| 00:00.25 | beek | evening jaytee |
| 00:00.35 | jaytee | evening beek |
| 00:00.38 | jaytee | how's things? |
| 00:01.00 | beek | Good. New PRIs are up and running. The only excitement was I couldn't make outgoing calls until I sent National 10-digit numbers. |
| 00:01.04 | beek | Easy fix in the dialplan. |
| 00:01.33 | fun330 | can either of you help me with my problem |
| 00:01.38 | beek | How are things your way? |
| 00:01.40 | jaytee | yeah, lot of providers are insisting on getting passed 10 digits |
| 00:02.20 | fun330 | i can't get my debug to show the trunk name, it will only show the ip address, i did a pastebin of my sip.conf and debug |
| 00:02.24 | jaytee | well, they're looking up because my boss just left for vacation. I'm migrating the last of the analog phones to ATA's and pulling the plug on the Nortel Option 11C on the 28th |
| 00:02.32 | fun330 | http://pastebin.com/d8179b93 |
| 00:02.39 | jaytee | he's been a major prick for the last 2 weeks |
| 00:03.07 | beek | That sounds like fun. Perhaps the boss's plane will get hijacked and he'll be made a hostage. |
| 00:03.32 | jaytee | he doesn't fly |
| 00:03.38 | jaytee | he's driving to Florida |
| 00:03.43 | beek | Damn. It was a good thought, anyway. |
| 00:03.56 | beek | What ATAs are you using? |
| 00:04.12 | jaytee | Linksys Sipuras |
| 00:04.18 | jaytee | 2102's and 8000's |
| 00:04.47 | beek | Ahh... I love those things. We're using 3102s at our tower sites now and running our repeater phone-patch traffic back over our RF WAN. |
| 00:05.04 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 00:05.27 | jaytee | it's weird because they've both got nat'd routers in them and the 2102's register and work fine without having nat=yes but the 8000 that has 8 ports has to have nat=yes and canreinvite=no or ports 3 through 8 won't work. |
| 00:05.45 | beek | jaytee: That's weird. |
| 00:06.30 | *** join/#asterisk Bonix (n=Bonix@200-195-41-212.isimples.com.br) |
| 00:06.35 | jaytee | not really, the first two ports use 5060 and 5061 to register, the rest increment the port number by 100 so port 3 would be 5160, 4 would be 5161 etc. |
| 00:07.21 | *** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler) |
| 00:07.30 | beek | What does that have to do with nat=yes? |
| 00:07.41 | beek | I'm missing the connection. |
| 00:08.15 | jaytee | because * sees all of them as the same IP |
| 00:08.50 | beek | ... but different ports. |
| 00:09.24 | jaytee | yeah, for sip registration but there's nothing specifically defined for the rtp streams so without the nat=yes you get no audio |
| 00:10.01 | beek | I understand that part. I just can't figure out why it will work without that on the first two ports and then fail thereafter. |
| 00:10.27 | beek | Wait a minute... it's been a long day. |
| 00:10.38 | beek | I see what you mean now... the 2102 doesn't require it but the 8000 does. |
| 00:10.39 | jaytee | dunno, but most devices that I've hooked to * with 2 connections using the same IP work fi the ports are 5060 and 5061 |
| 00:10.41 | beek | slaps head |
| 00:11.21 | jaytee | yet both are configured as a nat'd router which is the part I thought was weird, not that I needed that for the 8000. with all those ports that actually made sense. |
| 00:11.21 | beek | I'm looking at the 8000 now. Nice looking box. |
| 00:11.28 | jaytee | for the price it's nice |
| 00:12.27 | jaytee | plus I like the streaming media feature |
| 00:12.49 | Micc | the SAP8000 has support for SIP trunks in the latest firmware. |
| 00:12.58 | Micc | So you'll only use one registration for all 8 ports if you want. |
| 00:13.10 | jaytee | Micc, yeah I read that but does that feature work with *? |
| 00:13.19 | beek | My brother's company is getting ready to take the plunge. They have a 17 POTS lines and are going to move them to a PRI except for one or two for fax machines. |
| 00:13.23 | Micc | jaytee, yup, I have a customer using it right now. |
| 00:13.33 | jaytee | beacuse I don't think * supports SIP "trunks" |
| 00:13.49 | Micc | jaytee, just set the call-limit=16. |
| 00:14.04 | jaytee | two calls per port? |
| 00:14.11 | Micc | jaytee, yeah. |
| 00:14.28 | Micc | jaytee, it has built in call waiting and conference. |
| 00:14.33 | jaytee | Micc, this works with 1.4? |
| 00:14.40 | Micc | jaytee, yes. |
| 00:15.42 | Micc | It sames me a lot of time and special macros to do the rollovers. |
| 00:17.11 | Micc | saves |
| 00:18.19 | jaytee | yeah, even the 2102's support 3 way conference and call waiting |
| 00:18.39 | jaytee | and the streaming media feature is nice if you want to use one port as a streaming source for music on hold |
| 00:21.24 | stabler | im having issues with my inbound calls.. outbound are fine... when i make an inbound call i just get the providers vm service |
| 00:21.27 | stabler | http://pastebin.com/m32462045 |
| 00:21.34 | stabler | that is my sip debug |
| 00:22.25 | *** join/#asterisk yondaime (n=Yamato@unaffiliated/yondaime) |
| 00:24.24 | jaytee | 401 Unauthorized |
| 00:24.40 | stabler | this my sip and extensions conf |
| 00:24.42 | stabler | http://pastebin.com/m29ccf2b6 |
| 00:24.46 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.static.bezeqint.net) |
| 00:25.10 | *** join/#asterisk killown (n=Yamato@unaffiliated/killown) |
| 00:26.27 | stabler | any help would be appreciated |
| 00:29.42 | *** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler) |
| 00:31.24 | jaytee | stabler, check your sip.conf. your * server is blocking the call as 401 unauthorized |
| 00:31.52 | stabler | this is my sip and exten conf |
| 00:31.53 | stabler | http://pastebin.com/m29ccf2b6 |
| 00:31.56 | stabler | do you see anything? |
| 00:32.02 | stabler | im a complete noob |
| 00:32.14 | stabler | trying to soak up as much knowledge as possible |
| 00:32.40 | jamincollins | anyone know how (or if it's even possible) to get 1.4 to track the state of dynamic agents with Local/ channels? |
| 00:34.08 | jaytee | stabler, are you behind a router using NAT? |
| 00:34.15 | stabler | no |
| 00:34.25 | stabler | server has a public address |
| 00:34.41 | stabler | well public and private both.. it has two nics |
| 00:35.08 | stabler | but is connecting to the sip service via the public address |
| 00:35.40 | jaytee | connecting to MagicJack, right? |
| 00:35.55 | stabler | yea |
| 00:37.16 | jaytee | well, in that account you have insecure=very but at the top of the file you have tcpenable=no. tcpenable indicates you're running Asterisk 1.6 and insecure=very was deprecated and hasn't been used since 1.2 |
| 00:37.38 | eppigy | TRABAJO |
| 00:37.43 | stabler | so what should i use? |
| 00:37.50 | *** join/#asterisk shmaltz (n=chatzill@mail.dmaven.com) |
| 00:37.58 | shmaltz | hi everyone |
| 00:38.24 | eppigy | hello shmaltz |
| 00:38.31 | voxter | anyone use redfone tdmoe products, or anybody's tdmoe products? |
| 00:38.35 | stabler | jaytee: i am indeed running 1.6 |
| 00:38.36 | shmaltz | anything new going on here? |
| 00:38.47 | jaytee | NO TRABAJO, es tiempo por la siesta! |
| 00:38.51 | shmaltz | long time not been here |
| 00:39.39 | stabler | jaytee: what should i use in place of the insecure=very? |
| 00:40.08 | shmaltz | stabler, what are you trying to do? |
| 00:40.09 | jaytee | insecure=port,invite or insecure=invite,port. probably the latter |
| 00:40.55 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 00:41.04 | jaytee | shmaltz, he's getting a 401 Unauthorized from his "ITSP", MagicJack :-) |
| 00:41.35 | shmaltz | jaytee, getting it back? or on inbound calls? |
| 00:41.49 | jaytee | on inbound calls from what I understand |
| 00:41.57 | jaytee | ask him |
| 00:42.04 | shmaltz | stabler, you have to create 2 settings in sip.conf |
| 00:42.17 | jaytee | <stabler> im having issues with my inbound calls.. outbound are fine... when i make an inbound call i just get the providers vm service |
| 00:42.17 | jaytee | < |
| 00:42.33 | *** join/#asterisk el_critter (n=critter@190.39.203.141) |
| 00:42.49 | *** join/#asterisk trnzmeta (n=bleh@secure27.lnk.telstra.net) |
| 00:42.58 | jaytee | <stabler> this is my sip and exten conf |
| 00:42.58 | jaytee | <stabler> http://pastebin.com/m29ccf2b6 |
| 00:42.58 | jaytee | <stabler> do you see anything? |
| 00:42.58 | jaytee | <stabler> im a complete noob |
| 00:43.15 | eppigy | noobles |
| 00:43.23 | shmaltz | one a peer for inobund make sure host has some value that matches the registration in there without a username/pass, make also sure that the context is restricted |
| 00:43.25 | shmaltz | and the other a user for outbound put the username/pass in there |
| 00:43.25 | jaytee | shmaltz, that's his sip.conf and extensions.conf pastebin |
| 00:44.09 | shmaltz | stabler, you there? |
| 00:44.12 | stabler | yea |
| 00:44.23 | stabler | im alittle lost |
| 00:44.25 | shmaltz | what is the problem? inbound or outbound? |
| 00:44.29 | stabler | inbound |
| 00:44.35 | stabler | outbound are fine |
| 00:44.46 | stabler | when i call the # i just get the providers vm service |
| 00:44.59 | shmaltz | stabler, you first have to register |
| 00:45.17 | shmaltz | are you seeing the 401 in the debug messages? |
| 00:45.24 | stabler | here is my sip debug |
| 00:45.26 | stabler | http://pastebin.com/m32462045 |
| 00:45.48 | stabler | this is my sip/ext conf |
| 00:45.49 | stabler | http://pastebin.com/m29ccf2b6 |
| 00:47.35 | shmaltz | stabler, which IP is the one that asterisk is sending back 401 to? |
| 00:47.41 | shmaltz | in the debug output |
| 00:48.08 | shmaltz | this IP doesn't match any host in sip.conf: |
| 00:48.09 | shmaltz | 66.61.139.61 |
| 00:48.16 | shmaltz | how do you think asterisk will allow it? |
| 00:48.43 | stabler | thats the ip of one of my users |
| 00:49.01 | shmaltz | then which output from that debug window is the one you are having a problem with? |
| 00:49.09 | shmaltz | Which line here: |
| 00:49.10 | shmaltz | http://pastebin.com/m32462045 |
| 00:49.34 | stabler | it doesnt really seem like the inbound call triggers anything |
| 00:49.39 | shmaltz | exactly |
| 00:49.45 | shmaltz | it's not hitting your box, |
| 00:49.50 | shmaltz | stabler, you have to first register |
| 00:49.57 | shmaltz | add a register line in your sip.conf |
| 00:49.59 | stabler | what do you mean |
| 00:50.03 | stabler | register first |
| 00:50.31 | shmaltz | I beleive this will do: |
| 00:50.33 | shmaltz | register => xxx:xxx@66.104.96.198 |
| 00:50.52 | stabler | i have this register => xxx:xxx@proxy1.columbus.talk4free.com:5070 |
| 00:50.57 | shmaltz | the first part before the : should be the username and the second should be the pass |
| 00:51.33 | shmaltz | stabler, what does sip show registry give you? |
| 00:51.55 | shmaltz | that looks ok, since that resolves to the same IP |
| 00:52.28 | shmaltz | sorry I don't know how to check that in 1.4 or 1.6 but figure it out |
| 00:52.33 | *** join/#asterisk docelmo (n=vircuser@pool-151-199-175-67.lyn.east.verizon.net) |
| 00:52.37 | stabler | http://pastebin.com/m6d00df97 |
| 00:52.40 | *** join/#asterisk erth64net (n=erth64ne@69-30-67-191.dq1sn.easystreet.com) |
| 00:52.46 | stabler | that is sip show registry |
| 00:52.48 | stabler | http://pastebin.com/m6d00df97 |
| 00:53.20 | shmaltz | stabler, but from we can see in the debug it's not hitting you |
| 00:53.23 | shmaltz | do the following: |
| 00:53.27 | stabler | i have a feeling im missing some dumb in my conf's |
| 00:53.28 | shmaltz | sip no debug |
| 00:53.35 | shmaltz | then do |
| 00:53.43 | shmaltz | sip debug peer magicjack |
| 00:53.50 | shmaltz | then issue a sip reload |
| 00:53.57 | shmaltz | pb the output |
| 00:54.13 | shmaltz | then make an inbound phone call in pb the output again |
| 00:54.50 | *** join/#asterisk Kumbang (n=unknown@125.163.83.153) |
| 00:55.58 | jaytee | suspects shmaltz's sip_fu is strong |
| 00:56.20 | shmaltz | wanders what sip_fu stands for |
| 00:56.41 | jaytee | shmaltz, it's like kung_fu for SIP debugging |
| 00:56.47 | *** join/#asterisk bkw_ (n=brian@freeswitch/developer/bkw) |
| 00:56.54 | shmaltz | :) |
| 00:57.00 | shmaltz | jaytee, love doing that |
| 00:57.11 | shmaltz | I recently signed up with a hosted PBX provider |
| 00:57.14 | shmaltz | quite a big one |
| 00:57.18 | shmaltz | they use meta switch |
| 00:57.34 | shmaltz | I got my Polycom phone plugged it in, and it's so wonderfull it works |
| 00:57.51 | shmaltz | but for $19.99 I'm not just going to let that Polycom 501 rot on my desk |
| 00:58.13 | shmaltz | so I used some little nifty script to reveal to me the sip password in the 501s web interface |
| 00:58.40 | shmaltz | plugeed those values into my sip.conf |
| 00:58.42 | shmaltz | and viola I got 6 new inbound channels on my asterisk box for $19.99 |
| 00:59.17 | shmaltz | here is the script, just creat a bookmark in firefox/ie that when pressed will give you a popup revealing the char behind the ****** |
| 00:59.27 | shmaltz | javascript:(function(){var s,F,j,f,i; s = %22%22; F = document.forms; for(j=0; j<F.length; ++j) { f = F[j]; for (i=0; i<f.length; ++i) { if (f[i].type.toLowerCase() == %22password%22) s += f[i].value + %22\n%22; } } if (s) alert(%22Password is:\n\n%22 + s); else alert(%22No passwords%22);})(); |
| 00:59.54 | shmaltz | next thing was to test their security |
| 01:00.05 | shmaltz | using the menu on the phone, I logged into the admin menu |
| 01:00.12 | shmaltz | checked how they provision the phone |
| 01:00.14 | shmaltz | FTP |
| 01:00.16 | shmaltz | nice |
| 01:00.50 | shmaltz | firing up my packet sniffer I just captured the username/pass used with FTP |
| 01:01.09 | shmaltz | and there it was thousands and thousands of sip usernames and passwords that can actualy make phone calls |
| 01:01.40 | shmaltz | all the Polycom config files were in one flat folder using the same ftp user/pass for all the files |
| 01:03.22 | stabler | http://pastebin.com/m619cfa1d |
| 01:03.46 | stabler | thats my sip debugs |
| 01:04.03 | stabler | just doesnt seem like the call is making it |
| 01:06.24 | shmaltz | stabler, can you check your firewall logs? |
| 01:07.22 | stabler | i keep seeing these |
| 01:07.22 | stabler | Outbound Registration: Expiry for proxy1.columbus.talk4free.com is 120 sec (Scheduling reregistration in 105 s) |
| 01:07.36 | stabler | but never one for inbound |
| 01:07.52 | stabler | is that normal |
| 01:08.26 | shmaltz | stabler, no it's not |
| 01:08.35 | shmaltz | stabler, firewall logs |
| 01:08.54 | *** join/#asterisk xuser (n=xuser@unaffiliated/xuser) |
| 01:08.54 | stabler | i currently have everything open for testing |
| 01:09.51 | shmaltz | stabler, have you followed this: |
| 01:09.53 | shmaltz | http://www.gruups.com/magicjack/ |
| 01:10.22 | stabler | yes |
| 01:10.38 | stabler | thast why i had the insecure=very in there |
| 01:10.47 | stabler | which is now changed |
| 01:11.07 | shmaltz | it seems taht you need to change the fromuser to match your serial number and also in the registration you have to do that |
| 01:11.11 | shmaltz | have you done that? |
| 01:11.33 | shmaltz | ok I gtg |
| 01:11.36 | shmaltz | cya around |
| 01:11.40 | shmaltz | bye |
| 01:11.46 | shmaltz | is logging out |
| 01:12.04 | shmaltz | logged out :) |
| 01:15.20 | *** join/#asterisk dlewis (i=457e665b@about/security/staff/dlewis) |
| 01:15.27 | stabler | sigh.. my help is gone |
| 01:20.28 | *** join/#asterisk andresmujica1 (n=andresmu@ubuntu/member/andresmujica) |
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| 01:26.11 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
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| 01:49.43 | *** join/#asterisk lilkid (n=chatzill@87-194-38-230.bethere.co.uk) |
| 01:50.17 | lilkid | why wouldn't i be able to use the CLI command file convert? |
| 01:50.21 | lilkid | just says no such command |
| 01:51.45 | jaytee | what syntax are you using? it's simply file convert tt-weasels.gsm tt-weasels.ulaw |
| 01:53.08 | *** join/#asterisk advorak (n=advorak@c-76-102-231-132.hsd1.ca.comcast.net) |
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| 01:57.52 | stabler | jaytee, do you have any other ideas on my issues? |
| 01:58.13 | jaytee | stabler, nope |
| 01:58.23 | stabler | darn |
| 01:58.35 | stabler | i have a feeling its someting dumb in my conf files |
| 01:58.43 | jaytee | sorry, but I don't deal with sip providers much. I use PRI to the PSTN |
| 02:00.31 | *** join/#asterisk pheezer (n=user@d.42inc.com) |
| 02:02.21 | *** join/#asterisk [intra]lanman (n=intralan@freeswitch/developer/intralanman) |
| 02:09.28 | Pan3D | stabler: so, when you launch *, you see the outbound registration go through, correct? |
| 02:09.39 | stabler | yes |
| 02:09.39 | Pan3D | but not the inbound? |
| 02:09.44 | stabler | correct |
| 02:09.46 | Pan3D | k |
| 02:10.10 | stabler | do you need me to repost my conf and sip debug? |
| 02:10.15 | Pan3D | Does your SIP debug show the outgoing SIP to the provider? |
| 02:10.27 | Pan3D | does it try to authorize? |
| 02:10.40 | stabler | give me a second ill make an outbound call |
| 02:10.48 | stabler | and record the output |
| 02:11.30 | Pan3D | well, it should when you start the * service. I'd restart via CLI and capture that in the SIP debug. |
| 02:11.54 | Pan3D | in other words, start by loading things fresh |
| 02:12.26 | Pan3D | You should see the out and in attemps in the SIP debug |
| 02:12.40 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-6e81e1a8960bb3de) |
| 02:13.07 | Pan3D | btw, I'm no * master, but you've been working hard all day to find this, so it's worth trying to help. |
| 02:14.06 | *** part/#asterisk pheezer (n=user@d.42inc.com) |
| 02:14.08 | stabler | http://pastebin.com/m501d724b |
| 02:14.13 | stabler | there is an outbound call |
| 02:14.22 | stabler | that is the sip debug from it |
| 02:15.10 | stabler | this is my sip/ext conf |
| 02:15.10 | stabler | http://pastebin.com/m29ccf2b6 |
| 02:15.16 | stabler | maybe you can find something in that |
| 02:15.41 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-6cc1347497bca6a9) |
| 02:15.47 | stabler | this is another sip debug/registration |
| 02:15.48 | stabler | http://pastebin.com/m619cfa1d |
| 02:16.09 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 02:17.08 | stabler | is get his |
| 02:17.10 | stabler | *this |
| 02:17.12 | stabler | [Feb 17 21:16:20] NOTICE[26010]: chan_sip.c:15333 handle_response_register: Outbound Registration: Expiry for proxy1.columbus.talk4free.com is 120 sec (Scheduling reregistration in 105 s) |
| 02:17.19 | stabler | but never one for inbound |
| 02:17.22 | drmessano^ | What exactly is the problem? |
| 02:17.29 | stabler | inbound calls |
| 02:17.50 | stabler | they instantly goto the providers vm |
| 02:18.20 | stabler | outbound works perfectly |
| 02:21.42 | drmessano^ | Are you showing it being registered? |
| 02:23.10 | stabler | yes |
| 02:23.28 | stabler | when i do a sip show registry |
| 02:23.31 | *** join/#asterisk TrentCreek (i=kvirc@ppp-70-254-112-199.dsl.hrlntx.swbell.net) |
| 02:23.34 | stabler | i have one entry |
| 02:23.42 | stabler | i also get these |
| 02:23.43 | stabler | [Feb 17 21:22:44] WARNING[26010]: chan_sip.c:2794 retrans_pkt: Maximum retries exceeded on transmission NDkzMzVlZWI1N2FjYTAxYmUzY2MxZjc4MWE5NjU4NTI. for seqno 122 (Non-critical Request) -- See doc/sip-retransmit.txt. |
| 02:23.52 | stabler | i dont know what that is |
| 02:24.50 | *** part/#asterisk dlewis (i=457e665b@about/security/staff/dlewis) |
| 02:25.52 | *** join/#asterisk Mw3 (n=mw3@ip59934bd1.rubicom.hu) |
| 02:27.30 | stabler | any ideas? |
| 02:32.18 | *** join/#asterisk LemensTS (n=customgt@adsl-75-42-150-213.dsl.stlsmo.sbcglobal.net) |
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| 02:39.50 | brunner | how has a fully-working call screening system, complete with call queues, a web interface, and a meetme conference for interfacing with the radio station mixer |
| 02:40.19 | brunner | so, thanks to the devs who made this possible |
| 02:40.53 | brunner | s/how/now |
| 02:42.11 | brunner | oh, and a mysql db for tracking the status of the calls for the web interface |
| 02:45.47 | *** join/#asterisk angryuser (n=gdobrovo@LPuteaux-151-42-35-99.w193-251.abo.wanadoo.fr) [NETSPLIT VICTIM] |
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| 02:45.53 | *** join/#asterisk ManxPower (n=manxpowe@69.sub-75-254-6.myvzw.com) |
| 02:46.15 | *** join/#asterisk drmessano (n=nonya@pdpc/supporter/active/drmessano) |
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| 03:05.21 | jayrod422 | can anybody shed on any tips for how they measure call quality on their * boxes.. im thinking of logging the jitter in the cdrs and having something croned that reads the cdrs and alerts me when things go to shit |
| 03:05.52 | carrar | If customers call and say it sounds like crap |
| 03:05.57 | carrar | then you need to fix it ;) |
| 03:06.24 | jayrod422 | thats what im doing now |
| 03:06.33 | carrar | not working? |
| 03:06.34 | jayrod422 | and it f*cking sucks |
| 03:06.36 | carrar | heh |
| 03:06.42 | carrar | then sounds like you have other issues |
| 03:06.46 | jayrod422 | i want some kinda nagios thing |
| 03:06.48 | jayrod422 | its carriers |
| 03:06.50 | jayrod422 | they suck |
| 03:06.52 | jayrod422 | but they are cheap |
| 03:07.02 | carrar | have nagios check their registration latency |
| 03:07.03 | jayrod422 | when shit goes belly up |
| 03:07.10 | jayrod422 | tried that |
| 03:07.30 | jayrod422 | what happens in carriers put a sbc out there thats sip in sip out |
| 03:07.30 | carrar | whats the real issue why their calls sound bad? |
| 03:07.35 | jayrod422 | and when their vendors go crap up |
| 03:07.43 | jayrod422 | its a chain reaction |
| 03:08.05 | jayrod422 | jitter and packet loss |
| 03:08.09 | carrar | So monitor the Status ms time |
| 03:08.13 | carrar | in sip show peers |
| 03:08.43 | carrar | How you gonna monitor jitter and packetloss |
| 03:08.51 | jayrod422 | well |
| 03:08.56 | carrar | seems like a lot of overhead |
| 03:09.05 | jayrod422 | im looking at when a caller hangs up |
| 03:09.10 | jayrod422 | in the exten => h |
| 03:09.16 | jayrod422 | logging the jitter to the db |
| 03:09.36 | carrar | and latency? |
| 03:09.47 | jayrod422 | pings i guess |
| 03:09.51 | jayrod422 | what else can you do |
| 03:09.57 | carrar | haven't tried |
| 03:10.09 | carrar | where are you pulling those variables out of? |
| 03:10.11 | jayrod422 | what i really want is to be able to detect when shits crazy |
| 03:10.16 | jayrod422 | and automatically reroute |
| 03:10.23 | jayrod422 | and route back automatically when things are better |
| 03:10.29 | jayrod422 | the jitter ones |
| 03:10.30 | thehar | smokeping |
| 03:10.49 | thehar | use it. love it. |
| 03:11.02 | jayrod422 | i have it |
| 03:11.12 | jayrod422 | i use nagios for the same functionality |
| 03:11.30 | carrar | looks like mtr kinda |
| 03:11.33 | carrar | without the graph |
| 03:11.42 | thehar | smokeping is god |
| 03:11.55 | jayrod422 | have you seen this |
| 03:12.14 | jayrod422 | http://pastebin.com/d3cd88029 |
| 03:12.16 | carrar | does it measure jitter? |
| 03:12.26 | jayrod422 | yep |
| 03:12.34 | carrar | smokeping does? |
| 03:12.49 | thehar | i have not seen that |
| 03:13.04 | thehar | i must |
| 03:13.05 | carrar | jayrod422, thats a nice chunk for a agi |
| 03:13.25 | carrar | I'll try that |
| 03:13.28 | jayrod422 | im thinking of taking that output and logging it to cdrs |
| 03:13.31 | carrar | could be interesting stats |
| 03:13.35 | jayrod422 | but im not sure it will work |
| 03:13.44 | carrar | I could write a agi for ya |
| 03:13.51 | carrar | dump it into a table |
| 03:13.58 | carrar | with other channel vars |
| 03:14.08 | carrar | so you can trace it down |
| 03:14.08 | thehar | this information is passed in CHANNEL? |
| 03:14.21 | jayrod422 | yes |
| 03:14.24 | thehar | mmmm |
| 03:14.31 | thehar | i want said agi |
| 03:14.47 | carrar | looks easy to write |
| 03:15.00 | jayrod422 | so your saying dump to agi |
| 03:15.07 | carrar | no |
| 03:15.12 | carrar | use AGI to dump it to a db table |
| 03:15.21 | jayrod422 | ah ok |
| 03:15.22 | carrar | with other variables from the channel |
| 03:15.28 | thehar | deadagi? |
| 03:15.29 | carrar | I do that for incoming calls |
| 03:15.37 | jayrod422 | i was thinking just put it into cdrs |
| 03:15.42 | carrar | track a bunch of stuff |
| 03:15.49 | carrar | naw |
| 03:15.50 | jayrod422 | that way i can see what calls were crazy |
| 03:15.55 | carrar | don't put anything in cdr |
| 03:16.05 | carrar | just create a new table |
| 03:16.18 | carrar | and put enough info in it so you can correlate it to the cdr table |
| 03:16.40 | jayrod422 | that can work |
| 03:16.44 | carrar | only thing in the cdr you would want is the duration |
| 03:16.56 | carrar | and even that might be in the channel when you call the agi |
| 03:19.16 | thehar | i've not written agi |
| 03:19.18 | thehar | hrm. |
| 03:19.21 | carrar | I have |
| 03:19.27 | carrar | I can share it once I write it |
| 03:19.31 | thehar | i would lourve |
| 03:19.40 | carrar | I use postgreSQL |
| 03:19.49 | carrar | and perl for the agi |
| 03:19.51 | thehar | that's fine i can change to mysql |
| 03:19.54 | carrar | yeah |
| 03:20.07 | thehar | perl is standard at my workplace |
| 03:20.20 | thehar | altho i have an asshat junior admin trying to push ruby |
| 03:20.50 | carrar | jayrod422, where did you find those rtpqos variables? |
| 03:21.00 | carrar | you have to hunt through the source? |
| 03:21.12 | carrar | or someone else did that once already |
| 03:21.29 | jayrod422 | um |
| 03:21.32 | jayrod422 | i forget |
| 03:21.34 | jayrod422 | google |
| 03:21.35 | carrar | heh |
| 03:21.41 | carrar | google knows all |
| 03:21.56 | thehar | ^_^ was just going to say heh |
| 03:23.02 | jayrod422 | http://lists.digium.com/pipermail/asterisk-dev/2008-November/035096.html |
| 03:25.17 | *** join/#asterisk Micc (n=Michael@c-76-121-255-52.hsd1.wa.comcast.net) |
| 03:25.28 | Micc | why are the members of one of my queues marked as (invalid) ?? |
| 03:26.01 | *** join/#asterisk moy (n=moy@CPE001cdfec4cee-CM00080dab8485.cpe.net.cable.rogers.com) |
| 03:28.54 | Micc | how can I prevent that from happening? |
| 03:31.31 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 03:33.16 | drmessano | Google knows NOTHING |
| 03:33.26 | thehar | google knows all |
| 03:33.36 | drmessano | Google knows NOTHING |
| 03:33.40 | thehar | google knows all |
| 03:33.50 | drmessano | But feel free to repeat the same thing, meaninglessly |
| 03:33.54 | thehar | yes |
| 03:34.05 | drmessano | Great retort |
| 03:34.09 | drmessano | You showed me |
| 03:34.18 | thehar | *gobble* |
| 03:34.28 | drmessano | Is that sexual? |
| 03:34.32 | thehar | no |
| 03:34.44 | drmessano | Felt that way |
| 03:34.55 | thehar | sounds like you have a problem that needs solving |
| 03:35.07 | drmessano | Now THAT sounds sexual |
| 03:35.15 | thehar | perhaps. perhaps not. |
| 03:35.19 | thehar | i'll go with not. |
| 03:35.27 | drmessano | What sort of problem? |
| 03:35.54 | thehar | i don't smell it and i'm near redwood and that north. =P |
| 03:35.58 | thehar | ugh MISTYPE |
| 03:36.19 | thehar | damned split screen always beats me |
| 03:36.31 | drmessano | .... |
| 03:36.50 | drmessano | Should I give you a few mins to work this out on your own? |
| 03:36.56 | thehar | it's been resolved |
| 03:37.10 | thehar | irssi thought it should be sticky again |
| 03:37.19 | drmessano | You dont seem that convinced |
| 03:37.32 | drmessano | Maybe you need to use X-Chat.. It runs great on Ubuntu |
| 03:37.45 | thehar | ew |
| 03:37.59 | drmessano | Kubunutu, Edukububuntu, Gobukunbuntu, and the other variants |
| 03:39.06 | drmessano | I prefer AsterKubuEdubuntu.. Because nothing makes someone learn the value of a good education like running Asterisk a KDE Ubuntu box along with everything else |
| 03:39.18 | thehar | Die. |
| 03:40.19 | drmessano | KooKooKaChoobunutu rocks too.. I AM TEH WALRUS... I AM THE PENGUIN.. KOO KOO KACHOO |
| 03:42.04 | carrar | jayrod422 |
| 03:42.13 | carrar | This seems to work better |
| 03:42.14 | carrar | exten => h,1,Noop(all = ${CHANNEL(rtpqos,audio,all)}); |
| 03:42.26 | carrar | NoOp("SIP/5060-b6c97430", "all = ssrc=787709222;themssrc=3998802898;lp=0;rxjitter=0.000061;rxcount=137;txjitter=0.000000;txcount=137;rlp=0;rtt=0.000000") |
| 03:42.29 | thehar | ooohh neato |
| 03:42.30 | drmessano | I wish I could send to AMI from the Windows Command line |
| 03:42.48 | carrar | yours want not giving real numbers |
| 03:42.52 | carrar | just intergers |
| 03:42.54 | carrar | (sp) |
| 03:43.17 | drmessano | Hmmm |
| 03:43.49 | Micc | how do I define an agent device? |
| 03:43.58 | Micc | It seems that agents.conf only defines agents. |
| 03:44.06 | Micc | As in a group or number. |
| 03:44.24 | drmessano | Like any other SIP device |
| 03:44.25 | Micc | But how do I make static members that are devices or like Local/701@salon-agents ? |
| 03:44.37 | Micc | drmessano, yeah but in which file? queue.conf? |
| 03:44.41 | drmessano | Like any other SIP device |
| 03:44.47 | Micc | member = Local/701@salon-agents? |
| 03:45.09 | Micc | Theres no example for anything like that. The only examples are for Agent/@1 or Agent/501 |
| 03:45.27 | Micc | So can I make an agent SIP/701? |
| 03:45.58 | Micc | how do I associate a device with an agent in a conf file? |
| 03:46.02 | drmessano | If you're still talking about device definitions, all devices are defined the same |
| 03:46.15 | drmessano | Per protocol |
| 03:47.01 | Micc | ah, I see an example. I do it in queue.conf |
| 03:50.00 | carrar | jayrod422, still here? |
| 03:50.04 | Micc | So if I'm not using login/login theres no need to define an agent. |
| 03:50.30 | Micc | So why did some of my agents turn into (Invalid) ??? |
| 03:54.54 | *** join/#asterisk ZaVoid (n=zavoid@nj-76-6-39-193.dhcp.embarqhsd.net) |
| 04:02.14 | Micc | why is this still a problem in 1.6 when its been known since 1.2? |
| 04:02.15 | Micc | http://www.freepbx.org/trac/ticket/3118 |
| 04:04.09 | kb3ien | finge features tend to do that. |
| 04:04.52 | kb3ien | somethings get fixed faster than others, i'm noticing ${CALLERID} remains a thorny issue with the 1.4.x current. |
| 04:08.06 | *** join/#asterisk freakazoid0223 (n=matt@pool-71-246-17-74.phlapa.fios.verizon.net) |
| 04:08.18 | drmessano | Micc: Looks like no one could reproduce it |
| 04:08.23 | drmessano | Did you actually READ the report? |
| 04:09.00 | Micc | yes. |
| 04:09.15 | Micc | I can obviously reproduce it. |
| 04:09.23 | Micc | and its with Local/ channels |
| 04:09.29 | Micc | it doesn't happen with SIP devices. |
| 04:09.50 | Micc | and staticmembers doesn't change anythign. |
| 04:10.09 | Micc | I mean persistent members. |
| 04:10.32 | drmessano | So they start up invalid, but then go valid when you reload? |
| 04:10.34 | Micc | I think if they tried with local channels they would have reprod it. |
| 04:10.40 | Micc | yup. |
| 04:10.49 | Micc | and I'm shutting down with quit now |
| 04:10.59 | Micc | then doing service asterisk start |
| 04:11.15 | Micc | I mean stop now |
| 04:12.54 | drmessano | Which version of Asterisk? |
| 04:13.20 | jaytee | Asterisk version 5 rocks!!! |
| 04:13.48 | lilkid | any idea what WARNING[17800]: file.c:1041 ast_waitstream: Unexpected control subclass '20' means? |
| 04:14.08 | lilkid | there's nothing wrong.. but it's bugging me |
| 04:14.11 | jaytee | I've been getting that error from my queue |
| 04:14.41 | drmessano | Lovely |
| 04:14.49 | drmessano | Why is shit always broke |
| 04:14.53 | drmessano | ALWAYS |
| 04:14.55 | Corydon76-dig | jaytee: version 0.5.0? |
| 04:15.21 | jaytee | Corydon76-dig, it was a joke |
| 04:16.14 | lilkid | drmessano: Because life sucks |
| 04:16.31 | *** join/#asterisk Rabenklaue (n=Rabe@g226203062.adsl.alicedsl.de) |
| 04:19.25 | Micc | drmessano, I'm using version 1.6.0.5 |
| 04:19.38 | Micc | Here is another issue I'm having. |
| 04:20.18 | Micc | Now that I'm using the sip device as the agent, when an atxfer happens and it transfers into a voicemailbox, it doesn't answer until sometime after the unavailabel message. |
| 04:20.50 | Micc | I'll hear the hold music while its dialing, then it seems like the transfer to the voicemail takes it a little while to figure out its answered. |
| 04:21.56 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 04:22.18 | Micc | maybe I need to put an answer in the macro there. |
| 04:24.02 | *** join/#asterisk Kisu (n=k@daniel1117.broker.freenet6.net) |
| 04:27.27 | Micc | ok, that works. |
| 04:27.57 | *** join/#asterisk Kyosh (i=48595dad@gateway/web/ajax/mibbit.com/x-d1af8b88b4a77522) |
| 04:28.21 | *** join/#asterisk stuffcorpse (n=rick@121.98.136.241) |
| 04:36.55 | lilkid | how can I make the default greeting for voicemail..uh..less american? |
| 04:37.30 | lilkid | e.g. # isn't the pound key.. |
| 04:37.38 | lilkid | in uk-ish |
| 04:38.42 | *** join/#asterisk thansen (n=thansen@c-67-171-115-155.hsd1.ut.comcast.net) |
| 04:38.52 | *** join/#asterisk jeffik (n=jeffik@69-196-165-181.dsl.teksavvy.com) |
| 04:39.57 | thehar | replace the spool files |
| 04:41.58 | lilkid | thought so, though how do I know which ones are used for the voicemail? |
| 04:42.55 | thehar | there is a text file in that directory that explains what is what |
| 04:42.57 | thehar | i forget where now |
| 04:44.34 | lilkid | thanks |
| 04:44.35 | *** join/#asterisk Nasra (n=Nasra@CPE001217b1920e-CM00111ade9528.cpe.net.cable.rogers.com) |
| 04:45.19 | *** join/#asterisk _gm (n=gmustafa@115.186.106.37) |
| 04:45.33 | thehar | <PROTECTED> |
| 04:45.40 | thehar | in src :) |
| 04:45.56 | [TK]D-Fender | lilkid: There are several UK replacement sets for the complete set |
| 04:48.31 | jaytee | brits call it the hash key I think |
| 04:48.44 | thehar | heh |
| 04:48.48 | jaytee | and they call drug stores "the chemist" |
| 04:49.03 | jaytee | and they eat blood pudding and jellied eels |
| 04:49.15 | lilkid | [TK]D-Fender: Where may I find them? also what directory would I be replacing? |
| 04:49.59 | lilkid | it's just hilarious non british people talking about british customs lol |
| 04:49.59 | [TK]D-Fender | lilkid: http://www.google.ca/search?hl=en&q=asterisk+UK+prompts&btnG=Google+Search&meta= |
| 04:50.14 | lilkid | thanks |
| 04:51.24 | drmessano | Jaytee ditched me |
| 04:51.45 | eppigy | :[ |
| 04:51.56 | jaytee | no, my firefox took a nose dive and augered in |
| 04:52.25 | jaytee | and I had to actually do a kill -9 on it |
| 04:52.58 | [TK]D-Fender | that's what she said... |
| 04:53.17 | drmessano | init 6 |
| 04:56.45 | *** join/#asterisk russellb (n=russell@asterisk/digium-open-source-team-lead/russellb) |
| 04:56.45 | *** mode/#asterisk [+o russellb] by ChanServ |
| 04:56.52 | carrar | oh crap |
| 04:56.54 | carrar | he's back |
| 04:56.57 | carrar | shhhh |
| 04:57.26 | carrar | forget everything I said |
| 04:57.50 | thehar | hi russellb |
| 04:58.49 | lilkid | the asterisk sounds directory at /usr/share/asterisk/sounds ? |
| 04:58.50 | russellb | waves |
| 04:59.09 | thehar | hehe |
| 04:59.42 | carrar | lilkid: /var/lib/asterisk/sounds/ |
| 05:01.35 | lilkid | i'm using debian, seems that everything is at /usr/share/asterisk/... instead of /var/lib/asterisk/... |
| 05:01.40 | lilkid | strange |
| 05:02.08 | carrar | did you compile from source? |
| 05:02.36 | lilkid | no, used apt |
| 05:02.47 | carrar | that might be way |
| 05:03.05 | carrar | you get what someone else perferred |
| 05:03.11 | carrar | compile from source is best |
| 05:05.05 | lilkid | fair enough, I will when I get the time and work out how to |
| 05:05.44 | carrar | takes about maybe 2 mins longer |
| 05:06.10 | carrar | unless it's running on a 286 |
| 05:07.33 | lilkid | 2 minutes if you know what you're doing ;) . i don't think i've ever compiled anything before.. |
| 05:08.33 | russellb | apt-get install build-essential ; wget http://downloads.digium.com/pub/telephony/asterisk/asterisk-1.4-latest.tar.gz ; tar -xvzf asterisk-1.4-latest.tar.gz ; cd asterisk-1.4.<something> ; ./configure ; make ; make install |
| 05:08.35 | russellb | there you go :) |
| 05:08.40 | carrar | 20 misn if you've never done it before |
| 05:08.49 | carrar | heh |
| 05:08.53 | carrar | ok now back down to 2 |
| 05:09.29 | thehar | russellb: you have a button for that don't you |
| 05:09.35 | russellb | i do not. |
| 05:09.38 | thehar | lies |
| 05:09.43 | russellb | i use svn :) |
| 05:09.46 | thehar | haha |
| 05:10.11 | russellb | and i have various aliases and variables that make my commands shorter |
| 05:10.21 | russellb | like, I have RA=http://svn.digium.com/svn/asterisk/ in my environment |
| 05:10.26 | russellb | so I can ... svn co $RA/trunk |
| 05:10.28 | thehar | hehe |
| 05:10.39 | russellb | svn co $RA/branches/1.4 |
| 05:10.44 | russellb | lots of stuff like that, heh |
| 05:11.16 | lilkid | thanks russellb :] , and ok you win carrar |
| 05:11.24 | russellb | lilkid: you're welcome |
| 05:11.34 | russellb | thehar: alias inst="sudo make install" |
| 05:11.45 | russellb | i'm laaazy :( |
| 05:12.02 | thehar | haha |
| 05:12.05 | thehar | wow |
| 05:12.10 | thehar | inst really is faster than install? |
| 05:12.23 | russellb | $ inst, instead of $ sudo make install |
| 05:12.24 | [TK]D-Fender | lilkid: no need to compile anything. Its a bunch of friggen sound files, jsut untar and put them where they belong |
| 05:12.26 | thehar | i'm sure you type just as fast if not faster than me |
| 05:12.27 | russellb | much less typing :-p |
| 05:12.39 | [TK]D-Fender | What's with all this package nonsense? |
| 05:13.50 | russellb | do the hardcore people from other projects recommend that you go with source, too? |
| 05:14.07 | russellb | it seems like the asterisk community is different in that respect, but maybe not .. |
| 05:14.10 | thehar | apting asterisk is not a good ideal silly |
| 05:14.22 | thehar | s/ideal/idea/ |
| 05:14.33 | russellb | i don't mind packages at all, if the source isn't modified |
| 05:14.45 | thehar | me either .. except asterisk |
| 05:14.48 | russellb | but a lot of packages (debian included) include a lot of patches that I either don't agree with, or just plain aren't safe |
| 05:14.52 | drmessano | I mod teh source |
| 05:15.06 | lilkid | [TK]D-Fender: lol i was saying that my path is /usr/share/asterisk/... instead of /var/lib/asterisk/... then people explained that i should compile rather than using apt |
| 05:15.20 | drmessano | removes every 14th z from the package |
| 05:15.31 | russellb | differences in directory structure and such, that's all fine ... |
| 05:15.40 | russellb | but applying 1 MB of patches ... I'm not cool with that. |
| 05:15.41 | [TK]D-Fender | lilkid: wE'RE TALKING ABOUT sound files HERE, NOT SOURCE CODE |
| 05:15.52 | russellb | [TK]D-Fender: i don't even know what we're talking about |
| 05:15.57 | thehar | i do |
| 05:16.03 | drmessano | russellb: Hacking in code to fix problems that may not even exist anymore.. always much lulz |
| 05:16.13 | [TK]D-Fender | russellb: He hant's UK English sounds instead of Allison. End of story |
| 05:16.20 | russellb | i like to interject into random conversations with opinions without knowing what the conversation is about |
| 05:16.20 | lilkid | [TK]D-Fender: keep up! we moved on from the sound files! |
| 05:16.26 | russellb | [TK]D-Fender: oic! |
| 05:16.30 | drmessano | "We added this to compensate for x in Asterisk 1.07 in DRMux |
| 05:16.32 | russellb | drmessano: yeah :-/ |
| 05:16.45 | [TK]D-Fender | lilFeel free to expand this into "rip my distro apart" now :) |
| 05:16.46 | drmessano | But hey, lets still apply it to 1.6 |
| 05:17.01 | thehar | ugh no distro wars |
| 05:17.12 | carrar | Atari OS! |
| 05:17.18 | Pan3D | Chevy is waaay better than Ford |
| 05:17.20 | russellb | fedora is for n00bz man |
| 05:17.34 | drmessano | Ubuntu is for primal tribesmen |
| 05:17.37 | russellb | fedora has some of the best packages for asterisk around |
| 05:17.40 | russellb | they were very well done |
| 05:17.53 | russellb | of course, asterisknow has the best asterisk packages :-p |
| 05:18.03 | Pan3D | lol |
| 05:18.04 | lilkid | using windows gives you a bigger penis |
| 05:18.10 | carrar | Where are the secret Allison porn sound files |
| 05:18.10 | thehar | is asterisknow your porn russellb ? |
| 05:18.14 | russellb | heh |
| 05:18.18 | russellb | something like that |
| 05:18.22 | drmessano | You mean "AsteriskNOW 1.2" or "AsteriskNOW 1.5b1 vaporware edition" |
| 05:18.23 | thehar | lol |
| 05:18.29 | thehar | vap0r |
| 05:18.40 | russellb | drmessano: 1.5 ftw |
| 05:18.48 | russellb | my data structures book is my porn |
| 05:18.52 | thehar | wow |
| 05:18.57 | Kyosh | secret allison porn files eh? heheheh |
| 05:18.58 | russellb | totally kidding. |
| 05:19.00 | Pan3D | hawt |
| 05:19.01 | drmessano | russellb: I'm using 1.5b2 right now... in the year 2525 |
| 05:19.13 | russellb | beta 4ever |
| 05:19.14 | thehar | i hope year 2525 asterisk now is more than 1.5 |
| 05:19.20 | drmessano | Thats right |
| 05:19.34 | russellb | blames ... thehar |
| 05:19.47 | thehar | what are you hating for? |
| 05:19.50 | russellb | <3 |
| 05:20.03 | thehar | russellb: please tell me that astricon isn't in phx again this year |
| 05:20.11 | russellb | i believe it is. |
| 05:20.13 | drmessano | I will give Qwell some credit... I mean, he got one beta release out the door.. Can't say that about Windows 2000 Home Edition or the XboX 361 |
| 05:20.15 | thehar | fuck. |
| 05:20.15 | thehar | seriously? |
| 05:20.23 | russellb | yeah. |
| 05:20.24 | thehar | =( |
| 05:20.28 | thehar | i don't want to go to phoenix again |
| 05:20.34 | russellb | thehar: *points at marketing* |
| 05:20.35 | Qwell | drmessano: actually, I spun ISOs today |
| 05:20.40 | drmessano | Wut? |
| 05:20.41 | russellb | Qwell: oh snap!! |
| 05:20.49 | thehar | have it in slc :D |
| 05:20.50 | Qwell | gotta figure out how to release them though |
| 05:21.01 | russellb | Qwell: torrentz man |
| 05:21.10 | drmessano | PUT THEM ON THE INTERWEB |
| 05:21.14 | drmessano | THATS HOW |
| 05:21.16 | drmessano | Christ man |
| 05:21.18 | drmessano | Listen |
| 05:21.23 | drmessano | Go down the hall |
| 05:21.25 | russellb | give it to me so i can make a keygen |
| 05:21.26 | carrar | When is Seattle Astricon!!! |
| 05:21.33 | thehar | i'll gives you some QMOE for frees courtesy of XM if marketing moves it to slc |
| 05:21.41 | drmessano | Theres this dude named Russell or something.. he's the "Head Open Source..." something or another.. |
| 05:21.52 | drmessano | He can tell you how to release something |
| 05:21.54 | drmessano | Its not hard |
| 05:21.56 | thehar | release |
| 05:21.59 | russellb | looks around |
| 05:22.15 | russellb | i can has codez? |
| 05:22.23 | drmessano | Step 1: Make the code |
| 05:22.24 | thehar | kan |
| 05:22.32 | drmessano | Step 2: Upload |
| 05:22.38 | drmessano | Step 3: THERE IS NO STEP 3 |
| 05:22.48 | thehar | kill step 2 by writing code on server |
| 05:22.55 | drmessano | WIN |
| 05:22.56 | thehar | MAXIMIZE time |
| 05:23.06 | thehar | in vim |
| 05:23.27 | drmessano | Qwell: So is this Beta 2 or AsteriskNOW 1.5b1.01.0001 ? |
| 05:23.38 | drmessano | Are we looking at a Wine release cycle here? |
| 05:23.39 | Qwell | 83 |
| 05:23.40 | thehar | time to watch movie with my wifey |
| 05:23.59 | drmessano | .009.0009.0000ohnevermind |
| 05:24.35 | thehar | waves bye to russellb |
| 05:24.36 | thehar | & |
| 05:24.37 | drmessano | Someone was asking me the other day if there was ever gonna be another beta.. |
| 05:24.39 | russellb | cya thehar ! |
| 05:24.42 | drmessano | You know what I told them????? |
| 05:24.46 | russellb | Qwell: it's just "83" ?? |
| 05:24.48 | drmessano | "TRY TRIXBOX" I said |
| 05:24.52 | denon | russellb: the intarweb is a scary place: http://mine.icanhascheezburger.com/view.aspx?ciid=3455320 |
| 05:24.55 | drmessano | Can you believe that? |
| 05:24.57 | Qwell | russellb: AsteriskNOW83 |
| 05:24.59 | *** join/#asterisk fiddur (i=fiddur@c042.rit.se) |
| 05:25.10 | *** kick/#asterisk [drmessano!i=north@pdpc/sponsor/digium/Qwell] by Qwell (Qwell) |
| 05:25.11 | *** join/#asterisk drmessano (n=nonya@pdpc/supporter/active/drmessano) |
| 05:25.20 | drmessano | SAD FACE :( |
| 05:25.25 | *** kick/#asterisk [drmessano!n=russell@asterisk/digium-open-source-team-lead/russellb] by russellb (russellb) |
| 05:25.26 | *** join/#asterisk drmessano (n=nonya@pdpc/supporter/active/drmessano) |
| 05:25.28 | russellb | giggles |
| 05:25.40 | russellb | denon: nice :) |
| 05:25.50 | drmessano | ....... |
| 05:26.45 | denon | russellb: I wonder how many gigabits of lolcat and facebook traverse the net each year |
| 05:27.06 | denon | we build this multi-billion dollar global network, and teenagers fire up myspace |
| 05:27.25 | russellb | likes facebook |
| 05:27.28 | drmessano | Wait now |
| 05:28.01 | drmessano | What about the facebook data thats called from twitter called from facebook called from myspace called from ZOMG BRAIN FREEZE |
| 05:28.02 | *** mode/#asterisk [-b %watchy!*@*] by russellb |
| 05:28.35 | denon | drmessano: no doubt! freakin social networking in circular loops trying to keep themselves in sync with everything else |
| 05:29.05 | drmessano | Nevermind the LOLCATZ images on Facebook linked from MySpace and uploaded to Flicker called from Twitter via Facebook MySpace Twitter.. Um, where was I? Dad is that you? |
| 05:29.13 | denon | and what's worse, is clients that post the same useless crap to a dozen social networking sites at once .. |
| 05:29.28 | denon | I mean, it wasn't hardly worth saying in an IM, but now it's on 50 servers around the globe |
| 05:29.29 | drmessano | MUST UPLOAD SAME CAT PIC TO 100 SITES, BRRRRB |
| 05:30.14 | drmessano | Best. Cat. Pic. Ever. Today. Hour. Last 10 mins. |
| 05:30.16 | drmessano | **MEMO** CATS DO CUTE SHIT. GET OVA IT |
| 05:30.45 | drmessano | Someone asked me the other day if I was on twitter.. |
| 05:30.59 | denon | I thought that stuff was illegal her |
| 05:31.00 | denon | here |
| 05:31.08 | denon | or is it just by prescription |
| 05:31.14 | denon | snickers |
| 05:31.46 | Qwell | drmessano: no, but really, I need to rebuild one package, respin the ISOs, and figure out how to upload them. hopefully tomorrow |
| 05:32.03 | russellb | Qwell: yay! |
| 05:32.06 | drmessano | Probably.. Maybe.. Well, Friendfeed updates my twitter which updates my facebook which updates my myspace which updates my IM which SMS's me then BlackBerry messages me to tell me I updated my twitter via friendfeed over Facebook through myspace.. So AM I ON TWITTER???? MAYBE. |
| 05:32.10 | russellb | Qwell: nice job working on all of that stuff |
| 05:32.26 | Qwell | russellb: you didn't see what package I made today |
| 05:32.33 | russellb | Qwell: not yet, I guess not :) |
| 05:32.34 | *** join/#asterisk voxter (n=voxter@S0106001c1025ca09.vc.shawcable.net) |
| 05:32.38 | Qwell | starts with an O and ends with an R2 |
| 05:32.44 | russellb | Qwell: wow |
| 05:32.49 | russellb | Qwell: you're seriously ahead of the game! |
| 05:32.50 | Qwell | I is teh ready |
| 05:32.53 | drmessano | Office Live Communications Server 2007 R2? |
| 05:33.00 | Qwell | didn't build or test it, but...shrug. was easy |
| 05:33.06 | russellb | Qwell: right. |
| 05:33.19 | russellb | Qwell: I think I might merge that this week |
| 05:33.20 | [TK]D-Fender | drmessano: Twitter? People on it are alright. Ok, half right. The FIRST half :) |
| 05:33.23 | Qwell | DO IT |
| 05:33.26 | russellb | no reason to wait |
| 05:33.36 | russellb | it compiles, _tons_ of people are already using it |
| 05:33.38 | Qwell | sounds like it passed review pretty well |
| 05:33.46 | russellb | and i can't find anything wrong with it in review |
| 05:33.48 | drmessano | Hmm |
| 05:34.04 | russellb | Qwell: i just get nervous on things that big :) |
| 05:34.13 | Qwell | whats the worst that could happen? |
| 05:34.19 | drmessano | "trixbox" |
| 05:34.20 | Qwell | well, no, don't answer that |
| 05:34.28 | russellb | it's broken and then people look at me and say "WTF DID YOU DO?!?!" |
| 05:34.34 | drmessano | You could run a make and end up with trixbox 2.6 |
| 05:34.58 | drmessano | With a giant jumping Kerry G Bonzi Buddy |
| 05:35.02 | russellb | Qwell: but hopefully given how many people have been running it already, it's ok ... |
| 05:35.05 | Qwell | yeah |
| 05:35.17 | drmessano | "It looks like youre trying to make a call.. OOOK OOOOK" |
| 05:35.23 | voxter | whatcha talkin about? something new coming? |
| 05:35.31 | drmessano | voxter: sounds like it |
| 05:35.32 | russellb | voxter: R2 support :) |
| 05:35.48 | denon | drmessano: http://mine.icanhascheezburger.com/view.aspx?ciid=3455370 |
| 05:36.02 | denon | blames russell for sending me to this stupid lolcat site :) |
| 05:36.04 | voxter | r2 d2!! :) or olcs ? |
| 05:36.07 | drmessano | denon: ROFLLLLLLLL |
| 05:36.18 | drmessano | Holy shit.. thats is SOOO WIN |
| 05:36.28 | drmessano | Downloading for safe keeping |
| 05:36.30 | russellb | heh |
| 05:36.46 | russellb | someone brought a really big server to astricon in a huge suitcase to get help installing asterisk |
| 05:36.53 | denon | drmessano: heh, or you could just make it again in 10 seconds like I did |
| 05:36.53 | carrar | all this talk about lol |
| 05:36.54 | carrar | http://osburn.com/lol_flame_thrower.gif |
| 05:37.04 | voxter | russellb: oh man. the dude with the acid glasses. |
| 05:37.15 | denon | russellb: wow, never heard of vmware? |
| 05:37.16 | drmessano | ~howcaninstall |
| 05:37.18 | voxter | voxter: he even brought a 19 inch lcd too! |
| 05:37.19 | drmessano | Ok good |
| 05:37.27 | russellb | denon: it was hilarious |
| 05:37.30 | russellb | voxter: yeah! |
| 05:37.36 | thehar | russellb: I REMEMBER that dude! |
| 05:37.38 | thehar | hahaha |
| 05:37.41 | denon | nice .. |
| 05:37.42 | thehar | i was in the codezone when he was in there |
| 05:37.43 | drmessano | ~howcaninstall |
| 05:37.44 | jbot | http://mine.icanhascheezburger.com/view.aspx?ciid=3455370 |
| 05:37.46 | voxter | you dont even know how awesome it was. |
| 05:37.52 | voxter | i wish someone got some pictures |
| 05:37.59 | denon | drmessano: heh, guess I shoulda put my name on it |
| 05:38.00 | denon | claim to fame |
| 05:38.04 | thehar | i was too busy fetching cookies from the hall |
| 05:38.10 | thehar | to picture |
| 05:38.32 | thehar | & for reals |
| 05:38.33 | drmessano | "gotta have my pops" INDEED, SIR |
| 05:38.39 | thehar | addicting you people are |
| 05:38.56 | russellb | oh snap, thehar wrote on my wall! |
| 05:39.05 | denon | russellb: I bet ssh from his laptop was a life changing experience .. or will be, someday |
| 05:39.11 | voxter | damn graffiti artists. |
| 05:39.14 | russellb | denon: someday, yes :) |
| 05:39.34 | voxter | I wonder if he took that monster tower thru security as carryon |
| 05:39.39 | voxter | That would have been an amusing conversation. |
| 05:40.21 | drmessano | "Whats in the box?" "10 year old technology" |
| 05:40.24 | drmessano | "Carry on then" |
| 05:40.34 | Kyosh | anyone know where i can download a complete NPANXX list/database? |
| 05:40.35 | denon | voxter: the security people always have something interesting to say when I carry on my 7960 |
| 05:40.50 | carrar | Kyosh, how about parts? |
| 05:40.54 | russellb | you should put the 7960 on your belt and say it's your cell p hone |
| 05:41.02 | carrar | Kyosh, http://www.localcallingguide.com/ |
| 05:41.04 | voxter | denon: haha i've taken a mixed bag of phones a few times. i usually just get a funny look like 'dude you need some meds' and they let me pass. |
| 05:41.06 | Kyosh | carrar, as much as i can get would be great |
| 05:41.09 | drmessano | Zack Morris Cellphone |
| 05:41.11 | denon | Kyosh: nanpa.com |
| 05:41.18 | *** join/#asterisk GameGamer43 (n=GameGame@cpe-66-67-181-68.rochester.res.rr.com) |
| 05:41.26 | Kyosh | nanpa,com doesnt provide a list/database |
| 05:41.32 | drmessano | http://www.dialaphone.co.uk/blog/wp-content/uploads/2007/09/zack.jpg |
| 05:41.49 | voxter | ive got that exact phone on my desk at work |
| 05:41.52 | carrar | telcordia sells lots og npanxx goodies too at a high price |
| 05:41.57 | voxter | it works too, except they took all the amps towers down. |
| 05:42.02 | Kyosh | too high a price, i know |
| 05:42.03 | denon | russellb: last time, flying out of california or somewhere, the security girl is like "man, thats the EXACT SAME phone we use here. ." |
| 05:42.04 | voxter | I thought about converting it into a bluetooth headset |
| 05:42.08 | denon | and I'm goin .. yeah, its a common phone .. |
| 05:42.10 | voxter | speaking of which, john todd STILL hasnt sent me mine |
| 05:42.11 | drmessano | http://i65.photobucket.com/albums/h214/bakedspam/zack-thumb-715639.jpg |
| 05:42.14 | drmessano | THAT IS WIN |
| 05:42.20 | denon | and she's like, "no, but its EXACRTLY the same .. " |
| 05:42.26 | denon | I think she thought I swiped her phone or something |
| 05:42.38 | denon | I hated to tell her that all 7960s do look exactly the same |
| 05:42.49 | denon | she must have thought life has some really incredible ironies |
| 05:43.02 | drmessano | Are you a pilot? |
| 05:43.09 | denon | no |
| 05:43.13 | drmessano | "No, but I screw like one.. Call me later" |
| 05:43.21 | denon | heh |
| 05:43.22 | voxter | denon: haha. "your blue 1998 toyota camry looks JUST LIKE MINE, OMGWTF" |
| 05:43.27 | *** join/#asterisk mintos (n=mvaliyav@nat/redhat-in/x-6055fe43f583dacb) |
| 05:43.48 | Kyosh | carrar, thats not really a list/database |
| 05:43.59 | Kyosh | i'd like to have like a huge excel or something |
| 05:44.04 | Kyosh | even a csv would be nice |
| 05:44.22 | drmessano | Kyosh: Who doesnt want a huge excel? They have pills for that now |
| 05:44.23 | denon | just start dialing numbers and ask them where they live |
| 05:44.24 | Kyosh | just so hard to find something |
| 05:44.30 | Kyosh | hehe |
| 05:44.43 | carrar | Kyosh, you can Email the owner |
| 05:44.53 | carrar | maybe spend more then 5 seconds on that site |
| 05:45.03 | voxter | russellb: do you haz link to the r2 stuff? or is it not public yet? |
| 05:45.12 | Kyosh | im still going thru it, but i dont see a downloadable file |
| 05:45.34 | russellb | voxter: libopenr2.org |
| 05:45.37 | carrar | be creative |
| 05:46.03 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 05:46.06 | voxter | cool, thanks. |
| 05:46.10 | russellb | sure, np |
| 05:46.48 | *** part/#asterisk TrentCreek (i=kvirc@ppp-70-254-112-199.dsl.hrlntx.swbell.net) |
| 05:47.13 | *** join/#asterisk Maliuta (n=scooby@kiev.lusan.id.au) |
| 05:47.19 | *** join/#asterisk CunningPike (n=arodgers@S01060014bf81366b.vc.shawcable.net) |
| 05:50.22 | voxter | any of you guys using * with xen? i was thinking of doing some multi tenant vhosting using xen.. |
| 05:50.27 | voxter | since freepbx has no multi tenant of course. |
| 05:50.35 | carrar | xen works great |
| 05:50.51 | voxter | best to toss in a TDM card and share it across all DOMu's for a timing source? |
| 05:51.32 | *** join/#asterisk nkohh (n=justin@unaffiliated/kohh) |
| 05:51.52 | russellb | voxter: there are patches around to use dahdi_dummy in xen |
| 05:51.57 | russellb | if they're not already merged |
| 05:52.08 | *** join/#asterisk frk2 (n=frk2@zivios/member/fkhan) |
| 05:52.18 | voxter | gotcha. ill give it a whirl in the next week or two. just wanted to see if people were doin' it. |
| 05:52.29 | russellb | yeah, i've heard of people doing it a number of times |
| 05:52.30 | voxter | seems to be a clean way to go to give people their own hosted freepbx |
| 05:52.31 | russellb | no personal experience, though |
| 05:52.32 | nkohh | hello everyone. is there a way to set up a multi-user VoiceMailMain extention? so that everyone could dial in to get their VM at the same extention, and then type in their mailbox number and passcode? |
| 05:52.54 | russellb | nkohh: yeah. Just have an extension that calls VoiceMailMain() with no arguments |
| 05:52.55 | voxter | nkohh: call voicemailmain() with no arguments and thats what it'll do. |
| 05:52.58 | nkohh | sorry, I hope i'm being clear. I'm a bit out of it tonight |
| 05:53.14 | voxter | like, exten => 8500,1,VoiceMailMain() |
| 05:53.17 | nkohh | russellb, voxter: you guys are fantastic. sorry for my ineptitude. i'm sure that's on voip-info if i'd looked hard enough. |
| 05:53.19 | voxter | anyone who dials 8500 will be prompted. |
| 05:53.24 | russellb | nkohh: no problem |
| 05:53.30 | nkohh | you guys are outstanding members of the comumunity. have a fantastic night. |
| 05:53.34 | russellb | i like questions that i have an answer to |
| 05:53.37 | nkohh | community! |
| 05:53.40 | voxter | russellb: i concur. |
| 05:54.31 | voxter | i guess ill go finish laundry now. two thumbs down. |
| 05:57.34 | lilkid | can anyone shed any light on this: |
| 05:57.37 | lilkid | res_musiconhold.c:515 monmp3thread: Request to schedule in the past?!?! |
| 05:57.39 | lilkid | ? |
| 05:58.47 | russellb | you can ignore it |
| 05:59.03 | *** join/#asterisk SunnyDP (n=scan@bas7-montrealak-1128545096.dsl.bell.ca) |
| 06:01.15 | lilkid | music on hold refuses to work because of it |
| 06:02.24 | mcnobody | Does anyone know about Local channel as a member of queue? |
| 06:04.01 | mcnobody | I have a problem with language of the announce message for the queue member. I can't find any way to configure that language for Local channel member. |
| 06:04.02 | *** join/#asterisk chikkis (n=chikkis@121.243.138.136) |
| 06:04.50 | carrar | CHANNEL(language) ? |
| 06:05.02 | carrar | can you set that? |
| 06:05.38 | mcnobody | carrar: Nope. That selectes the language of channel. The language of announce is selected by language of queue member device. |
| 06:05.51 | chikkis | hi everyone |
| 06:06.38 | mcnobody | carrar: For SIP/123 as a member this works well but when I have Local/123@context which sets something and then calls SIP/123 the language of announce message seems to be always english. |
| 06:12.29 | chikkis | can we make asterisk as service provider for sip trunks |
| 06:12.42 | carrar | You could |
| 06:12.53 | chikkis | okay |
| 06:13.17 | carrar | SER might be a better choice |
| 06:14.00 | chikkis | i wanted to build a practise lab for cisco call manager with an asterik(trixbox) as sip provider |
| 06:14.12 | chikkis | SER ??? |
| 06:14.19 | carrar | SIP Express Router |
| 06:14.55 | chikkis | i am kind new to the astrisk |
| 06:15.11 | chikkis | could give me some pointer for it |
| 06:15.19 | drmessano | 1. Dont use Trixbox |
| 06:15.20 | carrar | ~book |
| 06:15.21 | jbot | book is probably probably Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://tfot.leifmadsen.com or see ~buybook |
| 06:15.25 | carrar | read that |
| 06:15.28 | carrar | free pdf |
| 06:16.58 | chikkis | though i have played with trixbox as in installing and configuring |
| 06:17.11 | carrar | don't use that either |
| 06:17.30 | drmessano | trixbox is crap |
| 07:35.48 | *** join/#asterisk jbot (i=ibot@rikers.org) |
| 07:35.48 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.0.5 (2009/01/23), 1.4.23.1 (2009/01/23), *-Addons 1.6.0.1 (2008/12/02), 1.4.7 (2008/06/04), dahdi-linux 2.1.0.3, dahdi-tools 2.1.0.2 (2008/12/18), Libpri 1.4.7 (2008/08/05) -=- Related channels: #asterisknow #asterisk-gui #switchvox #freepbx #asterisk-bugs #asterisk-dev -=- jbot is back! |
| 07:37.52 | lanning | kb3ien, looks like they are building the capability into the routers. |
| 07:40.11 | *** join/#asterisk jbot (i=ibot@rikers.org) |
| 07:40.11 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.0.5 (2009/01/23), 1.4.23.1 (2009/01/23), *-Addons 1.6.0.1 (2008/12/02), 1.4.7 (2008/06/04), dahdi-linux 2.1.0.3, dahdi-tools 2.1.0.2 (2008/12/18), Libpri 1.4.7 (2008/08/05) -=- Related channels: #asterisknow #asterisk-gui #switchvox #freepbx #asterisk-bugs #asterisk-dev -=- jbot is back! |
| 07:40.12 | *** join/#asterisk Maliuta (n=scooby@kiev.lusan.id.au) |
| 07:45.05 | *** join/#asterisk jbot (i=ibot@rikers.org) |
| 07:45.05 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.0.5 (2009/01/23), 1.4.23.1 (2009/01/23), *-Addons 1.6.0.1 (2008/12/02), 1.4.7 (2008/06/04), dahdi-linux 2.1.0.3, dahdi-tools 2.1.0.2 (2008/12/18), Libpri 1.4.7 (2008/08/05) -=- Related channels: #asterisknow #asterisk-gui #switchvox #freepbx #asterisk-bugs #asterisk-dev -=- jbot is back! |
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| 08:15.54 | brunner | once I configure alsa.conf, how do I actually place a call? |
| 08:16.20 | rob0 | core show application dial |
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| 08:17.55 | jad_jay | Hello world |
| 08:18.11 | drmessano | SEGFAULT |
| 08:18.18 | jad_jay | :) |
| 08:18.45 | jad_jay | hard to debug hu? |
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| 09:06.34 | jad_jay | ~book |
| 09:06.35 | jbot | extra, extra, read all about it, book is probably Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://tfot.leifmadsen.com or see ~buybook |
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| 09:07.32 | jad_jay | ~wiki |
| 09:07.45 | ludan | hi guys |
| 09:07.56 | jad_jay | ola |
| 09:08.01 | ludan | how do I create a new admin for my asterisk gui? |
| 09:08.15 | jad_jay | dunno :) |
| 09:09.06 | ludan | ah, from manager.conf |
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| 09:12.03 | buzukelis | hollo! what is the best billing solution to asterisk now? |
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| 09:33.44 | jermey_g | if afer the bridge, audio disappear at one end, what could be wrong? |
| 09:34.14 | jermey_g | It seems there is some problem in codec negotiation, one side switches to ulaw and other to alaw |
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| 09:34.45 | jermey_g | while both phones support all the codecs in use. |
| 09:41.24 | tzafrir_laptop | ludan, a user that is a "manager" |
| 09:41.33 | ludan | found thanks :) |
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| 09:56.34 | tzafrir_laptop | this is what we call "brown out" |
| 09:57.08 | Dovid | haha |
| 09:57.08 | tzafrir_laptop | hmm.... what makes jbot play that message once a day at about that time? |
| 09:57.09 | Dovid | that was a good movie |
| 09:57.09 | tzafrir_laptop | Is this a cron jot at the server? |
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| 10:12.25 | *** join/#asterisk Jeff78 (n=merdze@118.100.164.103) |
| 10:12.38 | Jeff78 | hi |
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| 10:14.22 | tompaw | Morning everyone. |
| 10:14.52 | tompaw | Are there any reasons not to run Asterisk insidie VMWare VM? |
| 10:15.09 | tompaw | Assuming the VM is configured properly of course. |
| 10:16.40 | Jeff78 | good day! |
| 10:19.02 | Jeff78 | does anyone here face a problem where 2 voice cards (1 analog and 1 digital) are installed on a same pc and the span 1 and span 2 detected both card but position (span 1 or span 2) change after rebooting? |
| 10:25.17 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
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| 10:34.44 | jad_jay | hi |
| 10:34.53 | jad_jay | i have prb with freephonie |
| 10:35.05 | jad_jay | is so using it |
| 10:35.34 | jad_jay | can't reach, may i need the public and secret key |
| 10:36.54 | jad_jay | ~freephonie |
| 10:36.58 | xacatecas | hi, let's say I have Dial(SIP/100,15) and the Dial(SIP/100&SIP/101) ... is there any way to prevent the double missed call on SIP/100 if 101 answers the call? |
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| 10:57.53 | kippi | hey |
| 10:58.23 | kippi | is there away to use SIPAddHeader to disable missed calls on a linksys spa942 handset |
| 10:59.38 | lbt | tompaw: so long as the hardware passes through I think it's OK. I know that some asterisk hw generates a *lot* of interrupts and that may be something to watch out for... |
| 11:00.59 | lbt | I use the latest Xen and I don't think it handles PCI passthru at the moment - I'll be virtualising when it does |
| 11:03.56 | lbt | I'm having a problem with Festival on 1.6-rc1 .... some log/config info : http://pastebin.com/d49a2f03c so it creates a wav (which mplayer plays) but I get no sound and the Hangup doesn't get executed. Any suggestions? |
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| 11:17.44 | tompaw | lbt: thank you! |
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| 11:59.49 | kippi | is there away to use SIPAddHeader to disable missed calls on a linksys spa942 handset |
| 12:00.28 | kaldemar | what does linksys support say on that? |
| 12:03.11 | jad_jay | is there a good howto/manual to help for nat |
| 12:03.22 | jad_jay | and for sip peer |
| 12:03.43 | kaldemar | ~sipnat |
| 12:03.44 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 12:06.17 | xacatecas | hi, asterisk keeps on wanting me to use _X., however, this doesn't catch *ext values ... is there something other than _. that will catch the * as well or should I just use [*0-9] ? |
| 12:08.47 | kaldemar | use [*0-9] |
| 12:09.01 | jad_jay | kaldemar: thkS |
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| 12:14.51 | jad_jay | kaldemar: well maybe it failed because of my nat config, so one question is it normal to have this on the net http://paste.debian.net/28664/ |
| 12:15.27 | jad_jay | it tries to send packet to freephonie, then the same to the asterisk host |
| 12:16.20 | kaldemar | well, you obviously have configured something to do so. |
| 12:17.24 | jad_jay | hum |
| 12:17.33 | jad_jay | could it comes from asterisk? |
| 12:18.37 | kaldemar | yes it could if you have configured asterisk to register to itself. |
| 12:19.15 | *** join/#asterisk BBHoss (n=bbhoss@68.159.148.173) |
| 12:19.24 | jad_jay | hum in sip.conf ? don't think so |
| 12:19.43 | *** join/#asterisk McDarren (n=darren@shrubbery.chris.net.au) |
| 12:19.47 | jad_jay | so it came from my nat |
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| 12:21.25 | McDarren | Hello, do users in this channel provide asterisk-related help? |
| 12:21.57 | jad_jay | McDarren: no no it's all about bread making |
| 12:22.50 | McDarren | oh, that's a fortunate coincidence, as I was considering baking a loaf just now... |
| 12:23.28 | jad_jay | arf |
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| 12:24.11 | *** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim) |
| 12:24.13 | McDarren | I would like to know if it is possible to detect which party initiates an end call, and if so - how? |
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| 12:25.57 | Guza | hi can someone help me to install app_rxfax for asterisk 1.2 |
| 12:26.57 | Carlos_PHX | Guza: What's the issue you're having? |
| 12:27.10 | Guza | i dont know where to download this app |
| 12:27.13 | Carlos_PHX | Keep in mind that rxfax for 1.2 is very limited and not fully reliable. |
| 12:27.34 | Guza | hmm |
| 12:27.42 | Guza | how meny limited ? |
| 12:28.02 | Carlos_PHX | It's been a long time since I did a 1.2 install, but I believe it is part of the original distro and you just have to select it when you compile. I could be wrong. |
| 12:28.12 | Carlos_PHX | What is your goal with this? |
| 12:28.44 | Guza | i want to resive faxes by email |
| 12:29.56 | xacatecas | ta kaldemar |
| 12:30.02 | Carlos_PHX | What type of connection to the PSTN do you have? |
| 12:30.04 | xacatecas | what exactly is the issue with _. ? |
| 12:30.28 | Guza | i have 8 isdn line |
| 12:31.05 | Carlos_PHX | That should work fine. You'll need to recompile Asterisk. |
| 12:31.17 | kaldemar | xacatecas: it matches to special extensions such as i, t, and h. |
| 12:31.33 | xacatecas | kaldemar, oh. ouch. |
| 12:31.39 | xacatecas | yes, that can be "unexpected" |
| 12:31.52 | xacatecas | but in this particular case wouldn't matter. |
| 12:31.52 | kaldemar | yes, and undesired. :) |
| 12:32.00 | Guza | is there another solution without recompile ? |
| 12:32.43 | xacatecas | kaldemar, thanks again. |
| 12:32.59 | xacatecas | updated and committed in my repository. |
| 12:33.28 | Carlos_PHX | Guza: Not that I know of. That's the only way I know to include it. |
| 12:34.30 | Carlos_PHX | Fax via ISDN is pretty reliable as long as the cards and bus timing are stable. However our 1.2 server still has some fax machines it will not receive from. |
| 12:34.46 | kaldemar | McDarren: if you use option g with app Dial, dialplan execution will continue to next priority if the destination channel hangs up. |
| 12:34.50 | Carlos_PHX | The 1.6 version of fax is 100% good however. |
| 12:39.53 | Guza | thanks you |
| 12:40.09 | Guza | bye |
| 12:40.12 | *** part/#asterisk Guza (n=zagu@77.239.64.134) |
| 12:41.52 | jad_jay | kaldemar: well could you brief me shortly? I'm behind a nat, to register a sip provider what do i need as udp port visible |
| 12:42.37 | kaldemar | i can't brief you any better than the howto's. |
| 12:42.57 | jad_jay | the howto's aren't so clear |
| 12:42.59 | fun330 | can someone help me, i can't get my debug to show the trunk name instead of the ip address? i have a trunk in sip.conf that matches the ip address but the ip address always shows up |
| 12:43.22 | fun330 | i have a pastebin of both http://pastebin.com/d8179b93 |
| 12:43.28 | jad_jay | 5060 visible and redirected |
| 12:43.38 | jad_jay | kaldemar: 5060 visible and redirected |
| 12:43.52 | jad_jay | kaldemar: 10000:20000 visible and redirected |
| 12:51.32 | *** join/#asterisk botox93 (n=botox93@213.221.82.242) |
| 12:53.38 | kippi | is there away to use SIPAddHeader to disable missed calls on a linksys spa942 handset |
| 12:54.31 | Carlos_PHX | I have a potential customer that wants to connect 67 analog phones. It's a motel, so there is no cat 5 to the rooms and trying to avoid re-cabling expense. I don't know of any cost-effective way to connect to analog phones. Ideas? |
| 12:54.47 | kaldemar | kippi: have you asked linksys support? |
| 12:56.13 | jad_jay | kaldemar: Got SIP response 423 "Interval Too Brief" |
| 12:56.15 | kippi | kaldemar: haven't had much luck with linksys support, always seem to get the answer of no |
| 12:56.41 | *** join/#asterisk zeeesh (n=zeeesh@203.215.179.43) |
| 13:00.28 | *** join/#asterisk dlewis (i=c7340d67@about/security/staff/dlewis) |
| 13:02.05 | jad_jay | kaldemar: got it |
| 13:02.36 | jad_jay | i was needing defaultexpirey and nat=yes in general |
| 13:02.45 | jad_jay | thanks |
| 13:03.35 | *** join/#asterisk SuPrSluG (n=SuPrSluG@firewall-a.buf.ny.i-evolve.net) |
| 13:04.39 | McDarren | kaldemar: thanks :) |
| 13:04.44 | *** part/#asterisk dlewis (i=c7340d67@about/security/staff/dlewis) |
| 13:05.25 | *** join/#asterisk propellerhead (n=yogurt2u@host135.190-138-101.telecom.net.ar) |
| 13:07.28 | *** join/#asterisk MrNaz (n=mrnaz@ppp118-208-219-90.lns10.mel6.internode.on.net) |
| 13:09.47 | *** join/#asterisk DarkRift (n=dark@65.92.171.135) |
| 13:10.02 | JT | Carlos_PHX: but the rooms already have phone cabling, right? |
| 13:11.07 | Carlos_PHX | JT: Right, just need to take them to digital. |
| 13:11.45 | JT | Carlos_PHX: best way would be to get hardware FXS to SIP gateways or FXS to T1 channel banks |
| 13:12.05 | Carlos_PHX | Right, just trying to locate a not so expensive solution. |
| 13:12.23 | Carlos_PHX | Everything seems to be in the $1500 ++ range for 24 FXS. |
| 13:12.36 | JT | it's not that expensive, when you consider how many channels you are trying to terminate |
| 13:12.46 | JT | if they can't afford that, they're too damn stingey |
| 13:12.52 | Carlos_PHX | It's that expensive, when you consider the customer just won't buy it. |
| 13:13.01 | Carlos_PHX | Not worth doing, really. |
| 13:13.17 | JT | can still be worth doing |
| 13:14.00 | Carlos_PHX | There are no feature benefits in this case, just dollars. |
| 13:14.11 | Carlos_PHX | Replace aging phone system, replace T1-based lines. |
| 13:15.18 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 13:15.26 | Carlos_PHX | Well, there goes the channel. |
| 13:15.43 | JT | there is probably the potential for feature benefits, it just depends on how creative you are |
| 13:15.57 | Carlos_PHX | It's a motel, they are clear that they just want cheap dialtone. |
| 13:16.09 | Carlos_PHX | I'm usually very creative on features, but they basically don't care. |
| 13:16.21 | Carlos_PHX | "we're a Super-8, not the Hilton." |
| 13:16.27 | *** join/#asterisk maxhbp2005 (n=maxhbp20@122.169.12.23) |
| 13:16.37 | Carlos_PHX | "Our customers just need to call the bail bondsman and/or the drug dealer." |
| 13:16.59 | JT | then why sell them something they don't need or won't pay for? |
| 13:17.08 | Carlos_PHX | That's what I'm saying. |
| 13:17.31 | Carlos_PHX | If there's a way to do inexpensive service to analog phones, it works, if not, probably won't. |
| 13:17.37 | JT | little returns at the low end of the market |
| 13:18.06 | JT | it depends if you can save much on call costs or not |
| 13:18.08 | Carlos_PHX | If they had cat 5, simple and cost effective, but I'm sure it's just cat 3 to the rooms. |
| 13:18.18 | Carlos_PHX | We can save them $200/mo on internet and phone. |
| 13:18.25 | Carlos_PHX | And still make good money. |
| 13:18.30 | JT | simple and cost effective to do what, use sip phones? |
| 13:18.34 | Carlos_PHX | Yes |
| 13:18.42 | *** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
| 13:18.53 | JT | that would cost way more than sip gateways |
| 13:20.24 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 13:20.26 | Carlos_PHX | At $60/phone, not really. |
| 13:21.10 | JT | it's at least as expensive as using 24 port mediatrix or audiocodes units |
| 13:21.16 | JT | what phones btw? |
| 13:21.38 | Carlos_PHX | Linksys 901 |
| 13:22.07 | Carlos_PHX | I'll have to check out those options too. |
| 13:23.03 | maxhbp2005 | Hello All |
| 13:23.35 | JT | they're around $1500 |
| 13:24.07 | maxhbp2005 | I am using asterisk 1.6, i am working with phpagi script which is sending call and getting the duration after the call is completed, that code working fine in asterisk 1.4 with deadagi |
| 13:24.24 | maxhbp2005 | but in asterisk 1.6 when call is hangup the script is also hangup |
| 13:24.28 | Carlos_PHX | Yeah, so basically same as phones. |
| 13:24.33 | Carlos_PHX | Worth considering. |
| 13:24.41 | maxhbp2005 | so i am not able to run the another part of script |
| 13:24.42 | Carlos_PHX | We've never used one, so it's an unknown. |
| 13:24.46 | maxhbp2005 | can anyone help me |
| 13:24.55 | JT | phones have the cost of switches on top, especially if PoE |
| 13:25.09 | JT | also i think sip phones are too nice/fragile to give to motel guests |
| 13:25.25 | Carlos_PHX | That may be true. |
| 13:25.30 | [TK]D-Fender | And the 901 has no features.... hardly worth calling it a SIP phone. |
| 13:25.52 | [TK]D-Fender | 901 = phone w/ ATA soldered on... no CID even |
| 13:25.54 | Carlos_PHX | realizes he tossed out some Linksys bait |
| 13:26.13 | beek | Carlos_PHX: more like chum |
| 13:26.14 | Carlos_PHX | Sure, but it's still better than any Polycom. |
| 13:26.32 | beek | ducks |
| 13:26.41 | Carlos_PHX | Oh yeah, yesterday I replaced my last remaining IP 650 with a Linksys 925G. |
| 13:26.44 | [TK]D-Fender | feeds Carlos_PHX to the volcano and walks passively away |
| 13:26.44 | JT | yeah right |
| 13:26.55 | JT | better than a polycom, much lols to be had |
| 13:27.09 | [TK]D-Fender | JT : I'd believe him.. the 925G is a very nice looking phone. |
| 13:27.10 | Carlos_PHX | Heh, we've had that discussion here. |
| 13:27.22 | maxhbp2005 | is there any changes we have to give in deadagi call? |
| 13:27.26 | Carlos_PHX | We are replacing all the Polycoms with Linksys when we can. Don't sell Polycom any more. |
| 13:27.42 | JT | [TK]D-Fender: that the 901 is better than all polycoms? |
| 13:27.50 | Carlos_PHX | Yesterday I had to work at a site with Polycoms and remember why I hate them oh so much. |
| 13:28.05 | Carlos_PHX | JT: Nah, that was just obvious bait. |
| 13:28.12 | [TK]D-Fender | JT : Hell no.. its worse than ana analog phone on ATA :) |
| 13:30.52 | Carlos_PHX | So in all seriousness, that 925 is a hell of a phone. Only used it for half a day so far. Sound quality is very Polycom-HD-like. Not truly an HD codec, just sounds good. |
| 13:31.01 | *** join/#asterisk ZaVoid (n=zavoid@75.147.121.177) |
| 13:31.04 | Carlos_PHX | Looking forward to trying out wireless and BT. |
| 13:31.30 | *** join/#asterisk botox93 (n=botox93@213.221.82.242) |
| 13:32.29 | jermey_g | how do we read this |
| 13:32.31 | jermey_g | Peer User/ANR Call ID Format Hold Last Message |
| 13:32.31 | jermey_g | 192.168.20.243 2010 5e1ba0ee48036ba 0x4 (ulaw) No Init: INVITE |
| 13:32.31 | jermey_g | 192.168.20.162 2011 000000000014a00 0x4 (ulaw) No Rx: INVITE |
| 13:32.44 | jermey_g | the last column implies what? |
| 13:33.47 | *** join/#asterisk SunnyDP (n=scan@bas7-montrealak-1128545096.dsl.bell.ca) |
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| 13:57.03 | sheri_rao | i want to check my server, can anyone help me check H323 by giving me call |
| 13:57.33 | jer | is it possible to use chan_mobile with a usb connection to the phone? (using a nokia e51, reported to work really well) or do i need a bluetooth dongle? |
| 13:59.10 | *** join/#asterisk Torrieri (n=Torrieri@nelug/crew/torrieri) |
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| 14:02.34 | kaldemar | sheri_rao: good luck again |
| 14:04.00 | *** join/#asterisk MmixX (n=mix@61.14.191.142) |
| 14:07.11 | *** join/#asterisk telnettech (i=telnette@gw.percipia.com) |
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| 14:12.21 | disposable | is there a way to log the output in asterisk console to a file? /var/log/asterisk/messages isn't logging everything i want (sip debug) |
| 14:12.58 | tzafrir_laptop | disposable not directly, but see logger.conf |
| 14:13.15 | tzafrir_laptop | To see what is logged where: logger show channels |
| 14:13.30 | disposable | tzafrir_laptop: thank you |
| 14:13.37 | kaldemar | disposable: a command called script will help with that |
| 14:14.19 | kaldemar | if you can be attached to the cli while grabbing debug. |
| 14:14.36 | *** part/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
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| 14:29.23 | plundra | Quick question; what functions/ways do you use to control timespecific stuff in AEL? (Not AEL2) |
| 14:29.36 | plundra | Just want a basic openhours setup. |
| 14:30.03 | plundra | Don't really think includes => is the way to go in AEL, right? And ifTime isn't available. |
| 14:30.32 | *** join/#asterisk fcois93 (n=Administ@bagnolet.acropolistelecom.net) |
| 14:30.57 | [TK]D-Fender | plundra : Everything is available in AEL that is avaiable in basic dialplan as apps are concerned |
| 14:31.02 | fcois93 | anyone use A2billing ? |
| 14:31.23 | plundra | [TK]D-Fender: Hmm, mkay. Then I must be using it wrong :-D Thanks. |
| 14:31.25 | fcois93 | when I do a call, A2billing ask me again the number to call! |
| 14:31.45 | fcois93 | and, I want that it call my dial numer |
| 14:36.42 | plundra | What extensions is used in the context I use in my includes? |
| 14:40.04 | *** join/#asterisk neurosys (n=vinix@173.9.159.182) |
| 14:47.57 | *** join/#asterisk [intra]lanman (n=intralan@freeswitch/developer/intralanman) |
| 14:48.30 | Dovid | what would cause a channel lock on a PRI ? |
| 14:49.37 | tzafrir_laptop | many things |
| 14:49.49 | tzafrir_laptop | a lock held for long time? |
| 14:50.34 | Dovid | tzafrir_laptop: Not sure. speaking to cellcom now. it seems to only happen on incoming calls. they said when they try to send a call they get back that it is locked and that they cant send the call |
| 14:50.35 | jad_jay | Yes IT WORKS GREAT, Thks all of you |
| 14:51.54 | *** part/#asterisk mcargile (n=mikec@rrcs-24-173-156-170.se.biz.rr.com) |
| 14:51.58 | Dovid | tzafrir_laptop: carrier said between 5 and 15 minutes |
| 14:52.37 | [TK]D-Fender | plundra: HUH? |
| 14:53.48 | plundra | [TK]D-Fender: In the top of my context, I had an includes => othercontext|somefinetime|...; but it didn't seem to jump there, ever. |
| 14:55.00 | plundra | [TK]D-Fender: I'm trying out GotoIfTime now, which sounds more like what I want, and it kind of works, it jumps to where I want, although it does this _all_ the time, apparently :-D (Also, AEL warned about this not being the AEL way of doing it, I should use if/while/goto-keywords instead. |
| 14:56.15 | *** join/#asterisk Mog (n=mog@c-68-62-170-242.hsd1.al.comcast.net) |
| 14:56.15 | *** mode/#asterisk [+o Mog] by ChanServ |
| 14:56.27 | [TK]D-Fender | plundra: Pastebin what you've done that fails |
| 14:56.37 | *** join/#asterisk fskrotzki (n=fskrotzk@host198.textwise.com) |
| 14:59.01 | *** join/#asterisk hi365 (n=hi365@bzq-219-141-66.static.bezeqint.net) |
| 14:59.18 | hi365 | anyone have experience with asterisk <-> sonus equpment? |
| 14:59.56 | plundra | [TK]D-Fender: Not much to pastebin; GotoIfTime(14:00-16:00|*|*|*?closed|s|1); Always seems to jump. No matter if it's with in the timerange or not. |
| 15:00.20 | *** join/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 15:00.55 | kippi | is there away to use SIPAddHeader to disable missed calls on a linksys spa942 handset |
| 15:01.20 | [TK]D-Fender | kippi: What does their admin guide say? |
| 15:01.41 | *** join/#asterisk icebrew54 (i=proxy@static-71-117-242-28.ptldor.dsl-w.verizon.net) |
| 15:02.42 | *** join/#asterisk jmacz (n=jmacz@190.144.75.22) |
| 15:02.52 | kippi | [TK]D-Fender: can't see anything for SIP Headers and disabling missed calls, can only see how to switch them off for all calls |
| 15:03.17 | [TK]D-Fender | kippi: then thats likely all you can do. |
| 15:05.12 | *** join/#asterisk UQlev (n=yuriy@nb11-125.static.cytanet.com.cy) |
| 15:06.16 | tzafrir_laptop | Dovid, I believe it is the kind of lock held by an insect |
| 15:06.23 | fun330 | i can't get my debug to show the trunk name, it will only show the ip address, i did a pastebin of my sip.conf and debug |
| 15:06.50 | fun330 | http://pastebin.com/d8179b93 |
| 15:07.45 | Dovid | tzafrir_laptop: insect ? |
| 15:08.53 | tzafrir_laptop | bug |
| 15:09.08 | Dovid | tzafrir_laptop: Ok. In dahdi ? |
| 15:10.17 | Dovid | tzafrir_laptop: I am going through the full log. What word should I look for to find the lock ? i looked for lock but it came up nothhing |
| 15:10.54 | *** join/#asterisk jayrod422 (n=jayrod42@node2.164.136.64.1dial.com) |
| 15:11.03 | jad_jay | Is so know a good vu meter on gnome? |
| 15:11.08 | tzafrir_laptop | normally you won't see it |
| 15:11.15 | tzafrir_laptop | vu? |
| 15:11.28 | jad_jay | volume meter |
| 15:11.40 | Dovid | tzafrir_laptop: Is there any to have the log show it ? |
| 15:11.55 | tzafrir_laptop | aren't there a bunch of such panel applets? |
| 15:12.09 | tzafrir_laptop | Dovid, none that I can think of |
| 15:12.19 | tzafrir_laptop | How do you see that a lock is held? |
| 15:16.36 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 15:17.03 | Dovid | tzafrir_laptop: I don't cellcom see's it on their end. i didn't know abotu intesne debug which i just started so will see there. |
| 15:28.34 | *** join/#asterisk axisys (n=axisys@155.70.141.45) |
| 15:34.23 | *** join/#asterisk jameswf (n=james@unaffiliated/jameswf-home) |
| 15:37.39 | [TK]D-Fender | fun330: pastebin a call with SIP debug enabled |
| 15:38.59 | *** join/#asterisk askmeagain (n=ask@modemcable147.212-70-69.mc.videotron.ca) |
| 15:39.15 | *** join/#asterisk Deeewayne (n=dwayne@nat/digium/x-1ca33954aa476989) |
| 15:39.15 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 15:41.10 | *** join/#asterisk mmlj4 (n=jkelly@ip70-171-94-246.no.no.cox.net) |
| 15:41.16 | askmeagain | has anyone had a cisco gateway with FXO work with a sip trunk to asterisk ? |
| 15:45.15 | *** join/#asterisk SuPrSluG (n=SuPrSluG@firewall-a.buf.ny.i-evolve.net) |
| 15:45.20 | *** join/#asterisk cjk (n=cjk@vodsl-11080.vo.lu) |
| 15:48.28 | *** join/#asterisk moy (n=chatzill@74.12.124.158) |
| 15:50.53 | *** join/#asterisk ingenius (n=alektro@host143.200-117-156.telecom.net.ar) |
| 15:51.32 | *** join/#asterisk arpu (n=arpu@chello062178159144.10.14.univie.teleweb.at) |
| 15:55.52 | *** join/#asterisk lenne_dk (n=Leif@cpe.atm2-0-74391.hknxx4.customer.tele.dk) |
| 15:58.02 | lenne_dk | Hi. Had asteriks running, but server went strange. Installed on a fresh debian server, Asterisk 1.4.21.2~dfsg-3, but the extensions are not loaded. |
| 15:58.32 | lenne_dk | extensions reload command is missing |
| 15:58.53 | *** join/#asterisk zeeesh (n=zeeesh@203.215.179.43) |
| 15:59.06 | [TK]D-Fender | lenne_dk: "module reload pbx_config.so" |
| 15:59.51 | UQlev | lenne_dk: dialplan reload |
| 16:00.12 | UQlev | lenne_dk: instead of extensions reload |
| 16:00.18 | lenne_dk | <PROTECTED> |
| 16:00.18 | lenne_dk | graham2*CLI> module reload pbx_config.so |
| 16:00.18 | lenne_dk | [Feb 18 16:59:49] NOTICE[2605]: loader.c:590 ast_module_reload: The module 'pbx_config.so' was not properly initialized. Before reloading the module, you must run "module load pbx_config.so" and fix whatever is preventing the module from being initialized. |
| 16:00.49 | lenne_dk | No such command 'dialplan reload' |
| 16:01.27 | *** join/#asterisk russellb (n=russellb@asterisk/digium-open-source-team-lead/russellb) |
| 16:01.27 | *** mode/#asterisk [+o russellb] by ChanServ |
| 16:02.03 | UQlev | lenne_dk: type "help" |
| 16:03.15 | lenne_dk | Setting debug to 999: Parsing '/etc/asterisk/extensions.conf': Not found (Permission denied) |
| 16:03.26 | lenne_dk | I'll cgange that. |
| 16:05.02 | lenne_dk | That helped. I'd think such a fatal error should show up even at 0 debug... |
| 16:06.15 | *** join/#asterisk AppleBoy (n=AppleBoy@about/cooking/nakedchef/apple/tarts) |
| 16:06.21 | AppleBoy | file: you there? |
| 16:06.46 | file | yes |
| 16:06.56 | AppleBoy | how's it going? |
| 16:07.28 | file | good |
| 16:07.33 | file | but busy. |
| 16:07.49 | AppleBoy | ok, well I just noticed you fix the problem, so nevermind :) |
| 16:08.11 | Katty | hello all you beautiful people! |
| 16:08.18 | Katty | it's a gorgeous day outside! |
| 16:08.25 | file | AppleBoy: SSL cert? |
| 16:08.31 | AppleBoy | yep |
| 16:08.37 | file | yeah, fixed that this morning |
| 16:08.43 | AppleBoy | :) |
| 16:09.51 | *** join/#asterisk TrentCreek (i=kvirc@ppp-70-243-235-47.dsl.hrlntx.swbell.net) |
| 16:10.03 | AppleBoy | file, you're the best :) |
| 16:10.50 | TrentCreek | is a ROLM CBX II 9000 anygood? |
| 16:11.29 | Katty | snobs. |
| 16:11.31 | Katty | fine. |
| 16:11.33 | Katty | don't say hi. |
| 16:11.38 | Katty | SEE IF I CARE |
| 16:11.41 | beek | Hi Katty |
| 16:11.46 | Katty | cries. |
| 16:11.56 | eppigy | there there |
| 16:12.17 | beek | looks out window at dreary, rainy and cold environ and wonders where Katty's "gorgeous day" is. |
| 16:12.32 | rene- | it is quite nice a day here |
| 16:12.43 | *** join/#asterisk manxpower (n=Administ@router.asteriasgi.com) |
| 16:12.54 | rene- | it is windy, but not cold just perfect |
| 16:13.14 | Katty | hai manx |
| 16:13.31 | *** join/#asterisk c0rnoTa (n=c0rnoTa@80.251.113.56) |
| 16:14.00 | eppigy | i am exhausted |
| 16:14.58 | *** join/#asterisk Firass-z0r (n=asadf@c-67-201-205-34.reshall.wwu.edu) |
| 16:16.53 | *** join/#asterisk c0rnoTa (n=c0rnoTa@80.251.113.56) |
| 16:18.56 | *** join/#asterisk tobias (n=tobias@cpe-069-134-127-101.nc.res.rr.com) |
| 16:19.20 | *** join/#asterisk cody72 (n=nanook72@ip-64-202-118-8.7ticks.com) |
| 16:20.10 | cody72 | morning all, what is the best way to get asterisk to send voicemail emails to an IMAP server? |
| 16:20.24 | cody72 | we just changed from MS Exchange to Zimbra |
| 16:20.37 | *** join/#asterisk smelgin (i=user@adsl190-28-143-33.epm.net.co) |
| 16:20.43 | smelgin | hi everybody |
| 16:21.16 | smelgin | I have an Asterisk 1.4.22-rc5 with 1 FXO, and when I try to do a call, the system says "All lines are busy now" but I know the PSTN line isn't busy |
| 16:21.28 | smelgin | I've tried to restart zap (zap restart and dahdi restart) |
| 16:21.36 | smelgin | I trying restarting the machine |
| 16:21.49 | smelgin | I've tried restarting asterisk, wanrouter and zaptel |
| 16:21.53 | smelgin | and nothings happen |
| 16:21.57 | smelgin | what 's up ? |
| 16:22.05 | smelgin | and How can I solve the problem ? |
| 16:22.47 | freckle | smelgin: what does the CLI show when you dial in? |
| 16:22.59 | lenne_dk | cody: Is there a problem sending the voicemail as emails? |
| 16:23.26 | smelgin | <freckle> transcribe to you the text here? |
| 16:23.27 | cody72 | yeah it use to work till I changed to the new imap server |
| 16:23.49 | freckle | smelgin: use pastebin |
| 16:23.55 | cody72 | its not sending emails anylonger |
| 16:24.46 | cody72 | lenne_dk: I get this in the maillog version=TLSv1/SSLv3, verify=FAIL, cipher=DHE-RSA-AES256-SHA, bits=256/256 |
| 16:24.51 | [TK]D-Fender | ~pb |
| 16:24.53 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste, or , http://bin.cakephp.org/ |
| 16:24.53 | [TK]D-Fender | ^^^^^^^^^^^^^^ |
| 16:25.05 | freckle | cody72: IMAP is not a protocol for sneding emails, it is a SMTP issue by the sound of it, |
| 16:25.05 | lenne_dk | You don't send to an imap server, you send to a smtp-server, which then delivers to the imap. |
| 16:25.19 | lenne_dk | Can you increase the log on the smtp? |
| 16:25.46 | smelgin | <frenkle> misteriously has been resolved yet ... |
| 16:26.08 | smelgin | I don't understand what's happen |
| 16:26.14 | smelgin | but is resolved yet |
| 16:26.22 | cody72 | lenne_dk: the maillog is full of info, it knows who the mail relay server is |
| 16:26.33 | freckle | cody72: looks like the SMTP server wants to use TLS/SSL and the handshake is failing |
| 16:26.36 | smelgin | I'd like to understand what 's up |
| 16:27.02 | cody72 | freckle: can I give it a master account to login as? |
| 16:27.18 | [TK]D-Fender | smelgin: pastebin your complete failed call attempt's CLI output. |
| 16:27.26 | freckle | there is no account to login with SMTP |
| 16:27.52 | cody72 | freckle: to send emails from my server you have have to authenticate |
| 16:28.00 | *** join/#asterisk c0rnoTa (n=c0rnoTa@80.251.113.56) |
| 16:28.04 | freckle | ok |
| 16:28.16 | c0rnoTa | ツ |
| 16:28.18 | Deeewayne | waves to Katty |
| 16:28.28 | freckle | SMTP AUTH |
| 16:28.33 | lenne_dk | Can you authorize the asterisk's ip on the smtp? |
| 16:28.35 | *** part/#asterisk c0rnoTa (n=c0rnoTa@80.251.113.56) |
| 16:28.36 | cody72 | I was going to integrate imap storage into asterisk just don't have the time right now |
| 16:28.46 | cody72 | lenne_dk: did that |
| 16:29.00 | cody72 | freckle: where do Iput the SMTP AUTH? |
| 16:29.20 | freckle | you allow on the receiving server |
| 16:29.53 | lenne_dk | cody72: appearently not :-) It looks like a sendmail log, too long time I messed with that. |
| 16:30.22 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
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| 16:45.22 | ricko73 | is there an elegant way to do the following with analog outbound calling? |
| 16:46.14 | ricko73 | caller dials a three digit ext, that calls a phone number then sends a 3 digit DTMF to the remote system? |
| 16:46.47 | ricko73 | Using multiple w's is probably not very robust/reliable |
| 16:49.49 | smelgin | Thx D-Fender |
| 16:49.58 | smelgin | (for pastebin) |
| 16:50.34 | smelgin | but misteriously the problem has been resolved (I cannot put the log on this screen 'cause I have an elastix distro and the log is too large to analyze ...) |
| 16:57.14 | cesau | does specifying "callerid" in sip.conf not transfer to an itsp if "fromuser" is specified? |
| 16:57.21 | jer | anyone know of a way to use chan_mobile without bluetooth? or any other way i can connect a cell phone to my asterisk server via usb and make/receive calls and send/receive text messages (have a nokia e51 which is reported to work for all 4 of those items) |
| 16:57.56 | rene- | moy: hello |
| 16:58.32 | *** join/#asterisk erth64net (n=erth64ne@69-30-67-191.dq1sn.easystreet.com) |
| 16:59.03 | fun330 | i can't get my debug to show the trunk name, it will only show the ip address, i did a pastebin of my sip.conf and debug |
| 16:59.09 | fun330 | http://pastebin.com/d8179b93 |
| 17:02.54 | thehar | russellb |
| 17:02.57 | thehar | don't leave |
| 17:03.54 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 17:06.50 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 17:09.20 | *** join/#asterisk brutuz (n=brutuz@ip67-88-58-242.z58-88-67.customer.algx.net) |
| 17:10.37 | [TK]D-Fender | fun330: and I asked to see the COMPLETE call attempt with SIP debug enabled hours ago |
| 17:10.51 | *** part/#asterisk Mog (n=mog@c-68-62-170-242.hsd1.al.comcast.net) |
| 17:11.15 | [TK]D-Fender | cesau: typically no, but I might if you do "sendrpid=yes", and "trustrpid=yes" |
| 17:11.57 | *** join/#asterisk russellb (n=russellb@asterisk/digium-open-source-team-lead/russellb) |
| 17:11.57 | *** mode/#asterisk [+o russellb] by ChanServ |
| 17:12.52 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 17:13.12 | thehar | russellb: |
| 17:13.25 | *** join/#asterisk supjigator (n=shanebur@64.147.216.10) |
| 17:13.33 | supjigator | Hi. |
| 17:14.15 | brutuz | is there a CBT for asterisk? |
| 17:15.27 | Qwell | CBT? |
| 17:15.36 | *** join/#asterisk dmz (n=dmz@64.203.203.232.dyn-cm-pool-64.hargray.net) |
| 17:16.20 | E-bola | wonders why there arent any debian packages for 1.6 |
| 17:16.51 | Qwell | E-bola: same reason Etch still uses kernel 1.4 |
| 17:16.56 | Qwell | 2.4* |
| 17:17.01 | *** join/#asterisk CunningPike (n=arodgers@204.239.10.119) |
| 17:17.12 | *** join/#asterisk bminish (n=bminish@rb.brenbox.home.minish.org) |
| 17:17.14 | E-bola | etch doesnt use 2.4? |
| 17:17.15 | Gido-E | :-) E-bola see it as an Debian Feature |
| 17:17.54 | E-bola | well its fair enough its not in etch, as it pretty new |
| 17:18.00 | E-bola | but its not in unstable or testing either |
| 17:18.22 | Qwell | E-bola: there are 1.6 packages in AsteriskNOW :D |
| 17:18.50 | E-bola | hehe |
| 17:19.09 | E-bola | i dont really need 1.6, i'd just like to get it started since it would allow me to play with hooking up asterisk to exchange 2007 |
| 17:19.16 | E-bola | since 1.6 supports sip over tcp |
| 17:19.38 | *** join/#asterisk Mog (n=mog@c-68-62-170-242.hsd1.al.comcast.net) |
| 17:19.39 | *** mode/#asterisk [+o Mog] by ChanServ |
| 17:23.54 | *** join/#asterisk qdk (n=qdk@0x573c22aa.bynqu2.dynamic.dsl.tele.dk) |
| 17:25.31 | fun330 | [TK]D-Fender>: sorry must have scrolled by on me, this is for an inbound call what do you want to see the inbound call from start to finish? |
| 17:26.15 | LemensTS | Is there anyway to detect answering machines and voicemails with asterisk? im calling numbers from an agi script and was wanting it not to do something until the voicemail started... |
| 17:26.18 | [TK]D-Fender | fun330: I've already asked twice... should be rather clear |
| 17:27.48 | fun330 | okay hold on |
| 17:31.05 | kippi | anyone know where i can find a spa942 guide on how to set up LDAP? |
| 17:32.25 | Kobaz | hmmmm |
| 17:32.40 | Kobaz | i'm getting error 0x0, trying to boot the latest polycom firmware on a 320 |
| 17:34.42 | [TK]D-Fender | kippi: http://grimsy.blogspot.com/search/label/SPA942 |
| 17:35.07 | kippi | [TK]D-Fender: Thank you |
| 17:35.23 | *** part/#asterisk fcois93 (n=Administ@bagnolet.acropolistelecom.net) |
| 17:36.53 | *** join/#asterisk PTorres (n=PTorres@200.68.87.146) |
| 17:40.03 | *** join/#asterisk Whitor (n=Whitor@rrcs-24-97-4-146.nys.biz.rr.com) |
| 17:42.44 | brutuz | QWELL: computer based training |
| 17:43.27 | Whitor | Howdy all. Just wanted to say thanks for supporting such a great project |
| 17:43.32 | thehar | indeed |
| 17:43.50 | *** join/#asterisk HermesNeto (i=HermesNe@200.249.176.44) |
| 17:44.47 | brutuz | hi does anyone know of a site that will teach basic on how things work? |
| 17:45.02 | brutuz | i know the use of pbx.. |
| 17:45.09 | Whitor | google :) |
| 17:45.31 | brutuz | but what if you wanted to start a company.. what do you need from a telco |
| 17:45.40 | *** join/#asterisk otherwiseguy (n=otherwis@cpe-72-128-38-41.kc.res.rr.com) |
| 17:45.47 | cesau | is there a place where callerid needs to be set other than on the sip peer config in this example? http://pastebin.com/d50aed4a1 |
| 17:45.52 | Whitor | do you have a specific question ? |
| 17:46.21 | brutuz | assuming you need 5 lines.. how do you terminate it in your PBX |
| 17:47.04 | Whitor | brutuz, you can use a multiport FXO card like a digium wildcard TDM400p |
| 17:47.09 | brutuz | if im not mistaken each phone number will be 2 wires.. |
| 17:47.10 | Whitor | that will give oyu 4 |
| 17:47.23 | *** part/#asterisk PTorres (n=PTorres@200.68.87.146) |
| 17:47.43 | Whitor | then each line is essentially the same thing as a phone line into a regular handset ... |
| 17:48.00 | Whitor | if oyu needed more lines, I might suggest going with a t1 / pri setup |
| 17:48.10 | Whitor | or a fractional t |
| 17:48.27 | Qwell | brutuz: what? |
| 17:49.11 | brutuz | Whitor: how many lines/number can be assigned on a t1? |
| 17:49.24 | Whitor | 24 or 23 i don't remember |
| 17:49.26 | Whitor | t |
| 17:51.13 | brutuz | that t1 line will be terminated directly on the card attached to asterisk? |
| 17:51.33 | brutuz | or do i still need an appliance in between? |
| 17:52.01 | Whitor | you need something that can handle being a csu/dsu |
| 17:52.20 | Whitor | usually thats built right into the card you interface to yur t1 with |
| 17:53.04 | Whitor | Correct me if I'm wrong here please. I'm a relative newb too |
| 17:53.54 | brutuz | im assuming that from telco wire -- modem -- and then wire -- card (w/c is attached to asterisk ) |
| 17:54.23 | kaldemar | you don't need any modem in between |
| 17:54.34 | Whitor | ditto kaldemar |
| 17:55.35 | kaldemar | for example one of these will do for PRI: http://www.digium.com/en/products/digital/ |
| 17:56.22 | kaldemar | brutuz: have you read the book, the future of telephony? it offers a great amount of basic telephony knowledge in addition to being a great book on asterisk. |
| 17:56.36 | kaldemar | ~book |
| 17:56.36 | jbot | somebody said book was probably Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://tfot.leifmadsen.com or see ~buybook |
| 17:57.51 | *** join/#asterisk neurosys (n=vinix@sheltercorp.net) |
| 17:58.17 | *** join/#asterisk robwafle (n=robert@c-67-175-214-162.hsd1.il.comcast.net) |
| 17:58.26 | robwafle | hello all! |
| 17:59.19 | robwafle | does anyone knwo what the error "bearercapability notauth" means? |
| 18:01.35 | robwafle | anyone here? |
| 18:01.40 | Whitor | looks like you got unautherized bears in your config |
| 18:01.53 | robwafle | lol |
| 18:02.02 | Whitor | ok... idk really |
| 18:02.16 | Whitor | but I'm a newb so don't rely on me :) |
| 18:02.56 | russellb | Whitor, lol, nice |
| 18:03.07 | kaldemar | polar(ity) bears. you might want to tell some more about the setup, where you got that and what was happening when it occurred. |
| 18:04.01 | robwafle | i have a working asterisk box I've had for a year now |
| 18:04.15 | robwafle | and, I'm trying to attach via SIP instead of IAX today |
| 18:04.26 | robwafle | i added a new extension, and I can register with SIP |
| 18:04.36 | robwafle | but, when I try to make the phone call, I get this error on the client |
| 18:05.02 | robwafle | all on my local network |
| 18:05.09 | *** join/#asterisk brunner (n=chris@66.35.172.123) |
| 18:05.21 | robwafle | thanks kaldemar, hope that helps give you an idea |
| 18:05.21 | kaldemar | pastebin a SIP debug of a failed call |
| 18:05.42 | Whitor | kaldemar, nice pun |
| 18:06.32 | brunner | how many simultaneous ulaw SIP channels should I expect my 2GHz P4 to handle? |
| 18:06.36 | robwafle | pastebin a SIP debug of a failed call ? |
| 18:07.23 | brunner | or better yet, how does one go about stress testing an asterisk server? |
| 18:07.33 | brunner | I mean, how can I create 30 calls to test it with? |
| 18:08.00 | thehar | sipp? |
| 18:08.06 | brunner | yeah |
| 18:08.12 | thehar | no |
| 18:08.17 | kaldemar | go to asterisks cli (asterisk -r) set verbosity to a high level (core set verbose 10), enable SIP debug (sip set debug on) and then make a call, saving all output. |
| 18:08.18 | thehar | the application is called sipp |
| 18:08.21 | brunner | oh, SIPp |
| 18:08.22 | kaldemar | ~pb |
| 18:08.23 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste, or , http://bin.cakephp.org/ |
| 18:08.36 | brunner | nice, thanks |
| 18:10.12 | robwafle | thanks kaldemar! I'll try that. |
| 18:11.43 | robwafle | ok, so it says username mismatch on the CLI now.. let me see if I can fix that |
| 18:12.29 | robwafle | sweeeeet |
| 18:12.32 | robwafle | well, the problem was |
| 18:12.52 | robwafle | I am using Zoiper, and I had registered three ways |
| 18:12.55 | robwafle | one time via IAX2 |
| 18:13.00 | robwafle | and twice on SIP |
| 18:13.20 | robwafle | I deleted one of my SIP accounts, re-registered and i can make calls! |
| 18:13.46 | robwafle | so, nowi should .. set verbose 3, sip set debug off and call it a day! |
| 18:13.48 | robwafle | woo hoo! |
| 18:14.11 | robwafle | that probably won't be the case, but my wife is yelling at me to eat lunch |
| 18:14.27 | robwafle | i hope you'll be here in a minute because now I'm going to try to attach a sip device from peru |
| 18:14.31 | robwafle | wish me luck :) |
| 18:14.43 | kaldemar | good luck, with everything. :) |
| 18:16.13 | *** join/#asterisk mellow-yellow (n=mellow-y@mycomp.norris-stevens.com) |
| 18:18.15 | *** join/#asterisk muiro (n=muiro@cpe-173-89-177-15.neo.res.rr.com) |
| 18:18.23 | *** join/#asterisk lilkid (n=chatzill@87-194-38-230.bethere.co.uk) |
| 18:18.35 | muiro | Can I run more than one asterisk installation on a box? |
| 18:19.07 | muiro | I know I'd have to give each installation a sepparate install directory, to avoid socket and .pid problems |
| 18:19.20 | muiro | but can I bind an asterisk instance to a certain port or address? |
| 18:20.33 | lilkid | Is there a good relatively non-invasive billing system for * ? tried a2billing but it didnt feel very flexible.. |
| 18:20.47 | lilkid | with the complex dialplans i already have |
| 18:22.57 | jameswf | muiro: why do you need more than 1 ? |
| 18:23.55 | Shaun2222 | how can you dial a user in the users.conf... do you still have to specify dial(sip/) or dial(iax2/) or can you do dial(users/) and it find the channel they are on? |
| 18:26.35 | muiro | jameswf: multi-client |
| 18:27.17 | fun330 | d-fender: you still here? |
| 18:27.34 | jameswf | muiro: Contexts... one asterisk install can handle multiple offices |
| 18:27.42 | thehar | anyone suggest any cdr analyzers other than asterisk-stat? |
| 18:28.46 | muiro | jameswf: what about setting up of queues and users? |
| 18:29.04 | mintee | so.... |
| 18:29.13 | [TK]D-Fender | Shaun2222: still SIP/ ,etc |
| 18:29.14 | mintee | i just figured out some fax machines are retarded |
| 18:29.17 | [TK]D-Fender | Shaun2222: and.... |
| 18:29.20 | [TK]D-Fender | ~users.conf |
| 18:29.21 | jbot | users.conf is a flaming pile of sh1t that takes the fine control of several perfectly usable * config files and reduces them to the lowest common denominator and makes your system behave like a "toaster grade" PBX system. |
| 18:29.39 | mintee | I've always setup fax tone detection with a 3 second wait window before going to voice |
| 18:29.58 | [TK]D-Fender | fun330: Yes |
| 18:29.58 | mintee | and a client can't fax to me cause hismachine isn't sending the tone quick enough |
| 18:30.33 | brutuz | i was wondering why would someone use a channel bank equipment? |
| 18:30.57 | lilkid | [TK]D-Fender: By any chance, did you write the script for jbot? |
| 18:31.20 | [TK]D-Fender | lilkid: No, that'd be Tim Riker |
| 18:31.30 | [TK]D-Fender | lilkid: Actually no... he's jsut HOSTING it. |
| 18:31.34 | Whitor | mintee, don't you love standards ? |
| 18:31.36 | lilkid | Ah ok |
| 18:31.40 | [TK]D-Fender | lilkid: Don't recall the author |
| 18:31.44 | mintee | Whitor: there are so many of them ;) |
| 18:31.50 | [TK]D-Fender | lilkid: But its MY bitch :) |
| 18:32.04 | Whitor | and few are adhered to strictly :) |
| 18:32.16 | mintee | i bumped the wait time up to 5 seconds... waiting for him to resend to test |
| 18:32.17 | brutuz | and what is an inexpensive T1? |
| 18:32.24 | lilkid | [TK]D-Fender: Thanks anyway, and haha yeah i'll give you that |
| 18:33.07 | [TK]D-Fender | brutuz: One that that's cheaper than an expensive T1 |
| 18:33.15 | brutuz | anyone has idea what is an inexpensive T1? |
| 18:33.34 | mintee | < $450 |
| 18:33.46 | jameswf | muiro: everything is relitive, allot of the "Hosted PBX" services use asterisk |
| 18:33.58 | brutuz | D-fender: something a bit more intellectual |
| 18:34.02 | brutuz | :) |
| 18:34.20 | fun330 | [TK]D-Fender>: http://pastebin.com/d19304fb0 sorry took so long i had to wait for lunch break |
| 18:34.29 | jameswf | brutuz: you can get a bailout T1 from whitehouse.gov |
| 18:34.37 | jameswf | okay maybe not |
| 18:34.43 | icebrew54 | jameswf: you can get a bailout sense of humor from www.idiot.edu |
| 18:34.43 | [TK]D-Fender | brutuz: What is a T1? its SIGNALLING. typically electrical impules are somewhat FREE. |
| 18:34.52 | Whitor | jameswf, heh |
| 18:35.09 | [TK]D-Fender | brutuz: Perhaps you should be a LOT more specifiy about what kind of HARDWARE or SERVICE you talking about. |
| 18:35.13 | jameswf | icebrew54: financial aid availible? |
| 18:35.41 | Whitor | brutuz, yeah, t1,s are free... its the lines you have to rent from the telco |
| 18:35.59 | *** join/#asterisk lesouvage (n=lesouvag@82.73.69.76) |
| 18:35.59 | brutuz | i know what a T1 is.. |
| 18:36.02 | muiro | jameswf: so basically multiple clients would have to share from the same pool of extension numbers. And say I was using the live database configuration way, they'd have to share a database? |
| 18:36.24 | jameswf | If A$$ holes could fly this place would be an airport.... oh crap who invited the TSA |
| 18:36.27 | [TK]D-Fender | brutuz: Yes, you just lack the ability to communicate your needs |
| 18:36.29 | brutuz | i saw this card that they are using te410p |
| 18:36.55 | jameswf | muiro: NO contexts the others wouldn't even know about eachother |
| 18:37.02 | [TK]D-Fender | brutuz: "they"? Who is "they"? |
| 18:37.24 | jameswf | [TK]D-Fender: they=the man |
| 18:37.25 | lesouvage | jameswf: I like our phrase |
| 18:37.32 | lesouvage | ou=your |
| 18:37.41 | brutuz | i got confused on inexpensive T1? im assuming there is an expensive T1... |
| 18:38.14 | [TK]D-Fender | brutuz: You seem to not understand. Are you looking for a piece of HARDWARE, or T1 TELEPHONE SERVICE <- |
| 18:38.26 | mintee | brutuz: obviously a LEC will charge as much as a client will pay for a line |
| 18:38.37 | brutuz | http://www.digium.com/en/products/digital/te410p.php |
| 18:39.08 | [TK]D-Fender | brutuz: You are failing to properly state your needs. T1 is not a PRODUCT, it is a PROTOCOL. |
| 18:39.21 | jameswf | [TK]D-Fender: are you a twitter whore ? |
| 18:39.29 | mintee | brutuz: should ask here http://www.digium.com/en/company/contact.php |
| 18:39.52 | [TK]D-Fender | jameswf: No, I'm a big fan of BURNING social networking. |
| 18:40.21 | mintee | man, fuck this |
| 18:40.26 | jameswf | [TK]D-Fender: anti-social networking didn't work out to well |
| 18:40.28 | mintee | err, damnit, wrong channel again |
| 18:42.00 | brutuz | okie i got it... |
| 18:42.07 | [TK]D-Fender | jameswf: Facebook almost went draconian on everyone in as much as the knife is still in their hands as well. Trust = 0. Twitter? Do I seriously want to hear the random shit going through your typical retard's head? |
| 18:42.13 | brutuz | :) |
| 18:43.41 | jameswf | [TK]D-Fender: sometimes when I am bored i watct the /b/tards on 4chan :) |
| 18:44.32 | [TK]D-Fender | jameswf: I leave my "catch-up time" on the week's episodes of Cobert Report & The Daily Show for those times. |
| 18:44.47 | lesouvage | [TL]D-Fender: you should start a movement |
| 18:46.02 | lesouvage | asnm=anti social network movement |
| 18:47.31 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 18:49.01 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 18:49.07 | [TK]D-Fender | lesouvage: I get plenty of fibre... no problems with movements. |
| 18:49.17 | Whitor | I'd like to create a network of people who are against being a part of networks |
| 18:50.33 | LemensTS | No one knows how to place a call to an analog phone/cell phone and wait for it to hit voicemail to play a message? The part that i dont kno is how to get it to wait until the voicemail does the beep to start playing the message. |
| 18:51.08 | LemensTS | Should I use the followme app? Never used it, but heard it may help do this |
| 18:51.41 | outtolunc | think of it as non-silence followed by silence (vice where did the 'beep' go) |
| 18:53.05 | LemensTS | outolunc: Can I detect silence and non silence in agi? |
| 18:53.32 | outtolunc | in an agi, if you use something that uses a dsp like amd |
| 18:54.49 | *** join/#asterisk ingenius (n=alektro@host143.200-117-156.telecom.net.ar) |
| 18:55.39 | LemensTS | outtolunc: awesome didnt know about AMD. Ill read up on it thanks |
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| 19:09.17 | *** join/#asterisk manxpower (n=Administ@router.asteriasgi.com) |
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| 19:13.10 | *** join/#asterisk alexoc (n=alexoc@gate07.vulcanet.com.br) |
| 19:14.14 | alexoc | What is the best billing system for asterisk? A2billing, Astercc or Other? |
| 19:16.03 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 19:16.03 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 19:17.15 | *** join/#asterisk path_ (n=path@180-98-21-190.adsl.terra.cl) |
| 19:17.18 | lmadsen | hey all. If I wanted to get the value of GROUP_COUNT() from a script (perl, php, etc...), how do you think a good method would be to do that? I'm thinking possibly the Getvar in manager, but that might not return values of dialplan functions. I'm also wondering if an Originate might be useful... |
| 19:22.01 | manxpower | lmadsen: Set(MY_GROUP_COUNT=${GROUP_COUNT} Then just access it as a dialplan variable in your AGI |
| 19:22.33 | lmadsen | there is no AGI -- there is a script that connects to the manager |
| 19:22.42 | manxpower | lmadsen: It sucks to be you. 8-| |
| 19:22.48 | lmadsen | useful |
| 19:23.06 | manxpower | lmadsen: I suspect Manager is your best bet. |
| 19:23.20 | lmadsen | isn't that what I mentioned in my question? |
| 19:23.44 | [TK]D-Fender | lmadsen: I would make a small exten to take 2-3 vars for which group to count, and a key to search by, store the result in AstDB, and then you can poll/erase the key from AMI following |
| 19:23.49 | manxpower | lmadsen: *nod* Since there is no AGI, your way is the only way I can think of. |
| 19:23.55 | carrar | $blah = $AGI->get_variable("GROUP_COUNT"); |
| 19:24.03 | [TK]D-Fender | lmadsen: And yes.. Originate |
| 19:24.12 | lmadsen | [TK]D-Fender: thanks, sounds like I was on the right track |
| 19:24.20 | [TK]D-Fender | carrar: what part of "not AGI" did you miss? :) |
| 19:24.22 | lmadsen | carrar: again... I'm not using AGI |
| 19:24.26 | carrar | the 1st part |
| 19:24.33 | carrar | heh |
| 19:24.36 | [TK]D-Fender | ~cluebat carrar |
| 19:24.37 | jbot | ACTION pulls out a ClueBat (tm) and thwaps carrar. |
| 19:24.42 | lmadsen | ~!!! |
| 19:24.43 | jbot | well, !!! is BANG BANG BANG at http://www.starterupsteve.com/swf/Group_X_video.html |
| 19:24.43 | [TK]D-Fender | ClueBat(tm) NEVER MISSES!!!! |
| 19:28.55 | eppigy | hello |
| 19:28.57 | eppigy | i am dave |
| 19:29.08 | AlexOC | What is the best free billing system for asterisk? A2billing, Astercc or Other? |
| 19:29.21 | manxpower | I'm sorry, Dave. I can't do that. |
| 19:29.26 | manxpower | AlexOC: they all suck |
| 19:29.48 | *** join/#asterisk freckle_home (n=chatzill@84.45.168.57) |
| 19:30.06 | *** join/#asterisk af_ (n=getsmart@88-149-230-21.dynamic.ngi.it) |
| 19:30.30 | AlexOC | manxpower: What can I do so? Any suggestion? |
| 19:31.11 | manxpower | AlexOC: Most companies seem to write their own. Billing solutions are usually custom applications to meet the specific needs of the company. |
| 19:33.21 | AlexOC | it is true... but I need one solution with common features.. I just need to know, How is the minus worst free asterisk billing.. |
| 19:34.10 | *** join/#asterisk freckle_home (n=chatzill@84.45.168.57) |
| 19:34.48 | [TK]D-Fender | AlexOC: A2billing works and is maintained. |
| 19:36.16 | AlexOC | [TK]D-Fender: thanks |
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| 19:44.33 | lmadsen | [TK]D-Fender: thanks -- got it solved now |
| 19:44.42 | [TK]D-Fender | lmadsen: Cool. |
| 19:45.55 | lmadsen | basically did what you said -- 3 line dialplan with Local channel that does Set(DB(Usage/Total)=${GROUP_COUNT(total_calls)}), then used Originate and DBget from the manager. |
| 19:46.32 | [TK]D-Fender | lmadsen: Depending if you need to watch out for concurrency you could apss Originate a unique key for the DB value. Perhaps your case now doesn't require it. |
| 19:46.39 | *** join/#asterisk deadkode (n=Deadkode@208-58-59-94.c3-0.drf-ubr1.atw-drf.pa.cable.rcn.com) |
| 19:48.45 | deadkode | does asterisk 1.6.1-rc1 support loose routing of NOTIFY messages (for mwi subscriptions)? |
| 19:52.53 | lmadsen | [TK]D-Fender: possibly not, but I will mention that to the person I made it for |
| 19:53.30 | lmadsen | basically the script is a dialer, so it will be checking that before it originates a call, so most likely not |
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| 20:03.38 | brunner | how would I go about passing PRI channels through an asterisk box that has two T1 ports to my existing phone system? |
| 20:06.07 | *** join/#asterisk Arsenick- (n=rpurcell@modemcable026.33-70-69.static.videotron.ca) |
| 20:08.37 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 20:09.09 | *** join/#asterisk Linuturk (n=linuturk@fluxbuntu/developer/Linuturk) |
| 20:10.30 | [TK]D-Fender | brunner: "core show application dial" |
| 20:11.26 | brunner | I don't even own the PRI card yet. I'm asking for online resources. |
| 20:11.37 | [TK]D-Fender | brunner: "core show application dial" <- * CLI. |
| 20:12.24 | *** join/#asterisk regius (n=regius@c-ee6ee455.034-67-6c6b701.cust.bredbandsbolaget.se) |
| 20:12.56 | icebrew54 | anyone do 30fps video w/asterisk? |
| 20:12.57 | *** join/#asterisk pikachu2000 (n=pikachu2@196-209-199-37-rbry-bb-1.dynamic.isadsl.co.za) |
| 20:13.48 | regius | hi, where is asterisk trying to find the configfiles when running osx (installed from ports?). Is there any command to show the configifiles that are loaded |
| 20:13.58 | *** part/#asterisk pikachu2000 (n=pikachu2@196-209-199-37-rbry-bb-1.dynamic.isadsl.co.za) |
| 20:14.28 | Shaun2222 | drawing a blank whats the _ stand for on a exten => again? |
| 20:14.39 | [TK]D-Fender | Shaun2222: that its a PATTERN |
| 20:14.58 | Shaun2222 | so it |
| 20:15.09 | Shaun2222 | so it's required with things like _9X., |
| 20:15.40 | [TK]D-Fender | Shaun2222: Yes |
| 20:17.36 | brunner | [TK]D-Fender: how is that relevant to my question? |
| 20:17.51 | *** join/#asterisk path_ (n=path@180-98-21-190.adsl.terra.cl) |
| 20:18.37 | [TK]D-Fender | bruBecause every single call in * is just a call. Call comes in one port, it hits the DIALPLAN jsut like absolutely every other call. You want to send it to a PBX on another port? DIALIT, just like you would anything else. |
| 20:18.44 | [TK]D-Fender | brunner: Because every single call in * is just a call. Call comes in one port, it hits the DIALPLAN jsut like absolutely every other call. You want to send it to a PBX on another port? DIALIT, just like you would anything else. |
| 20:19.08 | [TK]D-Fender | brunner: There is no such thing as "pass through" anything in Asterisk. |
| 20:19.19 | brunner | [TK]D-Fender: okay, thank you |
| 20:19.30 | [TK]D-Fender | brunner: It is not a proxy or some invisible things between devices that think they are directly connected |
| 20:19.45 | Shaun2222 | [TK]D-Fender: is there a better way to get a extension number, for example to make the voicemail button work on my phones, i do the following 'GotoIf($["${CALLERID(num)}" != "${EXTEN}"]?4)' and the next exten after than is voicemailmain(${EXTEN}@default)... |
| 20:19.56 | Shaun2222 | but callerid num could be spoofed by a outside caller to match that exten |
| 20:21.10 | Shaun2222 | basically matching if the callerid is the same as the extension go into voicemailmain... the polycom phones when you hit messages->connect dial your own extension. |
| 20:21.21 | [TK]D-Fender | Shaun2222: Well "work" is grey. You could simply not pass it the box and have it prompt. If the call is coming from a SIP peer then you could have a var set in the Peer itself that will be accessible to the dialplan. |
| 20:21.33 | [TK]D-Fender | Shaun2222: You cuold look at the CHANNEL NAME as well, etc. |
| 20:22.03 | [TK]D-Fender | Shaun2222: and Polycom's dial what you tell them to when you hit the messages button |
| 20:22.20 | Shaun2222 | hmm, do they default to them self? |
| 20:22.25 | Shaun2222 | maybe i set it, haha |
| 20:22.33 | Shaun2222 | been awhile. |
| 20:27.03 | *** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose) |
| 20:28.06 | *** join/#asterisk goodjoke (i=1827a8fa@gateway/web/ajax/mibbit.com/x-52b1ba00091148b6) |
| 20:38.46 | icebrew54 | any suggestions for ekiga and decent audio quality? |
| 20:39.01 | icebrew54 | using gsm now and want to upgrade....any suggestions on a default ekiga install? |
| 20:39.11 | *** join/#asterisk ingenius (n=alektro@111-197-235-201.fibertel.com.ar) |
| 20:39.45 | [TK]D-Fender | icebrew54: G.711 |
| 20:40.10 | icebrew54 | ulaw? |
| 20:41.11 | [TK]D-Fender | icebrew54: Sure |
| 20:41.22 | icebrew54 | heh it's kinda crappy sounding |
| 20:41.37 | [TK]D-Fender | icebrew54: Whats on the OTHER END? |
| 20:41.38 | icebrew54 | ok, will try others...that's what the default was ;\ |
| 20:51.20 | *** join/#asterisk obnauticus (n=obnautic@about/windows/regular/obnauticus) |
| 20:57.28 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.static.bezeqint.net) |
| 20:59.39 | cesau | if i specify "callerid=" in sip.conf, does that caller id only apply for incomming calls? |
| 21:00.23 | thehar | only Set(${CALLERID(num)}) will only be passed out of your network |
| 21:02.33 | cesau | and if im calling out using originate sip/provider/number extension 2@blah -- i wouldnt have an opportunity to do so right? would a better approach be to originate local/number@context and have a dial command in the extensions after the set command? |
| 21:03.46 | *** join/#asterisk rbd (n=rbd@rrcs-96-10-27-206.se.biz.rr.com) |
| 21:04.24 | *** join/#asterisk lesouvage (n=lesouvag@82.73.69.76) |
| 21:06.45 | lesouvage | . |
| 21:08.06 | LeddyHM | what are ways that calls can be made if they are not made from a sip registration? |
| 21:08.34 | LeddyHM | Our log shows calls are going out/being made but I can't trace where they are originating |
| 21:08.45 | LeddyHM | no sip registrations, and no manager connections |
| 21:10.34 | *** join/#asterisk brunner1 (n=chris@66.35.172.123) |
| 21:10.43 | brunner1 | how can I tell what codec a channel is using? |
| 21:13.53 | *** join/#asterisk echinos (n=echinos@H211.C136.B196.A67.tor.heavycomputing.ca) |
| 21:16.16 | [TK]D-Fender | brunner1: "sip show channels" |
| 21:18.02 | LeddyHM | what about me tk? :) |
| 21:19.00 | jaytee | "sip show whiskey" |
| 21:19.08 | jaytee | "sip show bourbon" |
| 21:19.15 | jaytee | hmmm, must be out of booze |
| 21:19.27 | LeddyHM | I could use some of that |
| 21:20.26 | CrashSys | "sip show floor" |
| 21:20.33 | CrashSys | There it is |
| 21:22.10 | *** join/#asterisk joobie (n=joobie@mx01.anric.com.au) |
| 21:22.11 | LeddyHM | What are ways calls can be originated in asterisk? |
| 21:22.23 | LeddyHM | endpoint sip device and? |
| 21:22.51 | Corydon76-dig | LeddyHM: callfile, manager, cli |
| 21:23.43 | LeddyHM | trying to see if call files exist now but I got disco'd |
| 21:23.55 | CrashSys | Anyone ever had issues with getting X-Lite or Zoiper to work from a MAC to asterisk? |
| 21:26.56 | *** join/#asterisk bminish (n=bminish@rb.brenbox.home.minish.org) |
| 21:28.00 | LeddyHM | is there a command to see if any users/who is connected to the cli? |
| 21:28.11 | af_ | why ilbc directory is empty? |
| 21:31.38 | af_ | understood |
| 21:33.09 | *** join/#asterisk djbelieny (n=djbelien@72-48-61-165.dyn.grandenetworks.net) |
| 21:33.38 | Mog | licensing issues |
| 21:33.44 | Mog | there is readme about it |
| 21:33.48 | af_ | seen, thanks |
| 21:35.03 | djbelieny | i am writing a patch to app_mixmonitor.c which adds courtesy tone to the mixmonitor |
| 21:35.33 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) |
| 21:35.34 | djbelieny | can anyone help me with the best way to play a beep into a channel or audiohook ? |
| 21:35.56 | *** join/#asterisk umpc (n=Justin@unaffiliated/umpc) |
| 21:38.51 | *** join/#asterisk djbelieny (i=48303da5@gateway/web/ajax/mibbit.com/x-657351c9164de231) |
| 21:39.45 | djbelieny | Hey ALL: what's the best way to inject a beep into an audiohook, bridge or channel ??? |
| 21:39.48 | *** join/#asterisk mags2 (n=mags2@ampulex.whoi.edu) |
| 21:40.29 | djbelieny | ast_streamfile() becomes garbled since i am trying to do this from a mixmonitor thread... |
| 21:40.37 | djbelieny | any way to do this ? |
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| 21:53.47 | *** part/#asterisk djbelieny (i=48303da5@gateway/web/ajax/mibbit.com/x-657351c9164de231) |
| 21:55.58 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194) |
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| 22:04.47 | *** part/#asterisk mags2 (n=mags2@ampulex.whoi.edu) |
| 22:06.36 | icebrew54 | low call volume, how to troubleshoot? |
| 22:07.59 | eppigy | generate more calls? |
| 22:08.02 | eppigy | ZING |
| 22:08.51 | beek | icebrew54: http://tinyurl.com/cjy9el |
| 22:09.18 | [TK]D-Fender | Have problem, how do I fixa?! PLAESE!??!?! :$ |
| 22:09.38 | icebrew54 | I guess I'm looking at what ordering...I asked provider and they have no pointers |
| 22:11.08 | LemensTS | TK: try #freepbx |
| 22:11.16 | icebrew54 | beek: that didn't provide virtually any results |
| 22:11.40 | *** join/#asterisk path_ (n=path@180-98-21-190.adsl.terra.cl) |
| 22:11.46 | *** join/#asterisk n3hxs (n=HAMming@static-151-196-93-200.balt.east.verizon.net) |
| 22:11.55 | [TK]D-Fender | icebrew54: And you didn't provide any details, thus there is symmetry. |
| 22:12.16 | icebrew54 | low volume on SIP...PSTN is fine |
| 22:12.25 | icebrew54 | googling for adjusting SIP trunk volume and not finding much results |
| 22:12.55 | *** join/#asterisk bminish (n=bminish@rb.brenbox.home.minish.org) |
| 22:13.49 | [TK]D-Fender | icebrew54: DETAILS. Precisely what is on EACH end of the call? |
| 22:14.04 | Kobaz | yes give us all the juicy details |
| 22:15.49 | icebrew54 | ok, I tried with two different SIP clients....soft phone and PAP2....other end of that is SIP provider |
| 22:16.27 | icebrew54 | people calling into said SIP provider....then hitting our asterisk box |
| 22:16.31 | [TK]D-Fender | icebrew54: NAMES, try using NAMES. You successfully mentioned ONE (PAP2) |
| 22:16.34 | icebrew54 | so I we tried 2 different phones with same results |
| 22:16.54 | icebrew54 | pap2/ekiga to test if it was our phone, both same results |
| 22:19.35 | [TK]D-Fender | icethen your PROVIDER sucks |
| 22:19.56 | *** join/#asterisk justdave (n=dave@unaffiliated/justdave) |
| 22:20.33 | *** join/#asterisk _pepo_ (n=pepo@200.55.224.2) |
| 22:20.37 | _pepo_ | hi friends |
| 22:21.09 | _pepo_ | Is there some way to use postgresql without ODBC in Asterisk? |
| 22:21.21 | eppigy | say what |
| 22:23.30 | _pepo_ | sorry |
| 22:23.33 | _pepo_ | I want to use Postgres for my voicemail in Asterisk,so Is there some way to use Postgresql without odbc? |
| 22:25.45 | Corydon76-dig | Nope |
| 22:25.54 | _pepo_ | :( |
| 22:28.28 | [TK]D-Fender | So setup ODBC, whats teh big deal? |
| 22:30.15 | Kobaz | is there a way to reset a polycom to factory defaults without booting up and loading sip.ld |
| 22:31.31 | _pepo_ | I think that if use direct connection to pgsql it will be more efficient, like with mysql |
| 22:32.06 | Kobaz | if you're worried about efficiency, don't use a database at all and use the config files |
| 22:33.18 | _pepo_ | I am using the database to store sound files... the configuration is not problem |
| 22:33.42 | [TK]D-Fender | Kobaz: yes, and its very easy to find in the admin guide ... |
| 22:34.10 | Kobaz | [TK]D-Fender: apparently it's holding down buttons 1,3,5,7 |
| 22:34.20 | Kobaz | [TK]D-Fender: but that only works when the application is loaded on the phone |
| 22:34.47 | [TK]D-Fender | Kobaz: And it ISN'T loaded? |
| 22:35.02 | Kobaz | nope |
| 22:35.14 | Kobaz | it wont load, which is why i want to reset |
| 22:35.31 | Kobaz | error 0x0, loading.. and it won't download the firmware |
| 22:36.30 | Kobaz | i've spent more time futzing with configuring phones than any other item with asterisk |
| 22:37.36 | _pepo_ | maybe I can: I want to use Postgres for my voicemail in Asterisk,so Is there some way to use Postgresql without odbc? |
| 22:37.37 | Kobaz | polycom specifically |
| 22:37.38 | Kobaz | heh |
| 22:37.53 | cesau | When I change my caller id via: Set(CALLERID(all)="Us" <2165551212> -- it gets sent out as From: "Us" <sip:cgi71@172.18.55.20> -- any ideas why the number is getting overwritten? |
| 22:37.53 | _pepo_ | maybe I can: http://www.voip-info.org/wiki/view/Asterisk+RealTime+PostgreSQL |
| 22:37.56 | Kobaz | aastra/grandstream phones are way less delicate |
| 22:38.06 | Qwell | _pepo_: voicemail doesn't use realtime. |
| 22:38.13 | Kobaz | cesau: sip? |
| 22:38.16 | cesau | yeah |
| 22:38.25 | Kobaz | cesau: are you using trustrpid and sendrpid |
| 22:38.40 | cesau | not that im aware of =) |
| 22:38.55 | Kobaz | turn on sendrpid, and it will send the callerid |
| 22:38.58 | Qwell | _pepo_: and no, the efficiency gains you may see are trivial at best |
| 22:38.59 | _pepo_ | ok sorry |
| 22:39.10 | Kobaz | Qwell: voicemail uses realtime |
| 22:39.17 | Qwell | Kobaz: Wanna bet? |
| 22:39.21 | cesau | kobaz: awesome, thank you |
| 22:39.22 | _pepo_ | :) |
| 22:39.30 | *** join/#asterisk Doug52392 (n=Doug@c-66-31-177-196.hsd1.ma.comcast.net) |
| 22:39.36 | Qwell | Kobaz: You're telling the wrong guy he's wrong. :) |
| 22:39.50 | Kobaz | Qwell: if i update something in my voicemail table, it's immediatly available without a reload |
| 22:40.09 | Kobaz | if might not use 'realtime' specifically, but it is 'real time' |
| 22:40.23 | *** join/#asterisk bminish (n=bminish@rb.brenbox.home.minish.org) |
| 22:40.27 | Qwell | "use" != "is" |
| 22:41.59 | cesau | kobaz: it worked, thank you again! |
| 22:42.42 | *** join/#asterisk axarob (n=ebash@cpc3-barn8-0-0-cust288.brnt.cable.ntl.com) |
| 22:44.19 | Kobaz | np |
| 22:44.59 | *** join/#asterisk [gquit]bombadil (n=dana@CPE-70-94-44-157.wi.res.rr.com) |
| 22:47.41 | Doug52392 | Would it be possible to use a regular modem (as in, not a FXS/FXO card) to connect an Asterisk PBX to the PSTN? |
| 22:47.52 | [TK]D-Fender | Doug52392: No |
| 22:48.10 | manxpower | Doug52392: and the answer to your next question is also no |
| 22:48.12 | *** join/#asterisk killown (n=Yamato@unaffiliated/killown) |
| 22:48.47 | Doug52392 | What is the next question? |
| 22:50.00 | hardwire | any way to detect "RING" as the caller through a CAS/FXO T1 line over a wildcard? |
| 22:51.30 | *** join/#asterisk mazpe (n=mazpe@c-71-196-32-43.hsd1.fl.comcast.net) |
| 22:51.44 | mazpe | how do i can configure asterisk to Allow Anonymous Inbound SIP Calls? |
| 22:52.41 | manxpower | mazpe: that happens by default. |
| 22:52.50 | manxpower | they will land in whatever context= is in [general] |
| 22:53.56 | mazpe | manxpower: i need to change the setting to YES.. but for some reason FreePBX is not saving any changes made in General Settings |
| 22:54.04 | mazpe | i'm wondering if i can do this by hand |
| 22:55.11 | manxpower | mazpe: I know nothing about FreePBX. |
| 22:55.56 | mazpe | can it be done directly on asterisk? |
| 22:56.24 | [TK]D-Fender | mazpe: you've already been told YES. |
| 22:56.40 | mazpe | by default, i see. |
| 22:56.45 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 22:57.50 | *** join/#asterisk axisys (n=axisys@ip68-98-177-71.dc.dc.cox.net) |
| 22:59.25 | mazpe | so if by default asterisk allows anonymous inboud sip calls.. whats the parameter to disable it? |
| 22:59.53 | stabler | is there anyway to force x-lite to use port 5060 |
| 23:00.14 | stabler | or should that kind of thing be done on * side |
| 23:00.45 | Qwell | drmessano: guess what |
| 23:02.10 | [TK]D-Fender | mazpe: allowguest=no under [general] |
| 23:02.57 | *** join/#asterisk cp5 (n=samy@cpe-76-171-169-53.socal.res.rr.com) |
| 23:03.03 | cp5 | yo |
| 23:04.10 | *** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose) |
| 23:06.13 | cp5 | in asterisk 1.2, i'm running into an issue where sip channels don't close. the RTP ports seem to be left open until i run out of open file handles |
| 23:06.31 | cp5 | sip show channels will show thousands of open "Tx: ACK" messages to various phones (after many calls) and i have no way to close them |
| 23:06.45 | cp5 | any way to set a timeout on those? i'm not really sure why it's happening in the first place |
| 23:07.23 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 23:08.59 | mazpe | [TK]D-Fender thanks |
| 23:10.03 | *** part/#asterisk Doug52392 (n=Doug@c-66-31-177-196.hsd1.ma.comcast.net) |
| 23:10.07 | *** part/#asterisk Mog (n=mog@c-68-62-170-242.hsd1.al.comcast.net) |
| 23:10.19 | [TK]D-Fender | cp5: What version? |
| 23:10.29 | cp5 | [TK]D-Fender: 1.2.17 |
| 23:10.36 | [TK]D-Fender | cp5: upgrade |
| 23:11.27 | cp5 | [TK]D-Fender: i'd like to confirm the problem is fixed in an upgraded version, though i can't really test on this production system. i would test on another system, but the weird thing is i can't seem to reproduce the issue on another test system |
| 23:11.59 | [TK]D-Fender | cp5: Your test sample was also on 1.2.17? |
| 23:12.02 | cp5 | if i can reproduce it on another server (if there was a known issue with more information), then that would be great since i'd be able to reproduce, then test a newer version of asterisk |
| 23:12.06 | cp5 | [TK]D-Fender: yeah |
| 23:12.29 | [TK]D-Fender | cp5: 1.2 is not supported anymore, and you are on an OLD version of that series as well. |
| 23:12.40 | cp5 | [TK]D-Fender: kind of strange that i can't reproduce. using the same exact configs. only difference is i'm using SIP in test instead of ZAP, and different phones and network. can't take that production server down for testing |
| 23:14.39 | Qwell | heh, 1.2.17 was one of the only releases that I made |
| 23:15.14 | cp5 | nice |
| 23:15.34 | *** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim) |
| 23:20.00 | [TK]D-Fender | remembers.... and then reactivates the chan_skinny.so botnet! |
| 23:23.24 | *** join/#asterisk jeremy_g (n=a@c-1e94e755.021-183-73746f3.cust.bredbandsbolaget.se) |
| 23:25.15 | *** join/#asterisk SirThomas (n=tomc@mail.kendeco.com) |
| 23:27.27 | jeremy_g | ? |
| 23:27.35 | `g0rt | ! |
| 23:28.07 | jeremy_g | are there any know rtp syncrhonization problems with latest asterisk 1.6 |
| 23:30.58 | *** join/#asterisk k-man (n=jason@unaffiliated/k-man) |
| 23:31.12 | k-man | how do i connect to the asterisk process to monitor it? i forget |
| 23:32.52 | jeremy_g | asterisk -rc |
| 23:33.52 | *** join/#asterisk BeerSerc (n=dries@unaffiliated/beerserc) |
| 23:34.29 | *** join/#asterisk path_ (n=path@pc-15-190-86-200.cm.vtr.net) |
| 23:37.21 | manxpower | Asterisk -r (reconnect) -v (one or more verbose levels) |
| 23:37.29 | manxpower | -c is ignored when used with -r |
| 23:37.39 | manxpower | since -c says start it in the foreground, don't reconnect |
| 23:44.09 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 23:45.19 | *** join/#asterisk jm|home (n=jm|home@dilbert.jamiem.com) |
| 23:46.06 | jm|home | hello |
| 23:46.06 | jm|home | I have configured my Cisco 7912 for chan_sccp[2] but I have a 5-10 sec delay when I dial other [sip] phones connected to the Asterisk box |
| 23:46.06 | jm|home | what can I do to reduce that? |
| 23:47.42 | RypPn | jm|home edit sccp.conf and change digittimeout = to something more suitable? |
| 23:48.40 | manxpower | jm|home: I believe SCCP doesn't have a local dialplan (unlike SIP) You have ambiguous extensions in your dialplan. Example: exten => 1234,1,Whatever exten => 1NXXNXXXXXX,1,Whatever |
| 23:49.01 | *** join/#asterisk Bonix (n=Bonix@200-195-41-212.isimples.com.br) |
| 23:49.01 | manxpower | when you dial 1234 how does Asterisk know you are not dialing 12345551212? |
| 23:49.30 | jm|home | RypPn: manxpower: between you thank you |
| 23:49.43 | jm|home | manxpower: the 7912 lets you dial on-hook then select dial |
| 23:50.04 | RypPn | same as the 7940 and 7970 :) |
| 23:50.06 | Whitor | I believe there is a timout after each number input |
| 23:50.06 | jm|home | but like RypPn said, if I off-hook then dial and use digittimeout=1 it's being sensible |
| 23:50.12 | manxpower | jm|home: and you STILL get a delay? |
| 23:50.30 | jm|home | not now I've set digittimeout as per RypPn |
| 23:50.39 | RypPn | I use 3 btw |
| 23:50.47 | jm|home | is it seconds? |
| 23:51.00 | RypPn | doesnt feel like it |
| 23:51.09 | jm|home | it does here |
| 23:51.42 | RypPn | yeah, it is, just tried it, considering the problems with Playback thats nothing |
| 23:51.43 | jm|home | I pretty much never call off-hook though |
| 23:51.47 | jm|home | so I can edit away and then Dial |
| 23:51.57 | jm|home | RypPn: Playback probs? |
| 23:52.07 | RypPn | thats another gotcha with chan_sccp, dont use skip or noanswer on the end of Plackback |
| 23:52.14 | RypPn | Playback* |
| 23:52.19 | jm|home | hmm |
| 23:52.25 | jm|home | not sure that's in my dialplan anyway |
| 23:52.39 | jm|home | I did just have to convert all my gsm prompts to ulaw though :)( |
| 23:53.46 | RypPn | there are a few instances of it in the freepbx dialplan that needed carving up, but thats not for consumption in here really :) |
| 23:54.19 | manxpower | points to the sign on his wall. WARNING: THIS IS A TRIXBOX-FREE ZONE |
| 23:54.39 | LemensTS | is it more efficient to look up asm channel variables or to do a mysql query? i can send the variables over thru the scrip then look them up via asm, or i can just do a mysql query... |
| 23:54.40 | RypPn | who mentioned trixbox, this is handbuilt on a gentoo platform |
| 23:54.48 | jm|home | how does sccp handle NAT? |
| 23:54.59 | manxpower | trixbox, freepbx, they are all the same |
| 23:55.05 | RypPn | theres another viariable in sccp.conf to define the ip's |
| 23:55.18 | drmessano | heh |
| 23:55.27 | manxpower | jm|home: I didn't think it handled it at all, but I heard a rumor of at least one company using SCCP for consumers and most comsumers are NAT'd |
| 23:55.41 | jm|home | zoiks |
| 23:55.47 | Corydon76-dig | LemensTS: asm? |
| 23:56.12 | jm|home | manxpower: I'll try when I'm next on the remote site |
| 23:56.15 | LemensTS | asm->GetVar (phpagi) |
| 23:56.26 | Corydon76-dig | Oh, AGI |
| 23:56.27 | LemensTS | ami i think |
| 23:56.36 | manxpower | jm|home: there would have to be NAT support in the channel driver |
| 23:56.48 | RypPn | jm|home the variables are localnet= and externip= or externhost= |
| 23:56.58 | jm|home | RypPn: thanks :) |
| 23:57.06 | jm|home | if only there was a IAX firmware |
| 23:57.15 | Corydon76-dig | LemensTS: You're concerned about pure efficiency? |
| 23:57.22 | LemensTS | @Corydon: just have several hundred calls going out at once and wanted to be efficentent as possible. |
| 23:57.35 | *** part/#asterisk Chuggs (n=Chuggs@s142-179-186-158.ab.hsia.telus.net) |
| 23:57.59 | Corydon76-dig | Depends upon how many variables you have in the channel. Normally, AGI is more efficient, but the DB could be, if you have a ton of variables |
| 23:58.13 | LemensTS | Like 5 |
| 23:58.18 | Corydon76-dig | AGI, then |
| 23:58.31 | Corydon76-dig | I'm talking if you have on the order of 200 variables |
| 23:58.57 | LemensTS | Not to that point yet :) Thanks |
| 23:59.43 | LemensTS | BTW: $asm->GetVar($res,var2); you know how get more than one variable at a time thru it? |
