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| 00:44.51 | root52 | Ok, after working with HANGUPCAUSE I see what I think is my problem. Asterisk does not "hangup" when it recives a 404 from my SIP provider. Insted it waits until the SIP provider sends a SIP 408 request timeout causing HANGUPCAUSE to have 18. Any clue on how I can get asterisk to "hangup" when it recives the first 404 packet from the SIP provider? |
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| 00:47.34 | Kyosh | GotoIf and ChannelStatus |
| 00:47.38 | Kyosh | something like that |
| 00:50.52 | Kyosh | http://www.voip-info.org/wiki/view/Asterisk+variable+DIALSTATUS |
| 00:50.57 | Kyosh | not sure tho |
| 00:51.37 | *** join/#asterisk iluminator101 (~iluminato@unaffiliated/iluminator101) |
| 00:51.53 | iluminator101 | i am trying to setup asterisk with skype |
| 00:52.27 | iluminator101 | i have the digium skypeforasterisk, but i am running into few issues |
| 00:52.49 | Kyosh | root52: here you go, http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial |
| 00:53.35 | Kyosh | and 404 is a message, not a packet root52 |
| 00:54.24 | root52 | Kyosh: good point ;-) |
| 00:55.20 | Kyosh | but sadly after the Dial, i think it's going to wait for the 408 or 480 |
| 00:55.37 | Kyosh | what is the 404 message? whats it defined as? |
| 00:55.48 | root52 | not found |
| 00:56.06 | root52 | as in "that is not a real number that this server can find" |
| 00:56.21 | Kyosh | hmmm |
| 00:56.45 | root52 | I get like 3 of thoes before I get the 408 timeout and that is when the dial cmd exits |
| 00:57.23 | Kyosh | http://www.voip-info.org/wiki/view/Asterisk+variable+DIALSTATUS |
| 00:57.28 | Kyosh | its gotta match one of them |
| 00:57.35 | root52 | but by the time I get the 408 timeout HANGUPCAUSE is set to 18 (timeout) and DIALSTATUS is set to congesstiong. |
| 00:57.46 | Kyosh | but i wouldnt know how to hook the DIALSTATUS after the DIAL |
| 00:57.54 | Kyosh | hmm |
| 00:58.05 | root52 | it does, problem I don't want to wait for the 408 I want dial to exit after the first 404 |
| 00:58.11 | Kyosh | like creating an capturing a 404 event during the call, would be nice |
| 00:58.36 | root52 | haha yes time to put on my non-exsient developer hat |
| 00:59.04 | ChannelZ | iluminator101: like what |
| 00:59.15 | *** join/#asterisk hipitihop (~denis@203.132.229.236) |
| 00:59.27 | iluminator101 | is sip alias sip.skype.com |
| 00:59.35 | ChannelZ | huh? |
| 01:00.06 | ChannelZ | Skype For Asterisk is different than SIP for Skype |
| 01:00.45 | ChannelZ | (or Skype for SIP I guess it's called) |
| 01:01.35 | iluminator101 | so what sip alias would i use |
| 01:02.07 | iluminator101 | i setup up the skypeforasterisk trunk as digium instructed me to do, i provisioned the phone |
| 01:02.24 | iluminator101 | i reads failed on the phone |
| 01:02.52 | ChannelZ | I think you have two separate problems |
| 01:03.17 | ChannelZ | You can't get your SIP phone to register with Asterisk? |
| 01:05.10 | iluminator101 | i deleted all the vertical line codes |
| 01:06.17 | ChannelZ | that really has nothing to do with anything but ok.. |
| 01:14.15 | Kyosh | root52: why would you need to develop? i would hope its something you can handle in a dialplan |
| 01:14.22 | Kyosh | 'i would hope' |
| 01:14.51 | iluminator101 | freepbx error i am getting is Could not reload FOP server |
| 01:15.23 | Kyosh | FOP=FreeOuttaPee |
| 01:15.31 | Kyosh | or FreshOuttaPee |
| 01:23.02 | carrar | Oh man that sucks |
| 01:23.42 | jaytee | I don't use FOP, I'm a Dapper Dan man |
| 01:23.46 | carrar | How can we in this freebpx unrelated channel help? |
| 01:24.23 | Kyosh | oh i know i know |
| 01:24.26 | Kyosh | we cant! |
| 01:24.27 | Kyosh | :) |
| 01:24.55 | carrar | Dapper Dan still come in a round can? |
| 01:25.03 | jaytee | yep |
| 01:25.09 | carrar | thats HOT++ |
| 01:25.24 | carrar | PICS!!! of the Dapper Dan hair |
| 01:25.47 | jaytee | I don't really have a can :-) |
| 01:25.58 | carrar | WHAT |
| 01:26.00 | jaytee | and I shave my hed |
| 01:26.06 | jaytee | s/hed/head |
| 01:26.34 | carrar | You shaved a @ symbol on your head? |
| 01:26.43 | carrar | or * |
| 01:28.31 | jaytee | no, just shave my head down to 1/8" |
| 01:28.43 | jaytee | no patterns or symbols |
| 01:28.56 | carrar | How creative is that! |
| 01:28.58 | root52 | Kyosh: still thinking about it. I think the key to what I want is for asterisk to term the dial cmd after the first 404 message is recived |
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| 01:34.22 | *** part/#asterisk iluminator101 (~iluminato@unaffiliated/iluminator101) |
| 01:39.25 | Kyosh | yea |
| 01:39.37 | Kyosh | but how could asterisk listen for an event during a dial/call? |
| 01:40.34 | Kyosh | http://www.voip-info.org/wiki/view/Asterisk+variable+DIALSTATUS |
| 01:40.44 | Kyosh | There is also no way to look at a SIP response in the dialplan, even though it is reported on the debug console (if debug level >3). (Even though you can see every other SIP header with ${SIP_HEADER(<header_name>) you cannot see the actual response code.) |
| 01:40.44 | Kyosh | There should really be a FAIL result so that it is possible to distinguish between "CONGESTION" and a genuine call setup failure, but there isn't. |
| 01:40.49 | Kyosh | there we go |
| 01:40.55 | Kyosh | u aint the first with this problem |
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| 02:02.08 | manxpower | Kyosh: HANGUPCAUSE is more or less Q.931 cause codes. |
| 02:02.50 | manxpower | a 404 would (I think) be HANGUPCAUSE 1 |
| 02:03.08 | manxpower | There's an RFC defining what SIP responses map to which G.931 code |
| 02:03.11 | manxpower | Q.931, that is |
| 02:03.32 | manxpower | Use DIALRESULT only when you don't really care too much what happened to the call |
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| 03:05.36 | Kyosh | manxpower: wasnt my concern, i only helping root52. i also suggested looking into if asterisk has events that can be captured. but it doesnt seem that way. |
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| 03:22.59 | spenguin[work] | is this cheap or what https://www.voipbucket.com/shop/article_BT201/BudgeTone-201-SIP-Phone.html?sessid=GOnAd668Wgy4mkXZqx0RU9OwhEoKDY2pknZu9JtXOZWMCqoWP8lZpk73ZDpCbyJC&shop_param=cid%3D67%26aid%3DBT201%26 |
| 03:24.24 | Brookss | The Budget One |
| 03:25.13 | spenguin[work] | does it get any cheaper? |
| 03:25.21 | spenguin[work] | other than ebay |
| 03:25.33 | Brookss | Grandstream has a reputation for cheap/cheaply made phones so, idk... in the end it's how much you value your time vs grandstream issues |
| 03:26.05 | spenguin[work] | ok |
| 03:28.52 | jaytee | ~grandstream |
| 03:28.52 | infobot | i guess grandstream is the Yugo of VoIP hardware. Run. Run away now.. Though therealcircut says that they're not that bad |
| 03:29.55 | spenguin[work] | hrm k |
| 03:32.38 | [TK]D-Fender | ~gs |
| 03:32.39 | infobot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
| 03:36.57 | spenguin[work] | k |
| 03:44.12 | Naikrovek | polycom polycom polycom |
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| 03:56.27 | ChannelZ | That's one of the worst logos I've ever seen in my life |
| 04:05.46 | Naikrovek | what is |
| 04:05.52 | *** join/#asterisk joobie (~joobie@CPE-124-180-8-218.lns1.lon.bigpond.net.au) |
| 04:09.09 | boodu | bye |
| 04:09.26 | ChannelZ | voipbucket |
| 04:11.26 | joobie | got my monitoring system in realtime, showing the channel usage of DAHDI and 2 sip peers |
| 04:11.44 | joobie | kicks ass for keeping track of how many lines you're using and when peak is etc |
| 04:14.04 | TJNII | That logo is pretty atrocious. |
| 04:14.15 | Naikrovek | hah lol yes it is |
| 04:14.18 | Naikrovek | wow |
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| 04:45.13 | spenguin[work] | joobie: kewl |
| 04:45.18 | spenguin[work] | care to share how |
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| 04:53.39 | Kyosh | whats a good ata that supports T.38? |
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| 05:03.23 | Brookss | spa3102 |
| 05:03.37 | Brookss | Ive faxed over it |
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| 05:03.51 | Brookss | through asterisk/sip |
| 05:03.57 | Brookss | good quality |
| 05:04.50 | Kyosh | ulaw or t38? |
| 05:05.27 | Brookss | ulaw setting, but it uses t38 protocol on its own when doing fax |
| 05:05.40 | Brookss | so long as it's allowed on ata I mean |
| 05:05.51 | Kyosh | hmm |
| 05:05.55 | Brookss | heres what I also used http://www.future-nine.com/faq/index.php?action=artikel&cat=1&id=5&artlang=en |
| 05:06.04 | Kyosh | curious |
| 05:06.11 | Kyosh | i wouldnt think id try a sipura |
| 05:07.09 | Kyosh | that link doesnt show a device |
| 05:07.20 | Brookss | I think its just a config help |
| 05:07.41 | Kyosh | Can I use fax over your service? |
| 05:07.41 | Kyosh | The short answer is yes, you can. |
| 05:07.43 | Kyosh | thats it |
| 05:07.52 | Kyosh | ooo |
| 05:07.55 | Kyosh | gotta scroll way down |
| 05:09.24 | Brookss | :D |
| 05:09.57 | Kyosh | . Try to avoid faxing long documents. Try to fax pages in bunches of 2-3 at most to maximize chances of success. |
| 05:10.00 | Kyosh | dont like that one |
| 05:10.04 | Kyosh | ive done 10+ pages |
| 05:11.01 | Brookss | same, its jus a recommendation, but I've faxed more over it |
| 05:11.07 | Kyosh | k |
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| 05:21.46 | joobie | spenguin[work], via the AMI interface |
| 05:22.00 | joobie | spenguin[work], and a perl script that listens for the specific events.. which ties back to zabbix |
| 05:23.02 | joobie | i faxed over alaw |
| 05:23.04 | joobie | it was fine |
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| 05:33.13 | spenguin[work] | joobie: okay nice, I was thinking of playing around with some openflashchart |
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| 05:56.33 | joobie | spenguin[work], what's openflashchart? |
| 05:57.08 | joobie | spenguin[work], i wrote a flash app a few years ago which in realtime, plotted network usage per protocol on a flash line graph |
| 05:57.15 | joobie | it updated every half a second |
| 05:57.50 | joobie | it worked by listening for a tcp connection from the firewall, which had its own script to collect the data from iptables and then push it to the flash box |
| 05:57.53 | joobie | which then plotted |
| 05:58.07 | joobie | but hrm.. for asterisk, i was thinking a realtime view is probably not so important |
| 05:59.03 | joobie | rather if you integrate to monitoring system (such as zabbix or nagios), you can view historic trends and also send notifications if all lines are almost in use etc |
| 05:59.03 | joobie | benefit of this is you get the history trends which can help you when considering upgrades |
| 05:59.11 | joobie | BTW i used a perl module some other dood wrote to integrate to AMI |
| 05:59.18 | spenguin[work] | joobie: yeah realtime isnt really imp, its just to impress your bosses |
| 05:59.20 | joobie | it was pretty basic with this module |
| 05:59.24 | joobie | nod |
| 05:59.30 | joobie | realtime was important for that flash app i wrote |
| 05:59.43 | joobie | we had issues where people in the ofifce would start saying why is the internet so slow |
| 05:59.51 | joobie | with this app we could see very easily what was consuming the link |
| 06:00.56 | spenguin[work] | what flash app have you coded, joobie |
| 06:01.11 | spenguin[work] | I was hoping openflashchart would be more realtime |
| 06:01.24 | joobie | i wrote my own |
| 06:01.31 | joobie | was a few years ago.. |
| 06:01.53 | joobie | mine basically plotted a line graph.. each protocol had a line itself with its own colour |
| 06:01.54 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 06:02.03 | joobie | then i wrote the change every 0.5s |
| 06:02.06 | joobie | to a bitmap |
| 06:02.11 | joobie | shifted the bitmap left X pixels |
| 06:02.20 | joobie | then wrote to the end of the bitmap |
| 06:02.22 | joobie | shifted it again |
| 06:02.26 | joobie | and so forth |
| 06:02.44 | joobie | it gave the appearance of a scrolling graph and saved some i/o in using the bitmap |
| 06:03.23 | spenguin[work] | nice |
| 06:04.09 | joobie | i would use something like that spenguin[work] |
| 06:04.14 | joobie | just had a look at that site |
| 06:04.24 | joobie | the graphs dont look that crash hot |
| 06:04.49 | joobie | its very easy to do the above sorta thing in flash these days |
| 06:04.51 | joobie | bitmaps are native |
| 06:05.03 | joobie | and you can do some funky stuff |
| 06:05.15 | joobie | like i had sliding scales going for the y axis |
| 06:05.25 | joobie | so if the value was very high, it would dynamically scale the y axis |
| 06:05.31 | joobie | too low, go the other way |
| 06:05.42 | joobie | i duno if that openflash one you're looking at will be that fliexble |
| 06:05.51 | spenguin[work] | ofc2 has to read in the data, and then itll show the loading screen |
| 06:05.57 | spenguin[work] | totally beats the realtime idea |
| 06:06.05 | joobie | yer |
| 06:06.36 | joobie | literally man.. bitmap, reposition, write out what you want to the right, snapshot the bitmap |
| 06:06.43 | joobie | just do a loop like that and you can get it looking realtime |
| 06:08.29 | spenguin[work] | hrm |
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| 06:51.10 | J4zen | Goodmorning guys |
| 06:53.51 | J4zen | A client of mine has requested that id look into videophone solutions where he can start video-conference calls with at least three people. Ofcourse theres plenty of high-end expensive solutions from Tandberg and such, costing well over 3/4k $. I'm looking for something a tad smaller and less costly, it should be a bit larger than a traditional SIP-phone only with a larger display. Think along the lines of a Tandberg E20, but cheaper. Does anyone |
| 06:54.36 | J4zen | for reference, the tandberg E20: http://www.tandberg.com/personal-video-conferencing/video-voip-E20.jsp |
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| 07:07.36 | spenguin[work] | http://jcs.org/notaweblog/2010/04/11/properly_stopping_a_sip_flood |
| 07:07.45 | spenguin[work] | really cool, quick thinking |
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| 07:17.11 | petern_ | http://www.porticus.org/bell/images/1992videophone2500.jpg |
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| 07:50.12 | funtoo_nbu | is there a way to make musicon hold play a specific file every time, and then go to a random queue of files |
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| 07:57.09 | *** join/#asterisk voipnoob (~i@122.170.6.106) |
| 07:57.56 | voipnoob | Hi - I am doing an academic study on setting up a VOIP business at a high level |
| 07:58.06 | voipnoob | is this the right place to ask questions? |
| 07:58.29 | funtoo_nbu | probably |
| 07:58.46 | voipnoob | I am located in India |
| 07:59.15 | voipnoob | Assuming I have one or more asterisk servers setup, what are the next steps to setup a VOIP busines |
| 07:59.31 | voipnoob | i.e. how do I figure how many Asterisk Servers I need to have for 'n' number of users |
| 07:59.51 | voipnoob | and how do I go about getting SIP Trunking? |
| 08:00.04 | voipnoob | i am looking for all this info a high level |
| 08:00.58 | spenguin[work] | http://www.voip-info.org/wiki/view/Asterisk+dimensioning |
| 08:02.01 | spenguin[work] | that should help the first question |
| 08:02.09 | funtoo_nbu | spenguin[work]: ohh do me now plz :D |
| 08:02.53 | spenguin[work] | heh |
| 08:03.14 | spenguin[work] | if I have PRI signalling incorrectly set, would I recieve calls at all? |
| 08:03.23 | spenguin[work] | or theres only one way to find out? |
| 08:03.23 | spenguin[work] | :p |
| 08:03.49 | spenguin[work] | voipnoob: btw, Im from India too :) |
| 08:05.35 | funtoo_nbu | spenguin[work]: i want asterisk to play a greeting msg when placed on hold but then play a random file in the moh directory |
| 08:09.05 | spenguin[work] | well before hold, use playback |
| 08:09.24 | spenguin[work] | you can specify 'random' in the moh config |
| 08:14.36 | *** join/#asterisk JustERR (~justerr@78.108.73.46) |
| 08:14.54 | spenguin[work] | I get these errors in the logs, whenever theres a call comming in - "chan_dahdi.c:12581 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1" |
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| 08:15.14 | spenguin[work] | although the call is established sucessfully |
| 08:18.23 | JustERR | Hey guys, i've got a problem. My application is monitoring Asterisk events through AMI. The problem is that Asterisk drops my AMI client connection when the traffic load reaches about 4-5Mbit / 3-4 kpps. Does anyone have an idea on how to solve this? |
| 08:18.35 | funtoo_nbu | what do you mean play back? |
| 08:18.41 | funtoo_nbu | in extensions.conf? |
| 08:18.53 | funtoo_nbu | before the call is answered? |
| 08:21.03 | *** join/#asterisk raj-darkmystery (~test@114.143.184.114) |
| 08:21.12 | raj-darkmystery | hi Slugs_ |
| 08:22.39 | raj-darkmystery | sorry but someone accidently deleted the book you have provided me.. can you please send the link of the asterisk book |
| 08:24.44 | spenguin[work] | funtoo_nbu: nah, what I meant was you could have an exten that plays back the msg to the user before you have him on hold |
| 08:25.24 | funtoo_nbu | how? |
| 08:27.30 | spenguin[work] | funtoo_nbu: or better yet is you can transfer the call to an exten, that would playback the msg |
| 08:27.40 | spenguin[work] | then put the call on hold |
| 08:28.45 | funtoo_nbu | eep |
| 08:28.50 | funtoo_nbu | why so complex |
| 08:29.09 | spenguin[work] | not really complex |
| 08:29.30 | *** join/#asterisk maxagaz (~maxagaz@soho2.i-xanadu.com) |
| 08:30.35 | funtoo_nbu | k hold |
| 08:30.52 | funtoo_nbu | see i guess ive never looked into how asterisk even handles placing calls on hold |
| 08:32.32 | *** join/#asterisk sulex (~sulex@office.blindata.ch) |
| 08:34.14 | voipnoob | spenguin - tx |
| 08:34.20 | voipnoob | sorry got called away for a moment |
| 08:36.17 | *** join/#asterisk s4msung (~s4msung@dice.s4msung.de) |
| 08:36.41 | *** join/#asterisk raj-darkmystery (~test@114.143.184.114) |
| 08:37.46 | *** join/#asterisk raj-darkmystery (~test@114.143.184.114) |
| 08:37.57 | raj-darkmystery | hi Slugs_ you there? |
| 08:38.48 | voipnoob | spenguin - the link you pointed me to |
| 08:39.04 | voipnoob | seems to be more about deploying Asterisk on a LAN |
| 08:39.09 | voipnoob | I am wondering about WAN |
| 08:39.31 | voipnoob | where can i find Asterisk loads for WAN |
| 08:39.42 | spenguin[work] | you asked about how an asterisk box would scale across number of calls |
| 08:40.05 | voipnoob | yes, but that seems to be referring to calls inside a LAN |
| 08:40.16 | voipnoob | i.e. all the VOIP users are inside an intranet |
| 08:40.43 | voipnoob | i would assume that WAN requirements & load handling capabilities would be totally different |
| 08:40.50 | spenguin[work] | whats the difference? |
| 08:41.05 | spenguin[work] | with wan you are limited by your bandwidth |
| 08:41.22 | spenguin[work] | and itsp accounts |
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| 09:38.16 | tzafrir | http://www.acipia.fr/community/asterisk/asterisk-ejabberd-mod-client-asterisk/ - anybody tried it? |
| 09:48.05 | *** join/#asterisk pinoyskull (~pinoyskul@112.198.64.77) |
| 11:17.03 | *** join/#asterisk infobot (ibot@rikers.org) |
| 11:17.03 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.2.6, 1.6.1.18, 1.6.0.26 (2010/03/12), 1.4.30 (2010/03/12), *-Addons 1.6.2.1, 1.6.1.3, 1.6.0.5, 1.4.11 (2010/04/15), dahdi-linux 2.3.0 + dahdi-tools 2.3.0 (2010/04/13), Libpri 1.4.10.2 (2009/10/20) -=- Related channels: #asterisknow,#switchvox,#freepbx,#asterisk-dev,#asterisk-bugs |
| 11:18.59 | *** join/#asterisk neurosys (~neurosys@c-71-196-20-208.hsd1.fl.comcast.net) |
| 11:44.33 | *** join/#asterisk visz (vis@irkki.fi) |
| 11:44.36 | visz | hello |
| 11:44.54 | visz | is there a way to kill a zombie channel with status 'Ring' without restarting asterisk? |
| 11:45.29 | c0rnoTa | soft hangup |
| 11:45.36 | visz | yep, tried that |
| 11:47.14 | J4zen | I'm looking for a video-phone such as the Polycom VVX-1500, but it needs to support three-/fourway video-calls. Anyone know of such a device? |
| 11:51.45 | *** join/#asterisk slinksh0t (~slinksh0t@64.152.41.21) |
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| 11:57.07 | tzafrir | visz, sounds like a bug. Can you reproduce it? |
| 11:57.41 | *** join/#asterisk Tim_Toady (~moi@77.49.26.91.dsl.dyn.forthnet.gr) |
| 11:57.59 | tzafrir | petern_, can't ejabberd be also a gateway to MSN-messanger? |
| 12:02.09 | *** join/#asterisk rttrey (~trey@andc-office-fw.atlantic.net) |
| 12:03.17 | carrar | J4, get yourself a Videoconferencing MCU |
| 12:03.47 | *** join/#asterisk ccesario (~ccesario@189-19-6-236.dsl.telesp.net.br) |
| 12:03.50 | J4zen | like Tandberg solutions? |
| 12:04.04 | carrar | lots of MCU venders out there |
| 12:04.24 | J4zen | you don't know of any solutions where the phone/device provides native support for three-way calls? |
| 12:04.38 | J4zen | when you switch to MCU appliances the costs of such a solution skyrocket |
| 12:04.48 | J4zen | or am i mistaking? |
| 12:05.36 | carrar | maybe sipwitch? |
| 12:06.23 | carrar | no idea really |
| 12:16.31 | carrar | maybe http://www.gnugk.org/ |
| 12:19.18 | carrar | There is also Ekiga |
| 12:19.19 | *** join/#asterisk jpeeler (~jpeeler@asterisk/digium-software-dev/jpeeler) |
| 12:22.12 | J4zen | carrar: if i'm not mistaking those are only gateway/servers to 'support' three-way video conference calls. However you'd still need a device capable of initiating them, that's what im currently looking for |
| 12:22.29 | *** join/#asterisk [TK]D-Fender (~chatzilla@216.191.106.163) |
| 12:23.56 | visz | tzafrir, propably something to do with --without-dundi =P |
| 12:24.19 | tzafrir | visz, huh? |
| 12:24.46 | tzafrir | Why do you think this is related? |
| 12:26.36 | carrar | J4, try openmcu |
| 12:27.32 | carrar | Part of the h323plus project |
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| 12:34.39 | *** join/#asterisk LemensTS (~LemensTS@adsl-70-238-175-27.dsl.stlsmo.sbcglobal.net) |
| 12:34.53 | J4zen | very nice carrar, that might do the trick |
| 12:34.57 | LemensTS | how can you make a sip user reregister |
| 12:35.05 | LemensTS | besids sip reload |
| 12:37.44 | carrar | reboot the sip user (phone) |
| 12:37.54 | [TK]D-Fender | LemensTS: that doesn't make another client re-register |
| 12:38.08 | LemensTS | carrar: Tried that a few times, its not showing it reregistering |
| 12:38.15 | [TK]D-Fender | LemensTS: only thing that will is a decision on the client itself |
| 12:38.17 | Chainsaw | If it's a phone, you might be able to send it a sip notify. |
| 12:38.25 | LemensTS | SPA 2102 i think |
| 12:38.38 | [TK]D-Fender | LemensTS: If it isn't registering... then something is wrong. |
| 12:38.41 | petern_ | tzafrir, i guess. i never bothered figuring out how |
| 12:38.49 | [TK]D-Fender | LemensTS: bad networking or config on the client |
| 12:39.28 | LemensTS | TK: yea i had them reboot the router and the phone adapter several times. The SPA 2101 is registerd just fine. But they have no dial tone |
| 12:39.57 | LemensTS | I thought rebooting the SPA would make it unregister and re-register like it does when you reboot Xlite... |
| 12:40.03 | LemensTS | maybe it dont |
| 12:40.41 | *** join/#asterisk saisoma (~saisoma@client72.jdcc.edu) |
| 12:41.03 | Chainsaw | LemensTS: Not all SIP clients are that polite. On my doorbell, I had to tick a "unregister when rebooting" box. |
| 12:42.14 | petern_ | doorbell... |
| 12:43.49 | [TK]D-Fender | LemensTS: I've never heard of a device that wouldn't try if you pulled the power and reconnectede |
| 12:44.09 | [TK]D-Fender | "Unregister" = irrelevant |
| 12:44.28 | LemensTS | TK: Yea im sending them a new ATA I think the device is bad. |
| 12:44.29 | [TK]D-Fender | LemensTS: Maybe your side is the problem |
| 12:45.07 | LemensTS | TK: is there any way to flush registrations and make them reregister? |
| 12:48.44 | [TK]D-Fender | LemensTS: On the device? Doubt it. It will try on boot. |
| 12:48.56 | [TK]D-Fender | LemensTS: and probably on call/timeout |
| 12:51.19 | [TK]D-Fender | LemensTS: So you sat on CLI with SIP DEBUG while they rebooted the ATA? |
| 12:53.39 | LemensTS | TK: yea i just see options and 489's (489's because of 1.4 issue and linksys) |
| 12:54.02 | *** join/#asterisk af_ (~getsmart@88-149-240-255.dynamic.ngi.it) |
| 12:54.09 | J4zen | Silly question; a three-way call. Does that mean you can have a conversation with 2 other people (3 including yourself), or with 3 other people (4 including yourself)? |
| 12:55.01 | *** join/#asterisk muiro (~muiro@unaffiliated/muiro) |
| 12:55.12 | [TK]D-Fender | LemensTS: pastebin it all |
| 12:55.26 | [TK]D-Fender | J4zen: 3-way = 3 people |
| 12:55.39 | J4zen | in total. |
| 12:55.43 | [TK]D-Fender | J4zen: Yes |
| 12:55.54 | J4zen | Alright, Thanks [TK]D-Fender |
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| 12:56.16 | beek | mornin' [TK]D-Fender |
| 12:56.31 | [TK]D-Fender | beek: Mornin' |
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| 13:33.20 | cusco | hi... |
| 13:33.28 | Kyosh | i like to look at 3ways in a different way :) |
| 13:34.01 | Kyosh | hi cusco |
| 13:35.11 | cusco | I'm wondering... if we have two asterisk boxes placed in geographical different places, both have incoming pri lines.. now if calls come random trough pri-geo1 or pri-geo2, should we have a single queueing asterisk or both? or what I was really wonderibng was, having some dundi config that directs from queue to operator |
| 13:35.16 | cusco | hello Kyosh |
| 13:35.18 | *** join/#asterisk scottsmith7 (~ssmith@64.201.141.80) |
| 13:38.11 | Kyosh | cusco, you gave a scenario without telling us what you want to accomplish |
| 13:38.49 | Kyosh | do you want a single queue for both locations, shared across the 2 pbx's? because thats what it sounds like you want |
| 13:39.42 | Kyosh | just remember, each pbx sees itself as an individual and can pass calls between other pbx', but queue are independant of each pbx |
| 13:39.56 | Kyosh | but he's not even talking so i'll go lay down |
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| 13:43.44 | cusco | err |
| 13:43.45 | cusco | sorry |
| 13:44.03 | cusco | at first I thoguht about two different queues, both using the same realtime mysql information |
| 13:44.43 | cusco | that way I would get total redundancy.. if say, internet goes down at geo-1, geo-2 can still queue and has operators to work.. |
| 13:44.54 | cusco | it would also save some bandwidt |
| 13:45.30 | cusco | but then my boss thinks its best to have one queue only, he is afraid of implications of asterisk now knowing if some operators not queuing on this side are in a call from the other side's queue |
| 13:46.06 | cusco | so one queue only is doable (we have that right now, if peer is x, y or z, the dial goes trough a IAX2) |
| 13:46.21 | *** join/#asterisk saisoma (~saisoma@client72.jdcc.edu) |
| 13:46.32 | cusco | now with incomming pri also at geo2 I think I can do the same.. |
| 13:46.43 | cusco | then I remembered reading somehting about DUNDI |
| 13:46.55 | cusco | wich publishes the available route to a certain peer |
| 13:48.14 | Kyosh | first hurdle is a single queue shared across multiple asterisk boxes in different geo locations |
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| 13:48.34 | cusco | brb |
| 13:48.42 | cusco | I will read you later tho... |
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| 14:08.20 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:14.56 | *** join/#asterisk ManxPower-work (~manxpower@216.186.151.147) |
| 14:15.29 | ManxPower-work | What would cause asterisk to not have a version number? The only difference (MIGHT) be a missing library, but I have no idea what |
| 14:16.29 | Corydon76-dig | Lack of a .version file in the root directory |
| 14:16.30 | *** join/#asterisk moy (~chatzilla@bas1-unionville55-1177733627.dsl.bell.ca) |
| 14:16.44 | Corydon76-dig | root source directory |
| 14:17.01 | ManxPower-work | Corydon76-dig, what would cause that? |
| 14:17.19 | Corydon76-dig | A user who knows how to use the 'rm' command |
| 14:18.53 | *** join/#asterisk rgsteele (~rgsteele@207.106.239.81) |
| 14:18.55 | ManxPower-work | must have been a failure in the rsync |
| 14:34.16 | *** join/#asterisk kruemeltee (~Maddin@port-92-198-62-82.static.qsc.de) |
| 14:34.21 | kruemeltee | hello all together |
| 14:36.17 | *** join/#asterisk corretico (~laguilar@201.201.46.106) |
| 14:38.25 | kruemeltee | is there any other possibility to call a secondary connected telephone system ... I have one and in the past I dialled with the help of "Dial(SIP/Number@external-System)" but since firmware upgrade this way doesn't work anymore ... may they have changed something? |
| 14:39.05 | *** join/#asterisk slinksh0t_ (~slinksh0t@64.152.41.21) |
| 14:39.12 | kruemeltee | maybe they need another SIP header correctly set? But I don't know which one ... |
| 14:39.29 | kruemeltee | are there other possibilitys like the "old" way? |
| 14:40.02 | visz | kruemeltee, instead of 'external-System' try the ip-address |
| 14:40.24 | visz | if you had a trunk named 'external-System' |
| 14:40.49 | kruemeltee | so like "Dial(SIP/number@external_IP)"? ... I already tried ... and of course I have a trunk called "external-System" ;-) |
| 14:41.42 | visz | i have had similar problem with couple of servers, and using a ip-address fixed it |
| 14:42.17 | kruemeltee | the external system registered successfully at asterisk ... in the past and today ... but since this firmware upgrade this way doesn't work ... the hotline told me, they worked a lot at the SIP protocoll and there were many changes within this upgrade ... but they didn't tell me which changes ... |
| 14:43.02 | kruemeltee | the port too? Like "Dial(SIP/number@IP\;Port)"? |
| 14:43.27 | *** part/#asterisk scottsmith7 (~ssmith@64.201.141.80) |
| 14:44.02 | kruemeltee | sorry ... meant Disl(SIP/number@IP\:port) |
| 14:47.35 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 14:51.51 | Naikrovek | Dial(SIP/trunk-name/extension) |
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| 14:54.07 | visz | eww |
| 14:54.25 | visz | got a dialogic 3008 for test |
| 14:54.30 | visz | ships with windows xp =P |
| 14:54.33 | visz | media gateway |
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| 15:40.19 | *** join/#asterisk herzan (~herzan@12.51.112.34) |
| 15:40.26 | herzan | hey everybody |
| 15:40.33 | herzan | i need help with my asterisk server |
| 15:41.03 | leifmadsen | ~ask |
| 15:41.04 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 15:41.53 | herzan | ok the thing is I have an inbound route I call from outside the company, after that it connects to the DISA which reroutes to my outbound route |
| 15:41.56 | herzan | the outbound route has a pinset |
| 15:42.15 | herzan | the pinset is supposed to keep other people from misusing our telehpone service. |
| 15:42.31 | leifmadsen | following so far |
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| 15:42.46 | herzan | what I want to do is to let the outbound route not ask for pin if known cell phones or other known phone numbers call |
| 15:43.10 | herzan | if the caller is not recognized it is not supposed to ask for a pin |
| 15:43.14 | [TK]D-Fender | ~freepbx |
| 15:43.15 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there, or fpbx |
| 15:43.16 | [TK]D-Fender | ^^^^^^^^^^^^^^ |
| 15:43.18 | leifmadsen | herzan: match on callerID then I suppose (which isn't necessarily secure, but can work) |
| 15:43.20 | herzan | I mean it is supposed to ask for pin |
| 15:43.53 | herzan | match caller ID will not ask for pin anymore ? |
| 15:44.04 | leifmadsen | no, I'm saying match on callerID to skip the pin part |
| 15:44.10 | leifmadsen | if you're using a GUI, you're in the wrong room |
| 15:44.33 | herzan | i dont mind working on the conf files |
| 15:44.43 | herzan | if you tell me which ones i should take a look at |
| 15:45.35 | [TK]D-Fender | herzan: there is no confi file, and these scripts addons, etc aren't supoprted here |
| 15:45.55 | [TK]D-Fender | herzan: These are GUI config issues. You aren't in control of your dialplan. Tehir scripts are. |
| 15:45.57 | [TK]D-Fender | their* |
| 15:46.22 | herzan | ok thank you. where could i ask the question then? freepbx? |
| 15:46.54 | leifmadsen | herzan: the problem is that as soon as you modify anything in the GUI again the changes you make will be overridden -- you have to make the changes to the PHP scripts that build the asterisk configuration files |
| 15:46.57 | leifmadsen | see #freepbx |
| 15:47.12 | *** join/#asterisk QbY (~kelvin@c-24-126-145-123.hsd1.ga.comcast.net) |
| 15:47.45 | QbY | is it possible to tell asteirsk to log (to file) every step for a particular call.. have a very busy server, but need to see every step for a call... |
| 15:48.19 | leifmadsen | QbY: uncomment 'full' in logger.conf |
| 15:48.27 | leifmadsen | then 'logger reload' from the CLI |
| 15:48.52 | QbY | leifmadsen: yes, i know how to do that.. but what i want to do is to capture for just one call going through the server |
| 15:49.00 | leifmadsen | not possible -- use 'grep' |
| 15:49.19 | QbY | ok |
| 15:49.49 | herzan | ok thank you! |
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| 15:54.35 | boch | is it possible to search a database key from the value of it ? |
| 15:55.22 | *** join/#asterisk Faithful (~Faithful@202.6.145.116) |
| 15:55.35 | [TK]D-Fender | boch: parse out "database show" |
| 15:55.52 | [TK]D-Fender | boch: there is no * function for this, so you'll need to script it |
| 15:55.57 | boch | [TK]D-Fender, yes but i mean right from dialplan |
| 15:56.00 | boch | okey, thanks |
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| 16:10.54 | leifmadsen | voxter: ping? |
| 16:10.59 | leifmadsen | M17235 |
| 16:11.01 | MuffinMan | [new] [Asterisk] Core/General 0017235: [patch] asterisk dsp always reports detected DTMF length to be 0ms reported by frawd https://issues.asterisk.org/view.php?id=17235 |
| 16:11.07 | leifmadsen | any chance at all that would resolve your DTMF issues? |
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| 17:20.45 | voxter | leifmadsen: checking |
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| 17:27.30 | leifmadsen | voxter: coolio |
| 17:27.34 | voxter | leifmadsen: hmm.. im really not certain if this would help.. I think this is only applicable when asterisk is already told to attach a length variable to the event, in my case, the problem isnt that asterisk doesnt know how to do it, its that when it sees SIP/RFC2833 on both ends, it decides to not come up with its own values and just pass along what it got |
| 17:27.54 | leifmadsen | voxter: gotcha -- kinda was thinking that myself, but thought I'd check |
| 17:28.04 | voxter | leifmadsen: ya no worries, thanks for pointing it out man! |
| 17:28.09 | leifmadsen | np! |
| 17:29.50 | Naikrovek | anyone know a good network inventory spreadsheet template |
| 17:29.53 | Naikrovek | not sure how to go about this |
| 17:32.28 | leifmadsen | Naikrovek: hmmm... might check Google Docs and see if anyone has created a template. I have a friend who has done that a few times. I think he used some software to get the initial inventory off the network as well through auto-searching. |
| 17:32.51 | Naikrovek | google docs is a good idea |
| 17:32.58 | Naikrovek | google proper isn't helping much |
| 17:33.01 | leifmadsen | aye |
| 17:40.56 | p3nguin | Is Asset Tracker no good for that purpose? |
| 17:41.07 | Naikrovek | boss wants spreadsheet |
| 17:41.20 | Naikrovek | i say webapp would probably be better suited but whatever |
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| 18:15.11 | roe | so my snom 370 has started dialing letters instead of numbers all of a sudden, I don't really see the 'alpha/numeric' toggle button. Any thoughts? |
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| 18:17.15 | pabelanger | roe: web interface? |
| 18:17.35 | roe | what about it? |
| 18:18.01 | pabelanger | pabelanger: look for toggle button there. |
| 18:18.05 | pabelanger | heh |
| 18:18.54 | roe | yea, that is where I don't see it |
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| 18:21.14 | roe | nm, found it, that was stupid |
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| 18:47.12 | pabelanger | Anybody using CEL for reporting yet? |
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| 18:51.17 | Naikrovek | oh i need to audit more |
| 18:51.49 | Naikrovek | 4 oracle databases running, none of which are use for dev, all used for production, which is counter to the license agreement at oracle's site. |
| 18:51.54 | Naikrovek | time to cut some power |
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| 18:54.19 | leifmadsen | pabelanger: not yet :( |
| 18:55.50 | pabelanger | leifmadsen: I'm adding it to my LiveCD of Asterisk trunk this week. Looks very interesting. |
| 18:56.23 | leifmadsen | yes it does -- it is designed to resolve a lot of the issues with CDRs |
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| 19:01.09 | *** join/#asterisk manxpower (~ewieling@216.186.151.147) |
| 19:01.16 | manxpower | ~answers |
| 19:01.17 | infobot | [~answers] Asterisk Book: http://astbook.asteriskdocs.org & SIP w/ NAT: http://www.aocomputing.net/?p=3 & Wiki: http://www.voip-info.org & Docs: /path/to/src/asterisk/doc & 1.4 .vs. 1.6.0 .vs. 1.6.1: /path/to/src/asterisk/UPGRADE*.txt |
| 19:01.18 | manxpower | ~mailinglist |
| 19:01.19 | infobot | [~mailinglist] The Asterisk mailing lists can be found at http://lists.digium.com , http://www.asteriskguru.com/archives Search the archives by adding "site:lists.digium.com" to your Google search. |
| 19:02.42 | leifmadsen | ~manswers |
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| 19:19.45 | Naikrovek | manswers? |
| 19:19.50 | Naikrovek | lol |
| 19:21.53 | p3nguin | lol |
| 19:22.39 | p3nguin | naikrovek: I pulled a few sheets with asset info on them. What exactly are you looking for? |
| 19:23.00 | Naikrovek | p3nguin: never done an inventory system/spreadsheet before. not sure how to go about it |
| 19:23.11 | Naikrovek | an example that will get the wheels turning is all i need, really |
| 19:26.14 | p3nguin | Do you know if you want one sheet per asset tag or one sheet with multiple assets? |
| 19:30.51 | Naikrovek | one sheet, multiple assets |
| 19:31.27 | Naikrovek | i suppose i can list installed hardware and software somewhere else |
| 19:31.58 | Naikrovek | this just screams database to me but i dunno |
| 19:32.02 | Naikrovek | boss wants spreadsheet |
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| 19:37.03 | *** join/#asterisk doctorray (~ray@static-71-177-137-76.lsanca.fios.verizon.net) |
| 19:39.18 | doctorray | MixMonitor closes the filestream when a call is parked... is there any way to make it continue |
| 19:42.17 | leifmadsen | doctorray: I don't think so -- you can try using AUDIOHOOK_INHERIT() but once the channel that initiated the call recording goes away, I think it's just gone |
| 19:44.33 | doctorray | I suppose I could put another mixmonitor command in append mode in the parkedcallstimeout context, as well as writing my own parked call context that re-initiates recording |
| 19:44.52 | doctorray | I don't really need a recording of them on hold, but I do need it for when it picks back up again |
| 19:45.02 | p3nguin | It also stops recording if the call is transferred, which is annoying. |
| 19:45.58 | leifmadsen | p3nguin: that's the point of AUDIOHOOK_INHERIT() |
| 19:46.06 | doctorray | p3nguin: I just tested that in my setup and it continued recording after a transfer |
| 19:46.32 | p3nguin | Can you tell me where/how that is used? |
| 19:46.41 | leifmadsen | p3nguin: core show function AUDIOHOOK_INHERIT |
| 19:47.16 | leifmadsen | doctorray: I tested that today and noticed that the transfer has to be initiated by the called channel though and not the originating channel |
| 19:47.34 | p3nguin | So it needs to be added to every extension that dials phones, yes? |
| 19:47.36 | doctorray | great.. :) |
| 19:47.56 | leifmadsen | doctorray: go ahead and try, but that's what I came up with today (which actually makes sense to me). |
| 19:48.41 | leifmadsen | M17244 |
| 19:48.43 | MuffinMan | [feedback] [Asterisk] Applications/app_mixmonitor 0017244: MixMonitor fails to record atxfer calls reported by Samael28 https://issues.asterisk.org/view.php?id=17244 |
| 19:48.48 | leifmadsen | doctorray: see the note I added near the end |
| 19:49.10 | leifmadsen | p3nguin: added to whatever dialplan will initiate a MixMonitor() |
| 19:49.14 | p3nguin | Oh, what I asked about isn't right. |
| 19:49.19 | leifmadsen | i.e. add 1 additional line before MixMonitor() |
| 19:49.20 | p3nguin | yeah, I just read that. |
| 19:50.01 | doctorray | i see |
| 19:51.41 | p3nguin | Does it matter if the inherit line goes before or after the MixMonitor line? |
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| 19:51.57 | doctorray | Can the call parking context be modified to set that, such as a exten => 700,s,blah blah before it actually parks it |
| 19:52.43 | leifmadsen | p3nguin: not sure -- just put it before |
| 19:53.07 | p3nguin | core show ... indicated to put it after, so that was why I didn't know if it mattered. |
| 19:53.19 | leifmadsen | ah odd -- probably doesn't matter then |
| 19:53.27 | leifmadsen | try it and find out which one works :) |
| 19:53.54 | doctorray | thanks for the help. I'm gonna test after lunch |
| 19:54.03 | [TK]D-Fender | doctorray: No. So don't include the context directly and just make your own. |
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| 19:55.21 | doctorray | [TK]D-Fender: I'll try. I may come back later and ask about that. thanks |
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| 20:59.50 | p3nguin | If a context has no BackGround()s or WaitExten()s in it, there's no reason to have an 'i' exten, yes? |
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| 21:14.17 | manxpower | "i" is for "I"VRs |
| 21:14.30 | manxpower | just remember that and you'll be good. |
| 21:14.40 | manxpower | if it's not an IVR chances you don't need it. |
| 21:15.54 | p3nguin | It's part of an IVR, which is the reason I'm asking. But I can't think of any reason 'i' would ever be able to be reached if no caller input is accepted via BackGround or WaitExten. |
| 21:16.53 | *** join/#asterisk paulc (~paulc@unaffiliated/paulc) |
| 21:17.25 | leifmadsen | when you say IVR you probably mean auto-attendant :) |
| 21:17.31 | p3nguin | perhaps |
| 21:17.46 | p3nguin | Because this section does not allow any interaction. |
| 21:17.50 | leifmadsen | don't worry, that'll get all explained in the new book |
| 21:17.58 | leifmadsen | that's not the difference |
| 21:18.03 | p3nguin | oh |
| 21:19.02 | leifmadsen | auto-attendant is the menu you typically hear and can dial extensions from and press numbers to generate DTMF to go to other parts of the menu. And IVR is typically something similar, but gets information from an external source (such as from a database) and returns data dynamically. |
| 21:19.15 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 21:19.31 | p3nguin | hmm |
| 21:19.54 | p3nguin | So then most people who say IVR are probably using the term incorrectly. |
| 21:20.02 | leifmadsen | most people just substitute the terms interchangeably |
| 21:20.06 | leifmadsen | right |
| 21:20.46 | leifmadsen | for example: when you call Pizza Pizza and it says, "Would you like the exact order as last time?" and you press 1 for yes and you're done, that is an example of IVR |
| 21:20.47 | paulc | chatline = IVR |
| 21:20.52 | paulc | my daily bread and butter :) |
| 21:20.53 | ariel_ | no some of us know what an ivr is, and I use them all the time, for collecting info, like cc, or pins for different routes |
| 21:21.19 | leifmadsen | Asterisk just makes the line between IVR and auto-attendant blurred because Asterisk is so good at both |
| 21:21.32 | leifmadsen | however, it's just a term, and who cares? :) |
| 21:21.47 | ariel_ | your correct |
| 21:21.50 | leifmadsen | you're* |
| 21:21.54 | p3nguin | Okay, then the context in question is pre-processing for an auto-attendant. No user input is able to be accepted, thus no 'i' exten should be needed. Sound right? |
| 21:22.08 | leifmadsen | right |
| 21:22.27 | leifmadsen | if you're not accepting input from a user then there is no possibility of them dialing something wrong |
| 21:22.38 | leifmadsen | thus you don't need 'invalid' |
| 21:22.44 | p3nguin | That was my thought exactly. |
| 21:22.53 | leifmadsen | and now I leave |
| 21:23.01 | p3nguin | Would the same be true for t and T? |
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| 21:24.27 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 21:25.02 | doctorray | leifmadsen: it appears that setting the AUDIOHOOK_INHERIT to yes will continue a MixMonitor into call parking and picking up the parked call, but only if the called channel parks it |
| 21:25.48 | doctorray | which, for my particular application, will probably be alright |
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| 21:26.56 | p3nguin | Can a channel other than the called channel perform the park? |
| 21:27.18 | p3nguin | Aren't there only two channels involved in the call anyway? |
| 21:27.25 | p3nguin | calling and called |
| 21:27.38 | p3nguin | and I'm sure you don't allow the calling channel to park himself. |
| 21:31.13 | doctorray | if the caller is in the building with a sip phone, then technically they could park a call they made |
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| 21:39.17 | doctorray | and now for something completely unrelated |
| 21:40.04 | doctorray | I'm using the MySql CDR addon, and it won't record the caller ID name field, in the database as 'clid' -- haven't been able to find much online as to troubleshooting that. |
| 21:40.13 | doctorray | does anyone have a direction to point me in? |
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| 21:46.46 | doctorray | I think I foudn it |
| 21:46.48 | doctorray | found* |
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| 21:49.50 | [TK]D-Fender | ~book |
| 21:49.51 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 21:51.13 | doctorray | thaaanks TK |
| 21:53.16 | *** join/#asterisk sysreq (~sysreq@unaffiliated/sysreq) |
| 21:53.37 | [TK]D-Fender | doctorray: Wasn't actually for you... |
| 21:55.24 | leifmadsen | doctorray: ya, if the called channel does the parking I would expect it to work |
| 21:55.51 | doctorray | [TK]D-Fender: good deal. :) |
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| 21:56.46 | Get_The_Fish | hey, whats the deal with the patch file on downloads.asterisk.org for 1.6.2.7? Whats it for |
| 21:57.17 | [TK]D-Fender | Get_The_Fish: My guess would be patching some previous version into 16.2.7 |
| 21:57.45 | Get_The_Fish | that would be my guess as well. Hoping for a definite answer |
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| 22:01.45 | ecrane | reading RFC 3261 - It talks about something called a sip 'core' can someone describe to me what is a sip core? Is it any sip device (client, server, proxy, etc.?) |
| 22:01.45 | p3nguin | I would imagine they are talking about the protocol. |
| 22:01.55 | *** part/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 22:02.04 | leifmadsen | Get_The_Fish: it'd be a diff between 1.6.2.6 and the changes in 1.6.2.7 |
| 22:02.10 | leifmadsen | note that 1.6.2.7 is not yet out |
| 22:02.16 | Get_The_Fish | why thank you sir |
| 22:02.22 | leifmadsen | so it'd be a diff between 1.6.2.6 and the latest release candidate |
| 22:02.33 | Get_The_Fish | yeah, I am testing an issue to see if it exists in the rc |
| 22:03.45 | ecrane | It's on page 20 of the RFC. Says "Core designates the functions specific to a particular type of SIP entity..............." but I'm having trouble understanding what they are talking about. |
| 22:10.20 | Get_The_Fish | leif, this patch looks like it patches from rc1 to rc2 |
| 22:10.46 | leifmadsen | Get_The_Fish: that's right -- it'd be the diff between the previous version |
| 22:10.54 | leifmadsen | so from rc2 the prev is rc1 |
| 22:10.55 | Get_The_Fish | ah ok |
| 22:10.59 | Get_The_Fish | right rigth |
| 22:11.01 | leifmadsen | the prev of rc1 is 1.6.2.6 |
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| 22:17.17 | Get_The_Fish | is there an easy way to get the options selected from a previously done "make menuselect"? |
| 22:30.12 | Naikrovek | Get_The_Fish: don't think so. perhaps they're in the old makefile or something |
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| 22:37.38 | drfreeze | I am getting a caller with a callerid of Restricted |
| 22:37.39 | drfreeze | <PROTECTED> |
| 22:37.46 | drfreeze | I thought collerid had to be a number |
| 22:39.47 | leifmadsen | Get_The_Fish: copy menuselect.makeopts |
| 22:49.08 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 22:55.20 | Get_The_Fish | why thank you again Leif. A fountain of knowledge, you. :) |
| 22:55.29 | leifmadsen | I know some stuff :) |
| 22:55.48 | Slugs_ | ~pb |
| 22:55.49 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood 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://bin.cakephp.org/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude. |
| 22:56.46 | Get_The_Fish | so, are you thinking about doing another rev of *tfot anytime soon here? |
| 23:03.11 | *** join/#asterisk jksM (jks@193.189.93.254) |
| 23:09.01 | *** join/#asterisk obnauticus (~obnauticu@about/windows/regular/obnauticus) |
| 23:15.18 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 23:21.41 | *** join/#asterisk cdose1 (~chris@ip72-219-50-148.br.br.cox.net) |
| 23:31.04 | cdose1 | i'm placing automated outgoing calls using call files and all my outbound calls immediately hang up after the end user picks up. what's up with that? |
| 23:33.06 | Corydon76-dig | cdose1: Are you using a codec that you don't have a translator for? |
| 23:34.03 | Corydon76-dig | You'll need 2 translators for each call placed via callfile |
| 23:36.39 | cdose1 | CoderForLife, sorry, I'm a bit new to this. I don't know what codecs you mean |
| 23:36.51 | cdose1 | Corydon76-dig, sorry that was meant for you |
| 23:37.33 | Corydon76-dig | G.729? |
| 23:38.29 | Corydon76-dig | Ulaw? Alaw? GSM? iLBC? ADPCM? |
| 23:40.13 | bmoraca_work | g.>9000 |
| 23:40.19 | cdose1 | Corydon76-dig, sorry, I installed asterisk via trixbox, it was set up for me. the asterisk/codecs.conf file has sections in it for plc and speex, if that means anything. I don't know where else to look for references to codecs |
| 23:40.38 | Corydon76-dig | cdose1: Go ask in #trixbox, then |
| 23:41.03 | Corydon76-dig | ~trixbox |
| 23:41.04 | infobot | hmm... trixbox is SH1TB0X. Basically a CRAPPY, closed source distro. STAY AWAY! |
| 23:41.58 | cdose1 | Corydon76-dig, yeah thanks, i already figured that. but I don't yet have the experience to set up an asterisk server myself. i find it's always good to jump in with something pre-made, and work your way up from there |
| 23:42.32 | Corydon76-dig | There are plenty of others. Try PBX-in-a-Flash, I've heard better things about that |
| 23:42.46 | cdose1 | Corydon76-dig, really, ok thanks |
| 23:43.01 | p3nguin | PiaF instead of AsteriskNOW? |
| 23:43.27 | *** join/#asterisk atis_work (~atis_work@193.238.212.171) |
| 23:43.28 | Corydon76-dig | p3nguin: as long as someone else is out there to support it... |
| 23:44.21 | cdose1 | p3nguin, you would recommend AsteriskNOW instead? |
| 23:44.24 | Corydon76-dig | I'm a CLI person myself |
| 23:44.50 | cdose1 | Corydon76-dig, i generally am myself as well, but PBXs are a completely new world to me |
| 23:44.54 | p3nguin | cdose1: I've tried both PiaF and AsteriskNOW, and I wouldn't recommend PiaF to anyone. |
| 23:45.32 | cdose1 | p3nguin, ok, what would you recommend? |
| 23:45.34 | Corydon76-dig | p3nguin: the guy who put together PiaF has been a real asshole towards me. I feel as though I need to return the favor |
| 23:46.12 | p3nguin | cdose1: Start with a good distro, add asterisk, read the book, start configuring, enjoy. |
| 23:46.32 | p3nguin | cdose1: if that is not an option, AsteriskNOW is actually pretty decent. |
| 23:46.38 | Corydon76-dig | There's something to be said about people who think law school is an appropriate substitute for years of programming |
| 23:47.24 | cdose1 | p3nguin, option 1 definitely is an option, i just wanted to "jump in" so to speak, and start prototyping an idea, before actually commiting to it and spending all the time to set up something more permenant |
| 23:48.02 | Corydon76-dig | cdose1: yeah, start by reading the book, first |
| 23:48.08 | Corydon76-dig | ~thebook |
| 23:48.08 | infobot | well, thebook is Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://astbook.asteriskdocs.org |
| 23:49.28 | cdose1 | Corydon76-dig, thanks |
| 23:50.42 | *** join/#asterisk Micc (~quassel@c-98-225-57-96.hsd1.wa.comcast.net) |
| 23:51.53 | Micc | I keep getting errors receiving faxes with digium's fax for asterisk. I have a SIP trunk that goes to my main server, then from there forwards over IAX2 to another server that is only 1ms away. Would it be better to use SIP? |
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| 23:53.59 | Micc | It has no problems sending faxes to SIP peers, but sending to them through IAX -> SIP peer seems to have issues, so I'm thinking SIP->SIP->SIP peer might work better. |
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