| 00:01.01 | aceio | looking at cli output i am missing some files |
| 00:02.45 | p3nguin | Create the files or change the names that are trying to be played. |
| 00:13.57 | ManxPower | the sound files should be mono, 8Khz, 16-bit |
| 00:20.55 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 00:33.51 | Jumpie | ManxPower |
| 00:33.56 | Jumpie | how do you do the link for that book again? |
| 00:34.02 | Jumpie | ~book ? |
| 00:34.03 | 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 |
| 00:34.07 | Jumpie | lol..good guess |
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| 00:54.36 | TJNII | Wow. wgetpaste is a nifty tool. |
| 00:55.10 | TJNII | Dude, and it's even written in bash. That's my kind of hackery. |
| 01:00.05 | p3nguin | I like pastebinit better, since wgetpaste doesn't seem to have support for pastebin.com. |
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| 01:26.48 | aceio | hi all |
| 01:27.16 | aceio | <PROTECTED> |
| 01:28.07 | aceio | http://pastebin.com/2k2mF7GU |
| 01:28.13 | aceio | need some help |
| 01:32.47 | p3nguin | aceio: This has already been addressed. |
| 01:33.07 | p3nguin | (1902.44) <p3nguin> Create the files or change the names that are trying to be played. |
| 01:33.10 | p3nguin | (1913.57) <ManxPower> the sound files should be mono, 8Khz, 16-bit |
| 01:35.45 | aceio | p3nguin:O really |
| 01:36.11 | aceio | p3nguin: were can i find it |
| 01:36.48 | PhoenixMage | finally got the Cisco 7925 working |
| 01:37.11 | p3nguin | It wasn't working before? |
| 01:38.17 | PhoenixMage | p3nguin: Trying to work out how to make it work |
| 01:38.31 | PhoenixMage | the sccp.conf isnt particularly well documented |
| 01:39.03 | p3nguin | Yeah, I agree with that. Also, the documenation on their site seems to be very outdated and useless. |
| 01:39.44 | p3nguin | Do you still need help with sccp.conf, or you've figured it out already? |
| 01:39.55 | PhoenixMage | I think I have it worked out |
| 01:40.15 | PhoenixMage | Just trying to workout what all the keywords mean for a line confiuration |
| 01:53.35 | PhoenixMage | how do you d/c the cli after running asterisk -r? |
| 01:54.41 | p3nguin | exit |
| 01:54.56 | p3nguin | or SIGINT should do it, too. |
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| 02:15.52 | aceio | PhoenixMage: thx you |
| 02:16.11 | aceio | and luck?? |
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| 05:05.12 | PhoenixMage | why put a Hangup() for an extension in your extensions.conf? |
| 05:13.28 | Jumpie | yea isnt it a built in routine |
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| 07:00.04 | Milad | in my asterisk, running dialplan crashed if a sound file dose not exist , |
| 07:03.31 | Milad | is it okey ? |
| 07:03.51 | ChannelZ | crashed? |
| 07:04.15 | Milad | I mean channel hangup :D |
| 07:04.58 | ChannelZ | hmm I think the Playback() should just fail but not terminate the channel |
| 07:05.07 | ChannelZ | Are you sure your priorities aren't messed up or it exited for some other reason? |
| 07:05.30 | p3nguin | Maybe some actual CLI verbose output would be helpful. |
| 07:06.06 | ChannelZ | s/aren't/are/ |
| 07:06.18 | ChannelZ | err no I was right the first time |
| 07:06.21 | ChannelZ | rubs his eyes |
| 07:06.51 | Milad | ChannelZ, http://pastebin.com/xu7dbSYu |
| 07:07.58 | p3nguin | What is on train-ivr, 5439, 5? |
| 07:09.02 | Milad | p3nguin, http://pastebin.com/qVCLg1ez |
| 07:10.45 | p3nguin | It makes sense that the AGI can't run since the Read() failed due to the lack of the sound file. |
| 07:11.17 | p3nguin | I bet if you remove the sound file and let the Read() go ahead and accept input, the AGI will run. |
| 07:11.58 | *** join/#asterisk gospch (~gospch@unaffiliated/gospch) |
| 07:13.07 | Milad | p3nguin, I have same problem in queue , I removed queue-thankyou file and when say announcement to callers and want to play queue-thankyou channel disconnected ! |
| 07:13.29 | p3nguin | What do you have autofallthrough set to? |
| 07:14.30 | *** join/#asterisk tzafrir_laptop (~tzafrir@local.xorcom.com) |
| 07:14.59 | ChannelZ | I'm curious about "-- User disconnected" |
| 07:16.43 | Milad | nope i don't have a any autofallthrough |
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| 07:20.05 | Milad | ChannelZ, it's with more debug http://pastebin.com/1mGkrpH5 |
| 07:23.34 | ChannelZ | What version of Asterisk is this? |
| 07:23.55 | Milad | 1.4.26.3 |
| 07:24.04 | ChannelZ | hmm |
| 07:24.47 | ChannelZ | It still seems like your phone is hanging up the call, not Asterisk. |
| 07:25.21 | ChannelZ | I'm running 1.6.1 but when I do a Read() statement with a missing sound, the call does not terminate automatically. If I hangup however, I get "-- User disconnected" as you are. |
| 07:27.30 | Milad | as I said also I had a problem on queue which comes from cisco as 5300 |
| 07:27.41 | ChannelZ | I see you didn't Answer() the channel, maybe something is going awry |
| 07:27.44 | Milad | yup it's so strange fore me |
| 07:28.08 | Milad | mm check it now |
| 07:29.42 | Milad | nope it did not work |
| 07:29.55 | Milad | anyway thanks for your clues |
| 07:30.26 | ChannelZ | It begs the question, what happens if the audio file you're specifying in Read() DOES exist? |
| 07:32.55 | Milad | it's run agi and ... |
| 07:34.55 | ChannelZ | So why exactly are you wanting to play a file that doesn't exist in your Read() ? |
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| 07:46.31 | *** join/#asterisk aceio (~5ad056d3@gateway/web/freenode/x-zxxegzlesdxszpnf) |
| 07:47.58 | aceio | i have a warning message Unable to open main-menu (format 0x8 (alaw)): No such file or directory |
| 07:48.22 | aceio | main-menu can't be found |
| 07:48.37 | ChannelZ | again, a missing sound? |
| 07:49.57 | aceio | ChannelZ: so were can i locate it |
| 07:50.17 | Milad | ChannelZ, :D , just checking what happen if my sound file missed ! but my real problem is about queue announce ! I don't want to play queue-thankyou ! |
| 07:50.43 | Milad | I solve my problem on queue by copy silence sound replace with queue-thankyou |
| 07:51.21 | Milad | but I want to know it's usual to disconnect channel if sound file not found |
| 07:54.17 | aceio | iam also missing the welcome sound |
| 07:55.31 | tzafrir_laptop | aceio, where is that actual file? What's its name and full path? |
| 07:55.35 | tzafrir_laptop | (main-menu) |
| 07:55.58 | *** join/#asterisk The-Bat (~The-Bat@122.182.0.38) |
| 07:57.18 | aceio | exten => _xxxx,n,Background(main-menu,m) |
| 07:57.33 | aceio | it setup like this in my dialplan |
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| 08:02.29 | tzafrir_laptop | aceio, where does the sound file main-menu actually reside? |
| 08:02.49 | tzafrir_laptop | (generally, please try to show us that it should work) |
| 08:05.31 | *** join/#asterisk aceio (~5ad056d3@gateway/web/freenode/x-rnvchxexfujwdarz) |
| 08:06.18 | aceio | the main-menu file i can't onthe system |
| 08:06.36 | ChannelZ | Did you install any of the sound sets? |
| 08:08.10 | aceio | no i didn't select |
| 08:09.30 | *** join/#asterisk pif (~ldm@LSt-Amand-152-33-2-89.w193-251.abo.wanadoo.fr) |
| 08:10.40 | aceio | so i have selected all the english sound core |
| 08:12.53 | tzafrir_laptop | aceio, why do you expect it to be? (why do you expect to have a file with that name) |
| 08:13.09 | tzafrir_laptop | Is it a file you recorded, or a file that should be included with asterisk? |
| 08:13.41 | aceio | according the asterisk yes |
| 08:16.34 | tzafrir_laptop | aceio, the sound file main-menu is part of the 'extra' sounds packages |
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| 09:14.45 | ethel29 | guys, can some one shed some light on this please, i was given a task to create a predictive dialer, according to the client they prefer gsm/g729 sip trunks now i'm worry about the CPU utilization because like vicidialer i use meetme + local channels to accomplish the task, now the question is, Is Quad core is enough or i should stick to my original plan buying TCB transcoder card |
| 09:15.35 | ethel29 | my topology LAN - g.711u -> SIP trunks GSM/G.729 -> ITSP |
| 09:16.07 | ethel29 | thanks |
| 09:22.04 | carrar | Any amount of g.729 transcoding is gonna be better off on hardware |
| 09:24.11 | coppice | ethel29: dunno how fast the Digium G.729 is, but a single core of a modern quad core should encode/decode more channels than one of those TCB cards |
| 09:24.44 | coppice | ethel29: if you really want a lot of channels, sangoma has a new transcoding card with high capacilty |
| 09:26.48 | *** join/#asterisk tlir (~tlir@bzq-84-110-98-116.red.bezeqint.net) |
| 09:30.24 | BANSAL | hello everyone .. |
| 09:30.37 | BANSAL | I am new asterisk and trying to install it .. |
| 09:30.47 | BANSAL | I am following http://www.howtoforge.com/how-to-install-asterisk-for-your-first-pbx-solution |
| 09:31.01 | BANSAL | but not getting any result ... |
| 09:31.46 | BANSAL | when I run ./configure command .. it shows some errors as configure: error: no acceptable C compiler found in $PATH |
| 09:31.47 | BANSAL | See `config.log' for more details. |
| 09:32.00 | BANSAL | can anyone help me ... |
| 09:34.14 | ChannelZ | sounds like you have no acceptable C compiler installed. |
| 09:35.07 | carrar | heh |
| 09:35.16 | BANSAL | ChannelZ: can you help me how do I do that .. I am using fedora 12 |
| 09:35.52 | ChannelZ | well you need to install gcc - however one does that with RPM, I haven't used it in ages |
| 09:36.10 | ChannelZ | ncurses-dev too probably.. autoconf.. |
| 09:48.04 | ChannelZ | you're welcome |
| 09:58.34 | tlir | what could be the reason that when ext enters a queue it starts recording the ext-queue channel? (I have queues set to not record, though the extension is set to record always, both directions) |
| 10:01.40 | *** part/#asterisk henk (~henk@netwichtig.de) |
| 10:06.48 | tzafrir_laptop | ChannelZ, for the record: yum install $PACKAGE |
| 10:06.58 | tzafrir_laptop | and also: s/-dev/-devel/ |
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| 10:09.01 | jamicque | Hi, is there any way to change SIP "To:" header when preforming dial? |
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| 11:10.11 | ethel29 | carrar: thanks for your answer and to you steve... |
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| 12:00.48 | *** join/#asterisk talntid (~t@173.160.174.126) |
| 12:01.35 | talntid | if I wanted to make an extension where... if I press "1" it adds 1 to a variable, and speaks it, or if I press 2, it subtracts one from a variable and speaks it, is this possible? |
| 12:02.31 | talntid | so if I press 1-2-1-2-1-1 it would say "1" "0" "1" "0" "1" "2" |
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| 12:10.21 | piparkuka | hi guys |
| 12:11.06 | piparkuka | i have a problem to install asterisk via yjm repository |
| 12:11.35 | piparkuka | it seems to be names problem |
| 12:11.50 | piparkuka | Missing Dependency: kernel-x86_64 = 2.6.18-164.15.1.el5 is needed by package kmod-dahdi-linux-fwload-vpmadt032-2.3.0-1_centos5.2.6.18_164.15.1.el5.x86_64 (digium-current) |
| 12:12.15 | piparkuka | kmod-dahdi-linux-fwload-vpmadt032-2.3.0-1_centos5.2.6.18_164.15.1.el5.x86_64 demands kernel-x86_64 instead of kernel.x86_64 |
| 12:12.44 | piparkuka | how can i contact with the person that can fix it? |
| 12:17.48 | *** part/#asterisk tlir (~tlir@bzq-84-110-98-116.red.bezeqint.net) |
| 12:29.03 | *** part/#asterisk MiserySoft (~MiserySof@host81-139-171-30.in-addr.btopenworld.com) |
| 12:38.27 | tzafrir_laptop | piparkuka, generally file a bug |
| 12:42.47 | tzafrir_laptop | talntid, Simply use standard channel variables |
| 12:43.09 | talntid | I am not familiar with them |
| 12:49.31 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 12:53.54 | tzafrir_laptop | talntid, see how the language is set in the demo extensions.conf |
| 12:54.17 | tzafrir_laptop | context [demo], see the action for '2' (or is it '3') that sets the language to French |
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| 13:04.36 | talntid | will do |
| 13:04.37 | talntid | thanks |
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| 13:30.18 | piparkuka | tzafrir, what do u mean - file bug? |
| 13:47.41 | *** join/#asterisk hluesea (~hluesea@88.247.127.66) |
| 13:47.46 | hluesea | hello |
| 13:47.56 | hluesea | i have a problem about iax2 |
| 13:48.34 | hluesea | and my service is running at this time i take these warnings and i can't connect the 2 servers via iax2 ; |
| 13:48.36 | hluesea | [May 16 16:45:54] WARNING[6109]: channel.c:4014 ast_request: No channel type registered for 'IAX2' |
| 13:48.36 | hluesea | [May 16 16:45:54] WARNING[6109]: app_dial.c:1518 dial_exec_full: Unable to create channel of type 'IAX2' (cause 66 - Channel not implemented) |
| 13:48.36 | hluesea | <PROTECTED> |
| 13:49.01 | hluesea | could you know the problem or solution ? |
| 13:51.33 | jaytee | hluesea, looks like one or both of your servers isn't registering with the other |
| 13:52.31 | hluesea | how can i solve this problem ? |
| 13:52.53 | hluesea | service is running and i looked up modules, modules.conf, iax2.conf-s |
| 13:52.58 | hluesea | but i can't findanything |
| 13:53.13 | hluesea | i need a quick solution, customers calling rapidly :( |
| 13:53.44 | jaytee | define an iax peer in each iax.conf file and add a register line for that peer in the other server's iax.conf file |
| 13:53.48 | jaytee | it's in the book |
| 13:53.51 | jaytee | ~book |
| 13:53.53 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 13:59.21 | tzafrir_laptop | piparkuka, I don't have time to look into this now, but the .src.rpm is at http://packages.asterisk.org/centos/5/current/SRPMS/dahdi-linux-kmod-2.3.0-1_centos5.2.6.18_164.15.1.el5.src.rpm |
| 14:01.22 | hluesea | <PROTECTED> |
| 14:01.29 | hluesea | but alittle time ago it gone |
| 14:01.57 | jaytee | hluesea, have you tried doing an IAX2 reload on both servers then? |
| 14:02.08 | hluesea | and i can't see any iax2 peers, i checked configurations and that is looking good for boot server |
| 14:02.12 | hluesea | i tried reload |
| 14:02.52 | jaytee | hluesea, what else changed since they were working? |
| 14:04.10 | hluesea | upps i iax2 reloaded and i take like error from 1 server :( http://pastebin.com/aP6cAWK7 |
| 14:04.39 | hluesea | i added g729 for iax2 protocol but i have seen this problem i changed again to ulaw |
| 14:06.51 | jaytee | hluesea, if you get a segfault everytime you reload IAX then there's something messed up in your configs most likely |
| 14:07.27 | hluesea | how can i fix it up ? |
| 14:07.53 | jaytee | if it was working before then undo whatever you did that broke it |
| 14:09.00 | hluesea | is it based on iax2.conf or chan_iax2.so problem ? |
| 14:10.03 | jaytee | could be either. did you change anything in iax.conf? |
| 14:10.18 | hluesea | yes and i undo that changes |
| 14:14.17 | jaytee | hluesea, are you running standard asterisk or asteriskNOW or trixbox? |
| 14:14.30 | hluesea | asterisk 1.6.0 |
| 14:14.54 | hluesea | so that i am asterisk channel :) |
| 14:15.52 | jaytee | hluesea, do you get a segfault everytime you reload iax2? |
| 14:16.05 | hluesea | no i take only that time |
| 14:16.16 | hluesea | and i can't anything from google request |
| 14:17.14 | *** join/#asterisk KavanS (~KavanS@static-173-50-141-22.ptldor.fios.verizon.net) |
| 14:17.24 | jaytee | hluesea, have you tried complete restarts of the servers? |
| 14:17.39 | hluesea | i am trying now |
| 14:18.09 | *** join/#asterisk centoslinux (~centoslin@s0021-0018.dsl.start.no) |
| 14:18.12 | jaytee | hluesea, also double check your iax.conf file on the affected server |
| 14:18.27 | jaytee | make sure that there are no invalid entries or typos |
| 14:18.47 | *** join/#asterisk JunK-Y (~junky@64.15.77.94) |
| 14:18.59 | hluesea | i checked 3 times and i checked it before backed up conf files to |
| 14:19.00 | hluesea | too* |
| 14:19.09 | hluesea | all of them is the similar |
| 14:27.24 | *** join/#asterisk ttwhy (~tekkno@p4FECEF1D.dip.t-dialin.net) |
| 14:30.22 | hluesea | <PROTECTED> |
| 14:31.01 | hluesea | is reinstalling asterisk solve the problem ? |
| 14:31.41 | jaytee | hluesea, it might but you don't seem to know what the problem actually is |
| 14:32.14 | jaytee | hluesea, do you get any errors when reloading iax2? |
| 14:32.15 | *** join/#asterisk retentiveboy (~pdugas@74-95-28-37-Atlanta.hfc.comcastbusiness.net) |
| 14:38.45 | hluesea | when i tried to reload iax2; i disconnect |
| 14:38.59 | hluesea | and yes i do not know what is the main problem |
| 14:39.07 | hluesea | every thing looking ok |
| 14:39.10 | jaytee | disconnect? |
| 14:39.36 | hluesea | i disconnect from asterisk |
| 14:41.01 | jaytee | hluesea, have you looked in /var/logs/asterisk/messages to see what's in there when you disconnect? |
| 14:42.14 | hluesea | it is empty |
| 14:43.18 | jaytee | then you must not have logging turned on in logger.conf |
| 14:45.06 | hluesea | opened |
| 14:45.14 | hluesea | ok i am looking now |
| 14:46.18 | *** join/#asterisk BANSAL (~BANSAL@117.199.118.186) |
| 14:50.41 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
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| 14:59.40 | hluesea | jaytee -- can u see anything from this messages.log http://pastebin.com/eZJNWnAf for interrupt iax2 working ? |
| 15:01.03 | *** join/#asterisk rare1980_ (~rare1980@119.158.30.251) |
| 15:01.10 | jaytee | hluesea, yes |
| 15:01.31 | jaytee | [May 16 17:55:15] WARNING[5491] chan_iax2.c: Unable to open IAX timing interface: No such file or directory |
| 15:01.32 | [TK]D-Fender | [May 16 17:55:15] WARNING[5491] config.c: parse error: No category context for line 1 of /etc/asterisk/iax.conf |
| 15:01.34 | [TK]D-Fender | [May 16 17:55:15] ERROR[5491] chan_iax2.c: Unable to load config iax.conf |
| 15:01.50 | jaytee | and that one too |
| 15:01.56 | hluesea | how can i fix it |
| 15:02.02 | hluesea | TK d fender welcome :) |
| 15:02.02 | jaytee | plus you have problems with dahdi as well |
| 15:02.13 | hluesea | i am not using dahdi for any channel |
| 15:02.16 | [TK]D-Fender | hluesea: It's telling you to your face the first line of your config si bad. GO LOOK AT IT |
| 15:02.21 | hluesea | actually i installed it for the timing issue |
| 15:02.31 | *** join/#asterisk Poincare (~jefffnode@213.219.184.23.wls.msr03bkc3.adsl.static.edpnet.net) |
| 15:03.06 | hluesea | D-Fender i have a iax2 connection problem for one server to another |
| 15:03.23 | hluesea | i know the first line sql problem |
| 15:03.30 | [TK]D-Fender | hluesea: You have an error in the first line of your config. End of story |
| 15:03.31 | hluesea | i am configuring it |
| 15:03.40 | jaytee | hluesea, and TK just told you the problem is on line 1 of your iax.conf file so go fix it |
| 15:04.04 | hluesea | ok, how can i fix the iax2 problem ? |
| 15:04.14 | rare1980_ | hi all,, ztdummy is loaded properly but still i can't use meetme for confernce... please help.... i am getting this error msg on asterisk cli app_meetme.c:778 build_conf: Unable to open pseudo device |
| 15:04.15 | jaytee | pastebin your iax.conf file |
| 15:06.09 | *** join/#asterisk uqlev (~yuriy@91.184.221.31) |
| 15:08.13 | rare1980_ | any help on this please? |
| 15:11.17 | hluesea | http://pastebin.com/S0rRw9v9 |
| 15:11.23 | hluesea | my iax2.conf files |
| 15:11.23 | *** join/#asterisk smooth_penguin (~smoove@59.95.15.122) |
| 15:12.44 | jaytee | hluesea, you are missing the [general] context in your iax.conf files |
| 15:12.45 | pabelanger_ | rare1980_: permission issue. Does |
| 15:12.48 | [TK]D-Fender | hluesea: Is that ALL of it? |
| 15:13.02 | hluesea | after the general part |
| 15:13.04 | pabelanger_ | rare1980_: does Asterisk have permission to open it |
| 15:13.12 | jaytee | groans |
| 15:13.14 | [TK]D-Fender | [11:12]<jaytee>hluesea, you are missing the [general] context in your iax.conf files <-- If you're missing this, please place your head firmly in front of the barrel |
| 15:13.21 | jaytee | lol |
| 15:13.36 | jaytee | ~book |
| 15:13.47 | 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 |
| 15:13.47 | hluesea | :D i didn't include it i am adding the bastebin now |
| 15:13.50 | jaytee | bastebin? mmmmm, turkey!!! |
| 15:13.56 | [TK]D-Fender | hluesea: You have a problem with your file. It says its bad from the VERY FIRST FUCKING LINE. aND YOU THINK ITS A GOOD IDEA TO filter teh evidence? |
| 15:14.02 | jaytee | hluesea, ^^^^^^^^ |
| 15:14.21 | [TK]D-Fender | bursts a vein |
| 15:15.22 | rare1980_ | pabelanger: how can i check it out if asterisk is giving permission to open it |
| 15:15.23 | rare1980_ | ? |
| 15:15.58 | hluesea | D-Fender that is the another problem as you see, i have a very emergency iax2 problem i am pasting the pastebin all files immediately |
| 15:17.10 | pabelanger_ | rare1980_: lsmod |
| 15:17.20 | pabelanger_ | rare1980_: ls -la /dev/zap/pseudo |
| 15:17.47 | rare1980_ | i get this |
| 15:17.49 | rare1980_ | crw-rw---- 1 root root 196, 255 2010-05-16 08:06 /dev/zap/pseudo |
| 15:18.59 | pabelanger_ | rare1980_: what user are you starting Asterisk with? |
| 15:19.21 | rare1980_ | asterisk 1.4.22 |
| 15:20.07 | pabelanger_ | rare1980_: user.... not version |
| 15:20.21 | hluesea | iax2.conf files here http://pastebin.com/H9yrvD0j |
| 15:20.31 | rare1980_ | ooh sorry |
| 15:20.38 | rare1980_ | i m starting it with root |
| 15:22.15 | *** join/#asterisk nighty^ (~nighty@x122091.ppp.asahi-net.or.jp) |
| 15:23.25 | rare1980_ | pabelanger: please could you tell me wht else could be? or do you think if this is coz of permission? |
| 15:23.32 | pabelanger_ | rare1980_: you need DAHDI |
| 15:23.32 | jaytee | hluesea, does this line actually exist in the file? 1 st server IAX2.conf |
| 15:24.04 | rare1980_ | pabelanger: i have need is zaptel |
| 15:24.06 | rare1980_ | :( |
| 15:24.28 | rare1980_ | that is why from last 3 days i m working on it. and could not reslove it |
| 15:24.29 | rare1980_ | :( |
| 15:24.34 | pabelanger_ | rare1980_: Zaptel is not supported after version 1.4.21. |
| 15:24.38 | pabelanger_ | install DAHDI |
| 15:24.41 | rare1980_ | hummmmmmm |
| 15:24.43 | hluesea | no i added for the pastebin for simplify the understanding |
| 15:24.45 | rare1980_ | ooh |
| 15:24.51 | rare1980_ | then this could be the reason |
| 15:25.08 | rare1980_ | can i down grade my asterisk ? |
| 15:25.29 | rare1980_ | to asterisk 1.4.21 |
| 15:25.32 | tzafrir_laptop | pabelanger_, actually, zaptel is supported for asterisk 1.4.x |
| 15:25.56 | tzafrir_laptop | but make sure that this version of asterisk is built vs. zaptel and not vs. dahdi |
| 15:26.10 | *** join/#asterisk LemensTS (~LemensTS@71.86.32.146) |
| 15:27.06 | tzafrir_laptop | rare1980_, to check that: strings /usr/lib/asterisk/modules/chan_dahdi.so | grep /dev |
| 15:27.25 | tzafrir_laptop | If you see there /dev/zap - it's built vs. zaptel . /dev/dahdi: vs. dahdi |
| 15:27.39 | rare1980_ | strings: '/usr/lib/asterisk/modules/chan_dahdi.so': No such file |
| 15:27.44 | rare1980_ | i got this |
| 15:28.14 | tzafrir_laptop | rare1980_, is that where asterisk is installed? |
| 15:28.22 | pabelanger_ | tzafrir_laptop: Even bug fixes? |
| 15:28.26 | rare1980_ | tzafrir_latop: and pabelanger: thanks for your help.. let me try installing older version of asterisk |
| 15:28.40 | rare1980_ | thanks |
| 15:28.45 | tzafrir_laptop | rare1980_, have you built that version yourself? |
| 15:28.51 | rare1980_ | yes |
| 15:29.02 | tzafrir_laptop | pabelanger_, zaptel itself is hardly maintained. But if it works, it works |
| 15:29.14 | rare1980_ | humm |
| 15:29.20 | rare1980_ | rite but i have to try to learn |
| 15:29.28 | rare1980_ | may be one day i can learn :) |
| 15:29.40 | pabelanger_ | tzafrir_laptop: Agree, if was looking at it from a support PoV :) |
| 15:29.54 | tzafrir_laptop | rare1980_, is that the modules directory? Are there any other files under /usr/lib/asterisk/modules ? |
| 15:30.16 | rare1980_ | let me check |
| 15:30.49 | rare1980_ | app_addon_sql_mysql.so app_saycountpl.so cdr_addon_mysql.so chan_ooh323.so format_mp3.so res_config_mysql.so |
| 15:30.54 | rare1980_ | these files are in this dir |
| 15:31.02 | tzafrir_laptop | Only those? |
| 15:31.04 | rare1980_ | yes |
| 15:31.14 | tzafrir_laptop | That is: only the ones from addons |
| 15:31.17 | tzafrir_laptop | That's odd |
| 15:31.27 | rare1980_ | humm |
| 15:31.34 | pabelanger_ | rare1980_: pastebin your asterisk.conf file |
| 15:31.38 | rare1980_ | sure |
| 15:31.40 | rare1980_ | hold on |
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| 15:33.40 | rare1980_ | pabelanger: http://pastebin.com/AMAzfcU9 |
| 15:33.50 | rare1980_ | here is the file on pastebin |
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| 15:38.47 | hluesea | TK -D-Fender , jaytee can you found any thing for the iax2 problem :( |
| 15:39.13 | jaytee | hluesea, nope. |
| 15:39.26 | jaytee | can't determine what's wrong from an "edited" pastebin |
| 15:40.20 | jaytee | hluesea, either your iax.conf file is corrupt or you aren't showing us the complete file MINUS the crap you typed in at the beginning |
| 15:40.47 | hluesea | i entered the my used confs end send pastebin link |
| 15:41.33 | hluesea | ok in here http://pastebin.com/H9yrvD0j my using configurations 1 st iax2.conf label include first server confs and 2nd iax2.conf labelled 2. server iax2.conf |
| 15:42.09 | jaytee | hluesea, your logs were telling you that the asterisk parser could not find a context category on line 1. I don't need to look at a repaste of the same crap |
| 15:42.55 | jaytee | hluesea, you've either got a corrupt iax.conf file or you have crap at the beginning of the file that isn't commented out and it's causing the iax load to fail |
| 15:46.07 | [TK]D-Fender | [11:01]<[TK]D-Fender>[May 16 17:55:15] ERROR[5491] chan_iax2.c: Unable to load config iax.conf <---- THIS DOESN'T SAY IAX2.CONF |
| 15:46.22 | [TK]D-Fender | SPHELLING FALE |
| 15:46.39 | [TK]D-Fender | reaches for his ClueBat (tm) |
| 15:47.29 | hluesea | i am reloading and surveying again |
| 15:49.40 | jaytee | hluesea, in simple english make sure your file is named iax.conf and NOT iax2.conf |
| 15:50.09 | hluesea | :) i looked again and i saw this is right |
| 15:50.13 | hluesea | name is iax.conf |
| 15:50.42 | hluesea | i am very paniced because of the service is running and i can't reach the other box :( |
| 15:51.08 | jaytee | panic doesn't help anything |
| 15:59.59 | tzafrir_laptop | rare1980_, have you built asterisk with modules statically compiled into it? |
| 16:07.06 | hluesea | i take now like this chan_iax2.c:12359 set_config: Ignoring bindport on reload notice and still not working |
| 16:07.24 | rare1980_ | tzafrir_laptop: i just compiled asterisk |
| 16:07.48 | hluesea | i changed bind port and result is same then i comment the beginning of the bind port line that time it said the same thing for the bindaddr |
| 16:07.51 | rare1980_ | make clean |
| 16:07.52 | rare1980_ | ./configure |
| 16:07.52 | rare1980_ | make menuselect |
| 16:07.52 | rare1980_ | make install |
| 16:07.52 | rare1980_ | make samples |
| 16:07.58 | rare1980_ | with these commands |
| 16:10.57 | *** join/#asterisk ManxPower (~manxpower@240.sub-75-254-91.myvzw.com) |
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| 16:17.12 | rare1980_ | pabelanger: did you have chance to look at my asterisk.conf file which in pastebin |
| 16:17.12 | rare1980_ | ? |
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| 16:32.25 | tzafrir_laptop | rare1980_, does 'asterisk -r' work? |
| 16:34.59 | rare1980_ | yes |
| 16:35.02 | rare1980_ | it works |
| 16:35.20 | tzafrir_laptop | what's the output of 'modules show like dahdi' ? |
| 16:35.52 | tzafrir_laptop | sorry: |
| 16:35.59 | tzafrir_laptop | what's the output of 'module show like chan_dahdi' ? |
| 16:52.05 | lmns972 | hello everybody |
| 16:52.22 | lmns972 | I am testing on a server asterisk. I have a problem with outgoing calls. I get this error message. |
| 16:52.35 | lmns972 | SIP/2.0 481 Call/Transaction Does Not Exist |
| 16:52.58 | lmns972 | I do not see it or just come |
| 16:53.15 | lmns972 | I spend an operator sip (freephonie) |
| 16:53.27 | lmns972 | Are you have a idea ? |
| 16:56.17 | lmns972 | For more details when making a call from the server to an external number, |
| 16:56.31 | lmns972 | the phone ringing when I pick up external communication is cut. |
| 16:59.05 | *** join/#asterisk pabelanger_ (~pabelange@CPE0013f7abc09a-CM0013f7abc096.cpe.net.cable.rogers.com) |
| 16:59.36 | lirakis | hrm ... i put together a macro that takes a single arg, an sip peer (like 2003).. it the macro rings the peer, then checks mailboxexists(${ARG1}@default) ... and should send to vm if the box is there |
| 16:59.40 | lirakis | ... but |
| 16:59.44 | lirakis | mailboxexists always fails |
| 16:59.53 | lirakis | .. when i am checking against a passed arg value |
| 17:00.36 | *** join/#asterisk simple64 (~Administr@66-227-172-142.dhcp.aldl.mi.charter.com) |
| 17:00.43 | lirakis | .. but it works fine, if i use $macro_exten and call it from a specifically matched extension (exten => 2003,1,.....) |
| 17:01.01 | [TK]D-Fender | lirakis: Perhaps you should actually show us the failed call.. |
| 17:01.06 | [TK]D-Fender | lirakis: and your dialplan. |
| 17:02.55 | lmns972 | I added logs : http://pastebin.com/yzn56bE4 |
| 17:03.52 | lirakis | [TK]D-Fender, http://pastie.org/962722 |
| 17:05.16 | [TK]D-Fender | lirakis: lirakis PRIORITY JUMPING? Holy crap.. what is it... 2003? |
| 17:05.49 | [TK]D-Fender | lirakis: Go read the apps instructions |
| 17:06.12 | lirakis | [TK]D-Fender, http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+MailboxExists |
| 17:06.15 | lirakis | ? |
| 17:06.20 | lirakis | or is there a more up to date resource |
| 17:07.17 | [TK]D-Fender | lirakis: "core show application mailboxexists"/ |
| 17:07.29 | [TK]D-Fender | lirakis: WIKI is useless for app instructions, always check the CLI instructions |
| 17:08.17 | lirakis | [TK]D-Fender, so ... do you suggest i check the status of VMBOXEXISTSSTATUS and gotoif ? |
| 17:08.51 | lirakis | i mean... i appreciate trying to clean things up, but - that isnt really the problem... the problem is that mailbox 2003@default ... does indeed exist |
| 17:09.21 | [TK]D-Fender | lirakis: The problem is you're using a 1954 manual for how to fix your 2009 model car |
| 17:09.29 | [TK]D-Fender | lirakis: ancient broken instruction <- |
| 17:10.13 | lirakis | [TK]D-Fender, the cli docs .. are basically the same |
| 17:10.44 | lirakis | [TK]D-Fender, also ... this example works fine - even if it is horrendus ;) http://pastie.org/962727 |
| 17:12.25 | [TK]D-Fender | lirakis: pastebin the CLI instrucitons. |
| 17:12.47 | lirakis | http://pastie.org/962729 |
| 17:13.48 | lirakis | so... i guess i need the j option |
| 17:13.57 | [TK]D-Fender | lirakis: Big print <- |
| 17:14.37 | lirakis | [TK]D-Fender, ... dont follow |
| 17:14.44 | pabelanger_ | tzafrir_laptop: ping? |
| 17:17.07 | lirakis | ... yeah .. so easy answer ... you need to pass "j" as it no longer jumps by default |
| 17:17.18 | [TK]D-Fender | lirakis: It says you need an option to do it the way you're trying to and you clearly didn't use that option. |
| 17:17.46 | [TK]D-Fender | lirakis: "no longer" was since 1.2. That was 4 branches ago |
| 17:19.18 | lirakis | [TK]D-Fender, understood. have you a better way of doing the mailbox check - since apparently priority jumping is evil? |
| 17:19.33 | lirakis | checking the var status and gotoif? |
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| 17:22.03 | tzafrir_laptop | pabelanger, pong |
| 17:23.51 | tzafrir_laptop | pabelanger_, pong |
| 17:24.02 | [TK]D-Fender | lirakis: what would YOU do witha returned variable that holds a status? |
| 17:24.10 | *** join/#asterisk madsara (~madsara@rtr0.circleeight.net) |
| 17:24.27 | lirakis | [TK]D-Fender, .. it just seems redundant if the app already has the jump ability built in |
| 17:24.38 | lirakis | [TK]D-Fender, i guess.. is there a reason not to use the jump? |
| 17:24.47 | madsara | Hey everyone quick question regarding mostly a dictionary item. An FXO card can be used for PSTN origination AND termination correct? |
| 17:24.59 | tzafrir_laptop | ~fxsfxo |
| 17:25.00 | infobot | [~fxsfxo] An FXO port expects to receive dialtone and receive ring voltage. You can connect it to a PSTN line from the telco. An FXS port expects to provide dialtone and provide ring voltage. You can connect a phone or a fax to it. |
| 17:25.26 | madsara | Understood, question is really related to the terms origination and termination. |
| 17:25.36 | madsara | Obviously I can dial out (origination) and accept inbound (terminatino). |
| 17:26.08 | tzafrir_laptop | sure. Assuming your PSTN provider doesn't have any strange policy with your line |
| 17:26.10 | madsara | I think I'm basically looking for a "yes, you're correct" to back me up, or a "No, you've got your terms wrong" |
| 17:26.24 | madsara | tzafrir_laptop: Excellent, just checking my terminology. Much appreciated. |
| 17:27.21 | madsara | Oh, that reminds me - anyone here using digium cards for PRI or are you SIP-ing to another "hardened" device (like an adtrans) that's basically dedicated to that PRI |
| 17:28.29 | tzafrir_laptop | madsara, yes, there are such people here. Try asking a more specific question |
| 17:29.44 | tzafrir_laptop | still wonders about madsara's originology as well |
| 17:30.43 | madsara | Yeah, a million different ways of doing things... I've noticed a lot of "voip providers" (basically companies that drop an Asterisk pbx at a business location) have the asterisk box only handling SIP, with a dedicated device used for PRIs but SIP-ing back to the pbx. |
| 17:31.05 | *** join/#asterisk BugKhaM (~BugKhaM@113.53.212.138) |
| 17:31.21 | pabelanger_ | tzafrir_laptop: when you are building packages in debian for Asterisk, you are using the svn-buildpackage command? |
| 17:31.26 | madsara | I was speaking with a CLEC the other day, and they stated that's the way they see a lot of providers doing things also. |
| 17:31.39 | tzafrir_laptop | pabelanger_, when building from svn: yes |
| 17:32.16 | pabelanger_ | tzafrir_laptop: And that is your teams preferred method for maintaining it under Debian? |
| 17:34.39 | [TK]D-Fender | [13:24]<lirakis>[TK]D-Fender, i guess.. is there a reason not to use the jump? <- because priority jumping was phased out half a decade ago |
| 17:35.15 | lirakis | [TK]D-Fender, ... who knew ? |
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| 17:37.21 | ManxPower | lirakis, I imagine all the UPGRADE*.txt files in your Asterisk source directory know! |
| 17:38.44 | lirakis | ManxPower, dd-wrt with optware ... no source :/ |
| 17:38.44 | lirakis | but i get it |
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| 18:12.05 | tzafrir_laptop | pabelanger_, it's the one I'm used to, and have a number of things built on |
| 18:13.23 | *** join/#asterisk Wildy (~simba@89.222.134.42) |
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| 18:28.14 | *** join/#asterisk uqlev (~yuriy@91.184.221.31) |
| 18:28.36 | ph8 | hi everyone, i'm doing an asterisk setup at home - what are the best phones to use? i'm got an old grandstream GXP2000 here but someone was telling me grandstream suck? |
| 18:28.46 | ph8 | at work we've got Mitels but i think they're an all in one package and you can't buy their phones seperately? |
| 18:30.12 | [TK]D-Fender | Polycom > All |
| 18:32.03 | ph8 | ahh that's the one |
| 18:32.08 | ph8 | how many others are there? |
| 18:32.39 | *** join/#asterisk Dovid (~annon@213.8.121.90) |
| 18:33.07 | [TK]D-Fender | ph8: Where are you located? |
| 18:33.43 | [TK]D-Fender | ph8: What kind of call usage? How many phones? |
| 18:37.33 | *** join/#asterisk simple64 (~Administr@66-227-172-142.dhcp.aldl.mi.charter.com) |
| 18:40.58 | ph8 | in the UK |
| 18:41.01 | ph8 | just a couple, for the family |
| 18:41.06 | ph8 | but i'm interested in corporate as well |
| 18:41.21 | ph8 | does asterisk do things like directory integration that i see in offices? |
| 18:41.30 | WIMPy | likes the Snom 3x0 series. |
| 18:43.02 | WIMPy | And for the phone directory I use the minibrowser. |
| 18:45.24 | [TK]D-Fender | ph8: What kind of phone voluem & use? |
| 18:45.35 | ph8 | just a couple, for the family |
| 18:45.41 | ph8 | but i'm interested in the corporate environment as well |
| 18:45.45 | [TK]D-Fender | ph8: In your case : Linksys |
| 18:45.52 | ph8 | o rly? didn't know linksys did phones |
| 18:46.25 | [TK]D-Fender | ph8: http://www.telephonydepot.com/Catalog/Cisco-IP-Phones;jsessionid=0a010c481f435a23d8fdb15244f5b2ef8a27f0b47520.e3eTaxiPc3mTe34Pa38Ta38Na3b0 |
| 18:46.49 | [TK]D-Fender | ph8: Just as a reference. Linksys is the most competitively priced option in the UK in recent searches |
| 18:46.56 | [TK]D-Fender | ph8: And they are decent |
| 18:47.09 | WIMPy | likes the Linksys design, but the software is crappy |
| 18:47.11 | ph8 | wow, good call |
| 18:47.27 | ph8 | how does asterisk work as business offerings? i think i've seen people building off asterisk solutions - but it's GPL? |
| 18:47.42 | ph8 | if they build their own admin panel say - wouldn't they have to release that under GPL also? |
| 18:49.03 | [TK]D-Fender | ph8: Who is "they"? |
| 18:49.12 | ph8 | well whoever was using the asterisk code base/building the solution |
| 18:49.36 | [TK]D-Fender | ph8: I can release a proprietay program that runs on Linux you know... |
| 18:49.50 | ph8 | ah i get it |
| 18:49.51 | [TK]D-Fender | ph8: Just because it talks to something open doesn't mean the add-on has to be |
| 18:50.02 | ph8 | the GPL applies if someone like me modified asterisk code and released it |
| 18:50.07 | ph8 | but not with things that interact with it? |
| 18:50.17 | ph8 | like an admin panel for example |
| 18:50.28 | [TK]D-Fender | ph8: And there is a metric ton of commercial add-ons for * as it is. status viewers, call reporting, billing, general GUI, etc |
| 18:50.44 | [TK]D-Fender | ph8: And a ton of Free/OSS as well |
| 18:54.18 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
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| 18:58.42 | *** join/#asterisk file (~file@asterisk/developer-and-muffin-lover/file) |
| 18:58.42 | *** mode/#asterisk [+o file] by ChanServ |
| 18:58.46 | *** join/#asterisk rare1980_ (~rare1980@115.186.16.221) |
| 19:00.09 | rare1980_ | hi all .. i have re-installed asterisk 1.4.16 , zaptel and libpri.... but on asterisk cli zap command is not working |
| 19:00.22 | rare1980_ | i can load ztdummy. i can start zaptel service |
| 19:00.36 | rare1980_ | but still zap command is not working on asterisk cli |
| 19:01.04 | rare1980_ | so please any help on this? |
| 19:01.59 | [TK]D-Fender | rare1980_: module load chan_zap.so |
| 19:03.00 | rare1980_ | [TK]D-fender: i just tried on asterisk cli and i got this msgs |
| 19:03.01 | rare1980_ | [May 16 15:02:32] WARNING[5407]: loader.c:363 load_dynamic_module: Error loading module 'chan_zap.so': /usr/lib/asterisk/modules/chan_zap.so: cannot open shared object file: No such file or directory |
| 19:03.01 | rare1980_ | [May 16 15:02:32] WARNING[5407]: loader.c:649 load_resource: Module 'chan_zap.so' could not be loaded. |
| 19:03.31 | [TK]D-Fender | rare1980_: apparently you forgot to compile and install * AFTER having installed the prerequisite for chan_zap to be built |
| 19:04.15 | *** join/#asterisk p3nguin (gpz5GvdFkf@mtop-mpls.a2infotech.com) |
| 19:04.52 | rare1980_ | [TK]D-Fender: wht should i do now? |
| 19:05.15 | rare1980_ | i did compile and install |
| 19:05.20 | [TK]D-Fender | rare1980_: recompile and install * and actually LOOK that chan_zap is going to get built |
| 19:05.20 | rare1980_ | asterisk by the way |
| 19:05.33 | rare1980_ | ok |
| 19:05.35 | [TK]D-Fender | rare1980_: AFTER ZAPTEL <---------------------------- |
| 19:06.14 | rare1980_ | yes.. i did the same... 1st i install libpri and zaptel and asterisk.. |
| 19:07.42 | rare1980_ | are these correct comands to compile asterisk |
| 19:07.43 | rare1980_ | make distclean |
| 19:07.43 | rare1980_ | make clean |
| 19:07.43 | rare1980_ | ./configure |
| 19:07.43 | rare1980_ | make menuselect |
| 19:07.43 | rare1980_ | make install |
| 19:07.43 | rare1980_ | make samples |
| 19:09.04 | [TK]D-Fender | rare1980_: What did menuselect say about chan_zap? |
| 19:09.49 | rare1980_ | u mean while isntall zaptel ? |
| 19:09.57 | rare1980_ | did i select chan_zap? |
| 19:11.00 | [TK]D-Fender | rare1980_: does it SAY that it is going to be built? |
| 19:11.41 | rare1980_ | i don't know recompilng it |
| 19:11.43 | rare1980_ | right now |
| 19:12.16 | [TK]D-Fender | rare1980_: MENUSELECT. This isn't COMPILING. I asked about MENUSELECT |
| 19:12.23 | *** join/#asterisk Maliuta (~scooby@kiev.lusan.id.au) |
| 19:12.28 | ChannelZ | he's saying you must build and install zaptel FIRST - then, when you configure asterisk, it should notice zaptel and tell you |
| 19:12.31 | *** join/#asterisk ccesario (~ccesario@189-19-6-236.dsl.telesp.net.br) |
| 19:12.56 | ChannelZ | if asterisk isn't noticing you have zaptel installed, it's not going to build it. |
| 19:13.09 | rare1980_ | humm rite |
| 19:14.19 | rare1980_ | let me check i m building ad install zaptel |
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| 19:21.04 | roe | I'm trying to get more info on this 'Corporate Directory' support the polycom phones boast. It says I need a license key. Does anyone know if there is a cost associated with that, and if so do you know about what it is? |
| 19:21.27 | [TK]D-Fender | roe: Of course there is... and contact your reseller |
| 19:21.43 | [TK]D-Fender | roe: Or built your own using the MicroBrowser |
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| 19:26.24 | roe | [TK]D-Fender, it is a bit stupid that there be an extra cost associated with what amounts to an LDAP search |
| 19:27.39 | roe | I'll send them an email though, thanks |
| 19:27.55 | [TK]D-Fender | roe: welcome to the wonderful world of "enterprise features" |
| 19:32.17 | *** join/#asterisk voxter (~voxter@S010600090f53ea17.vc.shawcable.net) |
| 19:35.00 | voxter | Hmm. anyone using chan_skype? mine segfaults when i load it, and im not sure why |
| 19:36.10 | voxter | oh. I see. |
| 19:36.18 | voxter | minimum 1.4.25 - im using 1.4.21 |
| 19:39.38 | ChannelZ | oopsies. |
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| 19:59.06 | idespinner | roe, the corporate directory feature amounts to about 10$ per phone |
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| 20:22.00 | roe | idespinner, thanks |
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| 20:49.42 | *** join/#asterisk lesouvage (~lesouvage@cc741036-b.groni1.gr.home.nl) |
| 20:52.54 | lesouvage | I use Record() in a routine and I just find out that after 33 seconds the connection stops with "== Spawn extension (im_record_menu, s, 8) exited non-zero on 'SIP/xxxxxxxxxx-082326c8'" What can cause the ending of the connection? |
| 20:55.03 | ph8 | exit |
| 20:55.06 | ph8 | lol whoops |
| 21:03.53 | [TK]D-Fender | lesouvage: Have you considered actually looking at the call? |
| 21:06.52 | *** join/#asterisk MhaddogM1 (~MhaddogM1@adsl-32-49-180.mia.bellsouth.net) |
| 21:07.46 | lesouvage | [TK]D-Fender: yes, I have checked /var/log/messages, set verbose to 20 and examine the cdr database. In the cdr database I can see that Record() stops after 33 seconds time after time. I have tested it with a mobile connection and with a sip connection and the problem is the same. |
| 21:10.44 | [TK]D-Fender | lesouvage: SIP DEBUG, core debug 10, complete call. Go dump it |
| 21:11.00 | lesouvage | [TK]D-Fender: thanks |
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| 21:16.41 | *** join/#asterisk nwks (~chatzilla@dslb-094-217-071-073.pools.arcor-ip.net) |
| 21:17.43 | nwks | hello, i ve got a problem configuring dahdi. When i change the genconf_parameters " pri_termtype SPAN/1 NT SPAN/2 TE" and executing dahdi_genconf all spans are in te mode |
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| 21:33.43 | nwks | i am using 2 Wildcard B410P cards one for internal isdn one for uplink to ntba , when i use dahdi_scan, all seems to be ok. but in dahdi-channels.conf, generated from dahdi_genconf the inernal isdn are mis configured. Therfore i changed the genconf_parameters but when i run dahdi_genconf the ports are still in te mode. when i manualy change the dahdi-channels.conf in signaling bri_net_ptmp... |
| 21:33.45 | nwks | ...and restart dahdi and asterisk the pbx on intisdn still gets an error when trying to phone out |
| 21:42.00 | lesouvage | [TK]D-Fender: This seems to be a relevant line " [May 16 21:42:59] WARNING[2770] chan_sip.c: Hanging up call 2141242010-83CDE46539971605605@xx.xxx.xxx.38 - no reply to our critical packet (see doc/sip-retransmit.txt). " For complete output see http://pastebin.com/c1HSYFX7 |
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| 21:51.42 | luke-jr | Voipjet broken? :/ |
| 21:52.02 | [TK]D-Fender | lesouvage: that isn't SIP DEBUG |
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| 22:22.16 | lesouvage | [TK]D-Fender: Sorry it took so long. De SIP debug is at http://pastebin.com/P6qANarn . I tried another ITSP and that solved the problem. It seems to be a ITSP related problem. |
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| 22:28.25 | lesouvage | So I still have the problem, because I have to use the ITSP that causes problems. |
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| 22:34.08 | *** join/#asterisk cosmicwombat (~cosmicwom@69.7.44.68) |
| 22:38.22 | lesouvage | I find this line in my SIP debug info "Scheduling destruction of SIP dialog '3b332512499d155829d71cc67b22a23c@127.0.1.1' in 32000 ms (Method: REGISTER)" It is (almost) the 33 seconds that the call last now. But what can cause this problem? |
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| 23:02.44 | *** join/#asterisk Shaaan (Un1x@CPE001cc031aa0d-CM0014045acc3c.cpe.net.cable.rogers.com) |
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| 23:03.16 | Shaaan | Hey, guys anyone here know where i can get a Block of Did's that offer good quality service and have minimum 2 channels per DID and unlimted incoming?? |
| 23:03.34 | Shaaan | And also, has anyone ever done one of those Verification Vault type things with Asterisk? |
| 23:06.16 | *** join/#asterisk jks (jks@193.189.93.254) |
| 23:10.05 | Naikrovek | Shaaan: most ITSPs (if not all) can give you DIDs cheaply, w/free incoming. |
| 23:10.11 | Naikrovek | ~itsplist-us |
| 23:10.20 | infobot | Here are some popular ITSPs (USA) starting with the more respected ones: http://www.teliax.com , http://www.voicepulse.com/connect/ , http://www.broadvoice.com , http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net , http://voip.ms and http://flowroute.com |
| 23:10.20 | Naikrovek | hrm. |
| 23:10.28 | Naikrovek | there it is |
| 23:10.33 | luke-jr | define cheaply |
| 23:10.35 | Naikrovek | Shaaan: ^^^^^^^^^ |
| 23:10.55 | Shaaan | thanks Naikrovek |
| 23:10.59 | Naikrovek | $1 per month per DID |
| 23:11.04 | Naikrovek | that's what I pay for mine |
| 23:11.06 | Shaaan | Nice, wich who? |
| 23:11.13 | luke-jr | anyhow. since Voipjet is apparently dead, any suggestions for decent 1c/min termination? |
| 23:11.17 | *** join/#asterisk boodu (~boodu@175.158.129.128) |
| 23:11.34 | luke-jr | Naikrovek: I pay nothing for origination. >.> |
| 23:11.52 | Shaaan | Naikrovek who do you have a DID with? |
| 23:11.54 | Naikrovek | luke-jr: sure, but i pay nothing for termination |
| 23:12.06 | Naikrovek | well, no per-minute fees anyway |
| 23:12.10 | luke-jr | Naikrovek: except your monthly cost |
| 23:12.24 | Shaaan | and does anyone know about verification vaults with asterisk where basicly someoen can call in start a recording and at the end of the recording it generates a Verification Recording number?? |
| 23:13.05 | boodu | hello |
| 23:13.09 | Naikrovek | Shaaan: voicespringvoip.com. $33/mo per channel, free incoming and outgoing to US, Canada and a few other places |
| 23:13.17 | [TK]D-Fender | Shaaan: And that number is generated by what? |
| 23:13.45 | Shaaan | [TK]D-Fender, here is a more complete overview of what im looking for.. http://www.predictiveconcepts.com/v3/index.php?option=com_content&task=view&id=19&Itemid=50 |
| 23:14.20 | [TK]D-Fender | Shaaan: Simple answer please... |
| 23:14.38 | luke-jr | Naikrovek: $33/mo isn't $1/mo |
| 23:14.42 | Shaaan | Basicly, i suppose asterisk could generate a random number calling it a Recording ID |
| 23:15.10 | Naikrovek | luke-jr: no shit |
| 23:15.27 | Naikrovek | luke-jr: $33/mo per channel, $1/mo per DID |
| 23:15.34 | [TK]D-Fender | Shaaan: This is about.... 2 lines of dialplan so far... |
| 23:15.36 | Naikrovek | unlimited calling in or out |
| 23:15.51 | luke-jr | Naikrovek: in other words, $34/mo minimum |
| 23:16.02 | luke-jr | since he wants 2 channels, $67 |
| 23:16.02 | Naikrovek | luke-jr: i see the calculator is working |
| 23:16.12 | Shaaan | [TK]D-Fender, i realise that but there is more to it then just that.. |
| 23:16.16 | luke-jr | you implied $1/mo got what he needed |
| 23:16.47 | [TK]D-Fender | Shaaan: Make numbered recordings. No big deal... |
| 23:16.47 | Naikrovek | i did not imply that. you inferred it. |
| 23:17.36 | Naikrovek | DIDs are independent of service |
| 23:17.44 | Naikrovek | if you don't have service, DIDs won't do you any good |
| 23:18.01 | Naikrovek | if you don't have DIDs, service won't do you any good |
| 23:18.07 | Naikrovek | you can pay per minute, or per channel |
| 23:18.10 | *** join/#asterisk centoslinux (~centoslin@s0021-0018.dsl.start.no) |
| 23:18.11 | Naikrovek | i pay per channel |
| 23:18.17 | Naikrovek | you can choose to pay per minute |
| 23:18.20 | Naikrovek | i don't care |
| 23:18.25 | luke-jr | whatever |
| 23:18.31 | luke-jr | $33/mo is a ripoff in any case |
| 23:18.48 | Naikrovek | for 10,000 minutes per month $33/mo is dirt cheap |
| 23:19.03 | Naikrovek | we have 8 channels and regularly hit 10k minutes on each |
| 23:19.11 | luke-jr | you're never going to max out all channels |
| 23:19.21 | Naikrovek | we maxed out 6 |
| 23:19.32 | Naikrovek | and we'll be maxing out 8 soon |
| 23:19.45 | Shaaan | [TK]D-Fender, can i PM you? |
| 23:19.55 | [TK]D-Fender | Shaaan: Is there a need? |
| 23:20.01 | Shaaan | [TK]D-Fender yes |
| 23:20.22 | [TK]D-Fender | Shaaan: for now... |
| 23:31.17 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 23:32.16 | carrar | http://www.youtube.com/watch?v=fKXk1VhAuvE |
| 23:39.45 | *** join/#asterisk supa_disko (~bleh@secure27.lnk.telstra.net) |
| 23:54.11 | [TK]D-Fender | carrar: YOU BASTERD |
| 23:54.31 | [TK]D-Fender | brains slowly melt |