| 00:01.01 | ChannelZ | It never really knows the MAC address. |
| 00:01.21 | ChannelZ | It's dealing in IPs |
| 00:02.39 | file | I deal in code |
| 00:03.37 | ChannelZ | I deal in guns and burritos |
| 00:04.31 | dolcea-xoom | Ok, then i'll use ip |
| 00:04.32 | factormystic | in asterisk 11, has the sip command changed from prior versions? I don't have any "sip" commands |
| 00:04.48 | factormystic | possibly also another misconfiguration, although, I do have a sip.conf |
| 00:04.54 | ChannelZ | chan_sip probably isn't loaded then |
| 00:04.56 | *** join/#asterisk droemel (~droemel@p4FCACBAB.dip.t-dialin.net) |
| 00:05.13 | jpsharp | Or didn't even get built. |
| 00:05.20 | dolcea-xoom | Check cli-aliases.conf |
| 00:05.22 | ChannelZ | If your sip.conf is so screwed up it got tripped up parsing it, it could cause the module to fail to load |
| 00:05.29 | factormystic | ok I do not have a chan_sip showing in module show like chan_ |
| 00:05.47 | ChannelZ | Is this still 1.8 or did you build 11? |
| 00:05.56 | factormystic | I built 11 from source |
| 00:06.33 | ChannelZ | and 'module load chan_sip' says what |
| 00:08.27 | factormystic | Unable to load module chan_sip |
| 00:08.28 | factormystic | Command 'module load chan_sip' failed. |
| 00:08.28 | factormystic | [Oct 13 20:06:41] WARNING[5471]: loader.c:410 load_dynamic_module: Error loading module 'chan_sip': /usr/lib/asterisk/modules/chan_sip.so: cannot open shared object file: No such file or directory |
| 00:08.28 | factormystic | [Oct 13 20:06:41] WARNING[5471]: loader.c:878 load_resource: Module 'chan_sip' could not be loaded. |
| 00:08.57 | file | taps foot at dev VM at home |
| 00:09.23 | factormystic | and manually checking, chan_sip.so doesn't exist in that directory |
| 00:09.26 | file | do you have libssl-dev package installed? |
| 00:09.30 | ChannelZ | Hmm.. it either didn't get built |
| 00:09.31 | factormystic | so I interpret that possibly as saying it didn't get built |
| 00:09.46 | jpsharp | chan_sip didn't get built. Betcha it didn't build because of that dependency between res_crypto and libssl-dev. |
| 00:09.51 | factormystic | in make menuconfig, chan_sip is XXX and apparently not configurable |
| 00:09.53 | ChannelZ | make menuconfig in the source dir and see |
| 00:10.21 | ChannelZ | That's strange.. it's only dependent on chan_local which isn't dependent on anything |
| 00:11.30 | jpsharp | If you're on a debian or ubuntu box, install libssl and libssl-dev and rerun ./configure and I bet you it will appear. |
| 00:11.59 | file | there is to be no betting in this channel! we aren't licensed |
| 00:12.32 | jpsharp | You can pay me in M&Ms. |
| 00:12.49 | factormystic | jpsharp: its ubuntu, doing that next |
| 00:15.57 | factormystic | ok after getting libssl-dev and rerunning ./configure I see that chan_sip is checked off in make menuconfig |
| 00:16.36 | jpsharp | There ya go. Run make && make install, ride off into sunset. |
| 00:16.55 | factormystic | ok so does XXX in make menuconfig mean "not available" then? |
| 00:17.05 | file | it means dependencies not met, so can't be built |
| 00:17.33 | factormystic | alright, since I'm in here again I see that chan_motif is xxx'd out so I'm still missing something |
| 00:18.09 | jpsharp | I've not built chan_motif, so I don't know what it requires. |
| 00:18.15 | file | probably missing the libiksemel-dev package |
| 00:19.00 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 00:19.01 | factormystic | bingo |
| 00:19.47 | factormystic | and I see that both of those just got built here in make |
| 00:20.44 | file | chan_motif has gotten some bug fixes in SVN that'll be in the release, so I'd advise you to grab it from SVN and not the RC |
| 00:21.26 | jpsharp | I should build a new VM and try out 11. |
| 00:23.10 | jpsharp | Now to talk the kidlet out of one of her minecraft instances so I can have enough resources to run a new VM. |
| 00:23.48 | factormystic | haha |
| 00:25.16 | factormystic | excellent, chan_sip and chan_motif are both showing up, now back to setting up the config |
| 00:25.30 | factormystic | file: noted |
| 00:36.52 | factormystic | well |
| 00:36.55 | factormystic | its almost working |
| 00:37.20 | factormystic | right now I'm getting an infinite scroll of: |
| 00:37.26 | factormystic | res_xmpp.c:3561 xmpp_client_thread: JABBER: socket read error |
| 00:37.26 | factormystic | res_xmpp.c:3502 xmpp_client_receive: Parsing failure: Hook returned an error. |
| 00:37.26 | factormystic | res_xmpp.c:3499 xmpp_client_receive: Parsing failure: Invalid XML. |
| 00:38.04 | file | check thy credentials |
| 00:41.20 | factormystic | hm, I did before pasting that, however, setting up a per-application password without any special characters and then reloading the config file seems to have solved it |
| 00:41.39 | file | I don't know what that means |
| 00:54.17 | *** join/#asterisk volga629 (~volga629@76-10-130-18.dsl.teksavvy.com) |
| 00:56.52 | volga629 | Hello Everyone, looking some recommendation about NAT config, asterisk box is have public ip, on the same box public ip terminates two ipsec tunnel which provides connectivity for clients. |
| 00:58.58 | cusco | so ... does asterisk also have a private ip? |
| 00:59.07 | volga629 | no |
| 00:59.45 | volga629 | I tried set to route on global config, but rtp traffic get lost |
| 01:00.35 | cusco | so... I don't understand |
| 01:00.47 | cusco | if you have a ipsec tunel, you have a private network, right? |
| 01:00.59 | volga629 | on another end of tunnel |
| 01:01.19 | volga629 | asterisk side is public only |
| 01:01.36 | cusco | so asterisk is not related to ipsec? |
| 01:02.09 | volga629 | asterisk sit on same box where tunnel is terminates |
| 01:02.40 | cusco | so that box must have a private ip and a public ip |
| 01:02.44 | cusco | rigth? |
| 01:02.52 | volga629 | no only public |
| 01:03.00 | cusco | .... |
| 01:03.11 | volga629 | terminate tunnel I don't need private |
| 01:03.41 | cusco | doens't it has at least two network interfaces? |
| 01:03.52 | volga629 | no only one |
| 01:04.38 | cusco | so how does the ipsec tunnel ends there? |
| 01:05.26 | volga629 | if you know openswan leftsourceip=pub ip leftsubnet=pubip/32 |
| 01:06.44 | cusco | sorry I dunno openswan, I am just trying to understand |
| 01:06.57 | *** join/#asterisk powerunits (b6b1bd47@gateway/web/freenode/ip.182.177.189.71) |
| 01:07.01 | powerunits | hello |
| 01:07.12 | cusco | skipping... asterisk always comunicates with public ip... |
| 01:07.13 | cusco | so? |
| 01:07.14 | powerunits | every one. please one quick question |
| 01:07.35 | cusco | volga629: have you tried rtp set debug |
| 01:07.41 | cusco | to seewhere rtp is going? |
| 01:07.58 | powerunits | how many concurrent calls asterisk can handle.... 1000 , 2000 or more then these... |
| 01:08.15 | [TK]D-Fender | As many as can dance on the head of a pin |
| 01:08.16 | powerunits | i mean max concurrent calls |
| 01:08.20 | volga629 | and side B private subnet, for this case I tried few nat setting like route and no nat, but didn't work |
| 01:08.23 | [TK]D-Fender | Or 314. I forget which... |
| 01:08.24 | jpsharp | Depends on what you're doing with the calls, how big your machine is, and the phase of the moon. |
| 01:08.59 | volga629 | no didn't tried rtp debug |
| 01:09.18 | volga629 | that might give better view hmm |
| 01:10.27 | cusco | volga629: try setting localnet line insip.conf |
| 01:10.35 | powerunits | jpsharp: machine has Processor AMD Opteron™ 6272 - RAM 64 GB DDR3 ECC .... and i dialing simple calls to external sip career.. |
| 01:10.43 | cusco | you can set it more than once for several networks |
| 01:10.51 | powerunits | and some calls are between extension to extension |
| 01:11.14 | cusco | volga629: I still don't understand your network, I do have a l2tp tunnel, and asterisk uses it sometimes, i have a subnet set in localnet |
| 01:11.15 | volga629 | localnet should be 127..... ? |
| 01:11.27 | cusco | localnet = 192.168.1.1 |
| 01:11.38 | cusco | localnet = 192.168.100.0/24 |
| 01:11.41 | cusco | localnet = 192.168.1.0/24 |
| 01:11.51 | [TK]D-Fender | NOT the first |
| 01:12.14 | jpsharp | powerunits: If you're not doing any transcoding of codecs, at least a hundred or so. Asterisk scaling is more of an art (maybe even voodoo magic) than hard science numbers. |
| 01:12.24 | volga629 | you mean private remote network, because asterisk machine is only public ip ? |
| 01:13.24 | cusco | volga629: not in my case but eve if it where, and it where able to comunicate with a private network, I would set it there |
| 01:13.57 | volga629 | ok I will try localnet=pubip/32 |
| 01:15.15 | [TK]D-Fender | total waste |
| 01:15.25 | [TK]D-Fender | Why would Asterisk need to know that *IT* is Local? |
| 01:15.32 | [TK]D-Fender | Since when is your Asterisk talking to ITSELF? |
| 01:15.38 | [TK]D-Fender | CRAZY PEOPLE |
| 01:16.23 | volga629 | I am just looking for right nat setting for my situation |
| 01:17.47 | [TK]D-Fender | volga629: You are guessing and not looking at the calls. Stop trying to guess the answer when you can't see the question. |
| 01:20.23 | volga629 | I tried use sip debug, but I can't see nothing special and I tired use tcpdump see if rtp traffic going right direction. I didn't tried rtp debug, |
| 01:20.28 | jpsharp | passes around pickles to throw to the wall. |
| 01:21.39 | volga629 | and call going IAX2 trough tunnel establish the call, but rtp get lost |
| 01:21.45 | factormystic | so, so close to getting this working |
| 01:22.19 | jpsharp | IAX2 doesn't use RTP. |
| 01:22.39 | factormystic | right now a call coming in is indeed routed to my sip soft phone (microsip), but attempting to answer the call doesn't actually pick up the line, and I don't know how to figure out what the issue is |
| 01:22.46 | factormystic | I don't even know if its an asterisk thing |
| 01:24.47 | volga629 | yes but on the it going to sip extension |
| 01:24.50 | volga629 | end |
| 01:28.25 | ChannelZ | factormystic: is that device communicating properly with Asterisk to begin with? IE have you made some test extensions that just playback a sound or something and tested it from that device? |
| 01:30.53 | factormystic | I can tell that the sip client is connected, but not other than that... that sounds like a good thing to test |
| 01:38.47 | volga629 | I checking debug and I see something like this allocating new SIP dialog for 52fa1ffc66567c971cd9f6ac74aa8e58@publicip:5060 - OPTIONS (No RTP) |
| 01:40.02 | volga629 | and what mean of Request 102: Match Found |
| 01:40.41 | [TK]D-Fender | volga629I tried use sip debug, but I can't see nothing special and I tired use tcpdump see if rtp traffic going right direction. I didn't tried rtp debug, <- Who said anything about YOUR EYES? |
| 01:42.26 | volga629 | yes because that I am right now going through again all logs to give example of conversation in debug mode |
| 01:43.29 | [TK]D-Fender | volga629: Logs? Go place a call NOW. This should have taken all of .... 1 minute |
| 01:51.34 | volga629 | unfortunately can't place call right now, nobody in office, 2 am in Europe. This is test from today evening http://fpaste.org/6r1j/ |
| 01:53.06 | [TK]D-Fender | Not even a whole call |
| 01:53.18 | [TK]D-Fender | Come back when you have something real and complete to sho us |
| 02:00.05 | volga629 | OK, I will do some full test in the morning, thank you |
| 02:03.19 | *** join/#asterisk beardy (~beardy@unaffiliated/beardy) |
| 02:07.51 | *** join/#asterisk MRH2 (~Thunderbi@188-220-37-249.zone11.bethere.co.uk) |
| 02:12.25 | *** join/#asterisk JackStorm (no@ip70-171-114-94.no.no.cox.net) |
| 03:01.10 | factormystic | ChannelZ: do you have a wait time or other special configuration to get gtalk to consider the call picked up |
| 03:05.36 | ChannelZ | Are you talking about Google Voice specifically? |
| 03:07.20 | factormystic | that's the ultimate goal yes |
| 03:07.28 | factormystic | to reduce the number of moving parts I'm doing a gtalk call to test |
| 03:08.09 | ChannelZ | well just google talk no, you shouldn't need to wait or do anything |
| 03:09.34 | ChannelZ | For Google Voice specifically, I do an Answer(1000) then SendDTMF(1) to get Google to accept the call and connect it. |
| 03:13.14 | factormystic | hm |
| 03:13.34 | factormystic | any other delays or special magic? google is still ringing after I pick up the call |
| 03:14.28 | factormystic | asterisk says: Locally bridging Motif/+[my number]-38e7 and SIP/microsip-00000004 |
| 03:14.50 | factormystic | and I can see the dialplan steps being executed |
| 03:16.12 | ChannelZ | hmm not that I know of, but it's possible motif might be broken. |
| 03:16.45 | factormystic | file did say that chan_motif had more fixes in svn, guess that's next |
| 03:35.49 | *** join/#asterisk Naikrovek (~wwhite@unaffiliated/naikrovek) |
| 03:36.52 | *** join/#asterisk Naikrovek (~wwhite@unaffiliated/naikrovek) |
| 03:40.51 | *** join/#asterisk Naikrovek (~wwhite@unaffiliated/naikrovek) |
| 03:41.14 | factormystic | well how about that |
| 03:41.42 | factormystic | rebuilt from source, now it's picking up |
| 03:41.53 | factormystic | hot diggity |
| 03:43.11 | ChannelZ | After fetching from svn you mean? |
| 03:43.15 | factormystic | yes |
| 03:46.23 | factormystic | interestingly, I don't need any kind of delay, just the dtmf 1 to pick up the call |
| 03:47.37 | factormystic | file, ChannelZ, others, thank you for the assistance |
| 03:59.41 | *** join/#asterisk Naikrovek (~wwhite@unaffiliated/naikrovek) |
| 04:02.07 | *** join/#asterisk jaxon007_ (~jaxon007_@14.97.224.178) |
| 04:03.58 | jaxon007_ | how can i increase the volume of an existing channel from AMI? |
| 04:37.50 | *** join/#asterisk Naikrovek (~wwhite@unaffiliated/naikrovek) |
| 04:42.43 | jaxon007_ | how can i increase the volume of an existing channel from AMI? |
| 04:54.26 | *** join/#asterisk kasanop (~kasanop@95-27-144-8.broadband.corbina.ru) |
| 04:56.22 | *** join/#asterisk fisted_ (~fisted@unaffiliated/fisted) |
| 05:12.26 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 06:09.11 | *** join/#asterisk FireAndIce (~administr@219.91.143.148) |
| 06:11.10 | FireAndIce | Hi everyone!! |
| 06:11.45 | FireAndIce | I'm trying to make a voice call from one android phone to another android phone using sip and asterisk. |
| 06:12.11 | FireAndIce | Asterisk is installed on my desktop. |
| 06:12.31 | FireAndIce | All these devices are behind a router.. |
| 06:13.16 | FireAndIce | What other configurations do I require in order to make a voice call |
| 06:13.19 | FireAndIce | ?? |
| 06:13.46 | FireAndIce | IMSdroid is sip client on the phones. |
| 06:26.00 | [TK]D-Fender | jaxon007_: run func volume against it. |
| 06:27.03 | *** join/#asterisk evilman_home (kvirc@2.92.147.93) |
| 06:27.11 | FireAndIce | I did not get you..? |
| 07:03.01 | *** join/#asterisk vlad_starkov (~vlad_star@ppp91-79-92-39.pppoe.mtu-net.ru) |
| 07:05.14 | *** join/#asterisk vlad_sta_ (~vlad_star@77.41.88.210) |
| 07:07.51 | jaxon007_ | @Fender: i did in such a way that , my channel was in confbridge so i redirect those channel to new context using channelRedirect and increase the volume there using volume function and again send those channel to context .. |
| 07:09.05 | jaxon007_ | sorry again send those channel to confbridge |
| 07:16.40 | jaxon007_ | @fender: i dont know how can run a function against channel?..thats why i follow this method..can u explain how run a function against existing channel?.. |
| 07:45.27 | *** join/#asterisk brdude (~brdude@c-24-7-76-160.hsd1.ca.comcast.net) |
| 08:01.22 | *** join/#asterisk vlad_starkov (~vlad_star@ppp91-79-92-39.pppoe.mtu-net.ru) |
| 08:04.30 | *** join/#asterisk vlad_sta_ (~vlad_star@77.41.88.210) |
| 08:09.25 | *** join/#asterisk danfromuk (~IceChat77@2.27.39.234) |
| 08:48.53 | *** join/#asterisk brian98 (~brian98@79.97.24.23) |
| 08:56.37 | *** join/#asterisk Vince-0 (~Vincent@41-133-16-59.dsl.mweb.co.za) |
| 09:05.29 | *** join/#asterisk vlad_starkov (~vlad_star@77.41.88.210) |
| 09:05.36 | *** join/#asterisk tzafrir_laptop (~tzafrir@local.xorcom.com) |
| 09:11.16 | *** join/#asterisk timahvo1 (~rogue@41.212.120.182) |
| 09:35.36 | *** join/#asterisk InHisName (~InHisName@96.25.37.57) |
| 09:53.49 | *** join/#asterisk AliRezaTaleghani (~taleghani@unaffiliated/AliRezaTaleghani) |
| 09:54.33 | AliRezaTaleghani | did u run A2Billing as "FastAGI" over a remote server? |
| 10:25.31 | *** join/#asterisk pa (~pa@unaffiliated/pa) |
| 11:51.39 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 11:51.39 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 10.9.0 (2012/10/08), 1.8.17.0 (2012/10/08), DAHDI-linux 2.6.1 (2012/04/20), DAHDI-tools 2.6.1 (2012/04/20), libpri 1.4.13 (2012/10/09) -=- Visit the official Asterisk wiki: wiki.asterisk.org |
| 11:52.24 | AliRezaTaleghani | :-/ I need a FastAGI geek |
| 11:52.42 | *** join/#asterisk deo (~dnepangue@112.198.79.108) |
| 11:52.46 | *** part/#asterisk deo (~dnepangue@112.198.79.108) |
| 12:33.24 | *** join/#asterisk Azrael808 (~peter@cpc17-walt12-2-0-cust657.13-2.cable.virginmedia.com) |
| 12:33.48 | *** join/#asterisk felipealmeida (~user@mvx-187-16-79-187.mundivox.com) |
| 12:51.03 | cusco | AliRezaTaleghani: what do you *really* need? |
| 12:55.11 | *** join/#asterisk vlad_starkov (~vlad_star@213.79.102.163) |
| 12:57.39 | *** join/#asterisk slav3_kitten (~kitten@unaffiliated/slav3-kitten/x-0866809) |
| 12:59.27 | *** join/#asterisk teloniusz (goldie@inferno.hell.pl) |
| 13:23.31 | volga629 | Hello Everyone, here sip debug of full conversation http://fpaste.org/wv3t/, also I tried set rtp debug |
| 13:29.13 | volga629 | nf_conntrack_sip 23504 1 nf_nat_sip Is possible that kernel module can cause for troubles ? |
| 13:33.17 | WIMPy | Yes, nf_nat_sip WILL cause trouble. Don't load it. |
| 13:39.01 | volga629 | yes I unloaded and at least I see right now that firewall allowing traffic, but RTP never going back to phone, I see only going from the the phone |
| 13:57.02 | WIMPy | Do you allow RTP comming in? |
| 13:57.09 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 13:58.09 | *** join/#asterisk gusto (~gusto@2001:a60:11ff:1200::42:4) |
| 13:58.16 | *** join/#asterisk timahvo1 (~rogue@41.212.120.182) |
| 14:01.35 | *** join/#asterisk acidfu (~inetrio.c@75-119-246-190.dsl.teksavvy.com) |
| 14:02.03 | *** join/#asterisk vlad_starkov (~vlad_star@213.79.102.163) |
| 14:08.31 | volga629 | yes |
| 14:08.37 | *** join/#asterisk vlad_starkov (~vlad_star@213.79.102.163) |
| 14:09.23 | *** join/#asterisk Neptu (~Neptu@c213-89-2-159.bredband.comhem.se) |
| 14:09.56 | volga629 | RTP die when conversation starts |
| 14:13.10 | volga629 | check_rtp_timeout: Disconnecting call 'SIP/24004-00000019' for lack of RTP activity in 31 seconds |
| 14:13.24 | volga629 | can't find where is the trouble |
| 14:33.16 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 14:33.16 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 14:39.23 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 14:42.28 | volga629 | http://fpaste.org/xyZp/ this rtp debug |
| 14:42.56 | volga629 | is start sending and after sent request and nwver back |
| 14:42.58 | volga629 | never |
| 14:52.08 | *** join/#asterisk sertaconay (~sertacona@unaffiliated/sertaconay) |
| 14:52.19 | kasanop | volga629: can you capture the SIP INVITE packet? |
| 14:54.22 | *** join/#asterisk hehol (~Adium@2a01:198:71d:0:21f:d0ff:fea1:568e) |
| 14:55.08 | *** join/#asterisk vlad_starkov (~vlad_star@213.79.102.163) |
| 15:10.14 | *** join/#asterisk wonderworld (~ww@dsdf-4db50d31.pool.mediaWays.net) |
| 15:10.47 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 15:15.18 | volga629 | res_rtp_asterisk.c: 0x4090ef0 -- Condition for learning hasn't exited, so reject the frame. |
| 15:17.58 | volga629 | here debug http://fpaste.org/8RLZ/ |
| 15:18.15 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 15:32.23 | *** join/#asterisk k610 (~Instantbi@host-78-129-3-116.brutele.be) |
| 15:47.12 | *** join/#asterisk vlad_starkov (~vlad_star@213.79.102.163) |
| 15:47.28 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 15:50.33 | *** join/#asterisk vlad_starkov (~vlad_star@213.79.102.163) |
| 15:59.19 | *** join/#asterisk TimeRider (~steve@188-220-34-144.dsl.cnl.uk.net) |
| 16:02.30 | *** join/#asterisk tzafrir_laptop (~tzafrir@local.xorcom.com) |
| 16:35.46 | *** join/#asterisk felipealmeida (~user@mvx-187-16-79-187.mundivox.com) |
| 16:53.06 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 17:14.26 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 17:16.23 | *** join/#asterisk danfromuk (~IceChat77@2.27.13.100) |
| 17:30.49 | *** join/#asterisk felipealmeida (~user@mvx-187-16-79-187.mundivox.com) |
| 17:39.40 | *** join/#asterisk felipealmeida (~user@mvx-187-16-79-187.mundivox.com) |
| 17:41.18 | *** join/#asterisk brdude (~brdude@c-24-7-76-160.hsd1.ca.comcast.net) |
| 17:47.04 | *** join/#asterisk vlad_starkov (~vlad_star@213.79.102.163) |
| 18:04.06 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 18:36.51 | *** join/#asterisk l2trace99 (~l2trace@70-119-141-5.res.bhn.net) |
| 18:56.20 | *** join/#asterisk dolcea-xoom (~androirc@232-197-235-201.fibertel.com.ar) |
| 19:11.22 | *** join/#asterisk slidesinger-lt (~jtatum@c-174-57-5-70.hsd1.nj.comcast.net) |
| 19:18.05 | *** join/#asterisk amessina (~amessina@2001:470:c1dc:7779:d6be:d9ff:fe8d:7c1e) |
| 19:31.43 | *** join/#asterisk timahvo1 (~rogue@41.212.120.182) |
| 19:49.25 | *** join/#asterisk danfromuk (~IceChat77@2.27.13.100) |
| 19:50.25 | *** join/#asterisk gandhijee_ (~akp@50-195-36-113-static.hfc.comcastbusiness.net) |
| 19:50.28 | *** join/#asterisk jetlag (~jetlag@pool-71-168-253-208.cmdnnj.east.verizon.net) |
| 20:02.52 | *** join/#asterisk tzafrir_laptop (~tzafrir@bzq-218-155-148.cablep.bezeqint.net) |
| 20:15.09 | *** join/#asterisk Azrael808 (~peter@cpc17-walt12-2-0-cust657.13-2.cable.virginmedia.com) |
| 20:16.10 | *** join/#asterisk sertaconay (~sertacona@unaffiliated/sertaconay) |
| 20:38.50 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 20:56.01 | *** join/#asterisk amessina (~amessina@2001:470:c1dc:7779:d6be:d9ff:fe8d:7c1e) |
| 20:59.38 | *** join/#asterisk elico (~Thunderbi@109.64.221.76) |
| 21:18.51 | *** join/#asterisk elico (~Thunderbi@109.64.221.76) |
| 21:48.38 | *** join/#asterisk vlad_starkov (~vlad_star@ppp91-79-64-160.pppoe.mtu-net.ru) |
| 22:18.02 | dolcea-xoom | Has anyone tested an usb phone with asterisk? |
| 22:20.14 | *** join/#asterisk j4m3s_ (~j4m3s_@pdpc/supporter/active/j4m3s) |
| 22:20.43 | WIMPy | Do you have any idea what "usb phone" could mean? |
| 22:21.17 | dolcea-xoom | The ones you attach to a pc, meant to use with skype |
| 22:21.20 | dolcea-xoom | Those ones |
| 22:21.53 | WIMPy | And what do you think what kind of device that is? |
| 22:21.57 | elico | dolcea-xoom: it's just hardware... |
| 22:23.18 | dolcea-xoom | So i should configure the system, not asterisk? |
| 22:23.47 | elico | you need some softphone that knows how to work with it |
| 22:23.55 | elico | for windows xlite can be nice |
| 22:24.19 | elico | for linux it's something else sine it needs to recognize the usb "phone" |
| 22:25.24 | WIMPy | Which is just a USB sound"card". |
| 22:26.13 | elico | exactly. |
| 22:46.28 | *** join/#asterisk iabosi (~iabosi@centauri.iabosi.com) |
| 22:47.12 | iabosi | anyone here have experience getting asterisk-gui working on centos? it seems i have configured everything and still cannot connect to the mini http server |
| 22:58.34 | *** join/#asterisk vlad_starkov (~vlad_star@ppp91-79-64-160.pppoe.mtu-net.ru) |
| 22:59.23 | *** join/#asterisk vlad_sta_ (~vlad_star@79.164.131.116) |
| 23:04.23 | [TK]D-Fender | iabosi: 2 things. #1 : Asterisk-GUI is for all intents and purposes "dead". Unless you in it for masochistic fun, it doesn't exactly have a future. #2. PASTEBIN is your friend. Show us your configs, that applicable modules are loaded, files are present in the right place, see if * has bound a port, etc.... |
| 23:04.24 | [TK]D-Fender | ~pb |
| 23:04.25 | infobot | 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/; or install pastebinit with yum or aptitude. |
| 23:04.27 | [TK]D-Fender | ^^^ |
| 23:05.39 | iabosi | well I was originally configuring my extensions and such via the configuration files but figured the gui may be interesting to work with. |
| 23:06.18 | iabosi | before I get this off the ground I would like to approach it in the best way, so would you suggest I continue just managing manually via config files? |
| 23:07.16 | [TK]D-Fender | I recommend not using one that is dead. |
| 23:07.34 | [TK]D-Fender | You need to be sure of your own priorities |
| 23:08.53 | iabosi | Great, thanks [TK]D-Fender |
| 23:14.42 | Kobaz | weird |
| 23:14.47 | Kobaz | asterisk supposidly isn't doing anything |
| 23:14.52 | Kobaz | but it's chewing 50% cpu |
| 23:14.57 | Kobaz | no channels |
| 23:20.19 | ChannelZ | It's sending all your files to spammers. |
| 23:20.45 | Kobaz | probably |
| 23:21.19 | ChannelZ | Or downloading porn. That could be a bonus. |
| 23:46.01 | *** join/#asterisk Nemus (~Nemus@c-67-161-251-38.hsd1.ut.comcast.net) |
| 23:46.19 | *** join/#asterisk Defraz (~Defraz@184-155-136-196.cpe.cableone.net) |
| 23:47.22 | *** join/#asterisk vlad_starkov (~vlad_star@79.164.131.116) |
| 23:48.41 | volga629 | Hello Everyone, this debug http://fpaste.org/dMak/ about RTP issue from yesterday |