00:02.35 | *** join/#asterisk gusto (~gusto@ppp-93-104-86-239.dynamic.mnet-online.de) |
00:27.05 | *** join/#asterisk vinhdizzo (~vinh@ip70-181-154-104.sd.sd.cox.net) |
00:59.13 | *** join/#asterisk lorsungcu (~anonymous@65.103.31.33) |
01:05.12 | *** join/#asterisk mintos (~mvaliyav@114.143.44.23) |
01:35.06 | *** part/#asterisk ghost75 (~trechber@dslb-178-002-154-031.pools.arcor-ip.net) |
02:02.32 | *** join/#asterisk mjordan (~mjordan@75.76.55.191) |
02:02.33 | *** mode/#asterisk [+o mjordan] by ChanServ |
02:13.50 | *** join/#asterisk tallest_red (~CNZ@ip98-169-207-41.dc.dc.cox.net) |
02:26.19 | *** join/#asterisk mintos (~mvaliyav@114.143.45.36) |
02:29.12 | *** join/#asterisk imcdona (~imcdona@c-71-227-200-25.hsd1.wa.comcast.net) |
02:34.05 | *** join/#asterisk mzb (~mzb@2001:44b8:512d:7501:f66d:4ff:fe90:9629) |
02:41.08 | *** join/#asterisk mzb (~mzb@2001:44b8:512d:7501:f66d:4ff:fe90:9629) |
02:50.30 | mnathani | How can I troubleshoot a SIP connection using my IP Phone with the Aterisk server? |
02:50.47 | mnathani | The phone does not seem to register |
02:50.53 | *** join/#asterisk mzb (~mzb@2001:44b8:512d:7501:f66d:4ff:fe90:9629) |
02:52.59 | *** join/#asterisk mzb (~mzb@2001:44b8:512d:7501:f66d:4ff:fe90:9629) |
03:08.30 | *** join/#asterisk ChannelZ (channelz@burner.com) |
03:25.24 | *** join/#asterisk Sorcier_FXK (~nssystem@unaffiliated/sorcierfxk) |
03:29.13 | *** join/#asterisk coppice (~chatzilla@123203240234.ctinets.com) |
03:57.44 | lorsungcu | mnathani: im fukcing wasted |
03:58.09 | lorsungcu | mnathani: but u thunk almost anyone would askj for some logs |
03:58.30 | lorsungcu | mnathani: so fucking pb some goddamn logs |
04:09.14 | *** join/#asterisk radic (~radic@dslb-088-065-151-048.pools.arcor-ip.net) |
04:26.13 | *** join/#asterisk bpriddy (~bpriddy@ipv4.host.stabbyspazzout.net) |
04:42.23 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
04:42.24 | *** mode/#asterisk [+o pabelanger] by ChanServ |
05:09.03 | *** join/#asterisk JuStIcIa_ (~artur0@190.166.210.87) |
05:53.12 | *** join/#asterisk evilman_home (kvirc@95-29-217-19.broadband.corbina.ru) |
06:25.25 | *** join/#asterisk gerhard7 (~gerhard7@195-241-233-43.ip.telfort.nl) |
08:18.09 | *** join/#asterisk kresp0 (~kresp0@81.61.25.130.dyn.user.ono.com) |
08:26.18 | *** join/#asterisk zemmali-voip (~zemmali@unaffiliated/zemmali-voip) |
08:45.06 | kresp0 | Hi all, I'm having no audio issues when calling from a sip phone registered on my * to another * echo test. Network map is like this: |
08:45.09 | kresp0 | sip phoneLAN my * pbx |
08:45.13 | kresp0 | 192.168.2.4 <---------> 192.168.2.1 VPN friend * pbx |
08:45.17 | kresp0 | <PROTECTED> |
08:45.20 | kresp0 | upss |
08:45.45 | kresp0 | i mean, network is like this: http://pastebin.com/NKHGsHkz |
08:46.50 | kresp0 | damit, screwed again, sorry!. http://pastebin.com/NGiyiD3z |
08:47.03 | kresp0 | My sip.conf: http://pastebin.com/ZzcWiqhP |
08:47.08 | kresp0 | And sip debug on friend pbx: http://pastebin.com/6TpQxcs7 |
08:47.29 | kresp0 | Tried "nat=force_rport,comedia" (old sip=yes) on peer definition at sip.conf, but still get "Peer audio RTP is at port 192.168.2.4:41000" on remote PBX. |
08:48.06 | *** join/#asterisk ectospasm (~ectospasm@unaffiliated/ectospasm) |
08:50.38 | *** join/#asterisk k610 (~K610@cable-78.29.241.186.coditel.net) |
08:59.55 | *** join/#asterisk nixhr (~nix@sunce2.iprojekt.hr) |
09:00.36 | *** join/#asterisk pigpen (~mark@216.177.172.50) |
09:17.13 | *** join/#asterisk ThomasLocke (~ThomasLoc@pdpc/supporter/active/thomaslocke) |
09:42.33 | Rhomber | i've found a nice place to get the prompts.. but they only provide in 3 formats.. gsm, alaw and slin |
09:43.26 | Rhomber | i'm guessing it's better to convert the slin to ulaw to have it as well rather than rely on on-the-fly re-encoding... what's the best way to do that? and i assume slin is the best file to use for that? being 128Kbps |
09:44.19 | *** join/#asterisk corehook (~corehook@2.133.28.103) |
10:02.07 | *** join/#asterisk zemmali-voip (~zemmali@unaffiliated/zemmali-voip) |
10:10.12 | *** join/#asterisk pigpen (~mark@fw.seamans.cc) |
10:10.13 | ChannelZ | unless you have thousands of calls or something, letting asterisk transcode is nothing |
10:10.24 | *** join/#asterisk zemmali-voip (~zemmali@unaffiliated/zemmali-voip) |
10:11.14 | *** join/#asterisk areski (~areski@80.174.255.7.dyn.user.ono.com) |
10:14.58 | *** join/#asterisk coppice (~chatzilla@123203240234.ctinets.com) |
10:26.50 | *** join/#asterisk timahvo1 (~rogue@41.212.120.45) |
10:27.53 | *** join/#asterisk Natureshadow (nik@shore.naturalnet.de) |
10:39.50 | *** part/#asterisk corehook (~corehook@2.133.28.103) |
10:43.04 | *** join/#asterisk Chotaire (chotaire@chotaire-home.vipri.net) |
10:54.07 | Rhomber | ah cool, fair enough then :) |
10:58.02 | *** join/#asterisk gerhard7 (~gerhard7@195-241-233-43.ip.telfort.nl) |
11:15.35 | *** join/#asterisk ghost75 (~trechber@dslb-088-064-218-152.pools.arcor-ip.net) |
11:23.13 | *** join/#asterisk kresp0 (~kresp0@81.61.25.130.dyn.user.ono.com) |
11:24.13 | *** join/#asterisk [sr] (~kvirc@pal-213-228-181-48.netvisao.pt) |
11:24.28 | *** join/#asterisk tzafrir_laptop (~tzafrir@local.xorcom.com) |
11:33.28 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
11:44.07 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
11:44.07 | *** mode/#asterisk [+o pabelanger] by ChanServ |
12:08.24 | *** join/#asterisk hehol (~Adium@2a01:198:71d:0:bcb3:b94f:3cda:9d8f) |
12:09.37 | *** join/#asterisk StaRetji (~LittleAll@91.142.129.1) |
12:09.45 | StaRetji | Howdy folks |
12:09.59 | kresp0 | hi |
12:10.36 | StaRetji | is there a way to set asterisk to allow only 1 consequent calls for any caller id in given time period? |
12:11.15 | StaRetji | for example, if 4455 makes a call, then in next 5 minutes can not make another call |
12:11.29 | Gugge | StaRetji: yes, using GROUP_COUNT and IFTIME |
12:11.45 | Gugge | ohh, then you need some kind of database to save the state in too |
12:11.45 | StaRetji | Gugge: thx man |
12:12.13 | StaRetji | seems like one hellawa task for me :D |
12:12.34 | StaRetji | let me google if there are examples |
12:17.11 | StaRetji | hm, google is not my friend this round :/ |
12:23.05 | StaRetji | basically, I need to limit cal attempts, 1 clid 1 call attempt in 5 minutes |
12:28.57 | *** join/#asterisk fling (~fling@fsf/member/fling) |
12:40.26 | *** join/#asterisk moy (~moy@UNVLON55-1176057127.sdsl.bell.ca) |
12:48.59 | *** join/#asterisk nicknam1232 (5c15c013@gateway/web/freenode/ip.92.21.192.19) |
12:52.14 | Rhomber | hmm, Voicemail(${mbx},u) ... the 'u' part isn't generating the right sound .. it's using the system sounds and not the language=au ones.. is there some config i might have overlooked, i changed language=au everywhere i could think of |
12:58.53 | kresp0 | StaRetji, try using GotoIfTime and astdb. Sorry I dont know how to extract current time using asterisk functions. But maybe you can use a shell comand (date +%s) to get the current unix time |
13:12.10 | *** join/#asterisk gavimobile (~user@bzq-218-196-30.red.bezeqint.net) |
13:13.30 | gavimobile | hey folks, I need some assistance here. sometimes people are waiting in the queue too long. I want to give them the option to press a number which will forward to a voicemail box |
13:14.10 | gavimobile | periodic-announce will periodically play my announcement while waiting in the queue && |
13:14.11 | gavimobile | periodic-announce-frequency = 45 ; every 60 seconds |
13:14.47 | gavimobile | but how do I give them the option to press a number? via context from queues.conf? |
13:20.31 | kresp0 | ; A context may be specified, in which if the user types a SINGLE |
13:20.31 | kresp0 | ; digit extension while they are in the queue, they will be taken out |
13:20.31 | kresp0 | ; of the queue and sent to that extension in this context. |
13:20.31 | kresp0 | ; |
13:20.32 | kresp0 | ;context = qoutcon |
13:21.18 | kresp0 | try that option gavimobile |
13:21.23 | kresp0 | on queues.conf |
13:22.31 | gavimobile | kresp0: I got it, now in addition, I would like to save the queue position number and have my pbx call them back when their turn comes |
13:22.46 | gavimobile | what's the logic here |
13:23.28 | gavimobile | in the dialplan which == queues context , I need a variable which saves the queue position of the caller I assume, but how do I save the position? |
13:25.47 | *** join/#asterisk vlad_starkov (~vlad_star@83.220.236.35) |
13:35.21 | gavimobile | maybe ill reword my question.. when going to the context specified in queues.conf, does that caller loose his position in the queue? |
13:45.22 | *** join/#asterisk nicknam1232 (5c15c013@gateway/web/freenode/ip.92.21.192.19) |
13:57.49 | kresp0 | that caller gets kicked out of the queue. Maybe you should send them to another queue then. |
13:58.22 | kresp0 | and in that other queue put the Dial to call them back |
14:00.48 | kresp0 | oh now I understand what you want to do |
14:02.18 | kresp0 | Caller -> queue -> press key to exit -> hung up call -> add call to an outgoing queue |
14:02.37 | kresp0 | they will be in order |
14:25.26 | gavimobile | kresp0: thanks, what do you mean by outgoing queue? |
14:26.15 | gavimobile | like a spool of outgoing calls? |
14:28.26 | *** join/#asterisk timahvo1 (~rogue@41.212.120.45) |
14:42.40 | kresp0 | a queue where members are outgoing calls |
15:08.09 | *** join/#asterisk coppice (~chatzilla@123203240234.ctinets.com) |
15:33.48 | *** join/#asterisk vinhdizzo (~vinh@ip70-181-154-104.sd.sd.cox.net) |
15:41.48 | *** join/#asterisk coppice (~chatzilla@123203240234.ctinets.com) |
15:42.57 | *** join/#asterisk Neptu (~Hej@c213-89-2-159.bredband.comhem.se) |
15:51.09 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
15:51.09 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
16:07.21 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
16:33.07 | *** join/#asterisk appleboy (~appleboy@about/cooking/nakedchef/apple/tarts) |
16:38.48 | *** join/#asterisk Ice_Strike (Ice_Strike@87.115.83.87) |
17:33.42 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
17:40.32 | *** join/#asterisk jblack (~jblack@90.sub-70-199-225.myvzw.com) |
17:51.53 | *** join/#asterisk LiuYan (~LiuYan@211.154.128.171) |
17:52.07 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
17:52.08 | *** mode/#asterisk [+o pabelanger] by ChanServ |
18:18.14 | *** join/#asterisk gusto (~gusto@ppp-93-104-66-12.dynamic.mnet-online.de) |
18:18.35 | *** join/#asterisk reber (~reber@37.161.38.71) |
18:19.48 | *** join/#asterisk k610 (~K610@cable-78.29.241.186.coditel.net) |
18:46.34 | *** join/#asterisk digilink (~digilink@unaffiliated/digilink) |
18:48.29 | *** join/#asterisk lorsungcu (~anonymous@65.103.31.33) |
18:53.47 | *** join/#asterisk kontinuity (~kontinuit@122.167.100.176) |
19:03.53 | *** join/#asterisk moy (~moy@173.239.155.74) |
19:15.53 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
19:15.55 | *** mode/#asterisk [+o pabelanger] by ChanServ |
19:25.52 | *** join/#asterisk reber (~reber@37.161.224.96) |
19:27.01 | *** join/#asterisk jzaw (~jzaw@loki.dzki.co.uk) |
19:28.03 | *** join/#asterisk moy (~moy@173.239.155.74) |
19:29.47 | *** join/#asterisk mjordan (~mjordan@75.76.55.191) |
19:29.48 | *** mode/#asterisk [+o mjordan] by ChanServ |
20:04.18 | *** join/#asterisk lorsungcu (~anonymous@65.103.31.33) |
20:04.36 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
20:04.37 | *** mode/#asterisk [+o pabelanger] by ChanServ |
20:46.20 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
20:46.20 | *** mode/#asterisk [+o pabelanger] by ChanServ |
20:49.34 | *** join/#asterisk Linkforsoad (~Linkforso@2001:1af8:fec1:0:64d1:b473:d090:758b) |
21:01.47 | *** join/#asterisk moy (~moy@UNVLON55-1176057127.sdsl.bell.ca) |
21:15.45 | *** join/#asterisk Neptu (~Hej@c213-89-2-159.bredband.comhem.se) |
21:16.49 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
21:16.50 | *** mode/#asterisk [+o pabelanger] by ChanServ |
21:17.36 | *** join/#asterisk ghost75 (~trechber@dslb-088-064-218-152.pools.arcor-ip.net) |
21:25.14 | *** join/#asterisk threesome (~threesome@ip-94-113-12-74.net.upcbroadband.cz) |
21:26.04 | *** join/#asterisk freeedrich| (friedrich@perplexa.be) |
22:10.13 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
22:10.14 | *** mode/#asterisk [+o pabelanger] by ChanServ |
22:11.39 | *** join/#asterisk kowala (~kowala___@93-103-132-243.static.t-2.net) |
22:14.08 | *** join/#asterisk tzafrir_laptop (~tzafrir@212.179.75.202) |
22:20.06 | *** join/#asterisk fisted_ (~fisted@unaffiliated/fisted) |
22:24.42 | kowala | hello. i have a question regarding asterisk configuration and existing modules. is there a resource module available that cloud connect to a rest service (instead of mysql or pgsql) and retrieve sip account data? is this even possible to achieve? |
22:25.26 | beefcafe | ~~ |
22:26.14 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
22:35.03 | [TK]D-Fender | The words are English... but the order and usage is incomprehensible |
22:39.39 | edong23 | im thinking that is "could connect" |
22:39.49 | edong23 | oh.. maybe not |
22:39.52 | edong23 | hm... |
22:40.05 | edong23 | kowala: where is the decoder ring? |
22:42.54 | *** join/#asterisk nantou (~phonetic@gateway/tor-sasl/martinphone) |
22:43.23 | kowala | sorry about my spelling. |
22:44.05 | kowala | the issue I have is that databases cant be exposed to asterisk and all data available for integration is available through web services |
22:46.01 | kowala | i need to make asterisk to communicate with those webservices for realtime data |
22:49.32 | [TK]D-Fender | kowala: Go look at the Database integration section of the book. That is what we have. |
22:49.32 | [TK]D-Fender | ~book |
22:49.32 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
22:49.33 | [TK]D-Fender | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
22:52.36 | kowala | at the moment I am thinking of writing a resource module, similar to the LDAP one |
22:52.37 | kowala | if there will be no other way |
22:53.33 | kaldemar | kowala: take a look at res_config_curl |
22:55.16 | kowala | [TK]D-Fender: thanks for the reply. I have the definitive guide 3rd edition. the database integration is not possible in my deployment scenario. |
22:55.21 | kowala | kaldemar: thank you. I will check it out. |
23:01.25 | *** join/#asterisk nicknam1232 (5c15c64d@gateway/web/freenode/ip.92.21.198.77) |
23:20.39 | *** join/#asterisk k1ng (~k1ng@unaffiliated/k1ng) |
23:20.59 | *** join/#asterisk mjordan (~mjordan@75.76.55.191) |
23:20.59 | *** mode/#asterisk [+o mjordan] by ChanServ |
23:23.47 | *** join/#asterisk rdm (~rdm@unaffiliated/qubix) |
23:30.57 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
23:30.58 | *** mode/#asterisk [+o pabelanger] by ChanServ |