| 00:23.48 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 00:23.48 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.8.2.3 (2010/01/26), 1.6.2.16.1 (2010/01/18), 1.4.39.1 (2010/01/18), *-Addons 1.6.2.3, 1.4.13 (2010/01/14), dahdi-linux 2.4.0 (2010/09/01), dahdi-tools 2.4.0 (2010/09/01), libpri 1.4.11.5 (2010/11/22) -=- Visit the new official Asterisk wiki: wiki.asterisk.org -=- http://xkcd.com/806/ |
| 00:24.56 | *** join/#asterisk awclin_ (~alinford@g0962184.demon.co.uk) |
| 00:29.05 | *** join/#asterisk TimeRider (steve@188-220-33-123.zone11.bethere.co.uk) |
| 00:31.50 | nestAr | is there a config option for polycom to toggle having to select a line number to answer a call? |
| 00:32.14 | nestAr | IE: when the phone rings, it doesn't answer the call on a off-hook, have to select the ringing line (even if there's only one. |
| 00:34.08 | *** join/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
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| 00:56.41 | *** part/#asterisk ruben23 (~RLACUMBA@121.97.111.142) |
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| 01:00.49 | *** part/#asterisk retentiveboy (~retentive@74-95-28-33-Atlanta.hfc.comcastbusiness.net) |
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| 01:05.18 | SaiSoma|AtHome | nestAr, there may be, but it's not the default on the polycom's I have |
| 01:05.28 | SaiSoma|AtHome | that would mean that it's been altered from the original |
| 01:05.45 | SaiSoma|AtHome | if you have a centralized setup, you should be able to tell what settings are being changed |
| 01:05.56 | SaiSoma|AtHome | in the xml |
| 01:06.37 | *** join/#asterisk neurosys (~neurosys@c-98-254-246-250.hsd1.fl.comcast.net) |
| 01:07.30 | carrar | nestAr, download the default configs from polycom and start from there |
| 01:07.37 | carrar | only change what you need to change |
| 01:07.59 | carrar | and use current firmware |
| 01:08.06 | carrar | sipapp & boot |
| 01:08.33 | *** join/#asterisk p3nguin (~xwQ5kwYl6@cpe-0210-4bff-fe2b-9074.dhcp.a2infotech.com) |
| 01:08.39 | neurosys | I feel somehow someone has thought of generating a call that dials all extens except the one that initiated the call. I'd hate to reinvent the wheel :P |
| 01:09.13 | carrar | pretty east to do with a conditional statement |
| 01:09.15 | carrar | easy |
| 01:09.50 | carrar | so if you are calling from extension 1001 |
| 01:09.55 | carrar | GotoIf($["${CALLERID(num)}" = "1001"]?hangup) |
| 01:10.06 | carrar | in your page group |
| 01:10.20 | neurosys | hmm. That could work... |
| 01:10.59 | carrar | page out to local "page" extensions |
| 01:11.11 | carrar | local/ |
| 01:11.30 | neurosys | I wouldnt use SIP with the alert header? |
| 01:11.38 | carrar | you woul |
| 01:11.38 | carrar | d |
| 01:11.52 | carrar | that would also come after that line |
| 01:12.04 | carrar | depending on the model of phone |
| 01:12.25 | neurosys | aastra's |
| 01:14.24 | carrar | they are what: Set(ALERT_INFO=info=alert-autoanswer) |
| 01:14.34 | carrar | they are what: Set(ALERT-INFO=info=alert-autoanswer) |
| 01:14.40 | carrar | no idea |
| 01:14.44 | neurosys | <PROTECTED> |
| 01:14.53 | neurosys | well... not exactly for aastr |
| 01:14.55 | neurosys | *aastra |
| 01:15.07 | carrar | SIPAddHeader(Alert-Info: info=alert-autoanswer) |
| 01:15.20 | *** join/#asterisk sourcode (~code@ppp-58-8-231-44.revip2.asianet.co.th) |
| 01:15.23 | neurosys | I can use the page/intercom. But obviously if I try to page all, it includes the calling phone :P |
| 01:15.29 | neurosys | which makes problems. |
| 01:15.38 | carrar | from page dial local/ |
| 01:15.45 | carrar | dial each on seperatly |
| 01:15.47 | carrar | one |
| 01:16.08 | carrar | each one will test to see if is it'self |
| 01:17.21 | carrar | Page(Local/1101@blah&Local/1102@blah&Local/1103@blah) |
| 01:17.28 | carrar | then you have extension 1101 |
| 01:17.39 | carrar | that tests to see if it's 1001 calling |
| 01:17.43 | carrar | then set the headers |
| 01:17.46 | carrar | the dial |
| 01:17.52 | carrar | also before all that |
| 01:18.00 | carrar | check to see if it's registered |
| 01:18.07 | carrar | no need to call unreg phones |
| 01:18.40 | neurosys | I didnt know there was a page function :P |
| 01:18.47 | carrar | then you can also dial 1101 directly for directed paging |
| 01:18.52 | carrar | to extension 1001 |
| 01:19.42 | carrar | core show applications |
| 01:19.48 | carrar | read each one of those |
| 01:19.53 | carrar | core show application page |
| 01:20.12 | carrar | good to know whats in your toolbox before you start a job |
| 01:20.38 | nestAr | carrar: they have stripped out the base config so much in the new version.. that's what i'm running. i'll look through the admin pdf |
| 01:21.02 | neurosys | carrar: Is this in 1.6? |
| 01:21.12 | carrar | yes |
| 01:21.20 | carrar | 1.4 and up |
| 01:21.23 | carrar | maybe even 1.2 |
| 01:21.38 | carrar | I don't have any 1.2 boxes anymore |
| 01:22.08 | neurosys | carrar: I dont even see that function in 1.8 |
| 01:22.20 | neurosys | is in bundled into another application? |
| 01:22.46 | carrar | it's in my 1.8 |
| 01:23.01 | neurosys | :/ |
| 01:23.20 | carrar | it's a application |
| 01:23.22 | carrar | not a function |
| 01:23.41 | *** join/#asterisk Sipster (~Sipster@modemcable045.5-200-24.mc.videotron.ca) |
| 01:23.50 | neurosys | hmm. I went back to souce and looked in the menuconfig.. |
| 01:23.56 | neurosys | i have this under aplications: XXX app_page |
| 01:24.02 | carrar | it's default |
| 01:24.12 | neurosys | depends on dadhi |
| 01:24.17 | carrar | yeah |
| 01:24.20 | carrar | you need a timing source |
| 01:24.21 | *** join/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 01:24.33 | carrar | cause it uses a conference |
| 01:29.08 | carrar | hrmm Friday |
| 01:29.50 | Tozz_ | nope, its saturday |
| 01:29.57 | carrar | NO!! |
| 01:29.59 | carrar | Friday |
| 01:30.18 | neurosys | I thought Dahdi was required only for use with fxo/fxs |
| 01:30.38 | carrar | or digital circuits |
| 01:31.00 | neurosys | right. pri, etc |
| 01:31.08 | carrar | and timing source |
| 01:31.10 | neurosys | But not SIP trunking |
| 01:31.30 | carrar | might be able to use the dummy driver for timing |
| 01:31.35 | carrar | I don't bother with that |
| 01:32.24 | neurosys | how can I tell what asterisk is using currently for timing? |
| 01:32.24 | carrar | http://www.russellbryant.net/blog/2008/06/16/asterisk-16-now-with-a-new-timing-api/ |
| 01:32.29 | carrar | give that a try |
| 01:33.06 | carrar | still may need it |
| 01:33.08 | neurosys | ah |
| 01:33.17 | carrar | page uses meetme |
| 01:33.26 | carrar | I beleive |
| 01:33.34 | neurosys | it does |
| 01:33.55 | neurosys | that post was almost 3 years ago. guess that API isnt done heh |
| 01:34.02 | carrar | just slap a T1 card in your box |
| 01:34.05 | carrar | and be done with it |
| 01:34.19 | neurosys | heh this is an embedded platform ;) |
| 01:34.20 | carrar | nice to have somtimes |
| 01:35.36 | carrar | bbl |
| 01:35.52 | neurosys | kk |
| 01:47.19 | russellb | the API is done (has been for a long time) |
| 01:47.39 | russellb | what were you guys talking about? MeetMe() ? |
| 01:55.02 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 01:55.37 | nestAr | lol, i have something majorly wrong.. |
| 01:55.49 | nestAr | i just noticed this phone doesn't even ring on an incoming call |
| 01:55.54 | nestAr | nor does it seize the line |
| 01:55.58 | carrar | russellb, yeah |
| 01:56.02 | carrar | a timing source for it |
| 01:56.30 | carrar | I always use a t1 card so never really paid too much attention to other sources |
| 01:58.05 | russellb | yeah, so MeetMe() doesn't just use DAHDI for timing |
| 01:58.12 | russellb | DAHDI contains a conference mixing engine |
| 01:58.19 | russellb | which is what MeetMe() is built on |
| 01:58.40 | russellb | In later versions of Asterisk, we have a new bridging infrastructure, and a new conferencing application built on top of that - ConfBridge() |
| 01:58.55 | russellb | right now is has fewer features built-in to it, though. We are addressing that in Asterisk 1.10. |
| 01:59.07 | carrar | woah |
| 01:59.09 | carrar | 1.10!! |
| 01:59.10 | russellb | We're adding HD audio conferencing and a whole bunch of new features that will make it way better than MeetMe(). |
| 01:59.25 | russellb | HD conferencing is working already, next up is adding a bunch of features |
| 01:59.28 | carrar | better then MeetMe()!!! |
| 01:59.31 | carrar | Thas impossible! |
| 01:59.33 | russellb | we were designing configuration interfaces for it today actually ... |
| 02:00.35 | carrar | so DAHDI dummy driver is good enough for meetme without a t1 card then? |
| 02:00.40 | russellb | yes |
| 02:00.48 | russellb | no hardware needed. |
| 02:00.56 | carrar | nestAr, there you go! |
| 02:01.00 | russellb | and actually in the latest versions of DAHDI, you just need "dahdi" loaded |
| 02:01.07 | russellb | the "dahdi_dummy" stuff was integrated into the main module. |
| 02:03.59 | neurosys | thx russellb |
| 02:04.04 | russellb | sure, np |
| 02:04.08 | neurosys | :) |
| 02:06.19 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 02:07.17 | carrar | any short cuts russellb for paging large groups? |
| 02:07.33 | carrar | like a pagegroup= in sip.conf |
| 02:07.42 | russellb | use multicast RTP |
| 02:07.45 | hardwire | odbc? |
| 02:07.47 | carrar | the a pagegroup(X) or something in the dialplan |
| 02:07.48 | hardwire | oh |
| 02:08.04 | russellb | multicast RTP is the most scalable paging method |
| 02:08.04 | carrar | toss them into bridge |
| 02:08.26 | russellb | not all phones support it, you'll have to check if yours do |
| 02:09.04 | carrar | cept for remote installs |
| 02:09.13 | russellb | well ... yes. |
| 02:09.23 | russellb | in which case paging endpoints is just plain not scalable. |
| 02:09.29 | russellb | next thing is to do paging groups ... |
| 02:09.33 | russellb | record the page and do it in groups |
| 02:09.38 | carrar | the pagegroup could simplify it |
| 02:09.51 | russellb | you could build your own groups in a database, event just astdb |
| 02:09.58 | carrar | yeah |
| 02:10.27 | hardwire | russellb: are there phones that immediately grok Multicat RTP? |
| 02:10.36 | russellb | immediately? |
| 02:10.46 | hardwire | well |
| 02:10.49 | hardwire | without much tampering |
| 02:10.51 | russellb | yeah, they just watch for it at all times basically |
| 02:10.57 | hardwire | hmm |
| 02:10.57 | russellb | well, you have to turn it on ... |
| 02:11.04 | hardwire | I wonder if the spa-94x grok it |
| 02:11.07 | hardwire | checks |
| 02:11.13 | russellb | i don't remember which ones do |
| 02:11.22 | hardwire | then you just use emcast or what not? |
| 02:11.27 | russellb | i remember we made it work with a few different ones when we were documenting it in ~newbook |
| 02:11.43 | carrar | <Multicast_Address ua="na"> 224.168.168.168:6061^M </Multicast_Address>^M |
| 02:11.48 | russellb | nah, Dial(MulticastRTP/IP) |
| 02:11.50 | russellb | something like that |
| 02:12.01 | russellb | it's in ... |
| 02:12.03 | russellb | ~newbook |
| 02:12.04 | infobot | Asterisk: The Definitive Guide. A preview of the upcoming release is available at http://ofps.oreilly.com/titles/9780596517342/. The book is being released under a Creative Commons license: http://creativecommons.org/licenses/by-nc-nd/3.0/us/ |
| 02:12.08 | russellb | "Parking and Paging" chapter |
| 02:12.12 | hardwire | I'll read up |
| 02:12.13 | russellb | at least it's supposed to be |
| 02:12.26 | carrar | russellb, if only the new book had a search :) |
| 02:12.35 | hardwire | wget | grep |
| 02:12.40 | russellb | google, "<term> site:ofps.oreilly.com" |
| 02:12.47 | carrar | oh |
| 02:13.15 | carrar | Glenn Beck said google is evil |
| 02:13.18 | carrar | heh |
| 02:13.27 | russellb | he's also a loon |
| 02:13.32 | carrar | hahaha |
| 02:13.58 | hardwire | what is a glenn beck? |
| 02:14.00 | hardwire | googls. |
| 02:14.29 | carrar | russellb, that web site is slow |
| 02:14.41 | russellb | yes, it is quite slow indeed... |
| 02:14.53 | russellb | I do not disagree, heh. |
| 02:14.59 | carrar | I could offer a dedicated host for it! |
| 02:15.07 | russellb | it's not ours |
| 02:15.28 | hardwire | oreilly. |
| 02:15.37 | carrar | doesn't oreilly have a apache how too book? :) |
| 02:16.11 | carrar | finds himself getting blocked from oreilly |
| 02:16.14 | carrar | heh |
| 02:16.27 | russellb | i'm wondering if it's just inefficient code |
| 02:16.38 | russellb | like they're parsing our docbook XML and generating the pages on every request |
| 02:16.46 | carrar | ouch |
| 02:16.55 | russellb | i have no idea, just guessing based on the speed :-p |
| 02:17.05 | carrar | memcached! |
| 02:17.17 | russellb | but it has really been helpful. We turned off commenting for now, but it was very helpful for collecting feedback |
| 02:18.37 | carrar | it's a great idea to find mistakes before printing |
| 02:18.53 | carrar | just too slow for me :( |
| 02:19.19 | *** join/#asterisk JonnyD_work (~Jon@cpe-071-075-036-057.carolina.res.rr.com) |
| 02:31.27 | carrar | russellb, How do I buy the pdf version? |
| 02:31.49 | carrar | it's not listed under ebooks |
| 02:31.51 | carrar | hrmm |
| 02:31.57 | russellb | it's not out yet |
| 02:32.10 | russellb | march or april |
| 02:32.49 | carrar | :( *sniff* |
| 02:33.32 | russellb | heh, I appreciate that :-) |
| 02:34.34 | russellb | O'Reilly is busy with it working their magic |
| 02:34.43 | carrar | I'd proof read a pdf version if I had it :) |
| 02:35.00 | russellb | appreciate it - copy editing was just completed, our copy editor was amazing |
| 02:39.32 | jaytee | I want my Kindle version and I want it NOW!!! :-) |
| 02:39.50 | carrar | !!!!! |
| 02:40.16 | jaytee | "Are we there yet? Are we there yet?" < somebody just slap jaytee :-) |
| 02:41.51 | jaytee | I don't own the entire "zoo" of O'Reilly books but I have several of the various critters and I want to get Kindle versions of all of them. |
| 02:43.14 | *** part/#asterisk Bridge| (bridgelife@netburst.org) |
| 02:46.04 | jaytee | russell, the section on installing fail2ban was much appreciated |
| 02:52.34 | russellb | jaytee: yay |
| 03:01.32 | *** join/#asterisk diemos (~root@173-13-138-49-sfba.hfc.comcastbusiness.net) |
| 03:01.50 | diemos | is there anyone well-versed in polycom config files that could give me a lesson or two? |
| 03:03.34 | neurosys | russellb: where is this fail2ban section? |
| 03:03.43 | russellb | Security chapter |
| 03:07.42 | *** join/#asterisk tris (~tristan@173-164-188-122-SFBA.hfc.comcastbusiness.net) |
| 03:10.27 | jaytee | diemos, what are you looking to do? |
| 03:16.04 | *** join/#asterisk joako (~joako@opensuse/member/joak0) |
| 03:17.24 | joako | I setup a SIP trunk between two asterisk servers, and they register w/o any errors but when I try to send calls I get a message " NOTICE[24179]: chan_sip.c:15456 handle_request_invite: Failed to authenticate user "13055129135"" where the "User" is the callerID number, how can I eliminate this error? |
| 03:19.49 | *** join/#asterisk bbryant (~brett@c-76-26-221-79.hsd1.sc.comcast.net) |
| 03:26.00 | diemos | jaytee: I'm trying to understand how the provisioning file structure works |
| 03:26.19 | joako | On the other asterisk the error is check_auth: username mismatch, have <THE_USERNAME>, digest has <s> |
| 03:26.49 | diemos | I see the main 0000000000.cfg links to sip.cfg and phone1.cfg |
| 03:27.51 | jaytee | diemos, try giving this whitepaper a quick read: http://www.polycom.com/global/documents/whitepapers/configuration_file_management_on_soundpoint_ip_phones.pdf |
| 03:29.34 | *** join/#asterisk shamelessn00b (~chatzilla@175.110.242.37) |
| 03:30.20 | shamelessn00b | Hi, my SIP clients are behind NAT, I just wanted to test my astersik configuration, can anyone register with my asterisk box and dial 1111, I just want to make sure that Asterisk itself is set up properly |
| 03:30.31 | jaytee | diemos, the 000000000000.cfg file is a template file. You create a copy of it where the name matches the MAC address of the phone. |
| 03:31.39 | joako | Does anyone know where I can find a sample working config for asterisk to register to another asterisk via SIP> |
| 03:32.21 | joako | Currently it's not working for me. I register with no errors but when I send a call for some reason it is interpreting the callerid as the username. |
| 03:32.36 | jaytee | joako, there's a section in the old book on that |
| 03:32.41 | jaytee | ~book |
| 03:32.41 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook, or http://ofps.oreilly.com/ |
| 03:37.04 | joako | Yes I don't understnad that at all. To register asterisk to NONasterisk register => username:password@host is OK and call will come in fine. Why to register asterisk to asterisk I need register => username:password@host/username? |
| 03:38.42 | *** join/#asterisk NightMonkey (debian-tor@pdpc/supporter/professional/nightmonkey) |
| 03:39.18 | joako | ... the problem is I use users.conf on the remote asterisk to register, not sip.conf |
| 03:45.51 | joako | I posted my sip config and errors: http://pastebin.com/UKahQVUz |
| 03:56.41 | *** join/#asterisk Bidik (~bidik@li267-109.members.linode.com) |
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| 04:46.54 | dtoronto | asterisk 1.8.3, distributed setup with 1 queue server and 2 extension servers, using tigase PubSub to distribute presence. I would appreciate any dialplan knowledge on getting calls from the queue server to extensions on the remote extension servers |
| 04:48.05 | dtoronto | I basically want calls to come in on the queues and using then forward to respective agents logged into those queues dynamically |
| 04:48.31 | *** join/#asterisk NightMonkey (debian-tor@pdpc/supporter/professional/nightmonkey) |
| 04:54.17 | joako | how do I configure IAX so callerid is sent? |
| 04:57.01 | *** join/#asterisk shamelessn00b (~chatzilla@175.110.242.37) |
| 04:57.55 | shamelessn00b | I needed some guidance on how to configure a soft phone which is behind a firewall (NAT) with asterisk |
| 04:58.21 | shamelessn00b | I have the client registered but I'm not getting any audio |
| 05:00.34 | dtoronto | @shamelessn00b do you have the rtp ports already open? |
| 05:05.46 | shamelessn00b | yes 10000-20000 |
| 05:06.48 | joako | what are you using as a router? |
| 05:07.14 | shamelessn00b | Motorola CPE |
| 05:07.22 | shamelessn00b | I already tried port forwarding |
| 05:08.28 | *** join/#asterisk tash (~tash@ip68-103-230-178.ks.ok.cox.net) |
| 05:08.33 | tash | good evening |
| 05:08.58 | tash | can I pass a variable *from* Asterisk *to* a PHP file? |
| 05:17.23 | shamelessn00b | tash: AGI |
| 05:26.04 | *** join/#asterisk draeath (~draeath@unaffiliated/draeath) |
| 05:27.36 | tash | shamelessn00b: was afraid you were going to say that |
| 05:27.58 | tash | for some reason, AGI scares me :P I don't have any experience with it and am not sure I have a lot of time to learn it. But, if I must, I must |
| 05:27.59 | shamelessn00b | :/ |
| 05:28.11 | tash | is it rather simple? |
| 05:28.17 | shamelessn00b | you should try Asterisk Java |
| 05:28.24 | shamelessn00b | are you good with Java? |
| 05:28.56 | tash | I don't know java at all |
| 05:29.06 | shamelessn00b | Perl? |
| 05:29.10 | tash | I'm more of a networking guy, learning Asterisk and have some php experience |
| 05:29.16 | tash | not much perl, but could learn I s'pose |
| 05:29.32 | tash | yeah, you're dealing with a newb...my toolbox isn't quite that large yet |
| 05:29.52 | shamelessn00b | its really simple, check out some samples |
| 05:29.59 | shamelessn00b | did you build your asterisk from source |
| 05:30.01 | tash | here's what I want to accomplish, maybe there is a better way ... this shouldn't take but a sec to explain |
| 05:31.37 | tash | my asterisk server dials a user, I expect the user to press 1 to confirm they heard the message...if they press 1, I want to be able to update a row in my db ... not sure how to do that. and for some reason I was thinking I could do System(wget http://mysite.com/confirm.php?blah=blah) ... and confirm.php would receive the $_GET vars I pass in the URL and use that to update the DB |
| 05:31.43 | tash | seems like a lot of work though |
| 05:32.01 | tash | that's why I was wondering if there was an easier way, like just passing a var to PHP via my dial plan |
| 05:32.51 | tash | Yes, i built my asterisk from source |
| 05:35.53 | shamelessn00b | you'd find some sample agis in /var/lib/asterisk/agi-bin folder |
| 05:37.43 | shamelessn00b | I'd still recommend you learn JAVA and start using Asterisk Java |
| 05:37.46 | shamelessn00b | its much simpler |
| 05:38.01 | shamelessn00b | once you know how to set up java and how jars work |
| 05:38.06 | shamelessn00b | thats the tricky bit |
| 05:38.23 | shamelessn00b | writing AGIs is really very easy |
| 05:38.51 | tash | maybe I should hire you to do something for me ;) |
| 05:56.53 | shamelessn00b | ~book |
| 05:56.53 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook, or http://ofps.oreilly.com/ |
| 06:18.07 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 06:18.07 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.8.2.3 (2010/01/26), 1.6.2.16.1 (2010/01/18), 1.4.39.1 (2010/01/18), *-Addons 1.6.2.3, 1.4.13 (2010/01/14), dahdi-linux 2.4.0 (2010/09/01), dahdi-tools 2.4.0 (2010/09/01), libpri 1.4.11.5 (2010/11/22) -=- Visit the new official Asterisk wiki: wiki.asterisk.org -=- http://xkcd.com/806/ |
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| 07:55.39 | sawgood | anyone home? |
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| 08:04.27 | JerJer | nobody here but us chickens |
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| 08:22.35 | ChannelZ | clucks |
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| 08:24.39 | x1nux | hi |
| 08:25.07 | *** part/#asterisk x1nux (~x1nux@unaffiliated/x1nux) |
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| 10:37.06 | benngard | is there a way to find out when a queue does a timeout, who the call was queued to? |
| 10:40.02 | benngard | i see SIP/0317998975 with penalty 1 (dynamic) (Ringing) has taken no calls yet when i watch the queue but would like to get SIP/0317998975 in someway in the dialplan |
| 10:42.52 | benngard | NoOp(${DIALEDPEERNUMBER}) <- gives me who call, but i am looking for the extension that queue sent the call to |
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| 14:44.51 | Kobaz | heh |
| 14:45.04 | Kobaz | in the online draft for the next asterisk book: Whiskey Tango Foxtrot |
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| 15:00.53 | russellb | Kobaz: :-D |
| 15:02.29 | *** join/#asterisk kuku (~kuku@c-24-13-139-34.hsd1.il.comcast.net) |
| 15:03.29 | kuku | Is there a way to grab recording and the cdr database from 3 asterisk servers, and consolidate it into one database, with a large file system for lookup (to take load off the systems that are dialing ) ? |
| 15:04.29 | fenrus | how about writing the CDR from the machines into a central database-server ? |
| 15:04.41 | fenrus | using the same server for all the machines |
| 15:06.10 | Kobaz | fenrus: and your central db server goes down and now what |
| 15:06.20 | fenrus | Kobaz, it does not.. |
| 15:06.27 | Kobaz | fenrus: you should do both, do local logging plus a central one |
| 15:06.33 | Kobaz | fenrus: it will at some point |
| 15:06.38 | fenrus | yea, that would be the best way of doing it ofc |
| 15:06.45 | fenrus | using some mysql cluster you'd be "safe" |
| 15:06.49 | Kobaz | as time approaches infinity... the liklyness of failure reaches 100% |
| 15:07.01 | fenrus | yep |
| 15:08.16 | fenrus | backing up the database is of course something you need to do - but that i figured was common sense..? |
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| 15:29.51 | nestAr | anyone got any hints on what to check on my IP650... it won't seize a ringing call on off-hook, and it's not ringing at all. |
| 15:30.04 | nestAr | despite a tone being selected and the volume being turned up. |
| 15:31.25 | killown | I created an ivr extension with a playback of a mp3, but the audio is very low, http://bpaste.net/show/13840/ , How can I adjust the volume of my ivr sound? or does it be an issue with sound quality/codec |
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| 15:36.31 | nestAr | killown: i think you can run it back through lame and adjust the gain |
| 15:36.37 | nestAr | or a similar program |
| 15:36.55 | killown | nestAr ok |
| 15:37.08 | nestAr | i'm sure it's just the source material |
| 15:40.33 | killown | nestAr, thank you |
| 15:57.11 | *** join/#asterisk edwin_quijada (~edwin_qui@186.120.67.121) |
| 15:57.15 | edwin_quijada | Hi! |
| 15:57.54 | edwin_quijada | there is way to know if I am registetered with my provider correctly? |
| 15:58.20 | edwin_quijada | I have an account with my ISP and I know if I am passing the correct parameters to coenction |
| 16:00.04 | nestAr | sip show peers |
| 16:00.46 | *** join/#asterisk JonnyD_work (~Jon@173.226.80.154) |
| 16:03.42 | edwin_quijada | nestAr: ISP/8658775555 19.19.19.19 5060 Unmonitored |
| 16:04.00 | edwin_quijada | it means that I have registered with my sip? |
| 16:04.33 | nestAr | does it show them online at the bottom? |
| 16:05.06 | nestAr | Unmonitored: 3 online, 0 offline |
| 16:05.06 | edwin_quijada | 4 sip peers [Monitored: 0 online, 0 offline Unmonitored: 2 online, 2 offline] |
| 16:05.13 | nestAr | looks like it. |
| 16:08.11 | edwin_quijada | when I try to do a call using this ISP I get Everyone is busy/congested at this time (1:0/1/0) |
| 16:08.47 | edwin_quijada | this is my dialplan to dial exten => _9.,1,Dial(SIP/${EXTEN:1}@ISP,30,r) |
| 16:11.18 | nestAr | should work.. i'm using Dial(SIP/${EXTEN}@vitel-outbound) |
| 16:14.46 | edwin_quijada | it seems the same but I get the error above this is my extension and SIP.conf http://pastebin.com/7TK0NZDF |
| 16:19.48 | nestAr | edwin_quijada: are you forcing a register? |
| 16:20.38 | nestAr | register => user:pass@some.host.com:5060 |
| 16:20.42 | nestAr | in sip.conf |
| 16:23.59 | edwin_quijada | no |
| 16:24.16 | nestAr | add that and sip reload |
| 16:24.18 | nestAr | try again |
| 16:24.20 | edwin_quijada | ok |
| 16:31.54 | edwin_quijada | I get this Failed to authenticate on REGISTER |
| 16:33.50 | nestAr | yeah, so the server doesn't like something about you.. |
| 16:37.43 | edwin_quijada | Look, I have an account with my ISP and he gives me a xlite softphone to use my IP line I told him that I need use it with my sterisk and he said me "NOT" so I checked that I send to ISP with Wcap |
| 16:38.08 | edwin_quijada | I am trying to use this line with asterisk |
| 16:43.39 | *** join/#asterisk edwin_quijada (~edwin_qui@186.120.177.61) |
| 16:43.45 | edwin_quijada | nestAr: sorry |
| 16:44.03 | edwin_quijada | <PROTECTED> |
| 16:44.15 | edwin_quijada | I am trying to use this line with asterisk |
| 16:44.15 | edwin_quijada | <edwin_quijada> I change the user agent and I put like xlite |
| 16:46.27 | *** join/#asterisk slidesinger (~slidesing@c-68-44-99-163.hsd1.nj.comcast.net) |
| 16:47.16 | nestAr | edwin_quijada: not sure.. |
| 16:47.26 | nestAr | hang out, maybe someone else can help |
| 16:47.30 | *** join/#asterisk ariel_ (~chatzilla@63.214.236.169) |
| 16:47.38 | nestAr | you can turn debuging on and see if anything pops out |
| 16:48.00 | edwin_quijada | sip debug on in cli? |
| 16:48.14 | nestAr | i believe so, unless the syntax has changed |
| 16:48.57 | nestAr | sip set debug |
| 17:00.11 | nestAr | i wish i could figure out why this phone won't ring |
| 17:09.29 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 17:24.10 | WIMPy | I usually use CFU to VM to keep it from ringing. |
| 17:46.26 | *** join/#asterisk mechbangirc (~mechbangi@mbl-65-157-187.dsl.net.pk) |
| 17:47.27 | mechbangirc | okay, does rtp follow same route as sip or is it direct connection b/w peers on wan? even if i have canreinvite=no? |
| 17:49.40 | jaytee | as I understand it if you have canreinvite=no then the rtp streams will pass through your * server. |
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| 17:51.55 | *** part/#asterisk gnrfan (~gnrfan@200.48.253.230) |
| 17:52.41 | WIMPy | yes |
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| 19:07.04 | *** join/#asterisk dzup2 (~alex@unaffiliated/dzup) |
| 19:10.05 | dzup2 | wtf? what is that link on topic have to do with asterisk? |
| 19:10.32 | *** join/#asterisk SimonMX (~simon@5e0ed2bd.bb.sky.com) |
| 19:11.24 | SimonMX | Hey folks, not entirely an Asterisk related question, but... I just bought some Cisco 3911 SIP phones and it turns out you need a special SIP firmware |
| 19:11.30 | SimonMX | Can anyone help me obtain it? |
| 19:12.09 | SimonMX | Not sure why it is advertised as a SIP phone, but cant do SIP out of the box :-/ |
| 19:13.10 | coppice | where did you buy them? |
| 19:13.32 | SimonMX | From www.ebuyer.com |
| 19:14.07 | SimonMX | In the small print, it says you need to buy a licence. However, that is actually extended support for Cisco, which gives you access to the downloads area on Cisco.com |
| 19:14.44 | SimonMX | So I can pay for extended support to get access to the firmware from Cisco (50% of the cost of the phone!) |
| 19:15.00 | SimonMX | Ideally was looking to get it working without forking out more cash |
| 19:15.18 | SimonMX | I rushed the purchase and didnt see the print :-( |
| 19:17.26 | nestAr | the link is funny.. |
| 19:17.36 | *** join/#asterisk SimonMX (~simon@5e0ed2bd.bb.sky.com) |
| 19:18.19 | SimonMX | Add to that the fact it doesn't ship with a power supply and it wasn't really a well thought out purchase! |
| 19:18.43 | coppice | most phones don't ship with a power supply these days. people use PoE |
| 19:19.14 | nestAr | i started regretting not buying at least 2 power supplies.. |
| 19:19.26 | nestAr | since i only have one ethernet port at my desk |
| 19:19.37 | nestAr | so testing has been a lot more work. |
| 19:19.43 | SimonMX | I guess that makes sense. I ordered a POE switch anyway, so that issue is easily resolved. |
| 19:20.17 | SimonMX | I ordered two phones at £50GBP each, seemed like a good deal. |
| 19:20.53 | SimonMX | So I am reluctant to spend £25GBP on a support subscription with Cisco |
| 19:21.20 | SimonMX | although I guess I only need to do that once |
| 19:26.48 | SimonMX | Interestingly, this site http://www.voip-info.org/wiki/view/Cisco+Phones+Table shows the phone as SIP only |
| 19:27.06 | SimonMX | presumably that must mean the firmware is already loaded |
| 19:27.17 | SimonMX | Guess I will find out on Monday when the POE switch arrives |
| 19:40.08 | Tozz_ | I would google for the firmware |
| 19:40.49 | Tozz_ | it's probably somewhere there. Anyone with a Cisco account risks getting his account blocked if he downloads firmware files for equipment that he does not have a license for. |
| 20:07.56 | *** join/#asterisk iprouteth0 (~iprouteth@unaffiliated/iprouteth0) |
| 20:16.42 | *** join/#asterisk killown (~killown@unaffiliated/killown) |
| 20:19.30 | iprouteth0 | hello all |
| 21:17.25 | *** join/#asterisk Buklov (~Buklov@mail.sapsun.su) |
| 21:35.35 | *** join/#asterisk hetii (~hetii@87.99.51.172) |
| 21:35.40 | hetii | Hello :> |
| 21:36.14 | hetii | i`m not sure if this is a proper # but i go this: |
| 21:36.28 | hetii | dahdi_hardware |
| 21:36.30 | hetii | driver should be 'qozap' but is actually 'hfcmulti' |
| 21:36.32 | hetii | pci:0000:01:08.0 qozap+ 1397:08b4 Junghanns QuadBRI ISDN card |
| 21:37.05 | hetii | lsmod also show me that i had isdn and hisax loaded. |
| 21:37.32 | hetii | so the question is with driver i should use and should i disable hisax and isdn on blacklist ? |
| 21:38.08 | *** join/#asterisk Dovid (~Dovid@213.8.121.90) |
| 21:38.18 | Dovid | is there any way to have asterisk just accept any registration that it gets ? |
| 21:44.10 | BlackBishop | any way I can make asterisk Verbose(some text) when a sip user ends a call (through a datacard) ? |
| 21:44.29 | BlackBishop | if the other party hangs up .. the last Verbose shows up too .. |
| 21:44.42 | BlackBishop | but if the sip party hangs up .. nothing happens :| |
| 21:44.52 | BlackBishop | as in.. it doesn't continue to the other steps |
| 21:45.20 | *** join/#asterisk fauxalliance (~fauxallia@142.162.197.28) |
| 21:49.55 | BlackBishop | any ideas anybody ? |
| 21:54.09 | *** join/#asterisk javier_cintron (~javier_ci@200-56-203-54-cable.cybercable.net.mx) |
| 21:58.59 | hetii | asterisk -rvvvvvvvvvv |
| 21:59.24 | *** join/#asterisk manji (~manjiki@adsl-142-168.adsl.ntua.gr) |
| 21:59.59 | Dovid | BlackBishop: Use the h extension |
| 22:00.10 | hetii | and sip debug help will show you additional help |
| 22:04.36 | *** part/#asterisk javier_cintron (~javier_ci@200-56-203-54-cable.cybercable.net.mx) |
| 22:04.48 | *** join/#asterisk javier_cintron (~javier_ci@200-56-203-54-cable.cybercable.net.mx) |
| 22:05.58 | BlackBishop | goodie |
| 22:08.50 | *** join/#asterisk javier_cintron (~javier_ci@200-56-203-54-cable.cybercable.net.mx) |
| 22:14.56 | BlackBishop | thankd dovid |
| 22:14.59 | BlackBishop | thanks* |
| 22:15.14 | BlackBishop | "exited non-zero on" |
| 22:15.19 | BlackBishop | what's this thingy ? |
| 22:15.27 | BlackBishop | how can I make it exit with a 0 ? :) |
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