see * |
| 20:03.15 | [TK]D-Fender | angryuser: go look at the registration attempt. You have NOTHING under general. no bindport, codec, ANYTHING. this is such a stripped down setup I wouldn't be surprised if things are bad throughout. |
| 20:03.22 | [TK]D-Fender | angryuser: You aren't LOOKING at the problem. |
| 20:03.30 | [TK]D-Fender | angryuser: you are looking at everything else. |
| 20:03.37 | andrewy | for matching a * in the dialplan, do I have to use a special symbol or can I just use * ? |
| 20:03.46 | angryuser | i have cutted it all |
| 20:03.49 | [TK]D-Fender | andrewy: just * |
| 20:03.59 | angryuser | i will pb it completed |
| 20:04.03 | [TK]D-Fender | angryuser: and you are not showing SIP debug for the reg attempt |
| 20:04.11 | Juggie | seanbright, any luck? |
| 20:05.10 | angryuser | there is not reg attempt .... i turned sip debug on for 192.168.0.26 (remote trunk) and no traffic besides i copypasted befor |
| 20:07.31 | [TK]D-Fender | angryuser: if you enabled SIP debug and restarted * or did a SIP reload it should re-reg |
| 20:08.00 | *** join/#asterisk j0 (n=dan@S01060016b6b541d2.va.shawcable.net) |
| 20:08.01 | [TK]D-Fender | angryuser: if not, fill in the stuff under [general] that you should have in a sane environment. |
| 20:08.25 | emiller | i need a brush up: What does the ":1" mean in ${EXTEN:1} |
| 20:08.42 | [TK]D-Fender | emiller: returnt he EXTEN stripping the first digit |
| 20:08.59 | emiller | thank you Fender! |
| 20:09.16 | seanbright | Juggie: no sir |
| 20:09.27 | seanbright | Juggie: but i don't believe in luck, so... |
| 20:09.28 | seanbright | ;) |
| 20:09.42 | outtolunc | believe in some pri debug output <G> |
| 20:10.21 | seanbright | on the phone with the provider now |
| 20:10.22 | Juggie | seanbright, you need to pastebin some details. |
| 20:10.37 | seanbright | Juggie: no i don't... see? this is me not doing that :) |
| 20:10.47 | Juggie | you should then :) |
| 20:10.47 | seanbright | we'll get it worked out |
| 20:10.56 | seanbright | sorry to interript |
| 20:10.59 | *** join/#asterisk bkruse (n=bkruse@pool-71-115-162-59.gdrpmi.dsl-w.verizon.net) |
| 20:10.59 | *** mode/#asterisk [+o bkruse] by ChanServ |
| 20:11.00 | seanbright | interrupt* |
| 20:11.46 | angryuser | http://pastebin.ca/1071716 the upper part of sip.conf nothing special my sip_peer_trunk reported as OK 1 ms , but ne reg traffic, all i got for this peer and destination is is this http://pastebin.ca/1071721 |
| 20:12.13 | angryuser | i turned debug on trunk and on dest Ip |
| 20:12.42 | angryuser | really strange, maybe i have some module not loaded |
| 20:13.03 | [TK]D-Fender | angryuser: Filter all the comments out completely. |
| 20:13.38 | angryuser | is there any way to save all * info on start to log ? |
| 20:13.45 | [TK]D-Fender | angryuser: And for the last time, registering has NOTHING to do with ping, qualify(options) packets, or your PEER entries. NOTHING |
| 20:14.14 | angryuser | well it's all sip trafice i got, no reg, just a ping |
| 20:15.12 | [TK]D-Fender | angryuser: your other [general] section had NOTHING in it. FIX that. then pay attention to what comes in on your other system when you restart the one that should be registering. And while you're at it, turn off the "qualify" just so ou don't confuse yourself further. |
| 20:18.36 | angryuser | yep i admit it say no more i found it ;) |
| 20:20.20 | angryuser | i thought that there in no other [sections] between [general] and sip clients config |
| 20:21.31 | angryuser | ok thank's fender very much, i am going to kill myself for such a stupid error |
| 20:22.59 | [TK]D-Fender | angryuser: You're welcome. |
| 20:23.00 | FinboySlick | calls dibs on his computer. |
| 20:24.59 | *** join/#asterisk javb (n=javb@190.166.119.217) |
| 20:25.21 | javb | is there a way so that i can test if a mailbox is full ? |
| 20:25.29 | javb | (voicemail) |
| 20:25.33 | [TK]D-Fender | javb: Go count the messages |
| 20:25.54 | *** join/#asterisk nny_1 (n=Scott_My@64.203.239.83) |
| 20:26.01 | javb | [TK]D-Fender, can you give me an idea of how ? |
| 20:26.05 | angryuser | oh btw, when a destination domain has 2 ip's assigned (lets say 2 voip farms swithcing sometimes) and when * de a look up find 2 ip's he tryes to register with the first one, butthat farm in not working, will * tyr to register to second ip ? |
| 20:26.18 | [TK]D-Fender | javb: they're just files. go look. |
| 20:26.59 | javb | [TK]D-Fender, ok, but from Asterisk, how to count files quantity in an specific folder? |
| 20:27.03 | [TK]D-Fender | angryuser: If you give it an IP there is nothing to look up. If you give it a HOST it should probably look it up each time it tries it. |
| 20:27.26 | [TK]D-Fender | javb: FROM asterisk? Not sure if there is a way. Likely something you'll have to script outside of * |
| 20:27.32 | nny_1 | Hi all.. working on a simple web interface for our clients to manage users in a system (change passwords, email address, sms) nothing too fancy or trixish. I am looking at http://www.voip-info.org/wiki/view/Asterisk+GUI+phpMyEdit as a start, but the last time it was edited was 2004, (gonna test and see how it works, maybe mangle it a bit.) anyone else have an opinion on a simple web interface for those features? |
| 20:27.45 | [TK]D-Fender | javb: I think there might be a ${VM_COUNT} though... does ring a bell. |
| 20:28.11 | *** join/#asterisk chandoo (n=chandra@ool-4353b9e5.dyn.optonline.net) |
| 20:29.06 | FinboySlick | I keep getting: 'sSMTP[10740]: 501 5.1.7 Bad sender address syntax' when voicemail tries to send mail out. I imagine it isn't giving sendmail what it wants but how would I go about debugging just what asterisks outputs to my mailer? |
| 20:29.16 | [hC] | [TK]D-Fender: interesting. tried FTP, HTTP, both failed. Finally got it working by using TFTP. |
| 20:29.30 | [TK]D-Fender | [hC]: screams auth failure to me... |
| 20:29.34 | [hC] | [TK]D-Fender: (keeping in mind FTP has worked fine on this site for over a year). |
| 20:29.38 | [TK]D-Fender | anyways, heading home, BBIAB |
| 20:29.43 | javb | [TK]D-Fender, :( ... thats wigth, i need something like that, like a variable... but nothing . .. cant find anything |
| 20:40.18 | *** join/#asterisk Shotygun (n=thorn@82.166.243.195) |
| 20:41.52 | Shotygun | Hola, quick question, does (queue)penalty affects rrmemory? |
| 20:42.08 | Shotygun | latest 1.4 |
| 20:47.25 | *** part/#asterisk Mawkee (n=Mawkee@200.152.178.136) |
| 20:48.19 | *** join/#asterisk korihor (n=korihor@190.199.171.145) |
| 20:48.52 | *** join/#asterisk tvbuddy (i=IceChat7@12.175.85.122) |
| 20:49.02 | *** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194) |
| 20:49.54 | [TK]D-Fender | Someone asked me somenig right as I left at 20 minutes ago, what was it? |
| 20:50.00 | [TK]D-Fender | something* |
| 20:50.21 | outtolunc | [13:29] <javb> [TK]D-Fender, :( ... thats wigth, i need something like that, like a variable... but nothing . .. cant find anything |
| 20:50.21 | outtolunc | [13:29] * [TK]D-Fender has quit IRC |
| 20:50.47 | mvanbaak | [TK]D-Fender: I asked you to give me a beer |
| 20:52.17 | [TK]D-Fender | outtolunc: Did someone else answer him? |
| 20:53.23 | *** join/#asterisk gentoorax (n=gentoora@cpc1-stkn11-0-0-cust392.midd.cable.ntl.com) |
| 20:54.00 | *** join/#asterisk rwaite (n=richard@rrcs-74-218-125-86.central.biz.rr.com) |
| 20:54.44 | *** join/#asterisk stybba (n=alpha@201.193.44.196) |
| 20:55.56 | outtolunc | nope |
| 20:58.41 | errr | I have 2 asterisk boxes. 1 in Texas 1 in Colorado. I have found a bunch of guides on how to link them with iax2, but not with sip, any have a link on how to link these 2 boxes with sip? |
| 20:58.58 | [TK]D-Fender | errr: lookup "asterisk dual servers" on the WIKI |
| 20:59.02 | [TK]D-Fender | ~wikis |
| 20:59.02 | jbot | [~wikis] VoIP Wiki covering Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE. http://www.voip-info.org (c) Arte Marketing Inc / CommPartners |
| 20:59.09 | errr | thanks [TK]D-Fender |
| 20:59.44 | [TK]D-Fender | outtolunc: Oh well.. I found it |
| 21:00.45 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 21:01.08 | *** join/#asterisk luisjose (n=ljd@nelug/coreteam/luisjose) |
| 21:02.39 | seanbright | Juggie: you still abount? |
| 21:02.41 | seanbright | about* |
| 21:03.20 | errr | [TK]D-Fender: maybe Im missing something but I am only seeing iax2 examples there, just a mention of Of course you can also use SIP or H.323 (but not MGCP) to interlink two Asterisk servers, however IAX is the most common approach |
| 21:04.14 | lesouvage | Does any of you know a proper network performance tool that can be used to localize and pin point network problem with an Asterisk implementation? |
| 21:04.44 | [TK]D-Fender | errr: its almost no differnt for SIP. You register to the other box, just like any ITSP |
| 21:04.53 | [TK]D-Fender | errr: or not. A call is a call is a call. |
| 21:05.24 | Peri | lesouvage: traceroute? |
| 21:06.13 | errr | [TK]D-Fender: hmm, ok well I am able to call the extension that are local to each box using my sip channel I made, but I just cant make calls to outside lines, maybe Im just using the wrong context or something |
| 21:06.30 | lesouvage | Peri: I just read about http://freshmeat.net/projects/gnetwatch/ but I have no idea about the maturity of this project. |
| 21:06.38 | [TK]D-Fender | errr: could be. Go look at what you're doing and what debug messages say |
| 21:06.47 | errr | ok thanks |
| 21:07.03 | seanbright | ok, so here is the PRI debug output mixed with some dialplan foo... |
| 21:07.04 | Peri | lesouvage: that tool sounds a lot like netsaint or cacti |
| 21:07.09 | seanbright | http://pastebin.ca/1071748 |
| 21:07.15 | seanbright | the issue being that i can't make outbound calls |
| 21:07.46 | *** part/#asterisk nny_1 (n=Scott_My@64.203.239.83) |
| 21:08.02 | lesouvage | err: I managed to set up a sip trunk this week with the instruction on http://www.panoramisk.com/90/sip-trunk-with-asterisk/en/ It is working like a charm within the same network. |
| 21:09.04 | lesouvage | Or, to avoid the word trunk, I managed to interlink 2 Asterisk servers using SIP. |
| 21:09.27 | errr | lesouvage: awesome, thanks |
| 21:09.55 | [TK]D-Fender | lesouvage: or to avoid the word "link" because there is nothing "continuous" about this, you setup auth creds so they can call each other :) |
| 21:11.11 | lesouvage | [TK]D-Fender: I really want to avoid annoying you with words like sip trunk but what word should I use instead of it :-| |
| 21:11.15 | seanbright | having a problem placing outbound calls over a PRI, here is the PRI debug output mixed with some standard CLI output. if any other information would be helpful, I can provide it --> http://pastebin.ca/1071748 |
| 21:11.28 | [TK]D-Fender | lesouvage: I jsut gave you some :) |
| 21:12.46 | [TK]D-Fender | seanbright: pastebin your zaptel & zapata |
| 21:14.43 | *** join/#asterisk bsaxon (n=bsaxon@146.sub-75-250-80.myvzw.com) |
| 21:15.09 | seanbright | [TK]D-Fender: http://pastebin.ca/1071752 |
| 21:15.12 | lesouvage | [TK]D-Fender: I entered the channeld at 23:00 local time (just 12 minutes ago) so I'm afraid I missed it. |
| 21:15.14 | seanbright | span 1 is the one with the problem |
| 21:16.21 | *** join/#asterisk doolph (n=doolph@190.141.69.38) |
| 21:17.45 | [TK]D-Fender | [17:09]<[TK]D-Fender>lesouvage: or to avoid the word "link" because there is nothing "continuous" about this, you setup auth creds so they can call each other :) |
| 21:17.47 | [TK]D-Fender | [17:11]<lesouvage>[TK]D-Fender: I really want to avoid annoying you with words like sip trunk but what word should I use instead of it :-| |
| 21:17.55 | [TK]D-Fender | lesouvage: it was the line right ABOVE IT. |
| 21:18.55 | [TK]D-Fender | seanbright: "pridialplan=national" <- try it |
| 21:19.33 | lesouvage | [TK]D-Fender: tat is not a word that is a complete line. You can replace to many words for complete lines. |
| 21:20.12 | [TK]D-Fender | lesouvage: "setup auth creds so they can call each other" <- |
| 21:20.18 | *** join/#asterisk CVirus (n=Satan@82.201.174.27) |
| 21:20.44 | [TK]D-Fender | lesouvage: There is no single all-encompassing magic word to replace your OTHER misplaced all-encompassing word of "trunk" or "link" :) |
| 21:20.56 | jeev | Fender, take a break and come chew some gum with me |
| 21:21.20 | lesouvage | err: if you the same extensions (say 5001) on the two boxes then you will run into trouble. I have no idea why but it looks like a bug. Took me an hour to find out. |
| 21:21.25 | [TK]D-Fender | jeev: not something I have to synchronize with anyone for, thanks :) |
| 21:21.28 | seanbright | [TK]D-Fender: no go. --> [Jul 14 17:20:45] WARNING[23108]: chan_zap.c:9348 pri_dchannel: PRI Error on span 0: We think we're the CPE, but they think they're the CPE too. |
| 21:21.49 | [TK]D-Fender | seanbright: What are you plugging into? |
| 21:21.54 | jeev | come on, it tastes great.. it may be laced with every imaginable STD that could happen to your mouth. |
| 21:22.09 | seanbright | [TK]D-Fender: a t1? |
| 21:22.22 | JenniferAkemi | sounds like you are both being user |
| 21:22.32 | JenniferAkemi | instead of a network and a user |
| 21:22.41 | seanbright | hmmm |
| 21:22.47 | seanbright | runs away for a moment |
| 21:23.15 | *** join/#asterisk Strom_C (n=strom@208.127.172.112) |
| 21:23.32 | lesouvage | seanbright: are you conecting to a telco or to pbx with a t1 conection (lets say are you setting up a trunk ;-) ) |
| 21:24.06 | outtolunc | i bet he is doing both (dropping asterisk box in the middle) |
| 21:24.55 | *** join/#asterisk jjshoe (n=jjshoe@72.37.252.50) |
| 21:25.09 | *** part/#asterisk gbr_ (n=gbr@200.103.96.98) |
| 21:25.15 | jjshoe | is there an internal limit on the length of cid name/num within asterisk? |
| 21:25.19 | lesouvage | Then you have to change mode on the internal side and if possible use E1 because it will give you 6 extra lines, 30 instead of 24 |
| 21:25.23 | *** join/#asterisk exothermc (n=miles@74.85.89.146) |
| 21:25.27 | [TK]D-Fender | jjshoe: probably |
| 21:26.21 | seanbright | [TK]D-Fender: digium te220 -> pri -> pstn |
| 21:26.51 | *** join/#asterisk thansen|laptop (n=thansen@19.243.sfcn.org) |
| 21:26.54 | [TK]D-Fender | seanbright: what exactly is it plugged into? |
| 21:27.13 | seanbright | [TK]D-Fender: a cable with multiple wires that are twisted... hehe... |
| 21:27.17 | *** join/#asterisk ZX81 (n=matt@202.20.97.211) |
| 21:27.28 | seanbright | [TK]D-Fender: could you state your question in a manner in which i might be able to answer it :) |
| 21:27.42 | [TK]D-Fender | seanbright: no, that's the WIRE that connects it to something else. |
| 21:28.04 | seanbright | [TK]D-Fender: a 25 pair to a smart jack |
| 21:28.19 | seanbright | telco tested clean to CSU |
| 21:28.24 | seanbright | possible we're still looped up? |
| 21:28.59 | jjshoe | The other question I have is the proper syntax for setting caller-id information from an agi. All I see is 'SET CALLERID' but nothing in regards to the newer format of CALLERID(name) or CALLERID(num) |
| 21:29.26 | [TK]D-Fender | seanbright: Silly quesition, why the hell are both of your cards ports using the same group #'s? |
| 21:29.53 | seanbright | [TK]D-Fender: so my outbounds go over both spans? |
| 21:30.07 | outtolunc | jjshoe, in 1.4.19 func_callerid.c had name and num at char[80] |
| 21:30.37 | [TK]D-Fender | seanbright: ok, well the error before said both sides were acting as CPE. Clearly not good. |
| 21:31.32 | seanbright | [TK]D-Fender: agreed. |
| 21:31.45 | seanbright | [TK]D-Fender: just bounced asterisk and don't see that error again |
| 21:32.57 | exothermc | Can asterisk play custom messages while a caller is in a queue every X minutes? |
| 21:33.29 | putnopvut | exothermc: yes. look at the periodic-announce settings in queues.conf.sample in the configs directory. |
| 21:33.43 | exothermc | putnopvut: great thatnks |
| 21:35.47 | seanbright | [TK]D-Fender: out of ideas? |
| 21:36.34 | [TK]D-Fender | seanbright: Ext: 1 Progress Description: Calling equipment is non-ISDN. (3) ] |
| 21:36.43 | [TK]D-Fender | seanbright: Call your telco to confirm their settings |
| 21:36.59 | seanbright | [TK]D-Fender: will do |
| 21:37.55 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 21:38.19 | [TK]D-Fender | seanbright: Also, you are passing them a 7-digi CID (typically not great), and a 13 digit number CALLED (wtf?). Care to clarify? |
| 21:38.50 | seanbright | [TK]D-Fender: hmm? where are you seeing that? |
| 21:39.04 | exothermc | putnopvut: is there anyway to have a group of say 10 recordings and cycle through the group with periodic-announce settings ? |
| 21:39.29 | putnopvut | exothermc: well, you can do that in Asterisk trunk and Asterisk 1.6.0, but not in any version before that. |
| 21:39.41 | putnopvut | For those versions, you can have a comma-separated list of files to play. |
| 21:40.36 | exothermc | is 1.6 feature frozen yet? |
| 21:40.59 | *** join/#asterisk [hC] (n=hardcore@S01060016b6b53c0c.vc.shawcable.net) |
| 21:41.08 | *** join/#asterisk Vec (n=Vec@78-86-163-97.zone2.bethere.co.uk) |
| 21:41.39 | seanbright | [TK]D-Fender: oh i see... yeah, i am not sure where that is coming from. |
| 21:42.16 | seanbright | [TK]D-Fender: the "Called number" is strange because i'm just doing "Dial(Zap/g0/8888888888)" |
| 21:42.57 | seanbright | not sure where it's getting another 3 digits |
| 21:44.54 | *** join/#asterisk emiller (n=ed@70.89.48.17) |
| 21:45.04 | [TK]D-Fender | seanbright: New PB, and DON'T mask #'s |
| 21:45.18 | seanbright | [TK]D-Fender: heh, ok |
| 21:45.19 | [TK]D-Fender | seanbright: Nobody cares who you are or who you're calling. |
| 21:45.33 | emiller | i was curious if there was a command to show me what line in my dialplan was met with a call. |
| 21:45.52 | [TK]D-Fender | emiller: look at your CDR |
| 21:46.09 | emiller | good idea. thanks for the advise Fender |
| 21:46.15 | jjshoe | outtolunc ? |
| 21:46.45 | outtolunc | jjshoe: you asked about a callerid length limit *in* asterisk, didn't you? |
| 21:47.55 | VIPCarrier | any one use MOR PRO??? |
| 21:47.59 | putnopvut | exothermc: 1.6.0 is feature frozen. New features will be introduced in 1.6.1 |
| 21:48.06 | seanbright | [TK]D-Fender: http://pastebin.ca/1071771 |
| 21:48.32 | [TK]D-Fender | putnopvut: Good sign :) |
| 21:48.51 | jjshoe | outtolunc yeah |
| 21:48.58 | jjshoe | outtolunc gotchya. |
| 21:49.07 | jjshoe | outtolunc any idea on the proper format for setting callerid name from an agi? |
| 21:50.33 | seanbright | wtf was that? |
| 21:50.59 | *** join/#asterisk bmonty (n=bmontgom@cpe-70-115-194-204.satx.res.rr.com) |
| 21:51.34 | lesouvage | seanbright: your phone is flying to the ceiling? |
| 21:51.44 | [TK]D-Fender | seanbright: Hrm....I don't see the meaning for the 99 error |
| 21:51.54 | bmonty | I have a card that claims to be a Wildcard X101P. Is there any way that I can use it to connect to a analog telephone and provide dialtone? |
| 21:52.11 | seanbright | [TK]D-Fender: line 112 |
| 21:52.29 | russellb | bmonty: no, that is an FXO card |
| 21:52.29 | [TK]D-Fender | seanbright: Yes, I saw that,. |
| 21:52.32 | russellb | ~fxofxs |
| 21:52.32 | jbot | fxofxs is, like, An FXO port (red Digium module) expects to receive dialtone and receive ring voltage. You can connect it to a PSTN line from the telco. An FXS port (green Digium module) expects to provide dialtone and provide ring voltage. You can connect a phone or a fax to it. |
| 21:52.38 | seanbright | lesouvage: huh? |
| 21:52.43 | seanbright | lesouvage: was that you? |
| 21:53.00 | mvanbaak | pokes lesouvage |
| 21:53.07 | bmonty | russellb: ok, thanks....any recommendations for what FXS card to get? |
| 21:53.25 | mvanbaak | bmonty: get a digium wildcard |
| 21:53.29 | mvanbaak | TDM<something> |
| 21:53.39 | errr | lesouvage: that link was most helpful. Now my offices are working like a champ. Thank you |
| 21:53.42 | russellb | i would encourage you to consider a digium card. However, there are other options out there, as well. |
| 21:53.43 | [TK]D-Fender | bmonty: Probably best to buy an ATA like the Linksys SPA-210 or PAP2T-NA |
| 21:53.51 | lesouvage | seanbright: I haven't called you, I was just joking because the line "wtf was that?" looks kind of odd |
| 21:53.56 | outtolunc | jjshoe: 'set CALLERID somestring <10digitshere>' |
| 21:53.58 | [TK]D-Fender | bmonty: certainly much cheaper and more flexible in the vast majority of cases. |
| 21:54.01 | bmonty | I want to put the card in my asterisk box |
| 21:54.09 | bmonty | [TK]D-Fender: I'll check it out thanks |
| 21:54.12 | mvanbaak | bmonty: http://store.digium.com/products.php?category_id=45 |
| 21:54.29 | bmonty | mvanbaak: ouch..too expensive |
| 21:54.33 | [TK]D-Fender | SPA-2102* |
| 21:55.21 | mvanbaak | bmonty: remember you sponsor the Asterisk development with buying the digium cards |
| 21:55.26 | bmonty | anyone have experience with the IAXy? |
| 21:55.28 | outtolunc | jjshoe: if you meant using something like asterisk-perl this is it slightly diff |
| 21:55.45 | mvanbaak | the profit they make with it ensures they can continue to support the opensource community |
| 21:55.51 | [TK]D-Fender | bmonty: http://www.telephonydepot.com/product_p/105-054-pap.htm |
| 21:56.36 | [TK]D-Fender | mvanbaak: And you linked to 500$ TDM cards for a single FXS port. |
| 21:56.37 | seanbright | alright, so i'm basically screwed then? |
| 21:56.40 | bmonty | mvanbaak: I've got no problem buying from digium...I don't have the cash for one of those cards |
| 21:56.41 | lesouvage | bmonty: if it is just for tryig and testing you could start with an x100p original or clown. If it is just for studying you could also use hfc based isdn2 modem cards, intall asterisk on two old pc's and set up a trunk between the servers. |
| 21:56.42 | mvanbaak | just look at it this way: how much money do you save by using asterisk? |
| 21:57.02 | [TK]D-Fender | bmonty: follow my link. |
| 21:57.03 | lesouvage | clown=clone |
| 21:57.03 | mvanbaak | lesouvage: noooooooooo |
| 21:57.09 | mvanbaak | the X100P is horrible ! |
| 21:57.30 | jaytee | "Why won't my FXO line hangup?" |
| 21:57.30 | bmonty | [TK]D-Fender: will that box just sit on the network and talk to my asterisk server? |
| 21:57.41 | [TK]D-Fender | bmonty: yes, and let you plug 2 PHONES into it. |
| 21:57.52 | [TK]D-Fender | bmonty: it will speak SIP to your * box over your LAN |
| 21:58.03 | bmonty | [TK]D-Fender: nice..thanks for the help |
| 21:58.13 | [TK]D-Fender | (or across the internet / anytwhere TCP/IP can get you) |
| 21:58.21 | lesouvage | mvanbaak: I started my asterisk adventure with selling x100p cards. I had 2 or three dead on deliveries that were replaced and you ahve to play around wth the gains bt they are not that bad as people say. |
| 21:58.57 | lesouvage | I lots of these cards and sometimes I had to help to get them up and running. |
| 21:59.10 | jaytee | lesouvage, I beg to differ, they ARE as bad as people say, you either just had better luck or you have lower standards. |
| 21:59.12 | mvanbaak | lesouvage: not that bad ? |
| 21:59.43 | mvanbaak | lesouvage: no callerid in .nl, bad echo, unstable, no support, buggy driver |
| 22:00.04 | seanbright | [TK]D-Fender: i figured it out. |
| 22:00.20 | [TK]D-Fender | seanbright: What did you find? |
| 22:00.22 | seanbright | [TK]D-Fender: if i 'Set(CALLERID(name)=)' |
| 22:00.31 | seanbright | [TK]D-Fender: it works perfectly... not sure why |
| 22:00.47 | [TK]D-Fender | seanbright: AH... I did actually see reference to that some time last week actually... |
| 22:00.53 | [TK]D-Fender | seanglad to hear though |
| 22:00.58 | seanbright | [TK]D-Fender: ditto |
| 22:00.59 | *** join/#asterisk Aaton (i=yazz@ghost.230volts.net) |
| 22:01.02 | seanbright | [TK]D-Fender: thanks for the help |
| 22:01.10 | [TK]D-Fender | seanbright: np |
| 22:01.10 | jjshoe | outtolunc yeah traced it to a corrupt install of Asterisk::AGI :\ |
| 22:01.26 | lesouvage | I'm not saying it was perfect, but in combination with a little device to have callerid it worked reasonable and for those without the budget to by Digium or Sangoma it is the only option available. |
| 22:01.36 | seanbright | heh... and now it's failing again! nooooooooooo |
| 22:01.37 | seanbright | haha |
| 22:02.02 | *** join/#asterisk codefreeze-lap (n=murf@216.166.159.235) |
| 22:02.06 | seanbright | well that was weird. |
| 22:02.07 | jaytee | echo was horrible and required frequent retuning using fxotune with the X100P cards. And several times they'd lock and the only thing to fix it was a coldboot of the system. I bought 2 TDM04B cards and they were much better. Now I'm running 2 T1's PRI and have zero problems with it |
| 22:02.30 | outtolunc | fyi: that callerid(name)= <blank> was not a pri fix, it was voiceroute(iirc) and gafachi |
| 22:02.58 | seanbright | well it didn't fix it |
| 22:03.02 | outtolunc | (last week) |
| 22:03.04 | seanbright | well... it did for 1 call |
| 22:03.08 | seanbright | now it's all busted up again :) |
| 22:03.10 | *** join/#asterisk damjan (n=damjan@217.16.95.15) |
| 22:03.41 | outtolunc | seanbright: you should just originate a call to watch it vice all that dialplan stuff i seen |
| 22:03.56 | *** part/#asterisk Aaton (i=yazz@ghost.230volts.net) |
| 22:04.26 | outtolunc | did you ever put each pri in it's own group? |
| 22:04.31 | lesouvage | jaytee: I agree with you that the Digium cards (or Sangoma) are much much better and that the x100p card is not good enough for professional use, but imho it is good enough for trying and learning . |
| 22:05.02 | seanbright | outtolunc: no, but i think it's fixed again... i did a full reboot and all of my calls are going through now |
| 22:05.02 | lesouvage | if the two options are x100p or nothing. |
| 22:05.47 | jaytee | lesouvage, sure for just learning they'll do fine and you'll learn alot!!! :-) Probably more than you wanted but it'll come in handy when you actually get good hardware and then it goes bad on you down the road you'll have all that troubleshooting experience behind you. |
| 22:06.34 | *** join/#asterisk pcrane (n=pcrane@202.49.106.158) |
| 22:06.51 | bmonty | [TK]D-Fender: do you know what it means when it says the PAP2T is "unlocked"? |
| 22:07.07 | outtolunc | not tied to vonage |
| 22:07.17 | bmonty | outtolunc: thanks |
| 22:08.45 | *** join/#asterisk Strom_M (n=strom@208.127.172.112) |
| 22:09.36 | [TK]D-Fender | bmonty: How many phones do you want to connect? |
| 22:09.50 | bmonty | I'm going to connect all of the wiring in my house |
| 22:10.39 | [TK]D-Fender | bmonty: Ok, and are you intinding on using any analog LINES? |
| 22:10.53 | [TK]D-Fender | bmonty: Or are you looking to basically convert your whole home to VoIP? |
| 22:11.16 | bmonty | [TK]D-Fender: I'm use a DID and all phones in the house will use VOIP |
| 22:11.26 | bmonty | /use/using |
| 22:12.07 | [TK]D-Fender | bmonty: And you're OK with having all phones share that port like a "party line" (normal telco style)? |
| 22:12.51 | bmonty | [TK]D-Fender: the way I see it, the setup would be the same as if I had the telco come and install POTS at the house...right? |
| 22:13.00 | [TK]D-Fender | bmonty: Yup. |
| 22:13.12 | [TK]D-Fender | bmonty: You seem to have a pretty decent grasp on this. Good. |
| 22:13.20 | bmonty | except dialtone would come from my * box instead of the telco :) |
| 22:13.24 | [TK]D-Fender | bmonty: thats the device for you most likely. |
| 22:13.49 | bmonty | yeah, my initial question was to make sure that I couldn't force the hardware that I already own to do what I wanted |
| 22:13.54 | [TK]D-Fender | bmonty: Unless you have some other extra special need, but that pretty much nails it. |
| 22:13.55 | doolph | anyone had tried run asterisk on a amd geode 500mhz and 256mb ram |
| 22:14.07 | [TK]D-Fender | doolph: Plenty |
| 22:14.22 | [TK]D-Fender | doolph: Same as a Soekris NET5501 |
| 22:14.31 | [TK]D-Fender | doolph: they are popular |
| 22:14.39 | bmonty | [TK]D-Fender: I don't think I have special needs :) THanks again for your help! |
| 22:14.55 | doolph | can it work all those trascoding and things? |
| 22:14.57 | [TK]D-Fender | bmonty: basicaly cut the outside demarc, and plug the ATA into an existing jack |
| 22:15.07 | doolph | I have a via 1.5ghz and it seems to broke |
| 22:15.08 | [TK]D-Fender | bmonty: Everyone else in parallel will run off it |
| 22:15.20 | bmonty | [TK]D-Fender: this is a new house, the telco hasn't even been near it ;-) |
| 22:15.28 | [TK]D-Fender | doolph: No idea about transcoding load... you'll have to really look around for that |
| 22:15.37 | [TK]D-Fender | bmonty: Is the wiring complete? |
| 22:15.56 | bmonty | [TK]D-Fender: except for running a line to my computer room, yes |
| 22:16.05 | [TK]D-Fender | bmonty: if not, FFS don't bother with Cat3, and just run cat5e EvERYWHERE |
| 22:16.16 | bmonty | [TK]D-Fender: already done |
| 22:16.25 | [TK]D-Fender | bmonty: I mean exclusively |
| 22:16.52 | [TK]D-Fender | bmonty: 0 point to any cat 3. you can cross-connect so much more on RJ45, and use RJ11 plugs in RJ-45 jacks |
| 22:17.31 | lesouvage | doolph: you have to change a line in the make file |
| 22:17.34 | [TK]D-Fender | bmonty: Just a lesson I passed back from advising others in remodeling jobs |
| 22:17.44 | lesouvage | doolph: wait a moment. |
| 22:18.13 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.193) |
| 22:19.02 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 22:19.22 | bmonty | [TK]D-Fender: the house was wired with cat5 and RJ-45 jacks. Whoever set that part up was smart, but they didn't have the cable terminated in to a patch panel or anyplace where I can put the asterisk box. That is the part I still need to fix. |
| 22:19.48 | [TK]D-Fender | bmonty: So its all loose where you SHOULD put a patch-panel though, right? |
| 22:19.50 | bmonty | I'm just procrastinating about climbing around in the attic in the middle of summer |
| 22:20.21 | bmonty | [TK]D-Fender: not really, the builder terminated it in to an alarm panel. I'm going to splice in to that and run it to a patch panel in another part of the house. |
| 22:21.02 | bmonty | If I get really ambitious, I might run one wire for each jack :) |
| 22:21.05 | [TK]D-Fender | bmonty: Well you do seem to have this all well in hand. Glad I could advise something scaled to your first-round solution requirements :) |
| 22:21.24 | marc7 | does anybody have suggestions for generating useful reports from an asterisk CDR log (in a mysql backend)? I'm starting to think that I might need to go to greater lengths to capture information than mining whatever's in the CDR log |
| 22:21.33 | bmonty | [TK]D-Fender: I appreciate your advice |
| 22:22.00 | lesouvage | doolph: be sure that variable PROC has the value i586 There is a bunch of if statements that can be replaced by PROC=i586 (copied and past this from my own epia based asterisk server. |
| 22:22.09 | bmonty | bye! |
| 22:22.11 | *** part/#asterisk bmonty (n=bmontgom@cpe-70-115-194-204.satx.res.rr.com) |
| 22:22.32 | marc7 | it's for billing purposes, and I'm trying to sum the billsec/duration values for both inbound/outbound calls that come from a specific sip peer |
| 22:23.17 | *** join/#asterisk angryuser (n=sdfsdf@88.140.144.209) |
| 22:25.40 | lesouvage | doolph: and the changes has to be made in Makefile in the asterisk source cde directory (normally /usr/src/asterisk but it depends where you downoaded your source code) |
| 22:25.57 | *** join/#asterisk ZX81_ (n=matt@202.20.97.211) |
| 22:26.42 | doolph | I am copying the info |
| 22:28.10 | *** join/#asterisk craigk (n=craigk@58.174.150.119) |
| 22:29.49 | unpaidbill | marc i use something like this: "select date_format(calldate,'%m/%d'),sec_to_time(sum(billsec)),sum(billsec),count(calldate) from cdr group by dayofyear(calldate) |
| 22:30.01 | unpaidbill | works fine |
| 22:30.33 | unpaidbill | you'd have to modify it to group by userid or whatever |
| 22:30.42 | unpaidbill | you should probably check out asterisk-stat |
| 22:31.10 | lesouvage | doolph: don't forget to do a "make clean" before make when building asterisk |
| 22:35.00 | doolph | ok |
| 22:35.07 | codefreeze-lap | marc7: if you did this, you would not be the first! |
| 22:35.19 | *** join/#asterisk wasabi (n=wasabi@ubuntu/member/wasabi) |
| 22:35.47 | wasabi | So is there any decent way for a user to log off of one phone, and onto another? Looks like most phones don't make roaming users like that easy. |
| 22:35.56 | wasabi | polycoms and stuff |
| 22:36.12 | wasabi | I'd sure think it was neat if I could ask for auth on boot. :) |
| 22:36.30 | marc7 | unpaidbill: nice, yeah, the tricky part is identifying which user... but that query will come in handy |
| 22:36.37 | marc7 | codefreeze: i'll give asterisk-stat a shot |
| 22:36.51 | marc7 | again, per-peer statistics are going to be a little trickier |
| 22:37.02 | unpaidbill | should just have to group by srcext or whatever |
| 22:37.42 | unpaidbill | im actually rewriting asterisk-stat to use ajax/jgraph |
| 22:37.56 | unpaidbill | jgraph looks so much nicer |
| 22:38.42 | beek | Anyone have a favorite supplier for the Net 5501 instead of Soekris? They're "out-of-stock." |
| 22:39.20 | seanbright | outtolunc: where was this cid name problem talked about? one of the mailing lists? |
| 22:39.30 | *** join/#asterisk ltd (n=z@pat.transact.net.au) |
| 22:41.09 | unpaidbill | beek: nope :( i buy pcengines boards |
| 22:41.17 | th1 | I have a nokia phone connected to my asterisk and I'd like to have "+" translated into 00 in my dialplan, so I can make international calls from my phonebook, how can I do this in extensions.conf? I tried exten => _+X,1,Goto(00${EXTEN}:2) but it doesn't work |
| 22:41.20 | outtolunc | nope, just helping someone in here, as i had a client on gafachi that had the same issue (the person with the prob was on voice*something* sip provider |
| 22:41.29 | beek | unpaidbill: Those are the "Alix" boards? |
| 22:41.42 | unpaidbill | yeah, alix boards |
| 22:42.00 | seanbright | outtolunc: ah. bizarre. |
| 22:42.04 | unpaidbill | and wrap, in years past |
| 22:42.17 | beek | unpaidbill: Where do you buy these? |
| 22:42.43 | unpaidbill | i've had good luck with these guys http://www.mini-box.com/s.nl/sc.8/category.19/.f |
| 22:42.58 | beek | UnixDog: Thanks very much -- I'll head over there. |
| 22:43.11 | ManxPower | th1: You are matching ONLY phone numbers that are TWO characters long |
| 22:43.14 | th1 | any extensions.conf experts? |
| 22:43.15 | unpaidbill | the alix boards are cheaper than soekris too |
| 22:43.25 | unpaidbill | as long as you dont care about SATA/pci slots |
| 22:43.32 | th1 | ManxPower, so I need a dot afterwards? |
| 22:43.48 | beek | unpaidbill: Sorry about that -- I picked up the wrong NIC. I'll check them out. Thanks very much. |
| 22:43.49 | ManxPower | th1: _+X.,1,whatever |
| 22:43.57 | ManxPower | . means "one of more characters" |
| 22:44.42 | th1 | hm ok |
| 22:44.42 | ManxPower | th1: It sounds like you have not read the Asterisk book |
| 22:44.48 | th1 | I had long ago ;) |
| 22:45.01 | th1 | this was a basic mistake, and I've tried with the right setting with the dot |
| 22:45.03 | ManxPower | read it again. This is one of the most basic things in Asterisk pattern matches. |
| 22:45.04 | th1 | but it still doesn't work |
| 22:45.13 | th1 | I do know that |
| 22:45.26 | th1 | as I said it was just an artifact of my having tried lots of different things ;) |
| 22:45.29 | ManxPower | th1: put an ACTUAL copy of the CLI output as well as the CURRENT extensions.conf line that you think should be matching. |
| 22:45.33 | ManxPower | ~pb |
| 22:45.34 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 22:45.38 | ManxPower | on pastebin |
| 22:48.13 | th1 | ManxPower, what do I need to enable sufficient logging in the cli? |
| 22:48.22 | ManxPower | th1: Oddly, I am not paid for this. I am paid for other things. Should I go back to those other things, or can we move forward? |
| 22:48.49 | ManxPower | th1: "asterisk -rvvv" unless your logging is screwed up |
| 22:49.34 | unpaidbill | i'l pay you in POGs for this |
| 22:49.58 | *** join/#asterisk DarkRift (n=dark@bas10-montreal02-1177585130.dsl.bell.ca) |
| 22:50.02 | ManxPower | Euros would be better, or pretty much most currency except USA 8-) |
| 22:50.16 | *** join/#asterisk lvl- (n=lvl@101.248.dhcp.lht.hhs.nl) |
| 22:51.15 | th1 | ManxPower, it doesn't show anything in the console when I try a call starting with + |
| 22:51.51 | ManxPower | th1: then I totally misunderstand what you are doing. I assumed that calls were coming in from the carrier with + |
| 22:51.52 | th1 | just the phone says "address not in use". but if I have "sip debug" on it does show something |
| 22:52.04 | th1 | ah no it's coming from my nokia handset |
| 22:52.08 | ManxPower | If you are trying to DIAL the + then you must configure your IP phone to allow that, it's not handled in Asterisk |
| 22:52.30 | ManxPower | at least put SOMETHING on a pastebin |
| 22:52.48 | th1 | well, it does send something to asterisk. 2sec |
| 22:52.51 | *** part/#asterisk wwalker (n=wwalker@pdpc/supporter/sustaining/wwalker) |
| 22:52.58 | ManxPower | actually never mind. I can help with stuff coming from a carrier, but not some oddball phone |
| 22:53.39 | jameswf | root |
| 22:53.47 | unpaidbill | toor |
| 22:54.31 | [TK]D-Fender | unpaidbill: that's the root login/pass for Slax :) |
| 22:54.46 | th1 | manxpower: http://pastebin.ca/1071811 |
| 22:55.03 | th1 | I don't think there's something wrong with the phone |
| 22:55.12 | th1 | it's sending "+4560000" and it doesn't work in asterisk |
| 22:55.30 | unpaidbill | why would you want to dial + in the first plac |
| 22:55.31 | th1 | but when I send 004560000 it works |
| 22:55.57 | th1 | unpaidbill, because most of the numbers in my phonebook have + because that's what the GSM networks prefer |
| 22:56.31 | [TK]D-Fender | th1: then put a pattern including the + in front in your dialplan |
| 22:57.59 | th1 | [TK]D-Fender, that's what I tried |
| 22:58.03 | th1 | on the pastebin post |
| 22:59.18 | [TK]D-Fender | th1: [full_access] --- Looking for +4560000 in full_access_monitored (domain 10.10.4.253;user=phone) |
| 22:59.38 | [TK]D-Fender | th1: [full_access] != [full_access_monitored] |
| 22:59.45 | [TK]D-Fender | th1: Pay more attention next time |
| 23:00.14 | th1 | [TK]D-Fender, full_access_monitored just sets up some recording of the call and then goto full_access |
| 23:00.25 | [TK]D-Fender | th1: Do *I* see that? no. |
| 23:00.37 | [TK]D-Fender | th1: now pastebin something I can trust |
| 23:02.59 | *** join/#asterisk eXistenZ (n=existenz@unaffiliated/existenz) |
| 23:03.23 | th1 | http://pastebin.ca/1071816 |
| 23:04.14 | [TK]D-Fender | th1: Nother has functionally changed. |
| 23:04.29 | th1 | ? |
| 23:04.36 | [TK]D-Fender | th1: [full_access_monitored] does not have an exten with a "+" in front! |
| 23:05.06 | th1 | but [full_access] does |
| 23:05.19 | [TK]D-Fender | th1: and your call isn't LOOKING IN THERE. |
| 23:05.46 | th1 | in [full_access_monitored] its last line is Goto full_access |
| 23:05.46 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 23:05.49 | th1 | it works for all other numbers |
| 23:05.54 | th1 | just not the ones with + in front |
| 23:06.11 | [TK]D-Fender | th1: exten => _X.,3,Goto(full_access,${EXTEN},1) <-- this line will nver get CALLED in the first place |
| 23:06.27 | [TK]D-Fender | th1: *IT* doesn't match a number with a + in front |
| 23:06.30 | th1 | you're wrong about that, because it works for all other calls |
| 23:06.34 | th1 | ah |
| 23:06.39 | th1 | bah ;) |
| 23:06.41 | th1 | so that's it then |
| 23:06.56 | th1 | so I change it to _.? |
| 23:07.19 | [TK]D-Fender | th1: no, you make ANOTHER exten that will match with a + and do what you will with it. |
| 23:07.39 | *** join/#asterisk Firass-VC22 (n=Firass-z@method.vikcomm.wwu.edu) |
| 23:08.13 | [TK]D-Fender | th1: and go re-read the instructions on Goto. exten => _+X.,1,Goto(00${EXTEN:1}) <- this tries jumping to a PRIORITY. |
| 23:08.14 | th1 | like this: exten => _+X.,4,Goto(full_access,${EXTEN},1) |
| 23:08.22 | [TK]D-Fender | th1: Your logic was broken, and so was your code. |
| 23:08.38 | *** join/#asterisk jordanb (n=jordanb@adsl-68-20-182-32.dsl.chcgil.ameritech.net) |
| 23:08.46 | jordanb | Hi. |
| 23:09.11 | jordanb | Is there a way I can provide console dial with a caller ID string to use? |
| 23:09.26 | [TK]D-Fender | th1: just make a new pattern match and effectively repeat the 3 lines you have for the one WITHOUT the +) |
| 23:09.33 | [TK]D-Fender | jordanb: No. |
| 23:09.54 | jordanb | :< |
| 23:10.21 | jordanb | You know of a good way for me to test my telemarketer blocking scripts then? |
| 23:10.47 | [TK]D-Fender | jordanb: make an exten that will blank out your CID and GOTO the part that would answer the call. |
| 23:10.58 | [TK]D-Fender | jordanb: or call in with something that should trigger it |
| 23:11.21 | jordanb | Hrmm ok. |
| 23:11.43 | [TK]D-Fender | jordanb: "how do I test if my boat will float? Put it in water and see if it sinks" |
| 23:12.01 | jordanb | That's an obnoxious response. |
| 23:12.20 | JT | ... |
| 23:12.22 | [TK]D-Fender | jordanb: Highly accurate depiction of testing methodology |
| 23:12.37 | JT | sounds like someone's a little self-involved |
| 23:12.52 | [TK]D-Fender | jordanb: "How do I know if I configured * correctly for my line? Place a call using it or try calling your line". |
| 23:13.11 | [TK]D-Fender | jordanb: Same thing. Either way don't take any of it like any kind of attack. It wasn't |
| 23:13.14 | th1 | [TK]D-Fender, cool, thanks, it works |
| 23:13.21 | [TK]D-Fender | th1: Good to hear |
| 23:14.05 | [TK]D-Fender | Ok, I'm out for a bit. Later all. |
| 23:14.30 | jordanb | [TK]D-Fender; I'll just take it as the typical IRC asperger who says obnoxious things wihtout realizing it then. |
| 23:14.42 | jordanb | But thanks for the advise about making a testing extension. I think that will work. |
| 23:15.19 | [TK]D-Fender | jordanb: You are indeed entirely too sensitive. But still glad my advise was of use. |
| 23:15.32 | [TK]D-Fender | good luck everyone. |
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| 23:18.48 | JT | ffs |
| 23:18.53 | JT | adviCe not advise |
| 23:19.13 | JT | sends a "cheer up emo kid" postcard to jordanb |
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| 23:21.01 | angryuser | going to sleep |
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| 23:47.40 | angryuser | .. |
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