| 00:01.31 | DrukenLPY | i want fios.... :( |
| 00:03.54 | CCFL_Man2 | i don't want it yet because i don't have a cisco 1841 router for it |
| 00:05.34 | CCFL_Man2 | i can't afford it yet :P |
| 00:07.34 | CCFL_Man2 | anyone got a cheap 1841? :P |
| 00:08.20 | booray | $510 on ebay |
| 00:08.22 | booray | three hours left |
| 00:11.57 | CCFL_Man2 | thats 75% of my paycheck :P |
| 00:12.36 | *** join/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-825859ff00ff60e7) |
| 00:12.38 | booray | I guess you'll have to settle for a $38 buffalo and load dd-wrt? |
| 00:12.57 | CCFL_Man2 | i use a 1721 right now |
| 00:13.03 | CCFL_Man2 | $350 i paid used |
| 00:14.22 | ManxPower | the 1720s are well worth it |
| 00:14.40 | CCFL_Man2 | 1721 has a max routing speed of around 12mbit |
| 00:15.06 | ManxPower | CCFL_Man2: That does not matter in any environment I've ever used one in. |
| 00:16.05 | CCFL_Man2 | ManxPower: ideally i want the most efficient and fastesr routing possible with nat |
| 00:17.34 | ManxPower | CCFL_Man2: what is your uplink speed? |
| 00:17.54 | ManxPower | well up/down speed. |
| 00:17.55 | jsidhu2 | where can i download the latest hud server? is it only a part of trixbox or what? |
| 00:18.11 | ManxPower | jsidhu2: perhaps the trixbox channel can help you. |
| 00:18.30 | jsidhu2 | perhaps someone might know here? |
| 00:18.47 | ManxPower | jsidhu2: I doubt it, almost nobody here uses a GUI. |
| 00:19.08 | jsidhu2 | Manx: then what do you guys use for the operator? Such as the front desk? |
| 00:19.14 | ManxPower | This is really #asterisk-nogui, but someone forgot to put in the -nogui part when the channel was created. |
| 00:19.33 | ManxPower | jsidhu2: We use the Polycom sidecar. Many people don't use a BLF with Asterisk. |
| 00:19.56 | jsidhu2 | is it working well? |
| 00:20.12 | ManxPower | jsidhu2: It seems to be, but users want to fuck over IT so don't report problems. |
| 00:20.22 | CCFL_Man2 | ManxPower: with fios it's 1mit up |
| 00:20.26 | ManxPower | then they scream to their manager when something is broken. |
| 00:20.44 | ManxPower | CCFL_Man2: If you have FIOS then perhaps a better router might be in order. |
| 00:20.53 | jsidhu2 | i still think the hudlite solution might be better.. but thats for the input |
| 00:20.58 | ManxPower | But as far as we can tell the polycom sidecars work just fine with Asterisk |
| 00:21.39 | ManxPower | jsidhu2: Go for it, but this is the wrong channel for HUD stuff. |
| 00:22.49 | CCFL_Man2 | ManxPower: like an 1841? |
| 00:25.23 | *** join/#asterisk iPod-nano (n=root@c-68-43-60-193.hsd1.mi.comcast.net) |
| 00:25.33 | iPod-nano | This is cool. |
| 00:26.10 | iPod-nano | I have a text-based IRC client. |
| 00:26.22 | iPod-nano | On my Asterisk box. |
| 00:28.36 | russellb | this text based IRC ... must be a new thing ... |
| 00:28.41 | russellb | someone submit it to slashdot! |
| 00:28.44 | russellb | :-D |
| 00:29.49 | *** join/#asterisk dasenjo (n=dasenjo@190.5.196.64) |
| 00:30.17 | dasenjo | Hi everybody! |
| 00:32.32 | dasenjo | I don't like Xlite so much, but I was trying to configuring it .. SIP works well .. but there is a problem with the silence tx, in the previous version it could be solved just enabling it (silence tx), but now it isn't possible .. |
| 00:32.33 | *** join/#asterisk rmayorga_ (n=rmayorga@unaffiliated/rmayorga) |
| 00:33.20 | dasenjo | xlite says that is actually "transmitting silence", but when I speak, I hear nothing, the received sounds cut off .. |
| 00:33.44 | dasenjo | I found ***7469 in the voip wiki as a menu to tweak the conf. |
| 00:34.03 | dasenjo | but did not find a parameter that helped me .. |
| 00:34.15 | dasenjo | so, someone has found it? |
| 00:34.59 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 00:35.06 | dasenjo | I'm gonna try the windows ekiga version ... but I'm curious and wanted to ask .. |
| 00:38.43 | *** join/#asterisk nighty^ (n=nighty@sushi.rural-networks.com) |
| 00:41.41 | riddlebox | if I install a TDM card, and do lspci what should I see? |
| 00:45.37 | *** join/#asterisk wunderkin (n=wunderki@ip72-223-86-126.ph.ph.cox.net) |
| 00:45.52 | admin0 | hi guys .. when I take in a call from my zap e1 card and terminate it to a provider, the codec that goes is g711 .. i am looking for a way to make sure that all calls from the card goes as g723 or g729 |
| 00:46.03 | flenders | ~seen JT |
| 00:46.06 | jbot | jt is currently on #asterisk-dev (7d 6h 22m 8s) #asterisk (7d 6h 22m 8s) #slug (7d 6h 22m 8s). Has said a total of 522 messages. Is idling for 12h 23m 38s, last said: 'talking about appelza's issue'. |
| 00:47.22 | flenders | admin0: you want to make calls through your E1 to use a different codec? |
| 00:48.12 | admin0 | i am accepting calls through the e1 and want to terminate via IP .. when I check the logs of the terminating gateway, I see the codecs that sent was gsm/g711 |
| 00:48.29 | admin0 | while the server works with either g723.1 or g729 |
| 00:49.12 | *** join/#asterisk Corydon76-dig (i=red@pdpc/supporter/sustaining/Corydon76-home) |
| 00:49.12 | *** mode/#asterisk [+o Corydon76-dig] by ChanServ |
| 00:49.38 | *** join/#asterisk Corydon76-home (i=purple@pdpc/supporter/sustaining/Corydon76-home) |
| 00:49.38 | *** mode/#asterisk [+o Corydon76-home] by ChanServ |
| 00:50.13 | admin0 | if it was a SIP, i would just do disallow=all, allow=g723.1 allow=g729 .. but in case of this zap card, i am out of ideas |
| 00:51.05 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 00:51.27 | Qwell | where's strom? |
| 00:52.17 | Qwell | somebody name this phone |
| 00:52.17 | Qwell | http://www.telephoneart.com/clipart/telephone9853.jpg |
| 00:52.26 | admin0 | i have the g723.1 as well as g729 codecs in the server |
| 00:52.38 | mog | its a black one Qwell |
| 00:52.59 | Qwell | mog: twilson gave a better answer.. |
| 00:53.07 | Qwell | "black with buttons." |
| 00:53.25 | mog | ooh |
| 00:53.27 | mog | i missed those |
| 00:53.36 | elixer | had those at my last job |
| 00:53.39 | elixer | they were beige though |
| 00:53.47 | Qwell | elixer: even uglier... I like it |
| 00:53.53 | elixer | same button configuration though |
| 00:53.57 | Qwell | avaya, right? |
| 00:54.02 | admin0 | flenders, any hints on how I can do it ? |
| 00:54.02 | elixer | sure |
| 00:54.05 | Qwell | :p |
| 00:54.08 | twilson | qwell: I'm nothing if not precise. :-p |
| 00:54.11 | elixer | :) |
| 00:54.26 | flenders | admin0: you can't use GSM on an E1 |
| 00:54.45 | admin0 | i am not using it . it comes automatically and i am trying to find a way to block it |
| 00:54.52 | flenders | it'll use whatever codec the PSTN uses |
| 00:54.59 | flenders | 711a or u |
| 00:55.00 | *** join/#asterisk smace (n=chatzill@200.149.32.178) |
| 00:55.35 | flenders | you can only change the codecs on ITSP connections |
| 00:55.46 | flenders | or internally, between your SIP phones, for example |
| 00:56.57 | flenders | admin0: when you say you terminate the incoming call with a provider, you mean carrier? |
| 00:59.44 | admin0 | yeah |
| 01:00.20 | *** part/#asterisk smace (n=chatzill@200.149.32.178) |
| 01:01.23 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 01:05.25 | flenders | admin0: yeah, so all calls coming into your E1 will use alaw or ulaw, whatever codec your country uses |
| 01:06.10 | admin0 | is there a a way to transcode |
| 01:07.35 | flenders | where are these calls going to? |
| 01:07.38 | flenders | a SIP device? |
| 01:18.35 | *** join/#asterisk fujin (n=aj@unaffiliated/fujin) |
| 01:19.22 | wishes | "ast_writestream: Unable to translate to format h263, source format unknown" however h263 format shows in up 'show codecs' and its allowed in sip.conf - what am i doing wrong ? |
| 01:20.53 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 01:22.41 | *** join/#asterisk Shadowfire__ (n=jeff@rrcs-67-79-144-150.se.biz.rr.com) |
| 01:23.26 | *** join/#asterisk Delvar (n=Delvar@77.240.56.18) |
| 01:26.05 | *** join/#asterisk |omni| (n=rob@c-67-185-56-106.hsd1.wa.comcast.net) |
| 01:31.34 | *** join/#asterisk |omni| (n=rob@c-67-185-56-106.hsd1.wa.comcast.net) |
| 01:36.43 | riddlebox | what do I all need to do to get a digium tdm cardto work? I have the module loaded? |
| 01:36.57 | jingles | there's a few modules that have to be loaded. |
| 01:37.01 | jingles | zaptel, wctdm |
| 01:37.05 | *** join/#asterisk dijungal (n=kdaniel@208.0.231.76) |
| 01:37.16 | jingles | make sure they're in your /etc/modules file. |
| 01:37.19 | riddlebox | jingles, I have all of them loaded, even ztdummy |
| 01:37.31 | jingles | no need for ztdummy if you're using tdm cards. |
| 01:37.36 | riddlebox | jingles, do I need to edit a *.conf file? and put something in there? |
| 01:37.41 | jingles | the cards have the timing stuff built in. |
| 01:37.45 | jingles | zapata.conf |
| 01:37.49 | jingles | to allocate your channels. |
| 01:37.54 | *** join/#asterisk tengulre (n=tengulre@222.90.66.10) |
| 01:38.00 | jingles | and zaptel.conf as well. |
| 01:38.12 | dijungal | i have an IAX agent in a queue that gets all her calls dropped |
| 01:38.15 | riddlebox | what do I need to put in it though, I tried reading through it but I couldnt see what I should do |
| 01:38.17 | jingles | mm... all good stuff, found in 'the book' or on voip-info.com |
| 01:38.30 | dijungal | all her calls hangup before she can even answer the call |
| 01:38.35 | tengulre | voip-info.org. |
| 01:38.41 | tengulre | good morning everyone! |
| 01:38.42 | jingles | tengulre : thanks. |
| 01:38.44 | tengulre | ;) |
| 01:39.14 | *** join/#asterisk Delvar (n=Delvar@77.240.56.18) |
| 01:39.16 | dijungal | tengulre: morning??? |
| 01:39.42 | dijungal | it's 21:39 here |
| 01:39.51 | dijungal | AST |
| 01:40.19 | tengulre | haha... |
| 01:40.37 | tengulre | it's 9:40 here. Wed. |
| 01:40.42 | tengulre | I m in office. |
| 01:41.13 | dijungal | where r u ????? |
| 01:41.17 | tengulre | my company running asterisk box that have 20 lines(FXO) |
| 01:41.22 | tengulre | CHINA. |
| 01:41.31 | dijungal | :O |
| 01:41.50 | tengulre | that's very interesting. |
| 01:42.11 | dijungal | u'r on the other side of the planet ... interesting |
| 01:42.26 | tengulre | I developt a iax client running per client. |
| 01:42.36 | *** part/#asterisk jingles (n=foo@39.183.dowl.anc.borealisbroadband.net) |
| 01:42.50 | tengulre | but I still have a problem. |
| 01:42.57 | tengulre | when asterisk as a call center. |
| 01:43.19 | dijungal | like? |
| 01:43.24 | tengulre | agent could not transfer current calls to other agent, I don't know why? |
| 01:44.37 | tengulre | for example, a calling incoming and an agent A pickup this call, but A could not answer this problem so A need transfer call to other agent B or Agent c. |
| 01:44.53 | tengulre | but she don't know how to do? |
| 01:45.13 | tengulre | she known press '*' to huangup current call,.. |
| 01:46.05 | dijungal | look to see what code it is in features.conf |
| 01:46.29 | dijungal | what phone r u using? |
| 01:46.54 | tengulre | iax2 soft phone. |
| 01:47.11 | tengulre | myself developt one |
| 01:47.20 | Nugget | i m using a cisco phone 2 b n a sip redct server with man e ga sjas ejs ekt through fom. |
| 01:47.22 | tengulre | use iaxclient libaray. |
| 01:48.31 | tengulre | dijungal: if u use hard phone, you can not got customer's info from exists OA & CRM . |
| 01:49.11 | tengulre | dijungal: iax2 only need open one port, but sip not. |
| 01:49.26 | tengulre | sip need open 10000-20000 ports for rtp protocol. |
| 01:49.52 | dijungal | true |
| 01:49.53 | tengulre | if you agent distru different area, |
| 01:50.18 | tengulre | your telecom will disabled your communication. |
| 01:51.15 | dijungal | k |
| 01:51.29 | tengulre | my iax2 is very simple for that, provider gsm, g711,iLbc, speex codecs |
| 01:51.41 | tengulre | no |
| 01:51.49 | tengulre | dijungal: talking here. |
| 01:51.51 | tengulre | ;) |
| 01:52.11 | tengulre | dijungal: no , I don't use AgentCallBack()... |
| 01:52.43 | dijungal | k |
| 01:52.54 | tengulre | in my agent application, not huangup function, only have dtmf funtion. |
| 01:53.26 | tengulre | so if an agent login in, he/she can not huangup self. |
| 01:54.43 | *** join/#asterisk ez` (n=ezw@c142.169.166-68.clta.globetrotter.net) |
| 01:55.28 | tengulre | digungal: which server are you using ? |
| 01:56.16 | dijungal | 1.2.19 |
| 01:56.33 | *** join/#asterisk asdx (n=foo@adsl-146-3.click.com.py) |
| 01:56.46 | dijungal | u? |
| 01:57.08 | tengulre | dijungal: I point your hard server. like cpu: 3.0Ghz... |
| 01:57.44 | tengulre | me use Cou2 3.0Ghz. RAM 2GB , HD: 200GB |
| 01:57.56 | tengulre | I use ;) |
| 01:58.07 | tengulre | haha.. sorry for my english. |
| 01:58.25 | tengulre | because my country language is not english. ;) |
| 01:58.38 | dijungal | lol.. i figured |
| 01:59.07 | tengulre | where are u? |
| 02:02.16 | dijungal | caribbean |
| 02:03.50 | tengulre | I like <<caribbean pirates>> film. ;) |
| 02:06.38 | *** join/#asterisk Mavvie (n=edwin@ppp59-167-4-80.lns1.syd7.internode.on.net) |
| 02:09.43 | riddlebox | jingles, can you help me a sec on this tdm card? |
| 02:12.08 | *** join/#asterisk ELBunce (n=erik@kde/developer/bunce) |
| 02:14.11 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 02:14.30 | *** join/#asterisk Defraz (n=t0tal@24-116-152-177.cpe.cableone.net) |
| 02:15.26 | riddlebox | can someone help me get this TDM card working, when I run dmesg I see this, http://pastebin.ca/702775 and my zap restart is below it? |
| 02:15.53 | *** join/#asterisk |omni| (n=rob@c-67-185-56-106.hsd1.wa.comcast.net) |
| 02:16.16 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 02:17.41 | outtolunc | you should put the module in the first slot, and redo your config's for only 1 channel |
| 02:18.04 | riddlebox | outtolunc, I was wondering why it came to me in the last slot |
| 02:22.18 | riddlebox | outtolunc, where do I tell the config only 1 channel, in zapata.conf, or /etc/zaptel.conf |
| 02:23.22 | outtolunc | both |
| 02:23.39 | riddlebox | ok I have only set both for channel=1 |
| 02:25.14 | outtolunc | you'll need to reload your drivers and check that it is seen |
| 02:25.48 | *** join/#asterisk linagee (n=linagee@about/linux/staff/linagee) |
| 02:29.29 | outtolunc | and to answer your question, the reason they put the fxo in the forth slot is because of issue that device has with module loading (the fxs ones must be first) |
| 02:30.06 | outtolunc | so if you ever install any fxs modules you have to shift that fxo to the right |
| 02:32.36 | riddlebox | ohh ok |
| 02:36.26 | outtolunc | so what do you have so far |
| 02:40.16 | riddlebox | hrmm hold on a sec |
| 02:40.44 | outtolunc | anyways, either put the modules to slot 1 and use fxsks=1 in zaptel and channel 1 in zapata, or move it back to slot 4 and use fxsks=4 in zaptel and channel 4 in zapata |
| 02:41.09 | riddlebox | I moved it to slot 1 |
| 02:41.19 | outtolunc | k |
| 02:41.20 | riddlebox | if I ever get a fxs card, I will just move it over |
| 02:42.21 | riddlebox | should I use fxs_ks for signalling? or fxs_ls? |
| 02:42.36 | outtolunc | yes, you 'talk' the reverse to the module type |
| 02:43.07 | outtolunc | sorry misread |
| 02:43.13 | outtolunc | ue fxsks |
| 02:43.15 | outtolunc | er use |
| 02:43.22 | riddlebox | always use fxsks? |
| 02:43.41 | outtolunc | for an TDM400P FXO module, yes |
| 02:44.11 | riddlebox | I still get the message I am sorry that is not a valid extension please try again |
| 02:44.16 | outtolunc | for a TDM400P FXS module you would use fxoks |
| 02:44.26 | outtolunc | we aren't there yet |
| 02:44.42 | outtolunc | what do you have in zapata |
| 02:45.20 | riddlebox | channel => 1 |
| 02:45.28 | riddlebox | I followed "the book" |
| 02:45.30 | outtolunc | what else? |
| 02:46.06 | riddlebox | context=internal |
| 02:46.13 | outtolunc | fxsks=4 |
| 02:46.13 | outtolunc | loadzone = us |
| 02:46.13 | outtolunc | defaultzone=us |
| 02:46.14 | riddlebox | signalling=fxs_ks |
| 02:46.16 | outtolunc | grr |
| 02:46.28 | outtolunc | yes.. signalling=fxs_ks |
| 02:46.45 | outtolunc | once you have that in, then rmmod wctdm |
| 02:46.49 | outtolunc | then rmmod zaptel |
| 02:46.55 | outtolunc | then modprobe zaptel |
| 02:47.03 | outtolunc | then modprobe wctdm |
| 02:47.07 | outtolunc | then ztcfg -vvv |
| 02:47.12 | outtolunc | and what does that say |
| 02:47.47 | riddlebox | I get module zaptel in use |
| 02:48.02 | outtolunc | then not all modules using zaptel are unloaded |
| 02:48.09 | riddlebox | hehe asterisk was running |
| 02:48.24 | *** join/#asterisk mordaunt (n=mordaunt@unaffiliated/mordaunt) |
| 02:48.54 | riddlebox | 1 channel configured |
| 02:49.04 | outtolunc | ok, now start asterisk |
| 02:49.15 | outtolunc | and do a dial(Zap/1/.....) test |
| 02:49.48 | outtolunc | make sure you do a |
| 02:49.54 | riddlebox | cool it worked |
| 02:49.55 | outtolunc | 'set verbose 3' on the CLI |
| 02:50.04 | outtolunc | k |
| 02:50.39 | *** join/#asterisk jmacz (n=jmacz@190.24.98.133) |
| 02:50.45 | riddlebox | hrmm came up on my cellphone as unknown though, thats odd, its a charter line, I shouldnt have to worry about callerid |
| 02:51.13 | riddlebox | thanks for the help outtolunc, I knew I was close, but couldnt quite figure it all the way out |
| 02:51.17 | outtolunc | well it being a pstn device i doubt you are going to get that working |
| 02:51.22 | outtolunc | no prob |
| 02:52.40 | outtolunc | but you can try setting zapata.conf to usecallerid=yes and set a callerid.... in zapata.conf for that channel (just above the channel => 1) |
| 02:53.14 | *** join/#asterisk flenders (n=fserto@unaffiliated/flenders) |
| 02:53.53 | riddlebox | outtolunc, when an incoming call would come in, it would come as Zap/1 right? |
| 02:54.46 | outtolunc | Zap/1-1 |
| 02:54.56 | riddlebox | yeah I just saw that |
| 02:54.56 | outtolunc | but yes |
| 02:56.53 | riddlebox | hrmm exten => Zap/1-1,1,Goto(incoming,s,1) |
| 02:57.04 | riddlebox | thats what I have in my context default |
| 02:57.28 | riddlebox | but I get this message when I call in, pbx.c:2450 __ast_pbx_run: Channel 'Zap/1-1' sent into invalid extension 's' in context 'default', but no invalid handler |
| 02:58.05 | outtolunc | hehe |
| 02:58.11 | outtolunc | that's not what you want |
| 02:58.59 | outtolunc | exten => _X.,1,Goto(incoming,s,1) OR exten => s,1,Goto(incoming,s,1) |
| 02:59.56 | outtolunc | later you will learn to set it differently in zapata.conf |
| 03:00.37 | riddlebox | so right now it is coming in as 's'? |
| 03:01.11 | outtolunc | 's' is a fallback too default type exten |
| 03:01.23 | outtolunc | _X. means any length digit exten |
| 03:01.41 | outtolunc | (greater than 1 digit that is) |
| 03:01.57 | outtolunc | just use 's' for now |
| 03:02.06 | riddlebox | I still got the same message |
| 03:02.56 | outtolunc | which is? |
| 03:03.19 | outtolunc | oh nevermind |
| 03:03.20 | riddlebox | but I get this message when I call in, pbx.c:2450 __ast_pbx_run: Channel 'Zap/1-1' sent into invalid extension 's' in context 'default', but no invalid handler |
| 03:03.38 | outtolunc | did you do an 'extensions reload' after mod'ing your dialplan |
| 03:03.55 | riddlebox | I did just a whole reload of asterisk |
| 03:04.40 | outtolunc | [19:46] <riddlebox> context=internal |
| 03:05.08 | outtolunc | did you add that exten => s,1... to both 'internal' and 'default' |
| 03:05.11 | riddlebox | in zapata.conf I have context=default |
| 03:05.31 | outtolunc | so <reinsert> 'default' |
| 03:05.35 | outtolunc | ? |
| 03:05.36 | *** join/#asterisk shido6 (n=shido6@74-130-56-203.dhcp.insightbb.com) |
| 03:07.53 | riddlebox | outtolunc, ok, it works great when I use _X |
| 03:08.20 | *** join/#asterisk yidiyuehan (n=yidiyueh@bb121-7-184-197.singnet.com.sg) |
| 03:08.22 | yidiyuehan | can any one direct me a link to configure ISDN2 card with asterisk? |
| 03:08.51 | outtolunc | l |
| 03:08.53 | outtolunc | er k |
| 03:09.08 | riddlebox | outtolunc, where would I change it in zapata.conf so that I could have a number or something instead of _X? |
| 03:09.42 | outtolunc | not sure i understand your question |
| 03:09.49 | outtolunc | if you have multiple DID's then |
| 03:10.01 | CoaxD | POTS bad!#$ |
| 03:10.15 | outtolunc | you just do exten => 40844444444,1,Yadda() |
| 03:10.22 | outtolunc | whatever your did is |
| 03:10.38 | outtolunc | then leave the _X.,1,... below it as a catch all |
| 03:11.05 | riddlebox | you wouldnt have to distinguish between say all four channels then? |
| 03:11.34 | outtolunc | no, if they all use the same context |
| 03:12.17 | riddlebox | ok, thats less typing that I would have to do then :) |
| 03:12.40 | *** join/#asterisk Strom_M (n=strom@208.127.172.112) |
| 03:12.40 | *** join/#asterisk Strom_C (n=strom@208.127.172.112) |
| 03:13.04 | riddlebox | I need to take a class on configuring asterisk, I know some stuff but not enough |
| 03:15.18 | outtolunc | just read the handbook at http://www.digium.com/handbook-draft.pdf |
| 03:15.22 | outtolunc | it's older but useful |
| 03:15.53 | *** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net) |
| 03:15.53 | riddlebox | I have the O'reilly book, but I think thats pretty old too |
| 03:16.05 | outtolunc | still useful |
| 03:16.41 | outtolunc | you can't imagine the amount of material i've read over the years regarding asterisk <G> |
| 03:16.46 | hypa7ia | riddlebox: the new edition will be out very soon |
| 03:16.48 | riddlebox | ohh yeah, its helped |
| 03:17.11 | riddlebox | hypa7ia, I cant wait, I will get it right away, |
| 03:18.46 | hypa7ia | :) |
| 03:19.38 | [hC] | hypa7ia: coming to astricon again? :) |
| 03:19.41 | *** join/#asterisk CunningPike (n=CunningP@S010600095b33697f.vc.shawcable.net) |
| 03:21.39 | hypa7ia | [hC]: not this year :( |
| 03:21.46 | hypa7ia | i'm writing my CISSP that weekend |
| 03:21.52 | hypa7ia | need to study :/ |
| 03:21.54 | [hC] | Awww, im gonna have to represent canada all by myself? |
| 03:21.58 | [hC] | well i guess with juggie and pike, too |
| 03:22.00 | [hC] | :) |
| 03:22.14 | hypa7ia | i'm sure JunK-Y will be there, and Juggie |
| 03:22.21 | [TK]D-Fender | _X. <---- ICK |
| 03:22.23 | [hC] | good luck on your exam :) |
| 03:22.40 | [hC] | [TK]D-Fender: do you have any wildcardish preferences, or are you icking because of it? |
| 03:22.46 | hypa7ia | thanks [hC] |
| 03:22.51 | outtolunc | [TK]D-Fender: it was to get him up quickly |
| 03:22.51 | hypa7ia | i'll need it, it's tough |
| 03:23.13 | [TK]D-Fender | DID's should be hardcoded, stubs put up for disabled DID's, everything explicit. |
| 03:23.20 | [hC] | i like vendor neutral certifications |
| 03:23.32 | [hC] | [TK]D-Fender: I agree :) |
| 03:23.45 | [TK]D-Fender | You don't set up multiple DID's tu use some ridiculous catch-all.... |
| 03:26.10 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 03:26.24 | [TK]D-Fender | outtolunc, I opted for LASERS last yeear :p |
| 03:26.34 | outtolunc | hehe |
| 03:26.36 | hypa7ia | [hC]: it is in theory. it's still crap. |
| 03:26.46 | *** join/#asterisk [s]Animat (n=info@d220-238-210-46.dsl.vic.optusnet.com.au) |
| 03:26.53 | [hC] | hypa7ia: hence why i am cert free. |
| 03:26.54 | [hC] | :) |
| 03:27.01 | hypa7ia | lol |
| 03:27.18 | [TK]D-Fender | [hC], You really have to do something about that halitosis though ;) |
| 03:27.22 | hypa7ia | i will have CISSP, Sec+, some MS crap, and CCNA by the end of the year |
| 03:27.31 | [s]Animat | hello :) Anybody know why after I answer an incomming call from external SIP provider, I can no longer receive calls after hanging up? |
| 03:27.42 | *** join/#asterisk Shadowfire_ (n=jeff@rrcs-67-79-144-150.se.biz.rr.com) |
| 03:28.06 | [hC] | [TK]D-Fender: zzzzing! :P |
| 03:28.32 | [TK]D-Fender | [s]Animat, you'll have to show a call coming in with SIP debug enabled and channel dumps on both ends. |
| 03:28.46 | [TK]D-Fender | [s]Animat, PASTEBIN is your friend.... |
| 03:28.48 | [TK]D-Fender | ~pb |
| 03:28.48 | jbot | A Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 03:28.50 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^ |
| 03:29.08 | [s]Animat | [TK]D-Fender - Sure :) |
| 03:30.17 | riddlebox | outtolunc, what would be the preferred way of having an inbound call come into a context? |
| 03:33.49 | *** join/#asterisk hijacked (i=JqkA@66.255.220.17) |
| 03:35.38 | [TK]D-Fender | riddlebox, umm... that was a completely broken thought. Try sneaking up on it from a different angle :) |
| 03:35.43 | [s]Animat | [TK]D-Fender: I just tried it with a differnt softphone and it works fine ... cool. |
| 03:35.57 | [s]Animat | I'll still get the log now though |
| 03:36.20 | [TK]D-Fender | [s]Animat, Easily 60% of the problems we hear about seem to quicly vansih when we ask for proof :) |
| 03:36.28 | outtolunc | sorry got other helping occuring in background.. |
| 03:36.42 | [s]Animat | [TK]D-Fender: It's still a problem with the original softphone :P |
| 03:36.49 | [s]Animat | [TK]D-Fender: Which log? |
| 03:37.22 | [TK]D-Fender | [s]Animat, well if you've isolated an issue with a soft-phone, pick another... |
| 03:38.17 | [s]Animat | [TK]D-Fender: Which syntax should I use for PasteBin ? |
| 03:38.26 | outtolunc | riddlebox: just using the context=.. you use in zapata.conf with a list of DID to slice off those calls to send to another context to be processed should get you started |
| 03:38.29 | [TK]D-Fender | [s]Animat, NONE works for me... |
| 03:39.35 | [s]Animat | [TK]D-Fender: If you have time, please have a look at http://pastebin.com/d73f3202f and possibly provide any criticisms :P |
| 03:39.48 | [s]Animat | anybody else want to criticize can too :P |
| 03:40.11 | [TK]D-Fender | [s]Animat, I was asking for CLI output with SIP debug enabled... not a debug log... |
| 03:40.27 | [s]Animat | oh :\ |
| 03:40.42 | [s]Animat | Not sure how to do that so thanks anyway :) |
| 03:40.42 | [TK]D-Fender | [s]Animat, Thats right, I want to see EXACTLY how this call was processed |
| 03:40.50 | [TK]D-Fender | [s]Animat, "asterisk -r" |
| 03:41.00 | [TK]D-Fender | [s]Animat, "*CLI> sip debug" |
| 03:41.09 | [TK]D-Fender | [s]Animat, Cut&Paste |
| 03:41.21 | [s]Animat | whoa nice |
| 03:42.19 | riddlebox | outtolunc, but you would want everyone or most people in the same context, so how would I say zap channel 1 rings only on extension 522 |
| 03:42.33 | *** join/#asterisk rmayorga_ (n=rmayorga@unaffiliated/rmayorga) |
| 03:43.53 | [s]Animat | ok it's still broken with the other softphone |
| 03:44.01 | [s]Animat | that debug is great though |
| 03:45.48 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 03:46.27 | outtolunc | riddlebox: i actually am one to separate as much as you can .. and for me the only thing i usually have in [default] is a NoOp(what you doing here!) |
| 03:47.04 | [s]Animat | [TK]D-Fender: Any better? http://pastebin.com/d6bdc00ad |
| 03:47.20 | riddlebox | so you would separate them by different contexts then |
| 03:47.25 | [TK]D-Fender | riddlebox, I would always send EACT channel into its own context, and if "all roads lead to Rome", then you do that with a single Goto in your dialplan |
| 03:47.28 | [s]Animat | By the end incommnin calls are not accepted. |
| 03:49.07 | riddlebox | I see, thanks its all clearing up for me, outtolunc like I said earlier, I just need to learn the correct way of doing things |
| 03:49.19 | riddlebox | anyways thanks for the help and goodnight |
| 03:49.31 | outtolunc | yep, but to be able to do that you first need a working ivr <G> |
| 03:49.41 | outtolunc | which was what i was helping you do |
| 03:50.00 | outtolunc | goodnight |
| 03:50.05 | [TK]D-Fender | [s]Animat, Are you running a soft-phone on the same box as *? |
| 03:50.18 | [s]Animat | [TK]D-Fender: Yes |
| 03:51.05 | [TK]D-Fender | [s]Animat, I think thats it. * and your soft-phone are FIGHTING over the SIP port. Make sure your soft-phone uses 5061 (or ANYTHING except 5060) so that * is free to bind to 5060. |
| 03:51.07 | riddlebox | outtolunc, I have done a few things, but mostly with sip providers, I have also written a few AGI scripts one to do custom recordings in mythtv |
| 03:51.57 | [s]Animat | [TK]D-Fender: That's a very valid point! Thank you so much. A question, did you read a book to learn *? |
| 03:52.46 | *** join/#asterisk bmg505 (n=leon@196.209.176.121) |
| 03:53.39 | [TK]D-Fender | [s]Animat, No, I learned before there was a book of any kind. |
| 03:54.00 | [s]Animat | [s]Animat: Did you just use trial and error, forums, online resources, etc. ? |
| 03:54.04 | [TK]D-Fender | [s]Animat, But if makes you feel any better, I'm currently mirroring it while Asteriskdocs is down :) |
| 03:54.16 | [s]Animat | Or by reading the source? |
| 03:54.25 | [TK]D-Fender | [s]Animat, trial and error, my natural instincts and voip-info.org |
| 03:54.46 | [s]Animat | Voip instincts, eh? I do believe I lack those. |
| 03:54.57 | [s]Animat | Thanks for the help, mate :) |
| 03:55.24 | [TK]D-Fender | [s]Animat, I am by no means a coder. Virtually everything was on the WIKI. A complete app list. I learned programming by picking apart every piece of syntac. once you understand the bits its my nature to know where to use the bits to acheive my goals |
| 03:55.51 | [TK]D-Fender | [s]Animat, in sip.conf for your soft-phone on the * box do "port=5061" and set it as such in there. |
| 03:56.08 | [TK]D-Fender | [s]Animat, then restart * and make sure IT binds it properly. Then start up your soft phone and test. |
| 03:56.25 | [s]Animat | Yeah, have done mate. Will restart and test momentarily. |
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| 04:07.54 | [s]Animat | [TK]D-Fender: Works like a charm :) |
| 04:08.02 | [TK]D-Fender | [s]Animat, Cheers |
| 04:08.33 | [s]Animat | Thanks :) |
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| 04:33.18 | gaijinlah | hello, i have digium TE120 PCI card for PRI, it works on some old (from around year 2002) dell machines |
| 04:33.31 | gaijinlah | does it work on PCI-X slots |
| 04:33.43 | gaijinlah | on SunFire servers for example? |
| 04:33.56 | gaijinlah | anyone has any idea? |
| 04:34.41 | Strom_M | gaijinlah: if you have all the parts at your disposal, why not just try it? |
| 04:35.32 | *** join/#asterisk outtolunc (n=me@c-67-174-216-60.hsd1.ca.comcast.net) |
| 04:52.42 | *** join/#asterisk TedNJ38 (n=HungLad@ool-4573adc7.dyn.optonline.net) |
| 04:54.53 | TedNJ38 | Can someone help me please? I live in USA, I have a trixbox server here and I have a SIP phone here as well. I want to purchase a voip landline in a specific country (Colombia), I want to tie it to my trixbox server so when I dial 9 and the phone number, I am making the phone call as if it were a local call in Colombia and when someone dials the number of that phone line, I want it to ring it here. What is that service called? |
| 04:55.18 | gaijinlah | Strom_M, in fact I tried |
| 04:55.22 | gaijinlah | and lspci shows the card |
| 04:55.33 | gaijinlah | but the card's lamps were off |
| 04:55.43 | gaijinlah | so i wonder if anyone else has any experience |
| 04:57.18 | tzafrir_laptop | gaijinlah, lsmod | grep zaptel |
| 04:57.18 | Strom_M | TedNJ38: what, pray tell, is a "voip landline"? |
| 04:57.48 | TedNJ38 | So, I should google for a voip landline in Peru? |
| 04:58.19 | tzafrir_laptop | DID? |
| 04:58.23 | Strom_M | TedNJ38: what is this "voip landline" thing you keep asking about? |
| 04:58.35 | TedNJ38 | Storm: I have no idea as of what it is called. |
| 04:58.39 | Strom_M | a landline, by definition, is not voip |
| 04:58.47 | TedNJ38 | I want to purchase a number in Colombia. |
| 04:58.55 | Strom_M | you want a DID then |
| 04:58.56 | TedNJ38 | I want to use it to make local phone calls within Colombia. |
| 04:59.02 | TedNJ38 | So, it is called a DID? |
| 04:59.10 | Strom_M | look for a colombian ITSP |
| 04:59.10 | TedNJ38 | I will google for DID then, thanks. |
| 04:59.22 | TedNJ38 | Thanks. |
| 05:00.36 | Strom_M | no no no. |
| 05:00.42 | Strom_M | look for a colombian ITSP |
| 05:02.06 | TedNJ38 | Ok. |
| 05:02.07 | TedNJ38 | Thanks. |
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| 05:05.35 | rob0 | Ahoy! Methinks a landline is fer landlubbers! Arrrr. |
| 05:05.47 | *** part/#asterisk TedNJ38 (n=HungLad@ool-4573adc7.dyn.optonline.net) |
| 05:09.14 | hypa7ia | rob0: ye be corrrrrrect in that assertion, me matey |
| 05:24.31 | rpm | in mexico, anyone know if there are any other dsl providers than telmex? |
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| 05:50.42 | ironhead_webby | hi all can anyone help with a fairly simple problem, I am connected to freshtel and with a softphone and I would now like to connect to freshtel via an asterisk server that I have installed on my network. I have googled for examples and tried quite a few. I can get the softphone to register with my asterisk server but I can't get it to do an outgoing call ( I am not needing incoming) Can some one help with my iax.conf and |
| 05:50.42 | ironhead_webby | extensions.conf ? |
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| 06:09.58 | trippss | can anyone here recommend a) a sip to pstn gateway appliance, and b) pri pci card for use with *? |
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| 06:12.22 | J4zen | Hi there |
| 06:12.33 | J4zen | Could i bother you for one minute? |
| 06:14.00 | J4zen | Is there any guide explaining the basic installation of asterisk itself, i have tried AsteriskNOW and played around with it no problem. Now with Asterisk i have it running on a QuadBRI card, but i fear i did something wrong. Could anyone lend me a hand for 5 minutes? |
| 06:14.10 | *** join/#asterisk unixdog (n=unixdog@adsl-69-234-212-135.dsl.irvnca.pacbell.net) |
| 06:14.53 | unixdog | anyone got a zapata.conf for a tdm400 with 4 fxo ports |
| 06:15.00 | unixdog | that I cn look at |
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| 06:25.19 | unixdog | anyone got a zapata.conf for a tdm400 with 4 fxo ports I can look at |
| 06:25.44 | unixdog | for the us |
| 06:27.40 | teolicy | Hello. I'm very new to Asterisk, but am proficient with Linux. I'd like to build a home IVR system as a hobby. I was unable at a glance to understand what kind of hardware should I buy, to make sure the costs are reasonable at all. Anyone can recommend a simple PCI card to use with a single (maximum dual) PSTN phone line? |
| 06:28.31 | tzafrir_laptop | unixdog, you, after running genzaptelconf |
| 06:28.36 | tzafrir_laptop | (or even zapconf) |
| 06:30.16 | unixdog | yes |
| 06:30.51 | tzafrir_laptop | J4zen, I'm not sure how well AsteriskNOW supports misdn hardware. Or digital zaptel hardware |
| 06:31.41 | awk | tzafrir well you on a roll do you mind answering my simple question |
| 06:31.50 | *** join/#asterisk yassaccan (n=yassacca@admin189.hgo.se) |
| 06:33.31 | teolicy | I'm thinking along the lines of a Digium TDM10B, but I'm not sure if even that is not an overkill, or if I should even look at a different manufacturer (not Digium). |
| 06:33.51 | awk | Sangoma all the way... |
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| 06:38.52 | unixdog | I have to use what clients want |
| 06:39.11 | unixdog | some want sangoma some want rhino some want digium |
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| 06:41.47 | teolicy | Cheers folks. Later. |
| 06:51.47 | tzafrir | awk, what is your question? |
| 06:59.07 | awk | I posted it in #asterisk-dev |
| 06:59.18 | awk | ok here is one for the charts |
| 06:59.48 | awk | anyone had this issue, cisco ip phone, 7912 series, and a hp procurve switch the phones wont turn on |
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| 07:16.49 | JT | awk: do the phones have power? |
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| 07:27.34 | J4zen | Anyone running asterisk / openPBX on fedora 7? |
| 07:28.31 | J4zen | or a Junghanns QuadBRI 2.0 PCI card? |
| 07:35.27 | modu | bye |
| 07:35.29 | *** part/#asterisk modu (n=modu@rue92-6-82-237-172-115.fbx.proxad.net) |
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| 07:38.04 | awk | never mind ive seen a review its an issue with cisco |
| 07:38.12 | awk | fuck waste of 150 phones |
| 07:39.03 | awk | ok looks like I can swop 2 pairs around |
| 07:42.44 | booray | hmm.. i could just nap in this |
| 07:42.48 | booray | rather than work like i intended |
| 07:48.09 | FlatFoot | morning all |
| 07:48.26 | FlatFoot | anyone used the Linksys SPA-941 at all ? |
| 07:48.38 | FlatFoot | any reports on quality etc |
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| 08:14.26 | awk | swapping 4 with 7, and 5 with 8 you could use a standard |
| 08:14.28 | awk | POE injector to power the phone |
| 08:14.33 | awk | pfft, cisco! |
| 08:14.57 | awk | <awk> hi, hmm, wondering in asterisk manager how can I get a start and end time logged |
| 08:14.57 | awk | <awk> I can see pretty much everything but no start/end times |
| 08:15.16 | awk | <awk> I know I could use something like getdate() or something to that affect at start and / end of an event |
| 08:15.16 | awk | <awk> but was hoping a raw data stream through the manager would show start/end times |
| 08:19.26 | luke-jr | awk: ice cream is good |
| 08:20.15 | Uatec | "ast_yyerror: If you have questions, please refer to doc/README.variables in the asterisk source." |
| 08:20.19 | Uatec | THIS IS BUSINESS EDITION |
| 08:20.24 | Uatec | i don't have the source you morans |
| 08:20.31 | luke-jr | so download it |
| 08:20.32 | luke-jr | moron |
| 08:20.37 | luke-jr | :) |
| 08:27.12 | tzafrir | Uatec, so kindly ask it from the people who sold it to you |
| 08:27.30 | *** join/#asterisk qdk (n=qdk@213.150.62.32) |
| 08:27.56 | Uatec | luke-jr, that's not the point |
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| 08:28.12 | Uatec | tzafrir, i was going to, but i've figured out the problem |
| 08:28.18 | Uatec | but no help to the error message |
| 08:28.43 | tzafrir | Uatec, http://svn.digium.com/svn/asterisk/branches/1.2/doc/ |
| 08:29.01 | tzafrir | That file does not really change that often |
| 08:30.01 | tzafrir | I think you should suggest them to include that documentation in the asterisk package, if it is not already included |
| 08:30.32 | Uatec | yeah, i might |
| 08:30.36 | Uatec | infact |
| 08:37.00 | yang | H |
| 08:37.35 | yang | hrm, DTMF payload type on my phone is 101 , does that match the RFC2833 type ? |
| 08:42.38 | *** join/#asterisk darviria (n=darviria@194-105-181-29.ifb.co.uk) |
| 08:43.13 | Zeeek | . |
| 08:43.14 | awk | why is it nobody ever asnwerds my question |
| 08:43.20 | Zeeek | Someone stole my CLI prompt! |
| 08:43.25 | awk | somebody must know a way to get start/end time through ami |
| 08:43.29 | awk | stfu |
| 08:43.52 | Zeeek | tzafrir why does my 1.4 prompt not come up after every dialplan action? |
| 08:44.05 | Zeeek | something ishorribly wrong |
| 09:03.28 | *** join/#asterisk act1v8 (n=stojance@unaffiliated/act1v8) |
| 09:03.32 | J4zen | should asterisk be running in /usr/src ( fedora 7 ) r should it actually be in /etc/asterisk/ ? |
| 09:03.40 | act1v8 | What is Asterisk? I really don't understand |
| 09:04.17 | tzafrir | Zeeek, hmm... |
| 09:04.39 | tzafrir | J4zen, asterisk should be running from /usr/sbin/asterisk |
| 09:04.53 | J4zen | Ok, thank you |
| 09:05.12 | act1v8 | I really don't understand. is it software to create your own telephone company? or is it software for your phone on the computer? |
| 09:05.21 | tzafrir | asterisk.conf is by default /etc/asterisk/asterisk.conf . And it lists the pathes asterisk uses. Otherwise its internal defaults are used |
| 09:05.41 | *** join/#asterisk defswork (n=andy@83.105.96.154) |
| 09:06.23 | tzafrir | act1v8, http://voip-info.org/ , http://asterisk.org/ might be good starting points |
| 09:06.55 | tzafrir | Generally: to create your own telephone company |
| 09:07.04 | *** join/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-01e94625a1ba77ee) |
| 09:07.16 | act1v8 | oookkk |
| 09:07.21 | tzafrir | or even your own little home PBX |
| 09:07.57 | act1v8 | cool :) |
| 09:08.05 | tzafrir | Zeeek, IIRC the prompt should be reprinted after each line of output |
| 09:09.57 | *** join/#asterisk ZX81 (n=matt@202.20.97.211) |
| 09:10.18 | ZX81 | is there a digium hardware support forum - or can anyone help me with a b410p problem? |
| 09:10.35 | ZX81 | by forum I mean freenode |
| 09:10.52 | ZX81 | I'm too dead to call 500 |
| 09:12.03 | ZX81 | I've got a real cool problem - if I call out a mISDN b410p line via SIP its fine (alaw/gsm) if I call via IAX2 it works with gsm but breaks up bad with alaw |
| 09:12.09 | ZX81 | stock installs |
| 09:12.37 | ZX81 | sounds like samples are going at a different rate |
| 09:13.03 | ZX81 | meh, anyone know what the time is in digium land? |
| 09:13.15 | gremzoid | my clock says 19:13 |
| 09:13.22 | gremzoid | but i'm in upside down land... |
| 09:13.27 | ZX81 | ah mine says 9:13pm |
| 09:13.28 | ZX81 | :) |
| 09:13.40 | gremzoid | kiwi? |
| 09:13.51 | ZX81 | i.e. yeah |
| 09:13.58 | ZX81 | :) |
| 09:14.36 | ZX81 | I've managed to somewhat reduce the problems with the card by changing polling to 64 but its still not the same as the sip call |
| 09:14.42 | gremzoid | ... is it normal to not get DTMF tones from a SIP trunk to an IAX exten? |
| 09:14.53 | ZX81 | I'd look at the sip end! |
| 09:14.54 | ZX81 | :) |
| 09:15.00 | ZX81 | what format dtmf? |
| 09:15.02 | ZX81 | info? |
| 09:15.04 | gremzoid | great... |
| 09:15.04 | ZX81 | rfc2833 |
| 09:15.11 | gremzoid | sip end is a siemens hg1500 |
| 09:15.18 | ZX81 | play around with the dtmfmode settings in sip.conf |
| 09:15.20 | gremzoid | it dosn't play nice or configure well :P |
| 09:15.31 | ZX81 | so set it to ulaw/alaw and inband :) |
| 09:15.42 | gremzoid | tried that... and finally got some DTMF (IE SIP trunk to IVR menu works) |
| 09:15.43 | ZX81 | in asterisk i mean |
| 09:15.47 | ZX81 | just |
| 09:15.48 | ZX81 | :) |
| 09:16.06 | ZX81 | but sometimes it detects multiple/no digits? |
| 09:16.26 | gremzoid | just an incoming call over the SIP trunk won't send DTMF to an IAX exten on the asterisk box (works the other way tho in the same call) |
| 09:16.47 | ZX81 | but sends to the ivr? |
| 09:16.52 | gremzoid | ya |
| 09:16.55 | gremzoid | weird huh? |
| 09:16.59 | ZX81 | I'd say impossible but |
| 09:17.00 | ZX81 | :) |
| 09:17.11 | ZX81 | no messages on remote site of iax connection? |
| 09:17.18 | ZX81 | you sending to a box or a client? |
| 09:17.39 | gremzoid | in the same call, i can press buttons on the IAX phone and hear them on the digital phone that hangs off the HG1500 |
| 09:17.54 | gremzoid | but not back the other way... |
| 09:18.07 | ZX81 | yeah, but the iax phone probably won't generate iax tones to the listener |
| 09:18.13 | ZX81 | the tones are out of band for iax |
| 09:18.22 | gremzoid | is it some bug in asterisk, or something siemens have done (or not) |
| 09:18.32 | ZX81 | I dunno, some phones might - but why would you want to hear dtmf tones in your ear |
| 09:18.39 | *** join/#asterisk tcseke (n=chatzill@22-36.adsl.etel.hu) |
| 09:18.48 | ZX81 | if the siemens gets it to the ivr then there is no prob there |
| 09:18.53 | gremzoid | well what if i wanted an IAX trunk |
| 09:18.54 | gremzoid | ? |
| 09:19.01 | gremzoid | which is a possibility soon... |
| 09:19.03 | ZX81 | yeah that would use the dtmf |
| 09:19.28 | gremzoid | reckon i should submit it as a bug? |
| 09:19.37 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 09:19.47 | ZX81 | have you got another asterisk machine to test a trunk on? |
| 09:19.49 | ZX81 | also |
| 09:19.55 | ZX81 | you can do a iax2 debug |
| 09:20.07 | ZX81 | and you should see the dtmf being sent to the iax2 phone |
| 09:20.19 | gremzoid | hmmm thats a possibility |
| 09:20.20 | ZX81 | oh except in 1.4 its now iax2 set debug |
| 09:20.38 | gremzoid | oh wait... i'm not at work... i can't actually _answer_ the phone :P |
| 09:21.24 | ZX81 | heh |
| 09:22.29 | ZX81 | the problem with the b410p that I was having is on a customer's site that uses a crappy VPN to access them - and I can't be connected to the regular net at the same time |
| 09:22.38 | ZX81 | so I have to do one thing at a time |
| 09:22.39 | ZX81 | grrr |
| 09:27.13 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 09:36.26 | *** join/#asterisk Delvar (n=Delvar@77.240.56.18) |
| 09:40.31 | *** join/#asterisk modu (n=modu@rue92-6-82-237-172-115.fbx.proxad.net) |
| 09:40.45 | modu | hi |
| 09:41.45 | modu | Can someone tell me why this kind of line is bad : exten => _ZXXX,1,Dial(SIP/${EXTEN}, 20) |
| 09:42.09 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 09:42.53 | JT | Z? |
| 09:42.58 | modu | yesterday someone say me that I should use one exten by phone |
| 09:43.03 | JT | space before the 20 maybe |
| 09:43.27 | modu | it's not the syntax but the concept |
| 09:43.59 | modu | someone say that if you have 100 phone you may add 100 lines in extensions with each phone |
| 09:44.35 | modu | for a administration pont of view it's crazy for me |
| 09:44.39 | modu | point |
| 09:44.43 | JT | it's the way to do it |
| 09:44.46 | JT | you use a database |
| 09:44.50 | JT | not text configuration |
| 09:44.52 | JT | for 100 users |
| 09:45.26 | modu | what way ? without _ZXX ?? |
| 09:46.07 | JT | your sip phone should not have the same id as the extension number |
| 09:46.16 | JT | it's not the best practice |
| 09:46.19 | thewiizle | hey |
| 09:46.21 | modu | why not ? |
| 09:46.24 | thewiizle | anyone got astcc working? |
| 09:46.40 | JT | if you want multiple line appearances, or to easily reasign extensions to phones |
| 09:46.45 | modu | if I define each phone with a correct callerid in sip.conf ? |
| 09:47.00 | JT | if you have 100 phones, use a database |
| 09:47.02 | JT | as i said |
| 09:47.32 | *** join/#asterisk tsurko (n=tsurko@78.90.100.214) |
| 09:47.53 | modu | I've not see database in /etc/asterisk/* (only for cdr) |
| 09:48.21 | tzafrir | modu, so generate a portion of sip.conf from a database |
| 09:48.34 | tzafrir | pick your favorite database |
| 09:48.44 | modu | my exten is only for internal calls |
| 09:48.54 | modu | only calls with 4 digits |
| 09:49.22 | tzafrir | (a plain text "table" file is also a database. Limited, but very trasparent) |
| 09:50.08 | modu | I don't understand why _ZXXX,1,Dial(SIP/${EXTEN} is bad for internal calls |
| 09:50.21 | modu | it's simplier, so less errors |
| 09:50.58 | tzafrir | modu, who said it's bad? |
| 09:51.25 | tzafrir | It's good because it's simple |
| 09:51.30 | modu | JT no ? |
| 09:52.01 | tzafrir | You may run into some limitations of it, because it's a bit too simplistic |
| 09:52.20 | modu | nd yesterday [TK]D-Fender |
| 09:52.35 | modu | yes I must add Voicemail |
| 09:52.59 | modu | but what other limitation ? |
| 09:53.49 | tzafrir | modu, if you like that simplistic approach, consider just uisng users.conf |
| 09:54.21 | modu | what limitation ? why simplistic ? |
| 09:55.01 | tzafrir | sometimes one device per extension just doesn't fit |
| 09:55.30 | modu | Yes but I can add specific numbers for pools |
| 09:55.31 | tzafrir | modu, another handy tool that might help you with automation is configuration templates |
| 09:55.43 | tzafrir | [template](!) |
| 09:55.50 | modu | automation is bad |
| 09:55.50 | tzafrir | some configu items |
| 09:55.59 | tzafrir | [phone1](template) |
| 09:56.31 | modu | I'm looking for a clean config file, and edit it with vi |
| 09:56.44 | tzafrir | "automation" is good - it saves work. Automation that limits you is bad |
| 09:57.03 | tzafrir | Right. So with templates you can write less |
| 09:57.09 | Zeeek | ITAD ROCKS!!!! |
| 09:57.24 | modu | I don't agree, because automation will generate a hudge config file |
| 09:57.42 | tzafrir | [local-phones](!) |
| 09:57.48 | tzafrir | some config values |
| 09:58.19 | modu | perhaps I don't understand |
| 09:58.26 | tzafrir | [123](local-phones) |
| 09:58.59 | modu | what does this line means ? |
| 09:59.24 | tzafrir | http://svn.digium.com/svn/asterisk/branches/1.4/doc/configuration.txt |
| 09:59.51 | tzafrir | (works just as well in 1.2) |
| 10:00.37 | tzafrir | Think of inheritence of classes, if you have some programming background |
| 10:01.00 | modu | this is for sip.conf ? not extension ? |
| 10:01.23 | tzafrir | just the same. This works for every asterisk config file |
| 10:01.51 | modu | yes but I want to have one line per phone in sip.conf (not a problem for me) |
| 10:02.02 | modu | but a simple config in extension.cnf |
| 10:02.20 | modu | I just don't want to have dupplicate config |
| 10:02.34 | J4zen | Does anyone have any guides that show the basic outline on how to setup a new asterisk installation with BRI cards ( or even without ! ) |
| 10:02.41 | modu | like phone n°1 define in sip.conf and in extension.conf |
| 10:03.22 | tzafrir | J4zen, for startes: http://updates.xorcom.com/astribank/bristuff/INSTALL.html |
| 10:03.49 | modu | lunch time, see you |
| 10:04.45 | tzafrir | anything there is not clear, or needs fixing? |
| 10:05.10 | JT | modu: you talk a lot of crap |
| 10:05.14 | JT | "automation is bad" ... |
| 10:05.23 | JT | automation is SMART |
| 10:05.35 | thewiizle | so |
| 10:05.37 | thewiizle | astcc |
| 10:05.40 | thewiizle | any ideas |
| 10:05.47 | JT | thewiizle: so no-one knows, ok |
| 10:06.10 | thewiizle | smoneone must |
| 10:06.18 | thewiizle | its in the cvs surely someone must have an idea |
| 10:06.37 | JT | yes surely everyone on irc knows everything about ever bit of asterisk |
| 10:06.43 | thewiizle | yes surely thats right |
| 10:06.58 | JT | you are deluded |
| 10:07.00 | modu | JT: automation is goof, but no for generate a config file that can be done whiht macro |
| 10:07.05 | thewiizle | and you are nieve |
| 10:07.05 | modu | good |
| 10:07.14 | thewiizle | nice to meet you |
| 10:07.16 | JT | thewiizle: please speak english in here. |
| 10:07.25 | thewiizle | JT, instead of what? |
| 10:07.30 | JT | <PROTECTED> |
| 10:07.40 | thewiizle | was that not english? |
| 10:07.53 | JT | that last word wasn't |
| 10:08.05 | thewiizle | urm |
| 10:08.09 | thewiizle | are you sure about that? |
| 10:08.12 | JT | yes |
| 10:09.05 | thewiizle | ah my bad, typo |
| 10:09.08 | JT | https://www.wsu.edu/~brians/errors/nieve.html |
| 10:09.08 | thewiizle | niave |
| 10:09.15 | JT | still wrong |
| 10:09.16 | modu | The only thing i'm looking for is to use generic macro in extension |
| 10:09.32 | modu | but everyone one seems to prefer have one line per phone |
| 10:09.50 | thewiizle | s/a/i, /si/a |
| 10:10.00 | thewiizle | Concept and principle remain |
| 10:10.04 | modu | I want a config file that can be easily check by an admin |
| 10:10.58 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
| 10:12.28 | modu | no one agree with that ? |
| 10:12.49 | thewiizle | modu to do what? |
| 10:13.15 | modu | use a simple extension.conf file with macro |
| 10:13.24 | modu | and not define one phone per line |
| 10:13.30 | thewiizle | to call extensions? |
| 10:13.40 | thewiizle | sip users |
| 10:13.41 | modu | to link phone <-> extension |
| 10:13.50 | thewiizle | so, 101 --> 102 for example |
| 10:14.01 | modu | non 101 = SIP/101 |
| 10:14.33 | thewiizle | exten => _XXX,1,Dial(SIP/${EXTEN}) ? |
| 10:14.39 | modu | and not have to define "exten 101,Dial(SIP/101)" |
| 10:14.48 | thewiizle | thatll match three digits and dial it |
| 10:15.07 | modu | yes but if I have 100 phone (1..100) |
| 10:15.25 | thewiizle | that line will dial them all |
| 10:15.27 | modu | I don't want to have 100 lines |
| 10:15.47 | thewiizle | put that in your extensions.conf under from-internal |
| 10:16.01 | modu | yes it's exactly what I ask |
| 10:16.01 | Zeeek | good heavens |
| 10:16.08 | tzafrir | modu, sure. Clarity is always good |
| 10:16.15 | thewiizle | i had the same problem |
| 10:16.16 | modu | (with my poor english :-) |
| 10:16.20 | thewiizle | that line seems to work for me |
| 10:16.47 | modu | ok, really lunch now :-) |
| 10:17.27 | Zeeek | well, 1.4.11 seems to be working |
| 10:21.08 | JT | thewiizle: astcc isn't even part of base asterisk |
| 10:23.54 | *** join/#asterisk IvanV3835 (n=Miranda@styx.mcn.ru) |
| 10:29.46 | *** join/#asterisk axscode (n=axscode@203.213.217.123) |
| 10:30.08 | axscode | how to configure all my ZAP channels will use g729? |
| 10:31.35 | *** part/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-01e94625a1ba77ee) |
| 10:33.14 | ai-a | axscode: isdn is either ulaw or alaw comming in from your provider. |
| 10:33.58 | JT | ie G.711 |
| 10:35.23 | santibiotico | the zap channel is g711, if u want, u can have your stations using g729 and use transcoding... |
| 10:38.05 | axscode | oh.. so if my SIP phones will use g729 calling the ZAP, meaning it will use transcoding? thast why i need the license? |
| 10:38.25 | axscode | thats why it will use the g729 i mean. |
| 10:39.35 | axscode | what file to configure to allow video calls? even if its pass through? |
| 10:40.25 | JT | axscode: yes you will need to transcode. |
| 10:53.29 | *** join/#asterisk `Sean (i=Un1x@99.229.52.244) |
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| 10:59.28 | *** join/#asterisk Delvar (n=Delvar@77.240.56.18) |
| 11:05.53 | ai-a | Why not use g711 thoughout the phone system ? |
| 11:06.15 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 11:20.46 | *** join/#asterisk geoaxis (n=hatim@unaffiliated/geoaxis) |
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| 12:10.56 | *** join/#asterisk mohsen (n=chatzill@81.31.160.140) |
| 12:11.34 | mohsen | hmmm. i have problems using g729 (registered) with asterisk. |
| 12:11.36 | mohsen | [Sep 19 00:54:35] WARNING[3367]: codec_g729.c:420 load_module: Failed to initialize G.729 copy protection! |
| 12:11.47 | mohsen | anyone knows the problem? |
| 12:14.46 | *** join/#asterisk s0ck (n=m@unaffiliated/s0ck) |
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| 12:17.36 | *** part/#asterisk CtRiX (n=CtRiX@aretha.navynet.it) |
| 12:17.38 | s0ck | topic #astricon! is pointing at the wrong channel |
| 12:18.24 | *** join/#asterisk jmesquita (n=jmesquit@c915230f.virtua.com.br) |
| 12:18.55 | *** join/#asterisk lirakis (n=lirakis@65.200.191.253) |
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| 12:19.00 | lirakis | morning |
| 12:22.38 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 12:24.23 | *** join/#asterisk pzn (i=foobar@201-13-109-92.dsl.telesp.net.br) |
| 12:26.20 | pzn | Hi! is it possible to use an externa program to control asterisk in this way? 1- it makes phones at sip:123@ip and sip:456@ip to ring; 2-when the first phone answers, it plays an .wav file; 3- when the secons phone answers, it stops .wav file and connect both lines. is this possible? will I have to hack inside asterisk? |
| 12:27.45 | pzn | the reason for this is like an "alarm system", when the external program detects an alarm situation, it rings the operator and the person at alarm location at both time and connect each other automatically. |
| 12:28.33 | [TK]D-Fender | pzn: lookup "call files", "asterisk auto answer", and "ami originate" on the WIKI |
| 12:28.34 | [TK]D-Fender | ~wikis |
| 12:28.35 | jbot | wikis is probably http://www.voip-info.org |
| 12:34.32 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
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| 12:45.08 | *** join/#asterisk VijayG (i=VijayG@58.68.47.115) |
| 12:45.37 | VijayG | Hello |
| 12:46.52 | pzn | [TK]D-Fender, thanks! |
| 12:47.37 | VijayG | i need to configure calling cards in my asterisk |
| 12:47.51 | VijayG | box |
| 12:48.10 | VijayG | anyway, i can set that in system, so that it can automatically dial access number and pin number |
| 12:48.19 | VijayG | or anybody knows of similar software |
| 12:50.12 | VijayG | hello |
| 12:50.18 | VijayG | anybody there? |
| 12:51.41 | *** join/#asterisk Polis_ttt (n=Polis_tt@194-237-172-225-no48.business.telia.com) |
| 12:51.43 | mohsen | VijayG: google for asterisk2billing and astpp |
| 12:53.57 | VijayG | i need to make calls using calling cards |
| 12:54.00 | VijayG | but i want a auto system |
| 12:54.10 | *** join/#asterisk Siya (n=djerk@217-195-248-251.dsl.easynet.nl) |
| 12:54.13 | Siya | ello |
| 12:54.18 | VijayG | it shouldn't be like i have to dial the access number and pin number again and again |
| 12:55.03 | VijayG | you think asterisk2billing or astpp will support this? |
| 12:55.05 | Siya | do I need SRV dns record in place or can I test with sip:ext@domain.com ? |
| 12:55.09 | tzanger | morning |
| 12:55.17 | mohsen | VijayG: asterisk2billing can authenticate using callerid |
| 12:55.30 | lirakis | VijayG: use send dtmf |
| 12:56.03 | lirakis | VijayG: are you saying you want a calling card system? .. or you want to automate calls and you have to use a calling card? |
| 12:56.54 | VijayG | i want to use calling card |
| 12:57.02 | VijayG | i want to configure that on my asterisk system |
| 12:57.23 | VijayG | so that anyone dialing through asterisk server, need not to dial access number or pin number |
| 12:57.25 | mohsen | then take lirakis advise. I misunderstood you |
| 12:57.36 | VijayG | ok |
| 12:57.52 | *** join/#asterisk michael-i (n=michael-@Lfcce.l.pppool.de) |
| 12:59.03 | [TK]D-Fender | VijayG: "show application dial" <-------------- |
| 12:59.10 | VijayG | ok |
| 12:59.21 | thewiizle | VijayG |
| 12:59.25 | thewiizle | A2Billing does it based on CLI |
| 12:59.48 | thewiizle | You can manually add CLis to a Card or after first auth it remembers it |
| 13:00.52 | [TK]D-Fender | thewiizle: You're on the wrong track for his needs..... |
| 13:02.33 | McDouglas | hmm, for some reason i cant send fax from our external fax machine |
| 13:03.03 | [TK]D-Fender | McDouglas: Try asking in #radioshack |
| 13:03.13 | [TK]D-Fender | :D |
| 13:03.17 | McDouglas | obviously, with asterisk |
| 13:04.01 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:04.08 | McDouglas | machine is plugged into an fxs port on a digium card |
| 13:04.31 | McDouglas | and the outgoing channel is provided by a digium b410p |
| 13:04.33 | Siya | do I need SRV dns record in place or can I test with sip:ext@domain.com ? |
| 13:07.36 | *** join/#asterisk Pelalil (n=thomas@jack.nibsc.ac.uk) |
| 13:11.16 | *** join/#asterisk solar_ant (n=solar@122.164.55.250) |
| 13:13.47 | *** join/#asterisk mitcheloc (n=mitchel@adsl-68-120-230-95.dsl.irvnca.pacbell.net) |
| 13:14.00 | JT | McDouglas: misdn, good luck with that |
| 13:14.02 | *** join/#asterisk Corydon76-home (i=indigo@pdpc/supporter/sustaining/Corydon76-home) |
| 13:14.02 | *** mode/#asterisk [+o Corydon76-home] by ChanServ |
| 13:14.12 | *** join/#asterisk Corydon76-dig (i=one@pdpc/supporter/sustaining/Corydon76-home) |
| 13:14.12 | *** mode/#asterisk [+o Corydon76-dig] by ChanServ |
| 13:14.13 | *** join/#asterisk deeperror (n=deeperro@69-215-202-195.ded.ameritech.net) |
| 13:15.45 | _x86_ | heh |
| 13:15.48 | _x86_ | #radioshack |
| 13:15.50 | _x86_ | that's awesome |
| 13:15.57 | *** join/#asterisk _ShrikE (n=ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 13:15.59 | [TK]D-Fender | pwned |
| 13:16.03 | _x86_ | ;) |
| 13:16.06 | _x86_ | jesus christ |
| 13:16.10 | deeperror | ? |
| 13:16.18 | McDouglas | JT: actually, i narroved the problem down: i plugged a phone into the fax's extension, and if i dial the other zap extension i get a busy tone |
| 13:16.23 | [TK]D-Fender | Siya: Dial(SIP/joe@domain.com) |
| 13:16.27 | *** join/#asterisk bintut (n=bintut@cm47.gamma178.maxonline.com.sg) |
| 13:16.35 | _x86_ | AT&T sucks for CAS T1 service |
| 13:16.38 | McDouglas | so it has something to do with my analog card |
| 13:17.01 | [TK]D-Fender | McDouglas: gee maybe you should PASTEBIN the call attempt so we have something worth commenting on :p |
| 13:17.05 | [TK]D-Fender | ~pm |
| 13:17.06 | jbot | hmm... pm is project manager, or private message, or perl mongers, or pathetic moron: when you see someone say pm, they're asking if you think that they're a pathetic moron, or something you don't do without asking permission |
| 13:17.16 | JT | ~typo |
| 13:17.16 | jbot | typo is, like, when someone tries to type really fast without knowing what he is actually doing.... so he makes typing mistakes, or rm -rf / |
| 13:17.32 | [TK]D-Fender | McDouglas: Oh... and NO we don't trust your dialplan and that last comment has already been struck from the record... |
| 13:17.39 | [TK]D-Fender | :p |
| 13:17.57 | McDouglas | [TK]D-Fender: i dont need pastebin for it, its only 2 lines |
| 13:17.57 | McDouglas | <PROTECTED> |
| 13:17.57 | McDouglas | <PROTECTED> |
| 13:18.07 | *** join/#asterisk billybongo (n=rich@85-189-96-153.rcg-global.managedbroadband.co.uk) |
| 13:18.19 | deeperror | ? |
| 13:18.19 | McDouglas | thats all i see in the console when i try dialing the other phone connected to the other fxo port |
| 13:18.28 | [TK]D-Fender | McDouglas: pastebin the full call attempt so we can see what-s dialed where, and your zapatal.conf & zaptel.con |
| 13:18.49 | [TK]D-Fender | McDouglas: And your dialplan. |
| 13:19.01 | JT | McDouglas: that is NOT a dialplan |
| 13:19.03 | McDouglas | [TK]D-Fender: that was the FULL call attempt |
| 13:19.04 | JT | etc :) |
| 13:19.22 | JT | McDouglas: extensions.conf zapata.conf zaptel.conf |
| 13:19.33 | JT | misdn.conf |
| 13:19.55 | McDouglas | http://pastebin.com/d5a82092a |
| 13:20.00 | *** join/#asterisk snook3r (n=snook3r@bzq-219-46-202.isdn.bezeqint.net) |
| 13:20.00 | McDouglas | extensions.conf |
| 13:20.32 | McDouglas | zapata.conf http://pastebin.com/d9ad7944 |
| 13:21.00 | thewiizle | woohoo |
| 13:21.03 | *** join/#asterisk ManxPower (n=manxpowe@243.sub-70-223-238.myvzw.com) |
| 13:21.07 | thewiizle | remotely provisioned my first 7960 |
| 13:21.23 | [TK]D-Fender | McDouglas: So like.... WHERE'S context=Internal ; Uses the [internal] context in extensions.conf <----------------- |
| 13:21.27 | thewiizle | all with details dynamically created from my users database :) |
| 13:21.42 | [TK]D-Fender | McDouglas: your Zap/1 & Zap/2 have NOWHERE TO GO! |
| 13:21.47 | webtech_m33 | http://paste.debian.net/37488 |
| 13:21.58 | *** join/#asterisk jfitzgibbon (n=chatzill@64.72.237.187) |
| 13:22.01 | McDouglas | [TK]D-Fender: oh damn.. forgot about that when i was palying with the sip restriction :( |
| 13:22.10 | [TK]D-Fender | McDouglas: .... schmuck :p |
| 13:22.20 | [TK]D-Fender | heheh |
| 13:22.28 | [TK]D-Fender | NEXT@!@@@!@ (c) BKW |
| 13:25.21 | tzanger | gavels hahahaha |
| 13:30.26 | lirakis | i am trying to restart asterisk and i am getting "load module cdr_pgsql.so failed" |
| 13:30.49 | lirakis | what package is that part of?? .. i have tried recompiling and isntalling asterisk-addons |
| 13:31.12 | ManxPower | lirakis: do you need Postgress CDR logging? |
| 13:31.29 | lirakis | ManxPower: yes .. stupid queuestats program requires it |
| 13:31.47 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 13:31.47 | *** mode/#asterisk [+o anthm] by ChanServ |
| 13:33.19 | [TK]D-Fender | lirakis: part of Astrisk main... |
| 13:33.24 | ManxPower | I can tell you how to disable it, but not how to fix it. |
| 13:33.25 | lirakis | yeah i just saw that.. |
| 13:33.26 | lirakis | crap |
| 13:33.31 | lirakis | yeah i know how to disable it |
| 13:33.36 | lirakis | .. i just need to recompile asterisk |
| 13:35.04 | *** join/#asterisk putnopvut (i=putnopvu@nat/digium/x-5a66641bfd138d37) |
| 13:39.30 | ai-a | any fax audio experts here ? we have fax over asterisk on local lan, and its failing.. recorded the audio... well, how am i suppost to know what failed from that :) |
| 13:40.09 | [TK]D-Fender | ai-a: Fax over VoIP.... lol |
| 13:40.12 | *** join/#asterisk Winkie (i=sd@87-194-8-125.bethere.co.uk) |
| 13:40.35 | Winkie | hey guys, we're trying to set the callerid with SIP but it only seems to like being set to what's in the sip.conf file, we can't set it to anything else! |
| 13:40.40 | Winkie | is this a common symptom or is it just me? |
| 13:40.56 | [TK]D-Fender | Winkie: Show us what you're doing and we'll see... |
| 13:41.03 | [TK]D-Fender | Winkie: PASTEBIN is your friend... |
| 13:41.05 | [TK]D-Fender | ~pb |
| 13:41.06 | jbot | A Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 13:41.14 | Winkie | [TK]D-Fender: one second, i'll get my workmate to do it |
| 13:41.53 | *** join/#asterisk jmesquita (n=jmesquit@c915230f.virtua.com.br) |
| 13:42.11 | tzafrir | ai-a, faxing over IP? good luck |
| 13:42.15 | ai-a | [TK]D-Fender: i know... we have 6 faxes, 1 modem and 1 franking machine on our voip and they are moaning they are not working good enough... we've plugged the modem / franking onto a analogue line for now. however its not cost effective to now buy in 8 analogue lines ontop of a isdn line. |
| 13:42.49 | JT | ai-a: yeah, don't do them over voip. |
| 13:42.55 | ai-a | tzafrir: the company (IP Cortex) that supplied the system install a Softfax at the same time, and say they shouldnt have a problem... We have switches thoughout the buildings. |
| 13:43.07 | JT | they are mistaken |
| 13:43.16 | ai-a | So we have found. |
| 13:43.32 | Winkie | the system i'm planning on using by the way is iaxmodem w/hylafax over a pri |
| 13:43.37 | tzafrir | ai-a, for starters make sure you use g711 |
| 13:44.15 | ai-a | Winkie: all points to VoIP is for VOICE not DATA. |
| 13:44.25 | JT | V is for Voice |
| 13:44.28 | Winkie | indeed |
| 13:44.47 | Winkie | we're simply going to use alaw (europe) and a seperate box via a crossover i think |
| 13:44.48 | ai-a | tzafrir: i turned off all fax.38 / echo cancellation/ jitter,, turned gains up / down.. and so on.. nothing makes them really work. |
| 13:45.13 | ai-a | Winkie: eh,,, we use ATA with aLaw.. it doesnt work. |
| 13:45.17 | Winkie | [TK]D-Fender: http://www.pastebin.ca/703219 |
| 13:45.30 | lirakis | Asterisk as just started giving me all these crazy errors... like "cannot find extension context 'internal' " ... internal is definitely there |
| 13:45.48 | [TK]D-Fender | lirakis: You know what to do... |
| 13:46.16 | lirakis | one moment |
| 13:46.40 | solar_ant | hi all |
| 13:46.47 | [TK]D-Fender | Winkie: you have to consider that maybe BUFFALO doesn't want you messing with your callerID and *IT* its overriding whatever * sends... |
| 13:47.00 | solar_ant | how can i connect huge number of pstn lines to 1 machine ? |
| 13:47.02 | Winkie | [TK]D-Fender: that's a fair point, one moment |
| 13:47.11 | solar_ant | huge number as in 50+ |
| 13:47.13 | Winkie | solar_ant: get an ISDN PRI |
| 13:47.21 | JT | solar_ant: couple of PRIs |
| 13:47.22 | solar_ant | Winkie: thanks |
| 13:47.23 | [TK]D-Fender | solar_ant: 2 in fact. |
| 13:47.24 | Winkie | unless you mean a huge number of strictly non isdn lines |
| 13:47.29 | lirakis | here is a pastebin from starting asterisk with 'asterisk -c' |
| 13:47.32 | lirakis | http://pastebin.com/d557b591c |
| 13:47.35 | Winkie | solar_ant: in that case get a couple of FXO banks |
| 13:47.47 | JT | in that case kill yourself ;) |
| 13:47.53 | [TK]D-Fender | lirakis: And the rest? |
| 13:48.03 | Winkie | agreeing with JT here |
| 13:48.15 | [TK]D-Fender | Screw Channel banks, jsut get 2x SIP gateways. |
| 13:48.19 | lirakis | ah .. i think i see some thing "ig.c:501 process_text_line: parse error: No category context for line 1 of /etc/asterisk/extensions.conf" |
| 13:48.21 | Winkie | [TK]D-Fender: buffalo is our other asterisk server but i'm getting him to check it ou |
| 13:48.26 | Winkie | huh they do FXO sip gateways? |
| 13:48.33 | JT | sure they do |
| 13:48.35 | [TK]D-Fender | Winkie: All kinds. |
| 13:49.10 | Winkie | well you learn something new every day |
| 13:49.15 | Winkie | i knew of 1 port ones like the IAXy |
| 13:49.18 | Winkie | incidentally jesus IAX sucks! |
| 13:49.38 | ManxPower | Winkie: IAX is cool, the IAXy sucks. |
| 13:49.42 | lirakis | some how my [general] line got deleted |
| 13:49.43 | Winkie | ManxPower: nah IAX sucks also |
| 13:49.48 | JT | iax is a toy |
| 13:49.52 | Winkie | it really is |
| 13:49.53 | ManxPower | Winkie: why do you say that? |
| 13:50.05 | Winkie | ManxPower: because we've been debugging for 2 weeks bizarre audio bugs which occur with IAX |
| 13:50.13 | Winkie | jitterbuffer resets / out of order frames etc |
| 13:50.26 | Winkie | we switched to SIP yesterday and except for this callerid issue it's just fine |
| 13:50.28 | ManxPower | Winkie: there were major IAX fixes in 1.4.x recently |
| 13:50.38 | Winkie | ManxPower: we're running SVN on one side, 1.4.11 on the other |
| 13:50.43 | lirakis | phew.... |
| 13:50.47 | lirakis | thank god.. |
| 13:50.48 | ManxPower | Winkie: we had only a few issues with IAX, we moved away from it for several reasons. |
| 13:50.56 | JT | iax is only good for edge cases |
| 13:51.07 | ManxPower | Winkie: I think you would need 1.4.x SVN on both sides. |
| 13:51.29 | Winkie | ManxPower: hehe, that crashes on the other side because of zaptel (not sure why, didn't debug) |
| 13:51.33 | ManxPower | In any case, we use SIP w/reinvites. |
| 13:51.43 | Winkie | yeah that's exactly what we're moving to |
| 13:51.47 | Winkie | just having an issue with callerid |
| 13:51.49 | ManxPower | can't do IAX between servers and have the SIP phones reinvite. |
| 13:51.54 | Winkie | we can do about 110 calls in 8mbit |
| 13:52.01 | Winkie | which isn't bad with g726 |
| 13:52.18 | Winkie | iax should be able to bring that down nearly 2mbit |
| 13:52.30 | [TK]D-Fender | Winkie: Far more... |
| 13:52.41 | Winkie | [TK]D-Fender: well ideally yes down to 3mbit |
| 13:52.48 | Winkie | uh, 3.6 |
| 13:52.48 | _x86_ | ewwww... g726? |
| 13:53.01 | ManxPower | _x86_: G726 is a perfectly fine codec. |
| 13:53.02 | JT | iax would change it from 8Mbit/s to 2Mbit/s, what? |
| 13:53.03 | Winkie | we switched back to g711, just tested a 32k codec :) |
| 13:53.09 | JT | G.726 is fine |
| 13:53.15 | ManxPower | great audio quality, less bandwidth than ulaw, no patent issues. |
| 13:53.17 | Winkie | JT: not 2mbit, an E1 is only 30 64k channels |
| 13:53.24 | _x86_ | i like GSM |
| 13:53.32 | JT | Winkie: i know what an E1 is |
| 13:53.33 | _x86_ | or speex |
| 13:53.38 | _x86_ | speex is great |
| 13:53.42 | Winkie | JT: but then how would it get to 2mbit? |
| 13:53.54 | JT | g.726 has superior audio to gsm |
| 13:53.58 | JT | Winkie: what's the question? |
| 13:54.18 | Winkie | JT: ah you misread what i said :) |
| 13:54.27 | Winkie | i said bring it down by about 2mbit or more |
| 13:54.30 | Winkie | not to :) |
| 13:54.55 | [TK]D-Fender | Winkie: Except... just imagine how fragile that trunk would be though :) |
| 13:55.00 | Winkie | also [TK]D-Fender there's nothing at the buffalo side screwing with things that i can see: http://www.pastebin.ca/703227 |
| 13:55.11 | JT | anyway, iax trunking is unreliable at any real volume |
| 13:55.12 | Winkie | [TK]D-Fender: we grouped it into 4 trunks automatically and still had audio issues :( |
| 13:55.17 | Winkie | yeah agreed entirely |
| 13:55.27 | JT | iax is a joke really for real purposes |
| 13:55.43 | Winkie | that's such a shame |
| 13:55.47 | [TK]D-Fender | Winkie: show BOTH sides sperately please. SIP on EACH |
| 13:57.14 | Winkie | [TK]D-Fender: they're identical, that is the remote side i think |
| 13:57.28 | *** join/#asterisk jk|47 (n=chatzill@205.143.79.134) |
| 13:57.40 | jk|47 | hey ppl |
| 13:57.42 | *** join/#asterisk anonymouz666 (n=anonymou@189.25.37.205) |
| 13:57.45 | [TK]D-Fender | Winkie: "Think" doesn't give me any warm & fuzzy feeling.... go check both ends. |
| 13:57.46 | Winkie | always ends up being 'Unknown': Executing [11158588@incoming-isher:5] NoOp("SIP/buffalo-082d2978", "Num=Unknown| Name=Unknown| ANI=Unknown| DNID=11158588| RDNIS=") in new |
| 13:57.51 | Winkie | [TK]D-Fender: haha fair enough |
| 13:58.08 | jk|47 | can anyone tell me does asterisk support ccxml and is it the engine for ccxml or ? |
| 13:58.20 | Winkie | ok yes it's exactly the same, no mention of callerid anywhere |
| 13:58.54 | [TK]D-Fender | Winkie: go pastebin up a call with SIP debug enabled, and while you're at it tell me what you DO see exactly on the other end. |
| 13:59.03 | [TK]D-Fender | Winkie: for CID whent he call is received. |
| 13:59.20 | Winkie | [TK]D-Fender: see above, it comes through as Unknown, debugging more now |
| 14:01.40 | [TK]D-Fender | Winkie: Try this : exten => _11158588.,n,Dial(SIP/buffalo/${EXTEN:0:8}) |
| 14:03.19 | Winkie | instead of @? |
| 14:03.37 | Winkie | also i have a rather large sip debug for both sides, want me to pb it all? |
| 14:04.35 | [TK]D-Fender | Winkie: Change as I suggested and test. PB the failure if any. |
| 14:04.43 | [TK]D-Fender | Winkie: And yeah, the whole mess. |
| 14:05.34 | Winkie | [TK]D-Fender: changed, no success |
| 14:05.44 | [TK]D-Fender | Winkie: ok, PB it up. |
| 14:05.47 | Winkie | you'll have to give me a bit to pastebin this, i'm working through two sshs + screens :) |
| 14:08.59 | Winkie | [TK]D-Fender: ok this is local to remote, let me know if you want me to add the remote one too: http://pastebin.ca/703251 |
| 14:11.46 | [TK]D-Fender | Winkie: Looks like what I'd guess was a PRI. Check your calling presentation. It may have a "blocked" flag thats carrying over. Also NoOp the CID values BEFORE the "Set" and again before placing the dial |
| 14:11.51 | *** part/#asterisk deeperror (n=deeperro@69-215-202-195.ded.ameritech.net) |
| 14:12.30 | Winkie | [TK]D-Fender: it is a PRI, do you mind telling me how i can check calling presentation? I'll get the NoOps done now |
| 14:12.42 | [TK]D-Fender | Winkie: In your zapata.conf |
| 14:12.48 | Winkie | ah |
| 14:13.16 | [TK]D-Fender | Winkie: I believe there's a way to change that within your dialplan, but I don't recall exactly where. |
| 14:14.26 | Winkie | [TK]D-Fender: the only thing i can find in the examples is usecallingpres=yes which is not set at all in our production config, the description for this is somewhat vauge though |
| 14:14.39 | Winkie | also sorry if i'm afk for a few seconds, keeping my eyes on some new kittens |
| 14:15.05 | JT | README.variables |
| 14:15.14 | [TK]D-Fender | Winkie: Check the new NoOp's, and WIKI up the callingpres info. I'm not an expert in this, but I think its a very probably lead for you. |
| 14:16.09 | [TK]D-Fender | probable* |
| 14:16.11 | Winkie | [TK]D-Fender: the NoOps are being done for me now, i'll check out the callingpres stuff now |
| 14:16.15 | Winkie | yeah it looks pretty interesting |
| 14:16.49 | *** join/#asterisk ELBunce (n=erik@kde/developer/bunce) |
| 14:18.58 | Winkie | [TK]D-Fender: NoOps were no use, it is certainly set before the dial |
| 14:19.22 | [TK]D-Fender | Winkie: Ok, onto "Plan B" then.... |
| 14:19.26 | Winkie | indeed |
| 14:19.38 | [TK]D-Fender | Winkie: Wish you luck... |
| 14:20.36 | *** join/#asterisk NOT_guru (n=chatzill@24-241-103-142.static.stls.mo.charter.com) |
| 14:22.13 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 14:23.23 | Winkie | [TK]D-Fender: and no luck so far :( |
| 14:24.04 | [TK]D-Fender | Winkie: tip : enable PRI debug and see exactly whats coming in. |
| 14:24.29 | Winkie | [TK]D-Fender: i'm not sure it's a zaptel issue, unless there's some major difference in handling between IAX and SIP, CID worked fine with IAX |
| 14:24.37 | Winkie | i'm still checking out callingpres now in more detail |
| 14:24.43 | *** join/#asterisk funxion (n=nunya@63.214.236.169) |
| 14:24.47 | ManxPower | Winkie: what IS your CID issue? |
| 14:25.00 | Winkie | ManxPower: we used to have an IAX trunk between two machines, callerid worked fine |
| 14:25.05 | [TK]D-Fender | Winkie: Just to see if a block flag si being carried over, check the PRI call as it comes in. |
| 14:25.11 | Winkie | we switched to quick SIP friends at both sides, callerid comes across as 'Unknown' |
| 14:27.36 | _ShrikE | Ive seen you guys talk about this before. With Cisco PIX, is it sip fixup you recommend turning off? |
| 14:28.20 | *** join/#asterisk krdian_ (i=krdian@killer.radom.net) |
| 14:29.23 | *** join/#asterisk saftsack (n=saftsack@pD9E0445C.dip.t-dialin.net) |
| 14:29.28 | Winkie | [TK]D-Fender: i would like to present you with a gold star |
| 14:29.33 | Winkie | your initial suspicions were correct |
| 14:30.10 | krdian_ | hello |
| 14:30.16 | Winkie | hi |
| 14:30.31 | tzanger | Winkie: [TK]D-Fender's assumptions are generally bang-on |
| 14:31.34 | *** join/#asterisk javar (n=javar@69.79.134.24) |
| 14:31.35 | Winkie | indeed it would seem so! |
| 14:32.56 | defswork | Are there any simple/small windows utils to monitor lines and display inobstrusively on screen ? |
| 14:33.50 | lirakis | defswork: .. asterisk cli |
| 14:33.52 | lirakis | ha ha |
| 14:34.32 | lirakis | defswork: the only purpose built app i know of is hud / hudlite |
| 14:34.33 | defswork | thanks for that |
| 14:34.57 | defswork | I've struggled to get hud working on some systems |
| 14:35.00 | lirakis | defswork: .. but it does more than monitor a line.. and it cost $$ |
| 14:35.12 | lirakis | defswork: asterisk flash operator panel |
| 14:35.39 | defswork | fop is a little bug |
| 14:35.42 | defswork | big* |
| 14:36.10 | ManxPower | _ShrikE: sip fixup should be called "break sip stuff you stupid cisco" |
| 14:36.31 | *** join/#asterisk mindCrime (n=chatzill@66.83.208.218.nw.nuvox.net) |
| 14:36.45 | _ShrikE | ManxPower: haha.. Thanks. |
| 14:37.47 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 14:40.28 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 14:42.02 | *** join/#asterisk Dan0maN_Work (n=dschuh@64.149.174.136) |
| 14:43.25 | *** part/#asterisk jk|47 (n=chatzill@205.143.79.134) |
| 14:44.59 | *** join/#asterisk zeppelin_ (n=zeppelin@201.41.0.244) |
| 14:49.25 | lirakis | i have a few sip peers that show up as "unmonitored" in sip show peers |
| 14:49.32 | lirakis | how can i change them to be monitored |
| 14:49.35 | Winkie | do they have qualify statements? |
| 14:49.40 | Winkie | (in their peer definition in sip.conf) |
| 14:49.46 | lirakis | hmm.. no.. |
| 14:49.46 | *** join/#asterisk etfonhomey (n=chatzill@12.169.248.226) |
| 14:49.49 | Winkie | then add them :) |
| 14:50.15 | lirakis | Winkie: okay... i am actually using FOP .. and i want to see calls coming in on them.. i am not sure if doing this will enable that.. |
| 14:50.24 | lirakis | Winkie: i guess i will find out |
| 14:50.26 | Winkie | FOP? |
| 14:50.34 | lirakis | Winkie: flash operator panel |
| 14:50.36 | Winkie | are you looking to see state changes? |
| 14:50.57 | *** join/#asterisk saftsack (n=saftsack@p54A76F88.dip.t-dialin.net) |
| 14:51.31 | lirakis | Winkie: yes |
| 14:52.50 | lirakis | Winkie: right now.. the sip peers ( for my providors ) .. do not show any state change.. just .. that they are there.. .. i would like to see the call information on them.. but .. nothing.. |
| 14:52.58 | *** join/#asterisk saftsack (n=saftsack@p54A76F88.dip.t-dialin.net) |
| 14:53.01 | lirakis | Winkie: even with qualify=yes and a reload |
| 14:53.29 | Winkie | lirakis: there is a way to do this but I can't for the life of me remember how |
| 14:53.38 | Winkie | i'm not sure if you still can, so i'd wait and ask someone more experienced :) |
| 14:53.54 | *** join/#asterisk tripps (n=ss@66.60.235.100) |
| 14:55.10 | *** join/#asterisk elixer (i=elixer@65.207.74.18) |
| 14:56.18 | *** join/#asterisk Aurs (n=Aurs@1ult2p8.ip.hipercom.no) |
| 14:56.23 | tripps | can anyone here recommend a) a sip to pstn gateway appliance, and b) pri pci card for use with *? |
| 14:57.55 | _ShrikE | lirakis: Do you have manager.conf set properly for FOP? |
| 14:57.57 | [TK]D-Fender | tripps: Sangoma A101d |
| 14:58.00 | *** join/#asterisk CunningPike (n=arodgers@204.239.12.183) |
| 14:58.36 | [TK]D-Fender | lirakis: PB it up... |
| 14:58.52 | [TK]D-Fender | Winkie: Were you able to override the presentation restriction ofrom your inbound channel? |
| 15:00.05 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 15:00.17 | tripps | [TK]D-Fender: is that the pri card? |
| 15:00.24 | [TK]D-Fender | tripps: Yes |
| 15:00.34 | tripps | good support in * I presume |
| 15:00.36 | [TK]D-Fender | tripps: 1-port with Hardware Echo Cancellation. |
| 15:00.46 | [TK]D-Fender | tripps: Works very well |
| 15:01.35 | Winkie | [TK]D-Fender: indeed we were, it worked just fine thank you |
| 15:02.17 | tripps | [TK]D-Fender: excellent thanks! do you have any recommendations for a sip to pstn gateway? |
| 15:02.23 | *** join/#asterisk ming_zym (n=ming_zym@124.254.54.4) |
| 15:03.25 | [TK]D-Fender | tripps: SIP/PRI gateways are very expensive per-port and you lose some control when you leave it external. I usually only recommend those when you're looking at a higher density highly redundent scenario |
| 15:03.34 | [TK]D-Fender | Winkie: Quite welcome |
| 15:06.30 | billybongo | anyone using a mysql cluster: how many boxen do you use? |
| 15:06.32 | [TK]D-Fender | etfonhomey: ping |
| 15:06.43 | billybongo | and why |
| 15:07.17 | tripps | [TK]D-Fender: roger that. thx again! |
| 15:07.19 | etfonhomey | What's up |
| 15:07.41 | etfonhomey | [TK]D-Fender Getting ready to head to a meeting. |
| 15:08.09 | tripps | [TK]D-Fender: assuming of course that I am looking at such a scenario and understand the costs, what would be your recommendations or perhaps a site to direct me to? |
| 15:08.42 | [TK]D-Fender | tripps: AudioCodes Mediant 1000 , Patton (ask others for their opinion on these). |
| 15:14.45 | tripps | [TK]D-Fender: very cool. thx. |
| 15:15.09 | *** join/#asterisk ShadowHntr (i=sentinel@wikipedia/Shadowhntr) |
| 15:15.28 | *** join/#asterisk Tili (n=tili@cm48.gamma244.maxonline.com.sg) |
| 15:17.29 | tripps | [TK]D-Fender: looks like the mediant 1000 allows you to actually install * on the box? |
| 15:17.57 | [TK]D-Fender | tripps: Not that I'm aware. |
| 15:18.15 | [TK]D-Fender | tripps: And I really wouldn't want to TRY even if I could... |
| 15:18.53 | tripps | gotcha |
| 15:26.15 | tripps | [TK]D-Fender: how in your opionion would * function as a SIP gateway? |
| 15:26.52 | [TK]D-Fender | tripps: I'd step back and look at that greater picture of what you want to do before answering something like that... |
| 15:28.01 | billybongo | so anyone using mysql with an asterisk cluster? |
| 15:28.11 | *** join/#asterisk techie (n=techie@c-67-181-184-170.hsd1.ca.comcast.net) |
| 15:28.18 | billybongo | with or without openser |
| 15:28.21 | billybongo | <PROTECTED> |
| 15:28.22 | tripps | [TK]D-Fender: how do you mean exactly? |
| 15:28.34 | billybongo | or should I be looking to postgresql? |
| 15:28.35 | [TK]D-Fender | tripps: What exactly are you looking to do? |
| 15:34.27 | *** join/#asterisk NightMonkey (n=NightMon@pdpc/supporter/sustaining/NightMonkey) |
| 15:35.47 | *** join/#asterisk saftsack (n=saftsack@pD9E0445C.dip.t-dialin.net) |
| 15:36.15 | *** join/#asterisk ivanfm (n=ivanfm@c906b486.virtua.com.br) |
| 15:38.35 | *** join/#asterisk luisjose (n=ljd@nelug/coreteam/luisjose) |
| 15:39.04 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 15:39.24 | Siya | [TK]D-Fender: outbound doesn't seem to be the problem |
| 15:39.31 | Siya | inbound is the issue |
| 15:39.56 | Siya | asterisk doesn't show anything inbound happening so was curious whether I really need the SRV dns record etc |
| 15:41.50 | [TK]D-Fender | Siya: So you want to RECEIVE un-authed calls to your * box? |
| 15:45.46 | *** join/#asterisk thansen|laptop (n=thansen@19.243.sfcn.org) |
| 15:48.14 | *** join/#asterisk mmreko (n=mrekowsk@pixout.appriss.com) |
| 15:48.30 | *** join/#asterisk |omni| (n=rob@c-67-185-56-106.hsd1.wa.comcast.net) |
| 15:50.18 | seldon75 | hi, I had a problem where all I get when I make an O/G call is loud white noise; so Corydon76 recommended I reload the wctdm module with nativebridge=0. Problem is, when I try to rmmod wctdm24xxp, it tells me "this module is in use" |
| 15:50.59 | Strom_M | seldon75: you have to stop asterisk first |
| 15:51.03 | seldon75 | ok |
| 15:51.05 | seldon75 | thought so |
| 15:51.23 | seldon75 | is it just "stop asterisk" and then afterwards "start asterisk" ? |
| 15:51.37 | Strom_M | at the console, "stop now" |
| 15:51.45 | seldon75 | ok |
| 15:51.49 | seldon75 | then to start..? |
| 15:51.51 | Strom_M | then later, start it however you started it last time |
| 15:51.58 | seldon75 | by rebooting ;) |
| 15:52.11 | Strom_M | /etc/init.d/asterisk start perhaps |
| 15:52.59 | seldon75 | thanks |
| 15:53.15 | *** join/#asterisk critch (n=critch@c-71-228-211-57.hsd1.tn.comcast.net) |
| 15:53.16 | *** join/#asterisk Defraz (n=t0tal@24-116-152-177.cpe.cableone.net) |
| 15:53.43 | critch | anyone experienced with colo-ing asterisk boxes in telco locations? need wiring answer |
| 15:54.01 | Strom_M | what kind of wiring answer |
| 15:54.05 | critch | Should I need a crossover cable from their smart jack to my asterisk box? |
| 15:54.18 | Strom_M | critch: probably not |
| 15:54.24 | tripps | [TK]D-Fender: well the thought has been to mitigate call quality issues when using a sip provider over the internet. in our case, many of our customers we've put on with a particular ISP have experienced poor call quality (lots of jitter, etc.). Any recommendations for a national ISP that provides QoS or otherwise has a SIP-friendly network? Perhaps someone on Level3 so we can stay on net (layer 2 perhaps) all the way to sip provider for pstn handoff? |
| 15:54.31 | Strom_M | use a straight-through cable |
| 15:54.48 | Strom_M | if that doesn't work, make sure you use a T1 crossover cable and not an Ethernet crossover cable |
| 15:55.02 | [TK]D-Fender | tripps: QoS over the internet.... LOL |
| 15:55.54 | critch | That is my opinion as well. They are telling me I do, but then I only get yellow alarm. If I use a straight through and they flip on their end we are good |
| 15:56.05 | *** join/#asterisk fujin_ (n=aj@unaffiliated/fujin) |
| 15:56.16 | Strom_M | critch: are you sure you're using a T1-specific crossover cable? |
| 15:56.35 | critch | Strom_M: yes, one I crimped and one they just made as well |
| 15:56.51 | critch | Smart jack is seeing green and we are seeing yellow |
| 15:57.03 | Strom_M | that sounds like you made an ethernet crossover cable |
| 15:57.13 | Strom_M | T1 crossover cable swaps pairs 1 and 3 |
| 15:57.18 | Strom_M | ethernet swaps pairs 2 and 3 |
| 15:57.26 | critch | correct, orange and blue swap |
| 15:57.34 | critch | plus their stripes |
| 15:57.48 | Strom_M | assuming you're using TIA-568-A and not TIA-568-B pinout |
| 15:58.17 | *** join/#asterisk hfb (n=hfb@pool-71-118-252-254.lsanca.dsl-w.verizon.net) |
| 15:58.18 | *** join/#asterisk CVirus (n=GoD@196.205.192.90) |
| 15:58.21 | critch | okay, either way. as you said pairs 1 and 3 |
| 15:58.37 | Strom_M | no, not "either way" |
| 15:58.43 | Strom_M | that pinout is crucial also |
| 15:58.44 | *** join/#asterisk pigpen2 (n=pigpen@207.71.48.222) |
| 15:59.04 | critch | yes, I know. I was backing away from nitpicking just because we are in agreement |
| 16:00.11 | pigpen2 | Hi all. I updated to polycom sip 2.1.1, now when I have stupid users take the receiver off hook and dial, for example *98, it just automatically sends the "*". |
| 16:00.13 | Strom_M | ok, i just want to be sure you're clear on that - most people I talk to are "computer people" and don't know their TIA-568-A and their 25-pair color code from a hole in the ground |
| 16:00.28 | pigpen2 | I cannot seem to find the new magical way to stop this from happening. ideas? |
| 16:00.44 | *** join/#asterisk MindTheGap (n=iote@200.251.73.5) |
| 16:00.47 | *** join/#asterisk Kigh (n=kai@ciphron.de) |
| 16:00.52 | critch | Strom_M: well, I don't know the 25 pair color code either, but I don't get to touch it either |
| 16:01.12 | tripps | [TK]D-Fender: not exactly over the net - that's why i'm looking for an on net provider we can deploy at client locations where we install * appliances where they can have QoS all the way to the SIP provider by keeping traffic on-net (perhaps layer 2) |
| 16:01.14 | Strom_M | yeah, but you do know at least the first five pairs of it, right? :) |
| 16:01.36 | [TK]D-Fender | tripps: So where does PRi fit into this? |
| 16:04.30 | Strom_M | critch: i'd be sure you're making a T1 crossover cable before exploring other options - make sure that when you identify pairs 1 and 3, those are on pins 4-5 and 1-2 of the 8P8C plug, respectively |
| 16:04.55 | critch | They just rewired their side. |
| 16:05.03 | critch | Now a straight through is happy |
| 16:05.16 | Strom_M | TIA-568-A pinout should look like GR GR/W OR BL/W BL OR/W BR BR/W |
| 16:05.19 | Strom_M | ah ok |
| 16:05.20 | outtolunc | that happens alot <G> |
| 16:05.59 | critch | thanks for the sanity check Strom_M |
| 16:06.09 | Strom_M | any time |
| 16:09.13 | tripps | for lack of good provider that fits my description and putting out fires at existing installs :) |
| 16:09.22 | tripps | [TK]D-Fender: last msg for you |
| 16:10.06 | *** join/#asterisk duckz (n=duckz@81.180.83.75) |
| 16:11.04 | *** join/#asterisk NOT_guru (n=chatzill@24-241-103-142.static.stls.mo.charter.com) |
| 16:12.57 | *** join/#asterisk theHub (n=theHub@69.177.93.21) |
| 16:18.07 | *** part/#asterisk pigpen2 (n=pigpen@207.71.48.222) |
| 16:19.33 | [TK]D-Fender | tripps: Unless you're setting up a redundant system with SER/etc I wouldn't bother with a SIP/PRI gateway |
| 16:21.54 | *** join/#asterisk jmls (n=jmls@62.49.235.130) |
| 16:22.01 | jmls | hey guys'ngals |
| 16:22.06 | jmls | 'n' |
| 16:22.20 | jmls | is there an opposite to the GROUP command ? |
| 16:22.24 | jmls | (function) |
| 16:23.02 | [TK]D-Fender | UNGROUP! |
| 16:23.19 | creativx | hehe |
| 16:23.25 | jmls | hadeha |
| 16:23.26 | jmls | ;) |
| 16:24.52 | jmls | I have a queue that I want to keep a count of the number of calls, so I add the channel to the XXX group |
| 16:25.04 | jmls | trouble is, that group overflows into the YYY group. |
| 16:25.28 | jmls | more trouble is, the YYY group is a standalone group of it's own, so I need to maintain a group count on that as well |
| 16:26.01 | jmls | but if XXX goes into YYY, I still have a count on the XXX group as well as the YYY group |
| 16:28.14 | [TK]D-Fender | jmls: Please restart your request... things are getting a bit blurry. What are you looking to count exactly, and do what based on this? |
| 16:28.24 | *** join/#asterisk SwK_ (n=SwK@user-69-73-37-99.knology.net) |
| 16:30.40 | *** join/#asterisk kink0 (n=k@161.pool62-37-205.static.orange.es) |
| 16:30.42 | kink0 | hello |
| 16:31.37 | jmls | [TK]D-Fender: I want to maintain a count of all active calls in each of the queues that I have, either answered or waiting |
| 16:31.51 | kink0 | any idea about this error : !! Unknown IE 124 (cs5, Unknown Information Element) |
| 16:31.59 | jmls | so I use GROUP()=<QueueName> |
| 16:32.26 | jmls | however, queue XXX overflows to YYY if they are not answered within a period of time |
| 16:32.40 | *** join/#asterisk Delvar (n=Delvar@77.240.56.18) |
| 16:32.44 | jmls | so, a call is in group XXX and YYY. |
| 16:32.47 | [TK]D-Fender | jmls: Parse out "show queues" |
| 16:33.11 | jmls | yeah, that's what I was trying to avoid. Messy |
| 16:33.33 | jmls | I don't understand why you can't "UNGROUP" for want of a better word |
| 16:35.02 | [TK]D-Fender | jmls: then use Chan_local to call your agents and place the group count in there. |
| 16:36.10 | jmls | I do use chan_local - but the count I want is the total number of calls waiting and talking in a queue. So I group before it goes into the queue_app |
| 16:36.26 | jmls | and not when I go to call the agent |
| 16:36.40 | [TK]D-Fender | jmls: If you want "waiting", then you're going to have to aprse. |
| 16:36.48 | [TK]D-Fender | parse* |
| 16:37.12 | jmls | no, all I need is the _total_ number of waiting and talking |
| 16:37.21 | jmls | not separate totals |
| 16:37.40 | jmls | it all works great except when a waiting call overflows |
| 16:38.01 | jmls | and therefore need to remove that channel from the group |
| 16:39.18 | *** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net) |
| 16:40.10 | thewiizle | i am |
| 16:40.13 | thewiizle | the fuckingdaddy.com |
| 16:40.39 | elixer | this is a family channel |
| 16:41.05 | thewiizle | sorry |
| 16:41.15 | thewiizle | the flippingdaddy.com |
| 16:41.27 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 16:41.56 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 16:44.07 | *** join/#asterisk candyban (n=nope@uu212-190-206-98.unknown.uunet.be) |
| 16:44.37 | *** join/#asterisk iPod-nano (n=root@c-68-43-60-193.hsd1.mi.comcast.net) |
| 16:44.48 | iPod-nano | Console-based IRC is so cool! |
| 16:45.17 | candyban | Hi guys ... is it possible to use CISCO phones with asterisk? 7912 and 7940 series |
| 16:45.43 | [TK]D-Fender | candyban: Yes, but they are not recommended |
| 16:45.54 | iPod-nano | From what I've heard, they've been able to reverse-engineer Cisco's protocol. |
| 16:45.55 | candyban | [TK]D-Fender: why? |
| 16:46.11 | *** join/#asterisk FCOJ (n=mordur@85-220-103-55.dsl.dynamic.simnet.is) |
| 16:46.12 | [TK]D-Fender | candyban: Licensed firmware, poor SIP implementation, higher cost. |
| 16:47.26 | iPod-nano | I didn't install a GUI on my Asterisk box, so I'm running console-based IRC and AIM clients. :-P |
| 16:47.31 | *** join/#asterisk Connor (i=Connor@198-144-174-5.knx.tn.nxs.net) |
| 16:47.40 | candyban | [TK]D-Fender: CISCO's appear to be the cheapest you can find (on ebay at least) ... I'd like to start experimenting so I would like 'cheap' stuff |
| 16:48.18 | *** join/#asterisk maraq (n=none@dsl-204.maraq.net) |
| 16:49.08 | [TK]D-Fender | candyban: Avoid the 7912. The 7940 is more workable. How much can you see them for? |
| 16:49.36 | candyban | [TK]D-Fender: between 30 and 69 euros |
| 16:49.55 | candyban | [TK]D-Fender: that's for the 7940 series (without power adapter though) |
| 16:50.25 | candyban | [TK]D-Fender: and that's quickly skimming (69 euros is with the "buy now" option) |
| 16:50.26 | [TK]D-Fender | candyban: Keep in mind it had better come with a Cisco PoE injector because those 2 models don't support 802.3af and don't have a standard wall-wart |
| 16:50.42 | [TK]D-Fender | candyban: Go for the 7940 only between those 2. |
| 16:51.04 | maraq | Hi, i have a Newbie question: i'm using a sipura2100 at the moment supplied by my phone-supplier here in the netherlands, is my assumption correct that i can tie in asterisk to pickup the line, then use the sipura ( or any other program ) to connect to asterisk ? |
| 16:51.17 | candyban | [TK]D-Fender: hmmz ... so a regular PoE switch won't cut it? |
| 16:51.41 | [TK]D-Fender | candyban: Nope, those are pre 802.3af standard running Cisco's proprietary standard |
| 16:52.19 | *** join/#asterisk codazoda (n=chatzill@70-96-185-203.directbb.com) |
| 16:53.13 | candyban | [TK]D-Fender: would a cisco PoE switch work or do they require the injectors? |
| 16:53.39 | codazoda | I've got a TDM404B. It seems to be working fine in that I've got all 4 lights on the back and 3 of the 4 ports answer. But, the first port always rings busy. A few minutes before I got this all working, I had static discharge into that port from the end of a phone cord I touched with my hand. Does a busy line likely indicate I fried the port? |
| 16:53.44 | [TK]D-Fender | candyban: I would presume that a Cisco PoE Switch would support it. Check the manuals to be sure |
| 16:53.56 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 16:54.11 | candyban | [TK]D-Fender: thanks a lot :) ... |
| 16:54.57 | russellb | codazoda: please contact support@digium.com |
| 16:55.17 | [TK]D-Fender | maraq: If you can unlock the SPA from your service provider and they don't try to block * for connecting to them sure. |
| 16:55.37 | codazoda | 'zap show channles' shows them all as 'inbound' in the 'default' context. |
| 16:55.47 | codazoda | russellb, you think they can help? |
| 16:55.53 | russellb | codazoda: of course |
| 16:56.03 | russellb | that is their job ... |
| 16:56.22 | codazoda | I doubt that shocking my card is covered under the warranty, if it turns out to be dead. :-) |
| 16:56.23 | candyban | [TK]D-Fender: which phones would you recommend? |
| 16:56.35 | [TK]D-Fender | candyban: ... |
| 16:56.37 | [TK]D-Fender | ~phones |
| 16:56.38 | jbot | rumour has it, phones is http://bani.anime.net/phones/. While personal preference will dictate which phone works best for you, general consensus on a rough order of quality and suggestibility is as follows: Polycom (any), Aastra 480i, Aastra 5i Series, Cisco 7940+, Linksys SPA-9XX, Snom, and finally everything else. Do not consider Grandstream phones. Ever. |
| 16:56.58 | bkruse | ~lart candyban |
| 16:56.58 | jbot | hooks into a hydrant and hoses candyban down |
| 16:57.10 | bkruse | sorry, I had to :D |
| 16:57.15 | bkruse | ~punch bkruse |
| 16:57.15 | jbot | ACTION hits bkruse like the hot kiss and the end of a wet fist |
| 16:57.22 | bkruse | LOL |
| 16:57.22 | russellb | codazoda: if you haven't already, try turning off the power to the box and powering it back up |
| 16:57.28 | maraq | [TK]D-Fender thanks :) i'm gonna try then ^^ |
| 16:57.31 | russellb | codazoda: you may have just got it in a bad state |
| 16:57.44 | codazoda | I did try to power off the box. |
| 16:57.46 | bkruse | jbot is nsfw |
| 16:57.56 | [TK]D-Fender | maraq: Try with just * to your ITSP and use a soft-phone first before screwing with your ATA. And don't get your hopes too high about unlocking it. |
| 16:58.29 | russellb | codazoda: k ... well i'm not sure about the warranty question. they can tell you, though. |
| 16:58.35 | [TK]D-Fender | bkruse: indeed, that was more than a little nasty... |
| 16:58.41 | candyban | [TK]D-Fender: we have polycoms at work and they work nice, but they are so expensive :( |
| 16:58.49 | [TK]D-Fender | bkruse: And news on MokoIAX? |
| 16:59.10 | [TK]D-Fender | candyban: Indeed their EU pricing isn't as nice as inNorth America.... Dunno why... |
| 16:59.11 | maraq | [TK]D-Fender: I'll have to start reading into what all this means first, ATA, ITSP.. all gibberish at the moment, but i'm excited at what this stuff can do :) |
| 16:59.25 | bkruse | [TK]D-Fender: digium work > mokoiax :[ |
| 16:59.29 | [TK]D-Fender | maraq: SPA-2100 = ATA, your provider = ITSP |
| 16:59.32 | bkruse | but I have started the gtk frontend |
| 16:59.41 | [TK]D-Fender | bkruse: Thats a given, and still not an answer :p |
| 16:59.45 | bkruse | still pretty active in the community, or trying to be |
| 16:59.46 | maraq | [TK]D-Fender: ah.. that was easy |
| 16:59.59 | bkruse | [TK]D-Fender: sean wants it to be released at the same time 02 is released |
| 17:00.24 | pots_line | . |
| 17:00.25 | [TK]D-Fender | bkruse: I was hoping more for the backend to integrate to the EXISTING dialer app. It'd be a petty plugin to choose the dialout source. |
| 17:00.39 | bkruse | [TK]D-Fender: thats whats going to happen |
| 17:00.48 | [TK]D-Fender | bkruse: EXCELLENT. |
| 17:00.51 | bkruse | when you configure it in the client, it will save in the mokodb and you can select it as a gateway |
| 17:01.08 | bkruse | gateways: gsm, cdma(but not), iax -> profile1, profile2 |
| 17:01.09 | bkruse | etc etc |
| 17:01.25 | [TK]D-Fender | bkruse: I've gone to a local 01 showing where the concensus seems to be that 02 wont be until mid Q1 08 with any realism attached. |
| 17:01.31 | bkruse | I have been reading up on the "proper" way to use the gtk wrappers they have (which are very minimal unfortunatly :/ ) |
| 17:01.44 | [TK]D-Fender | bkruse: What of CDMA? |
| 17:01.52 | bkruse | [TK]D-Fender: nothing of cdma |
| 17:01.56 | bkruse | cdma(but not) haha |
| 17:02.04 | bkruse | [TK]D-Fender: really? sean will be shipping me a couple for beta I hope |
| 17:02.16 | [TK]D-Fender | bkruse: Had my hopes up... my current provider is CDMA, and switching will be nasty on my pricing |
| 17:02.23 | bkruse | [TK]D-Fender: who is it? |
| 17:02.26 | [TK]D-Fender | bkruse: 02 beta? |
| 17:02.32 | [TK]D-Fender | bkruse: Bell Canada. |
| 17:02.34 | bkruse | [TK]D-Fender: im sure they do gsm also, pretty sure..... |
| 17:02.35 | bkruse | oh |
| 17:02.37 | bkruse | maybe not :] |
| 17:02.52 | bkruse | [TK]D-Fender: yes, when the 02s are actually running through before the "public" release of the phone |
| 17:03.01 | [TK]D-Fender | bkruse: Nope, Bell & Telus = CDMA, Rogers & Fido = GSM.. |
| 17:03.06 | bkruse | its going to be different than the 01 release, since 01 was purely for devs |
| 17:03.14 | bkruse | [TK]D-Fender: ya, I thought you meant in the states |
| 17:03.19 | [TK]D-Fender | bkruse: Yeah, bigger CPU, Wifi, etc |
| 17:03.33 | bkruse | right right, and the images have come a LONG way since release also |
| 17:03.41 | [TK]D-Fender | bkruse: I have to admit the screen is beautiful. The onscreen keyboard is a flaming pile of &@#^%@ however |
| 17:03.44 | bkruse | we will see, theres been talk of course, because they want to support everything... |
| 17:04.08 | bkruse | [TK]D-Fender: I know, someone did a python "proof of concept" multitouch keyboard, but did not realize the hardware just is not capable |
| 17:04.08 | [TK]D-Fender | bkruse: I say the Aug release firmware on the )!'s at that meeting. |
| 17:04.37 | bkruse | [TK]D-Fender: thats not bad, its still 01 firmware, its just...there are svn commits everyday |
| 17:04.47 | bkruse | you can download and even throw on the new image very easily |
| 17:04.55 | [TK]D-Fender | bkruse: Multitouch isn't as important as KILLING that wasted space between the keys and trying to cram EVERYTHING into 1 screen worth of it. I'm not opposed to paging for other keys |
| 17:04.56 | bkruse | prebuilt images, so you do not need the 2.2gig+ dev environment |
| 17:05.15 | bkruse | [TK]D-Fender: I believe they are, of course there has been talk for the hardware keyboard also |
| 17:05.30 | bkruse | like a slide out, but thats far out scope of 03 to be integrated, but possible clipon/addon |
| 17:05.47 | [TK]D-Fender | bkruse: there are plenty of BT mini keyboards out there... |
| 17:06.06 | [TK]D-Fender | bkruse: or usb if the split the mode on it |
| 17:06.20 | [TK]D-Fender | that'd be the most ccost effective. |
| 17:06.23 | bkruse | [TK]D-Fender: exactly |
| 17:06.34 | bkruse | well, I think someone was planning on actually making them, outside of FIC/moko |
| 17:06.36 | [TK]D-Fender | bkruse: Such promise..... |
| 17:06.38 | bkruse | to do a clickup to the USB |
| 17:06.59 | bkruse | would be neat, if I use it as my text messaging device, ill need it |
| 17:07.11 | bkruse | I want to do some experimenting with iax2 and text messaging :X |
| 17:07.37 | codazoda | Dropped in another TDM04B and that one answers on port one. So, I fried something on the card. I can hope that it's just the FXO. |
| 17:08.01 | [TK]D-Fender | codazoda: Should be an easy test |
| 17:09.09 | *** join/#asterisk [T]ank (n=ckwall@206.71.78.172) |
| 17:09.17 | [T]ank | i need some advice on something....... |
| 17:09.27 | Nugget | down, not across. |
| 17:09.33 | [T]ank | i am setting up some extensions that need to be recorded. |
| 17:09.48 | [T]ank | but i also want my queued calls to be recorded. |
| 17:10.02 | [T]ank | so i have set the queue to monitor |
| 17:10.15 | [T]ank | but if someone calls the extension directly i also have that set to monitor. |
| 17:10.20 | [T]ank | so i get two recordings. |
| 17:10.48 | [T]ank | is there any reason why I should record the queue instead of extension, or the other way around? |
| 17:10.56 | [TK]D-Fender | [T]ank: You should not be using the same extens for both purposes |
| 17:11.24 | [TK]D-Fender | [T]ank: Your queue should monitor itself and only dial out extens that ring the target devices and nothing more |
| 17:11.47 | [TK]D-Fender | [T]ank: Yes its a bit of duplication, but thats sanity for you.... |
| 17:12.08 | [T]ank | not sure i understand |
| 17:13.46 | [T]ank | maybe i do... |
| 17:14.10 | [T]ank | so i have the extensions set up in extensions.conf... but have been adding them dynamically to the queue as Local/exten@context/n |
| 17:14.41 | [T]ank | instead, i should add them ass SIP/device_name so that it does not call the dialplan commands. |
| 17:14.58 | [TK]D-Fender | [T]ank: Don't use your main extens. Make another set for Queue usage |
| 17:15.44 | [T]ank | so a hole second context for extensions? |
| 17:15.52 | [T]ank | one for direct dial, and one for queue dialed? |
| 17:16.10 | funxion | anyone familiar with this error -> channel.c: Didn't get a frame from channel |
| 17:16.36 | *** join/#asterisk famicon (i=redz@c51447ddc.cable.wanadoo.nl) |
| 17:17.18 | *** join/#asterisk snook3r (n=snook3r@bzq-219-46-202.isdn.bezeqint.net) |
| 17:17.19 | [TK]D-Fender | [T]ank: yup |
| 17:17.27 | [T]ank | ok... will try that. thanks |
| 17:17.46 | codazoda | A different FX0 doesn't solve the problem. Moral of the story, don't grab the end of a cord plugged into your Digium card and walk across the carpet. |
| 17:17.54 | *** part/#asterisk maraq (n=none@dsl-204.maraq.net) |
| 17:18.12 | *** join/#asterisk NightMonkey (n=NightMon@pdpc/supporter/sustaining/NightMonkey) |
| 17:18.58 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 17:19.21 | *** join/#asterisk Shido6 (n=wsofa6@204.126.120.132) |
| 17:23.32 | *** join/#asterisk Yourname`` (i=IM@unaffiliated/yourname/x-837320) |
| 17:23.52 | pots_line | . |
| 17:24.01 | pots_line | reseller is being a butt |
| 17:24.08 | pots_line | can't get firmware |
| 17:24.33 | pots_line | need Polycom 3.2.2 bootrom |
| 17:24.53 | Yourname`` | Hi, so I want extension 100 to show callerid 9175555555 when he dials out. And extension 200 to show callerid 4195554444 when he dials out. I use the "callerid=<"Agent 100"> 9175555555" and it doesn't work. They will both be dialing the same number, but dependng on WHICH agent is calling, the callerid needs to show. How do I do it? |
| 17:25.29 | pots_line | anyone have a reseller contact that is helpful in getting firmware |
| 17:27.38 | *** join/#asterisk saftsack (n=saftsack@pD9E0445C.dip.t-dialin.net) |
| 17:28.16 | pots_line | Set(CALLERID(num)="XXXXXXXXXX") |
| 17:28.50 | pots_line | exten => _1NXXNXXXXXX,2,Set(CALLERID(num)="##########") |
| 17:28.53 | pots_line | like so |
| 17:28.57 | pots_line | then dial |
| 17:29.04 | *** join/#asterisk qdk (n=qdk@0x50c627be.bynxx11.adsl-dhcp.tele.dk) |
| 17:29.06 | bkruse | [TK]D-Fender: 02 is scheduled for end of october |
| 17:29.12 | bkruse | but more likely januaray 08 |
| 17:29.40 | *** join/#asterisk luisavila (n=luisavil@194-79-71-122.net.novis.pt) |
| 17:29.47 | Qwell | 02 what? |
| 17:29.55 | pots_line | If you set the caller id as a variable in the sip.conf or iax.conf you won't need to do the set |
| 17:30.01 | pots_line | it will just inherit it |
| 17:30.12 | luisavila | in the zapata and zaptel |
| 17:30.23 | bkruse | GTA02 of the moko |
| 17:30.26 | Qwell | ahh |
| 17:30.31 | bkruse | Qwell: you will become friends with it. |
| 17:30.31 | Qwell | speaking of which |
| 17:30.32 | pots_line | can |
| 17:30.34 | *** part/#asterisk mohsen (n=chatzill@81.31.160.140) |
| 17:30.44 | Qwell | bkruse: whatever happened with that? |
| 17:30.56 | bkruse | Qwell: its still on, but the phone has not been released yet, haha |
| 17:31.03 | [TK]D-Fender | bkruse: Schedules were made to be broken :) |
| 17:31.04 | Qwell | ahh, they're doing 02 |
| 17:31.05 | pots_line | zapata |
| 17:31.05 | bkruse | I guess i can ask for some 01's |
| 17:31.08 | bkruse | [TK]D-Fender: true |
| 17:31.16 | pots_line | But why |
| 17:31.22 | Qwell | I wonder how many 01s they've sold so far |
| 17:31.34 | pots_line | unless you are all analog |
| 17:31.38 | [TK]D-Fender | pots_line: no "" around your CID # |
| 17:31.38 | pots_line | yuck |
| 17:31.54 | pots_line | works |
| 17:32.29 | pots_line | was answering the question |
| 17:32.31 | bkruse | Qwell: way more than they thought |
| 17:32.35 | Qwell | no doubt |
| 17:33.22 | [TK]D-Fender | Canadian celluar data plans = garbage :( |
| 17:33.33 | pots_line | use it to adjust outbound callerid on sip and iax trunking to carrier |
| 17:35.25 | pots_line | callerid in the sip and iax conf files is supposed to be callerid=NAME<NUM> |
| 17:35.48 | pots_line | Depending on ver . . . quotes _will_ piss of * |
| 17:35.59 | bkruse | Qwell: im excited, ill getcha one if i can, let me ping em actually |
| 17:36.19 | funxion | anyone familiar with this error -> channel.c: Didn't get a frame from channel |
| 17:38.01 | *** part/#asterisk bluebeard (n=jmls@62.49.235.130) |
| 17:38.48 | anthm | isn't that when ast_read_frame gets null which is one of the many ways a bridge loop will break? |
| 17:39.26 | funxion | I have no clue thats why Im asking |
| 17:39.37 | funxion | it seems to be the source of a lot of dropped calls for me |
| 17:39.52 | funxion | over both zap and sip channels |
| 17:40.02 | anthm | usually that is the result not the symptom |
| 17:40.23 | anthm | so you should look for something sooner that also happens every time |
| 17:41.18 | funxion | looking |
| 17:43.05 | *** part/#asterisk luisavila (n=luisavil@194-79-71-122.net.novis.pt) |
| 17:44.31 | anthm | you might want to turn on the full debug log too cos some of the stuff hides in there |
| 17:44.43 | anthm | there are a few gems that have no log at all associated with it |
| 17:45.47 | *** join/#asterisk luisavila (n=luisavil@194-79-71-122.net.novis.pt) |
| 17:46.01 | funxion | I've left full debug on |
| 17:46.11 | funxion | until I resolve some of thse issues |
| 17:49.28 | *** join/#asterisk shido6 (n=shido6@204.126.120.132) |
| 17:51.02 | *** join/#asterisk drwelby (n=mpfister@mail.enplan.com) |
| 17:53.45 | *** join/#asterisk Strom_C (n=strom@208.127.172.112) |
| 17:53.51 | *** join/#asterisk Strom_M (n=strom@208.127.172.112) |
| 17:53.55 | Siya | [TK]D-Fender: well from some hosts it would be nice |
| 17:54.20 | Siya | not sure whether accepting unauthed calls from 'anyone' would be such a good idea |
| 17:55.07 | Siya | I have family in the UK with a plustalk account so I can call them using a simple uri |
| 17:55.12 | [TK]D-Fender | Siya: no need for SRV, just a context and allowguest=yes |
| 17:55.21 | Siya | and I'd like to provide the same to them |
| 17:55.43 | Siya | a context? how do I match these calls to this context? |
| 17:57.05 | *** join/#asterisk litage|w (n=nick@70.55.220.203.static.comindico.com.au) |
| 17:57.21 | *** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.ct.comcast.net) |
| 17:59.00 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 17:59.20 | [TK]D-Fender | Siya: if they come in with no auth and match an exten in that context. |
| 17:59.57 | *** join/#asterisk yannj_fr (n=yannj_fr@vpn.intelunix.fr) |
| 18:00.39 | Siya | ic, so the context doesn't matter that much, just that extensions.conf knows the alphanumeric extension (the stuff before @domain.com) |
| 18:01.40 | Siya | that's the weird thing though I assumed this and I can dial the extension (with or without doamin) internally but I've not succeeded to do this from X-lite yet |
| 18:03.05 | Siya | rats! |
| 18:03.23 | Siya | [TK]D-Fender: thank you for your help, confirming that I did well |
| 18:03.51 | Siya | I was dialing <ext>@sip.domain.com and it should be <ext>@domain.com |
| 18:04.07 | Siya | cool |
| 18:04.10 | Siya | :) |
| 18:04.13 | *** join/#asterisk maraq (n=none@dsl-204.maraq.net) |
| 18:06.57 | *** part/#asterisk maraq (n=none@dsl-204.maraq.net) |
| 18:09.06 | *** join/#asterisk tuxd00d (n=tuxinato@128.187.174.112) |
| 18:09.19 | *** join/#asterisk CunningPike (n=arodgers@204.239.12.183) |
| 18:09.39 | *** part/#asterisk SwK_ (n=SwK@user-69-73-37-99.knology.net) |
| 18:09.52 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 18:11.01 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 18:11.42 | webtech_m33 | [TK]D-Fender: i got my TE405 to work... i move it from slot 2 to slot 1 and poof it started working |
| 18:12.04 | [TK]D-Fender | webtech_m33: Good to hear |
| 18:14.08 | *** join/#asterisk FXOL (n=porn@rrcs-64-183-254-126.sw.biz.rr.com) |
| 18:14.09 | drwelby | The "powers that be" have declared that we need to "upgrade" to an Asterix Appliance. Anyone made the transition from CLI to GUI and not regretted it? |
| 18:14.12 | FXOL | hello all |
| 18:14.30 | FXOL | Curious who might be able to help... haven't had any luck on the forums |
| 18:15.23 | FXOL | Where might I find the documentation for doing custom callplans, AGI, etc. |
| 18:15.33 | Qwell | drwelby: You could try out asterisknow, and see what you think of the gui. It's mostly the same as the asterisk appliance |
| 18:15.46 | FXOL | I've fudged my way through doing some custom stuff... but I am not finding any good references |
| 18:16.03 | Qwell | FXOL: check the wiki - there are a ton of examples there |
| 18:16.05 | Qwell | ~wikis |
| 18:16.06 | jbot | wikis is, like, http://www.voip-info.org |
| 18:16.17 | FXOL | I have... but no reference |
| 18:16.21 | FXOL | just random examples |
| 18:16.42 | Qwell | in asterisk try doing...umm... agi dumphtml, I think it is |
| 18:16.48 | [TK]D-Fender | drwelby: UPgrade? Ask them if they have any extra crack for resale.... |
| 18:17.05 | FXOL | I can post a sample of what I've written |
| 18:17.28 | JerJer | hey now people can ride a SLUT and not get in trouble with the significant other |
| 18:17.42 | [TK]D-Fender | FXOL: The dialplan apps all have usage pages through CLI. How you use them is up to you. |
| 18:17.50 | FXOL | but I don't see a reference to commands used, such as Playback, Read, GotoIf, etc |
| 18:17.53 | JerJer | http://www.foxnews.com/story/0,2933,297184,00.html |
| 18:18.05 | [TK]D-Fender | FXOL: "show application playback" <- in * CLI |
| 18:18.34 | [TK]D-Fender | FXOL: You've clearly missed the BIG PRINT. You may also want to download an read THE BOOK. |
| 18:18.36 | [TK]D-Fender | ~book |
| 18:18.37 | jbot | [book] Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 or Temporarily at http://www.aocomputing.net/AsteriskTFOT.pdf |
| 18:18.38 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^^^^^^ |
| 18:18.44 | Qwell | JerJer: somebody needs to put down the crackpipe |
| 18:19.04 | JerJer | or hit it harder :D |
| 18:19.06 | FXOL | ([TK]D-Fender): Forgive me... What Big Print? :P |
| 18:19.33 | *** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.ct.comcast.net) |
| 18:19.35 | [TK]D-Fender | FXOL: in the book, the WIKI has plenty of references on app usage and the * CLI, etc. |
| 18:19.47 | [TK]D-Fender | FXOL: Just typing in "help" in CLI, etc |
| 18:20.05 | [TK]D-Fender | FXOL: But regardless I've just handed you the key resources you were looking for. |
| 18:20.16 | FXOL | perhaps I'm not up on acronym's, yet... CLI? And I'll read that PDF :P |
| 18:20.55 | *** join/#asterisk GreggB (n=GreggB@66.206.86.107) |
| 18:21.43 | Nugget | CLI refers to the asterisk console (command line interface) |
| 18:22.05 | FXOL | hrmm |
| 18:22.21 | FXOL | whic is different from linux root login ;P |
| 18:22.25 | Nugget | yes. |
| 18:22.32 | FXOL | and how do I get to the CLI? |
| 18:22.33 | [TK]D-Fender | FXOL: Command Line Interface. As in what you get when you do "asterisk -r" to connect to * |
| 18:22.40 | Nugget | run "asterisk -r" |
| 18:22.41 | FXOL | aahhhh ;P |
| 18:22.45 | FXOL | lemme try it |
| 18:22.53 | [TK]D-Fender | FXOL: How on earth you could have used * and not known this is a mystery to me.... |
| 18:23.25 | FXOL | ([TK]D-Fender): only been playing with * about a week thus far... and doing minor dev for 2 days.... |
| 18:23.39 | FXOL | I learn by getting dirty rather then reading sometime ;) |
| 18:23.45 | [TK]D-Fender | FXOL: Ok, fine, sure, welcome abord! |
| 18:23.48 | FXOL | thanks |
| 18:23.54 | FXOL | not a newbie to code... just * :P |
| 18:24.11 | FXOL | [root@asterisk1 asterisk]# asterisk -r |
| 18:24.11 | FXOL | Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exi |
| 18:24.20 | Nugget | is asterisk running? |
| 18:24.30 | GreggB | Google isn't helping me much, so... Does anyone know of a way to detect red and yellow alarm states on my T1 (using zaptel w/ a Wildcard TE12xP Card). The log file /var/log/asterisk/messages doesn't consistently show when a circuit drops to an alarm state, and I'm seeking a way to auto-recover from such alarms (I'm getting red alarms about every 12-18 hours lately). |
| 18:24.30 | Nugget | the -r will try to attach to an already-running instance of asterisk. |
| 18:24.37 | FXOL | yup.. I'm on the phone ;P |
| 18:24.42 | [TK]D-Fender | FXOL: Start with this "show applications" and "show functions" and then drill each into detail to see how the bits work. In the BOOK, focus on chapter 5 (Dialplan basics) and read up on "asterisk variables" on the WIKI |
| 18:24.45 | Nugget | did you run it as root? |
| 18:24.55 | FXOL | I did a amportal restart as root, yew |
| 18:24.56 | FXOL | s |
| 18:24.57 | Nugget | ah, duh, I see. |
| 18:25.03 | FXOL | that a problem? |
| 18:25.08 | [TK]D-Fender | Nugget: Regardless if its running then root should be able to connect, no? |
| 18:25.12 | FXOL | okay.. in CLI now |
| 18:25.14 | *** join/#asterisk Igbothom_3rd (n=Hilton@office.quarkgroup.com.au) |
| 18:25.23 | [TK]D-Fender | ~amp |
| 18:25.23 | jbot | i guess amp is NOT supported here! People using it should join #freepbx (FreePBX is the new name of AMP) |
| 18:25.27 | [TK]D-Fender | ~freepbx |
| 18:25.28 | jbot | freepbx is probably unable to be supported here. It's a complex piece of dialplan and GUI, and can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 18:25.28 | Nugget | [TK]D-Fender: no, not in my experience. |
| 18:25.36 | [TK]D-Fender | Nugget: Ok... |
| 18:25.37 | FXOL | 177 applications registered |
| 18:26.13 | FXOL | 46 custom functions |
| 18:27.36 | *** join/#asterisk bmg505 (n=leon@196.209.176.121) |
| 18:27.42 | *** join/#asterisk deeperror (n=deeperro@69-215-202-195.ded.ameritech.net) |
| 18:28.22 | Nugget | [TK]D-Fender: I think I may have misunderstood your question. |
| 18:28.38 | Nugget | root should be able to connect to any running asterisk, no matter what uid it is using. |
| 18:28.50 | [TK]D-Fender | Nugget: SHOULDN'T root be able to access "asterisk -r" regardless of the user its running under <- |
| 18:28.58 | Nugget | yes, you're correct. |
| 18:29.01 | [TK]D-Fender | Nugget: Better :) |
| 18:29.07 | Corydon76-dig | Nugget: unless |
| 18:29.27 | FXOL | is there examples on how to "use" the applications? |
| 18:29.37 | FXOL | like the functions show |
| 18:29.54 | [TK]D-Fender | FXOL: As you're new here, know this : AMP/FreePBX/Trixbox/A@H are not supported here so your questions should not be regarding any configurations created or maintained by them. |
| 18:29.54 | Corydon76-dig | "core show application Foo" |
| 18:30.08 | Nugget | I was asking if FXOL was trying to run "asterisk -r" as non-root (I was blind and didn't notice that it was clear from his paste) and mis-read your question in that context. |
| 18:30.14 | deeperror | I've got a sip peer setup to voicepulse and it uses a hostname sfo.vp.com however they request that srvlookup=yes there is some kind of load balancing going on at their side. The first time i dial out to them i get 407 Auth required as the call is going to sip:1234567890@sfo.vp.com my dialplan trys to dial out the call again and this time it dials out and works to sip:1234567890@11.22.33.44 the srvlookup seems to work only |
| 18:30.22 | FXOL | ([TK]D-Fender): I don't think that applies... but ok |
| 18:30.34 | [TK]D-Fender | FXOL: "show application [appnamewithoutbraces]" |
| 18:30.45 | [TK]D-Fender | FXOL: "show function [functionnamewithoutbraces]" |
| 18:30.55 | Nugget | FXOL: extensions.conf is an "example" of how to "use" the applications. |
| 18:31.03 | FXOL | great |
| 18:31.05 | [TK]D-Fender | FXOL: it certainal does as you've already referenced amportal... |
| 18:31.08 | FXOL | thank you |
| 18:31.29 | FXOL | ([TK]D-Fender): I was indicating I restarted the server process |
| 18:31.48 | [TK]D-Fender | FXOL: Yes, but amportal brands you instantly. |
| 18:31.56 | FXOL | ([TK]D-Fender): brands me? :P |
| 18:32.08 | Nugget | amportal is not asterisk. |
| 18:32.18 | FXOL | I realize that ;P |
| 18:32.26 | FXOL | but doesn't it restart asterisk ? |
| 18:32.33 | Nugget | I have no idea. I've never used it. |
| 18:32.36 | FXOL | hehe |
| 18:32.38 | FXOL | it does ;P |
| 18:32.58 | Nugget | how did you manage to get amportal installed with absolutely no experience with asterisk at all? |
| 18:33.01 | FXOL | I'm just surprised that this references arent online |
| 18:33.10 | FXOL | (Nugget): quite easilly |
| 18:33.45 | FXOL | I've got a fully functional system w/ multiple phones and softphones... and already have 2 custom IVR apps... |
| 18:33.51 | FXOL | well.. semi custom |
| 18:33.55 | Nugget | but you've never heard of extensions.conf? |
| 18:33.59 | FXOL | yes |
| 18:34.04 | FXOL | that's where I have most of my custom items |
| 18:34.14 | Nugget | something just doesn't add up. |
| 18:34.18 | FXOL | like? |
| 18:34.53 | Nugget | your claim that you've written a custom dialplan without knowing what a dialplan is. |
| 18:35.00 | FXOL | no I didnt say that |
| 18:35.06 | Nugget | yes, you sort of did. |
| 18:35.15 | FXOL | I said I don't have a reference to all commands availalble in a dialplan |
| 18:35.27 | FXOL | I've built and AGI that pulls data from a web site too |
| 18:35.33 | [TK]D-Fender | FXOL: And you do now! Merry Christmas! |
| 18:35.35 | *** join/#asterisk tsurko (n=tsurko@78.90.100.214) |
| 18:35.37 | FXOL | but again... no reference to all functions available |
| 18:35.46 | lirakis | anyone know the pre recourded sound file that says essentially "no voicemail available for that extension" ? |
| 18:35.47 | FXOL | ([TK]D-Fender): yup :P |
| 18:35.54 | *** join/#asterisk jsmith (n=jsmith@000-190-367.area3.spcsdns.net) |
| 18:35.54 | *** mode/#asterisk [+o jsmith] by ChanServ |
| 18:36.10 | [TK]D-Fender | lirakis: Try to access one and watch the CLI :) |
| 18:36.48 | FXOL | see... learned something already |
| 18:36.48 | FXOL | <PROTECTED> |
| 18:36.49 | FXOL | :P |
| 18:36.55 | FXOL | simple enough.. but didnt have the refernce |
| 18:38.17 | FXOL | thanks again guys... |
| 18:38.27 | FXOL | this is why I like IRC over forums :) |
| 18:39.14 | Nugget | web forums are the ghettos of the internet. |
| 18:39.22 | deeperror | how should a peer be setup that requires srvlookup=yes? |
| 18:39.39 | *** join/#asterisk tsurko (n=tsurko@78.90.100.214) |
| 18:40.09 | deeperror | how should a peer be setup that requires srvlookup=yes if the lookup is not performed when referencing the peer name in the dial statement. |
| 18:40.41 | FXOL | one more question... |
| 18:40.42 | FXOL | <PROTECTED> |
| 18:40.51 | FXOL | are there references to available contexts? |
| 18:41.14 | FXOL | bah.. forget it ;P |
| 18:44.09 | [TK]D-Fender | FXOL: Indeed rather silly question, move along to reading the book :) |
| 18:44.32 | Nugget | egrep '^\[' /etc/asterisk/extensions.conf |
| 18:44.39 | Nugget | ^ reference to available contexts. :) |
| 18:45.36 | FXOL | (Nugget): nice ;P |
| 18:45.55 | syzygyBSD | of course, that doesn't work with included files... |
| 18:45.56 | [TK]D-Fender | Nugget: Yeah... and you know how much good THAT'LL do ;) |
| 18:48.28 | *** join/#asterisk DougVOIP (n=Dougg@208.230.232.54) |
| 18:48.33 | FXOL | nice |
| 18:48.34 | FXOL | <PROTECTED> |
| 18:48.35 | syzygyBSD | Nugget: asterisk -rx 'show dialplan'|egrep '^\[' |
| 18:49.27 | GreggB | Anyone know how to read the alarm state off a Wildcard TE12xP Card (zaptel) |
| 18:49.42 | syzygyBSD | zttool |
| 18:50.13 | syzygyBSD | that will just tell you what the state is, not why |
| 18:50.15 | GreggB | syzygyBSD: in a way which could be automated? |
| 18:51.02 | GreggB | My circuit keeps dropping to a red alarm, and all it's taken is running ztcfg -vvvv to bring it back online. |
| 18:51.22 | GreggB | So I'm trying to find a way to automate alarm state detection, and correction. |
| 18:51.27 | syzygyBSD | cat /proc/zaptel/1 ? |
| 18:51.39 | syzygyBSD | not sure if that would help, doesn't tell you red alarm |
| 18:51.48 | GreggB | Tried that - it doesn't report anything intresting |
| 18:51.56 | syzygyBSD | if it is constantly dropping to red alarm, something is probably misconfigured |
| 18:52.09 | tzafrir | Automated alarm correction? |
| 18:52.13 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 18:52.13 | *** mode/#asterisk [+o anthm] by ChanServ |
| 18:52.16 | deeperror | asterisk -rx "zap show status" |
| 18:52.18 | GreggB | Hmm, it's been working for months, and just started dropping to a red alarm about every 12-18 hours these past few days |
| 18:52.21 | tzafrir | alarms often report things beyond your control |
| 18:52.54 | syzygyBSD | like 'you have to get up and go to work now'... ya, I hate alarms |
| 18:53.18 | tzafrir | GreggB, you can check how zttool checks for alarms |
| 18:53.28 | GreggB | syzygyBSD: yea - I hate those too, I killed mine a few months back and never looked back :-) |
| 18:53.30 | tzafrir | one ioctl, basically |
| 18:53.54 | tzafrir | GreggB, I already have that information available in my sysfs zaptel branch |
| 18:54.36 | GreggB | tzafrir: code in development then? |
| 18:54.37 | *** join/#asterisk ManxPower (n=manxpowe@51.sub-70-222-13.myvzw.com) |
| 18:54.43 | tzafrir | right |
| 18:54.50 | *** join/#asterisk _Sam-- (n=sam@fresco.kneedraggers.com) |
| 18:55.16 | tzafrir | GreggB, but zaptel already notifies Asterisk of alarms |
| 18:55.20 | GreggB | tzafrir: cool - that's a nice long-term plan. I guess I need to dig around the zaptel development source then |
| 18:55.35 | tzafrir | You can have Asterisk report them in more meaningful ways |
| 18:55.45 | GreggB | tzafrir: I thought so too... I like deeperror's idea (dont know why I didnt think of that myself) |
| 18:55.47 | tzafrir | e.g: through the manager interface |
| 18:55.53 | _Sam-- | its been a while since i used meetme, and had a conference -- i have upgraded to 1.4 since the last time, but i dont know how to fix this... |
| 18:55.56 | _Sam-- | [Sep 19 14:55:33] WARNING[19929]: pbx.c:1797 pbx_extension_helper: No application 'MeetMe' for extension (default, 900, 1) |
| 18:56.18 | [TK]D-Fender | _Sam--: You didn't ahve a zaptel timing source installed before compiling *. |
| 18:56.33 | _Sam-- | i do have a zaptel timing source |
| 18:56.37 | deeperror | GreggB: how about something like this? http://www.pastebin.ca/703638 |
| 18:56.47 | ManxPower | _Sam--: not before installing Asterisk |
| 18:56.47 | FXOL | man... I may need a 3rd monitor... too much on my screens :P |
| 18:56.51 | _Sam-- | wcte11xp 24480 0 |
| 18:56.51 | _Sam-- | zaptel 221344 1 wcte11xp |
| 18:56.56 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 18:57.00 | _Sam-- | ive had them since much before installing asterisk. |
| 18:57.06 | [TK]D-Fender | _Sam--: Not set up at that point in time. or its not ready NOW and * didn't load MeetMe because of it. |
| 18:57.15 | ManxPower | _Sam--: Well then Asterisk did not see zaptel installed and so did not build MeetMe |
| 18:57.28 | [TK]D-Fender | _Sam--: try "module load app_meetme.so" |
| 18:57.34 | ManxPower | Pretty simple, really |
| 18:58.09 | _Sam-- | it cant load, it doesnt exist. |
| 18:58.17 | _Sam-- | [Sep 19 14:57:39] WARNING[31474]: loader.c:360 load_dynamic_module: Error loading module 'app_meetme.so': /usr/lib/asterisk/modules/app_meetme.so: cannot open shared object file: No such file or directory |
| 18:58.29 | ManxPower | _Sam--: so what you need to figure out is why Asterisk did not find Zaptel when you built it. |
| 18:58.33 | _Sam-- | what is the easiest way to fix? |
| 18:58.42 | GreggB | deeperror: aww damn - you just saved me even more work. Thanks! |
| 18:59.20 | deeperror | GreggB: just throw that command into a variable and check the variable for your alarm status in question if it validates run ztcfg haha |
| 18:59.20 | ManxPower | _Sam--: Well, since we don't know WHY the zaptel header and libraries were not found when you installed Asterisk, there may not be an "easy fix" |
| 18:59.24 | _Sam-- | would an older app_meetme.so from 1.2 asterisk work if i moved it to where it needs to be ? |
| 18:59.30 | ManxPower | _Sam--: no |
| 18:59.48 | _Sam-- | ok. i think i may know why. |
| 18:59.55 | GreggB | deeperror: cool, now onto getting the telco to fix the circuit itself :\ |
| 19:00.09 | ManxPower | _Sam--: I would rerun ./configure and look at the meetme application in make menuconfig or whatever it is called in 1.4 |
| 19:00.10 | _Sam-- | when i made asterisk 1.4 i dont know if i updated zaptel because i dont use the zap card for telephony any longer (its an idle card in the machine) |
| 19:00.11 | deeperror | good luck with that one will need a truck load of luck there |
| 19:00.21 | _Sam-- | maybe i need to make a new zaptel, before i make my new asterisk. |
| 19:00.38 | ManxPower | _Sam--: well that might be a reason. NEVER run different major versions of asterisk and zaptel on the same system |
| 19:01.00 | [TK]D-Fender | _Sam--: thats a real big no-no |
| 19:01.02 | _Sam-- | although in looking at my source directory, i do have the zaptel source extracted configured and installed (zaptel 1.4.4) |
| 19:01.10 | *** join/#asterisk solar_ant (n=solar@122.164.123.172) |
| 19:01.22 | GreggB | Yea Integra has been pretty good about things (I can call their main number, and actually get an english speaking tech who knows what he's doing) |
| 19:02.23 | _Sam-- | if older zaptel kernel modules are loaded do i have to rmmod or something to get the new ones loaded? |
| 19:02.25 | deeperror | live up to the name. not att :) |
| 19:03.12 | ManxPower | _Sam--: of course you do, just like any other kernel module. |
| 19:03.45 | _Sam-- | make install on zaptel exits with an error, i dont know if its benign, or means anything.... |
| 19:03.46 | _Sam-- | build_tools/genudevrules: line 1: udevinfo: command not found |
| 19:03.47 | _Sam-- | make: *** [devices] Error 1 |
| 19:04.00 | _Sam-- | plain make worked fine |
| 19:04.24 | ManxPower | you need to install the udevinfo command or uninstall udev |
| 19:04.39 | ManxPower | _Sam--: I think you might want to try a general Linux channel, as that is where your issue is. |
| 19:05.27 | ManxPower | I seem to recall the udevinfo command problem being discussed on the asterisk-users mailing list. The problem was that udev was partially installed by the OS install, but not fully installed. |
| 19:05.45 | _Sam-- | yeah, i dont use udev thats why im scratching my nads |
| 19:06.06 | ManxPower | _Sam--: well the make install is seeing some form of udev installed. |
| 19:06.09 | ManxPower | rpm -qa | grep udem |
| 19:06.15 | ManxPower | ..er... rpm -qa | grep udev |
| 19:06.20 | _Sam-- | dselect |
| 19:06.25 | _Sam-- | or apt-get |
| 19:06.46 | ManxPower | you'll have to figure out your distro's way of listing installed packages |
| 19:06.56 | _Sam-- | root@phone:/usr/src/ast2/zaptel-1.4.4# apt-get remove udev |
| 19:06.56 | _Sam-- | Reading Package Lists... Done |
| 19:06.56 | _Sam-- | Building Dependency Tree... Done |
| 19:06.56 | _Sam-- | Package udev is not installed, so not removed |
| 19:07.10 | *** join/#asterisk elixer (i=elixer@65.207.74.18) |
| 19:07.15 | ManxPower | good, now install any package with the string udev anywhere in the package name. |
| 19:07.19 | ManxPower | ..er... uninstall |
| 19:07.25 | _Sam-- | there are none. |
| 19:07.31 | _Sam-- | there are only 4 udev packages. |
| 19:07.34 | _Sam-- | the main one is 'udev' |
| 19:07.39 | ManxPower | well, I guess it sucks to be you then. |
| 19:07.40 | _Sam-- | and i dont got it, or any others on this machine. |
| 19:08.05 | _Sam-- | this machine has been running asterisk for 3 years, same hardware....i dont understand why installing zaptel 1.4.4 is a problem |
| 19:08.15 | _Sam-- | ive had 3 other versions of zaptel on here without this problem |
| 19:08.25 | _Sam-- | actually, 4 or more. |
| 19:08.45 | _Sam-- | no change to kernel, or machine. |
| 19:09.22 | lirakis | can i roll (cascade) calls from one queue to another if all the agents in the first queue are busy? |
| 19:10.01 | [TK]D-Fender | lirakis: yup. |
| 19:10.27 | lirakis | i guess.. i should ... maybe just set a time out |
| 19:10.36 | [TK]D-Fender | lirakis: Look at "show application queue" and the values for the QUEUESTATUS variable |
| 19:10.40 | lirakis | and then have the next queue be dialed? |
| 19:10.41 | ManxPower | _Sam--: how many versions of 1.4 zaptel have you installed? |
| 19:10.51 | _Sam-- | found it. |
| 19:10.51 | ManxPower | on this machine. |
| 19:10.52 | lirakis | [TK]D-Fender: okay |
| 19:10.55 | _Sam-- | there was /etc/udev |
| 19:11.00 | _Sam-- | i just mv'd it to udev.sav |
| 19:11.02 | _Sam-- | make install fine |
| 19:11.29 | _Sam-- | 'locate udev' yielded only 3 results |
| 19:11.31 | ManxPower | _Sam--: on RPM distros I would do an rpm -qilf /etc/udev to see what package created that |
| 19:11.57 | _Sam-- | i only have 2 RHEL boxes, and this isnt one unfortunately. i dont know the same commands for debian / apt. |
| 19:12.25 | ManxPower | I would NOT recommend running Asterisk on a distro you are not familiar with. |
| 19:12.35 | ManxPower | That's just asking for issues like this. |
| 19:12.38 | _Sam-- | thank you, but i wasnt asking for any recommendations. |
| 19:13.06 | ManxPower | now reinstall asterisk, starting with the ./configure |
| 19:13.17 | ManxPower | that should pick up and build meetme |
| 19:13.19 | FXOL | another question for you smart guys ;P |
| 19:13.54 | FXOL | is there a way to playback a file, but also monitor for a specific keypress response? |
| 19:14.09 | jsmith | FXOL: The Background() application? |
| 19:14.10 | Corydon76-dig | FXOL: Background |
| 19:14.15 | _Sam-- | thank you for the help, Manx. |
| 19:14.26 | FXOL | I looked at that... does that look for an extension only? or any keypress? |
| 19:14.26 | Corydon76-dig | or Read(), for that matter |
| 19:14.48 | ManxPower | FXOL: a key press is an extension in this case. |
| 19:14.52 | Corydon76-dig | Background is extension-only. Read is arbitrary input |
| 19:14.56 | FXOL | hrmm |
| 19:15.03 | FXOL | well.. I want to playback a file |
| 19:15.10 | FXOL | and not do anything special if no keypressed |
| 19:15.21 | Corydon76-dig | Well, Read has a timeout |
| 19:15.21 | FXOL | but if key is hit during it... branch off conditionally |
| 19:15.39 | ManxPower | Corydon76-dig: you can use read to playback a file while waiting for a keypress? |
| 19:15.46 | Corydon76-dig | ManxPower: yes |
| 19:15.53 | ManxPower | nifty. |
| 19:15.53 | FXOL | hrmm.. hows that work? |
| 19:15.54 | Corydon76-dig | The file is considered a prompt |
| 19:16.00 | ManxPower | "show application read" |
| 19:16.00 | Corydon76-dig | show application Read |
| 19:16.08 | FXOL | yea I know ;P |
| 19:16.23 | FXOL | <PROTECTED> |
| 19:16.38 | ManxPower | that must be the 1.4 read |
| 19:16.42 | FXOL | but doesn't that way until AFTER reading to get digits? |
| 19:16.52 | FXOL | wait = way |
| 19:17.06 | Corydon76-dig | Nope, it's during |
| 19:17.14 | FXOL | hrmm |
| 19:17.23 | Corydon76-dig | Why don't you try it? |
| 19:17.27 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-88-154-177-39.red.bezeqint.net) |
| 19:17.48 | FXOL | will do that ;P |
| 19:17.50 | FXOL | thx |
| 19:18.00 | FXOL | so just use play in place of background |
| 19:18.11 | FXOL | or playback for that matter |
| 19:18.14 | Corydon76-dig | No, use Read, in place of Background |
| 19:18.32 | Corydon76-dig | See the filename field? That's the file to play back as a prompt |
| 19:18.42 | [TK]D-Fender | FXOL: [|filename] <----------- |
| 19:18.43 | FXOL | right |
| 19:18.44 | FXOL | so example |
| 19:18.46 | FXOL | exten => s,n,Playback(custom/ClubGPID) |
| 19:18.48 | FXOL | oops |
| 19:18.49 | FXOL | sorry |
| 19:19.19 | Corydon76-dig | Read(gpid,ClubGPID,1) |
| 19:19.24 | FXOL | Read(MYVAR, filename, 1) ? |
| 19:19.49 | Corydon76-dig | and then you can check the value of ${gpid} |
| 19:19.54 | FXOL | right |
| 19:19.59 | *** join/#asterisk Lucky7 (n=Adam@207.200.28.175) |
| 19:20.02 | FXOL | simple enough |
| 19:20.11 | FXOL | didnt realize I could play bcak with read... thanks |
| 19:21.01 | _Sam-- | !seen zoa |
| 19:21.04 | FXOL | _Sam--, I don't remember seeing zoa |
| 19:22.44 | FXOL | (Corydon76-dig): how can I jump back to the main IVR? similar to 7777? |
| 19:23.05 | Corydon76-dig | Uh, Goto |
| 19:23.17 | FXOL | hehe |
| 19:23.49 | FXOL | Goto what tho? main IVR callplan? |
| 19:24.07 | Corydon76-dig | If that's what the label is, yes |
| 19:24.24 | FXOL | not sure.. will have to dig it up |
| 19:24.28 | FXOL | used the built in IVR |
| 19:24.58 | Corydon76-dig | exten => s,n(main IVR dialplan),NoOp()... Goto(s,main IVR dialplan) |
| 19:25.28 | [TK]D-Fender | FXOL: Ok, seriously, go sit down with the book and READ. |
| 19:25.40 | FXOL | I read through that book.. no refernce to this |
| 19:25.42 | Corydon76-dig | You thought I was joking, didn't you? |
| 19:25.44 | FXOL | newp |
| 19:25.57 | FXOL | it had some decent basic exmaples... |
| 19:26.07 | Corydon76-dig | Yes, you can define a label with spaces... |
| 19:26.18 | FXOL | and? |
| 19:26.21 | FXOL | I didnt ask about that |
| 19:27.13 | [TK]D-Fender | FXOL: The book shows all sorts of samples for IVR's, *'s standard extensions, usage of variables, etc. Stop dodging. |
| 19:27.51 | jsmith | FXOL: I wrote the book. Yes, it has examples. The second edition (in stores now!) has even more examples. |
| 19:27.57 | FXOL | I didn't "all sorts"... it was quite basic |
| 19:28.01 | FXOL | but ok |
| 19:28.23 | anonymouz666 | jsmith: TFOT2 will be PDF available? |
| 19:29.12 | jsmith | anonymouz666: Of course... we're debuting it next week at AstriCon! |
| 19:29.37 | jsmith | By the way, that was a subtle reminder... ASTRICON IS NEXT WEEK, PEOPLE! |
| 19:29.40 | FXOL | problem seems to be that when I change the IVR setup from the FreePBX admin... it could rename the dialplan in the .conf file |
| 19:29.42 | anonymouz666 | great |
| 19:29.55 | FXOL | but ok ;P |
| 19:30.16 | [TK]D-Fender | FXOL: FreePBX *owns your ass*. |
| 19:30.20 | FXOL | lol |
| 19:30.22 | jsmith | FXOL: That's one downside to the FreePBX gui... |
| 19:30.33 | FXOL | (jsmith): I see that ;P |
| 19:30.38 | [TK]D-Fender | jsmith: Yeah... the other ... is the FreePBX GUI :) |
| 19:31.01 | jsmith | Makes me remember why I gave up on my own GUI and started writing docs instead |
| 19:31.27 | *** part/#asterisk solar_ant (n=solar@122.164.123.172) |
| 19:32.37 | FXOL | so... this "should" work? :P |
| 19:32.37 | FXOL | GotoIf($["${SKIPID}" = "*"]?ivr-3,1) |
| 19:33.07 | alrs | jsmith: Is adhearsion the preferred AGI framework in the new book, or is it just one chapter of many? |
| 19:33.16 | [TK]D-Fender | FXOL: once execution gets there and that variable has any hope of containing the value you're looking for, sure |
| 19:33.19 | deeperror | how about asterisk world vs astricon? |
| 19:33.20 | jsmith | Sure, as long as ivr-3 is an extension in the current context |
| 19:33.26 | jsmith | alrs: Just one chapter of many |
| 19:33.29 | FXOL | hrmm |
| 19:33.42 | FXOL | no.. it's in the extensions_additional.conf |
| 19:33.51 | jsmith | deeperror: AstriCon is for Asterisk users. Digium/Asterisk World is for future Asterisk users |
| 19:34.12 | jsmith | deeperror: Digium/Asterisk World is more oriented at business-types investigating Asterisk |
| 19:34.26 | jsmith | deeperror: AstriCon is the official users conference for Asterisk... |
| 19:34.34 | deeperror | so would it be worthless to goto *world if i'm already implementing asterisk? |
| 19:34.45 | jsmith | deeperror: Obviously there's some overlap, but hopefully that helps clarify things |
| 19:34.59 | jsmith | deeperror: I'm not saying it's worthless... but you might get more out of AstriCon |
| 19:35.07 | FXOL | where is Astricon? |
| 19:35.13 | jsmith | Phoenix, AZ |
| 19:35.20 | FXOL | not too bad.. when? |
| 19:35.23 | russellb | next week! |
| 19:35.26 | FXOL | doh |
| 19:35.31 | jsmith | 25th through the 28th |
| 19:35.33 | FXOL | going be tearing up the roads in AR |
| 19:35.39 | jsmith | russellb: I'm excited too! |
| 19:35.44 | russellb | :-D |
| 19:35.47 | FXOL | (jsmith): they have a website? |
| 19:35.50 | russellb | astricon.net |
| 19:35.55 | deeperror | well i saw the talks at world are about callcenters and setting them up etc etc. I'm going out there in october...probably should have went to az instead haha |
| 19:36.10 | FXOL | woah |
| 19:36.10 | FXOL | http://www.astricon.net/?q=node/2 |
| 19:36.10 | jsmith | deeperror: Well, I'll be at both... Be sure to say hi. |
| 19:36.13 | Qwell | deeperror: it's not too late to go to astricon too |
| 19:36.13 | FXOL | Dallas :P |
| 19:36.18 | Qwell | FXOL: last year |
| 19:36.22 | FXOL | doh |
| 19:36.23 | FXOL | ye a;P |
| 19:36.24 | jsmith | FXOL: That's the page for last year's show |
| 19:36.24 | FXOL | damnit |
| 19:36.25 | deeperror | i know but it kinda is from work aspect haha |
| 19:36.26 | FXOL | :P |
| 19:36.31 | deeperror | maybe.... |
| 19:36.38 | Qwell | deeperror: try :D |
| 19:36.45 | Qwell | more users going is a good thing |
| 19:36.46 | deeperror | they pay for it let me ask |
| 19:36.47 | Strom_M | i may drop in for a day or two of astricon |
| 19:36.51 | FXOL | I'm a year late! :P |
| 19:37.02 | Qwell | FXOL: are you? or are you a week early? :) |
| 19:37.06 | Strom_M | if i can figure out accomodation |
| 19:37.07 | FXOL | year late ;P |
| 19:37.11 | FXOL | not in Dallas this year |
| 19:37.20 | Qwell | Strom_M: I'm sure a bunch of people have couches |
| 19:37.22 | FXOL | (jsmith): How should I change that command to deal with other context? |
| 19:37.35 | ManxPower | If they had it near or in Atlanta again, I'd go to Astricon |
| 19:37.41 | FXOL | <PROTECTED> |
| 19:37.46 | Strom_M | Qwell: surprisingly, i dont know anyone in phoenix |
| 19:37.50 | FXOL | s,1 I assume |
| 19:37.52 | Qwell | Strom_M: I mean attendees |
| 19:37.55 | Strom_M | phoneix |
| 19:38.09 | jsmith | FXOL: GotoIf($["${SKIPID}" = "*"]?some_context,some_extension,some_priority) |
| 19:38.15 | Strom_M | Qwell: hey, do you have a sofa? |
| 19:38.31 | Qwell | no idea.. I'm with Kevin, russellb, and putnopvut |
| 19:38.45 | FXOL | so... ?ivr-3,s,1 |
| 19:39.09 | wishes | are there any known major probs upgrading from 1.2 to 1.4? |
| 19:39.42 | deeperror | well its either one or the other hahaha and i've already got the tickets setup for october :( |
| 19:39.49 | ManxPower | wishes: most of them would be in....upgrade.txt or whatever they call it in 1.4 |
| 19:39.51 | deeperror | i guess next year will be at the con for sure |
| 19:39.55 | ManxPower | you should also look at the 1.2 upgrade.txt |
| 19:40.04 | _Sam-- | ManxPower: "locate udev" on machine yields nothing. zaptel 1.4 configure, there is nothing there that shows anything about udev when i configure, and when i make, i checked the make output, there is nothing there referencing udev either. however, anytime i make install zaptel, it creates /etc/udev with rules for zaptel in there, and modprobe zaptel fails. |
| 19:40.22 | ManxPower | _Sam--: ASTERISK ./configure |
| 19:40.27 | FXOL | (jsmith): ugh... I hit the * during playback... log shows the * hit... but doesn't exec. gotoif (at least not results I want) :P |
| 19:40.38 | ManxPower | now that you have removed the fake udev directory everything should build and install correctly. |
| 19:40.51 | _Sam-- | im not convinced my zaptel is correct, becaus i cant modprobe zaptel |
| 19:40.56 | FXOL | exten => s,n,Read(SKIPID,custom/ClubGPID,1) |
| 19:40.56 | FXOL | GotoIf($["${SKIPID}" = "*"]?ivr-3,s,1) |
| 19:40.57 | _Sam-- | shouldnt i be able to? |
| 19:40.57 | ManxPower | _Sam--: *shrug* Wait until the next zaptel release, the bug is supposed to be fixed there. |
| 19:41.06 | wishes | ManxPower: i wasnt overly interested in the changelog, but more of a 'any majorish bugs that will stop the server working until i can fix it' |
| 19:41.19 | ManxPower | wishes: Oh, I'm sure there are. |
| 19:41.22 | wishes | i might just make another server and test it |
| 19:41.38 | _x86_ | ugh |
| 19:41.38 | wishes | if i screw the phones over and cant get them back again ill be peeved :) |
| 19:41.40 | ManxPower | Never, EVER trust a Digium release to be stable. |
| 19:41.49 | _Sam-- | my 'make' of zaptel seems to make everything thing. the make install seems to fail. |
| 19:42.00 | wishes | but atm im having more problems because im on an older version and the docs dont work etc |
| 19:42.10 | _x86_ | I've got an asterisk box with (2) T1's going into it, (1) CAS T1 to the PSTN, and another (1) CAS T1 to an FXS channel bank |
| 19:42.11 | ManxPower | wishes: "docs don't work"? |
| 19:42.25 | wishes | i wasted like 4 hours just to figure out festival doesnt take ' or " like the examples and docs |
| 19:42.27 | ManxPower | most of the docs out there are for 1.0 or 1.2 |
| 19:42.31 | jfitzgibbon | if 1.4 was intended to be dropped in untested on top of 1.2.x, it would be called 1.2.something |
| 19:42.42 | ManxPower | uh, "show application festival" would have told you. |
| 19:42.42 | _x86_ | when the PSTN T1 goes down, and comes back up, one of my SIP phones gets 24 concurrent calls |
| 19:42.47 | _x86_ | what would cause this? |
| 19:43.06 | ManxPower | _x86_: A crappy dialplan. |
| 19:43.22 | wishes | ManxPower: nah it doesnt |
| 19:43.26 | ManxPower | but without a pastebin of it happening...... |
| 19:43.32 | _x86_ | ManxPower: can i PB it and have you review it for me? |
| 19:43.45 | ManxPower | _x86_: if it is hard work I'm not interested. |
| 19:44.17 | FXOL | (jsmith): Are the " around the * not right? |
| 19:44.43 | ManxPower | FXOL: if you have quotes on one side of the = you need them on the other side. |
| 19:44.48 | FXOL | exten => s,n,Read(SKIPID,custom/ClubGPID,1) |
| 19:44.49 | FXOL | GotoIf($["${SKIPID}" = "*"]?ivr-3,s,1) |
| 19:44.51 | FXOL | that's what I have |
| 19:44.52 | ManxPower | also, you really should have a SPACE on either side of the = |
| 19:44.58 | FXOL | I do |
| 19:45.23 | ManxPower | FXOL: as the priority AFTER the read do a Noop(SKIPID is ${SKIPID}) |
| 19:45.32 | FXOL | it seems to skip that condition |
| 19:45.32 | FXOL | <PROTECTED> |
| 19:45.32 | FXOL | <PROTECTED> |
| 19:45.32 | FXOL | <PROTECTED> |
| 19:45.39 | ManxPower | it should show you SKIPID is * |
| 19:45.52 | FXOL | not "user entered" ? |
| 19:46.20 | ManxPower | the -- lines are what Asterisk generates. |
| 19:46.23 | _x86_ | ManxPower: http://pastebin.ca/703723 |
| 19:46.35 | FXOL | right.. that's what is in the log |
| 19:46.40 | FXOL | so it see I'm hitting * |
| 19:46.43 | ManxPower | just remember if you don't want to be fed to the aligators, use pastebin.ca for pasts longer than a line or two. |
| 19:46.56 | _x86_ | ManxPower: fwiw, SIP/7796 is the extension always getting slammed when the PSTN T1 bounces |
| 19:46.59 | wishes | mmm aligators |
| 19:47.14 | ManxPower | _x86_: and the CLI pastebin of this happening? |
| 19:47.22 | _x86_ | ManxPower: 7796 is also the only extension in the inbound [receptionist] context |
| 19:48.04 | ManxPower | I also don't see the zaptel.conf |
| 19:48.16 | FXOL | (ManxPower): see a problem with what I have there? |
| 19:48.24 | _x86_ | ManxPower: Zap/25-49 dialing SIP/7796, 7796 is a polycom 501, and it can handle about 6 calls at a time, the rest all trigger voicemail, and end up leaving 4 second messages |
| 19:48.41 | _x86_ | ManxPower: right now, 7796 has 916 messages (the PSTN T1 has been bouncing all day) |
| 19:48.51 | ManxPower | _x86_: a 4-second SILENT message? |
| 19:49.06 | _x86_ | ManxPower: hmmm... no... dialtone |
| 19:49.12 | _x86_ | forgot to mention that part |
| 19:49.20 | _x86_ | 4 seconds of dialtone per vm |
| 19:49.23 | ManxPower | _x86_: You are forgetting to mention a lot of stuff. |
| 19:49.32 | ManxPower | what is the signaling for the PSTN T-1 channels? |
| 19:50.43 | *** join/#asterisk elriah (n=null@adsl-070-148-053-190.sip.bhm.bellsouth.net) |
| 19:50.53 | ManxPower | _x86_: when we had a CAS T-1, if the line bounced, it should show up as incoming calls on all the channels. |
| 19:51.06 | elriah | Hi all. Is it possible to SET the DNID variable in asterisk 1.2? |
| 19:51.23 | ManxPower | elriah: I doubt it, but try it and see. |
| 19:51.36 | elriah | I did, no luck. I figured as much, just thought I would ask. |
| 19:51.42 | ManxPower | DNID is generally the same as EXTEN unless there was a call forward or something like that. |
| 19:52.33 | elriah | hrm.. |
| 19:52.36 | elriah | Thanks, ManxPower. |
| 19:53.20 | *** join/#asterisk bkruse (i=bkruse@nat/digium/x-36c5ec67c1c6f582) |
| 19:54.34 | *** join/#asterisk atomicd (n=AtomicDa@74-206-0-80.static-ip.m.telepacific.net) |
| 19:56.25 | FXOL | (ManxPower): see a problem with what I had there? ... or am I annoying the crap outta you already? :P |
| 19:57.09 | ManxPower | FXOL: without seeing the that part of the dialplan on pastebin...... |
| 19:57.18 | FXOL | one sec |
| 19:57.58 | ManxPower | and make it fast, I have some outside cable plant work to finish before I leave for New Orleans |
| 19:58.46 | FXOL | (ManxPower): http://www.pastebin.ca/703740 |
| 19:58.53 | atomicd | Stupid question: When you turn on sip debugging, does it save the information to a file? If so, where? If not, how can you make it save to a file. |
| 19:59.19 | FXOL | (ManxPower): damnit.. screwd that up |
| 19:59.20 | FXOL | once sec |
| 19:59.21 | ManxPower | Well for one thing you forgot to put exten => whatver infront of the goto |
| 19:59.33 | FXOL | yea, so I did :P |
| 19:59.38 | ManxPower | you need to COPY AND PASTE |
| 20:00.17 | FXOL | lol |
| 20:00.18 | ManxPower | FXOL: you realise that you are going to hell for using a GUI, right? |
| 20:00.20 | FXOL | that's what I was missing :P |
| 20:00.28 | FXOL | lol |
| 20:00.31 | FXOL | you mean for the IVR? |
| 20:00.50 | ManxPower | no, I mean the _additional.conf stuff is from GUIs that were installed. |
| 20:00.58 | FXOL | ah... yes |
| 20:00.59 | FXOL | and? :P |
| 20:01.05 | ManxPower | We don't support GUIs here. |
| 20:01.09 | FXOL | that's fine |
| 20:01.23 | FXOL | I'm not modifying them... just wanted to redirect to one |
| 20:01.29 | ManxPower | We also don't support their config files. |
| 20:01.32 | FXOL | lol |
| 20:01.51 | FXOL | okay.. I'll move the IVR into my .conf file.. then you will "support" it? :P |
| 20:02.06 | FXOL | I'll probably end up doing that anyhow |
| 20:02.10 | FXOL | and not use GU |
| 20:02.12 | FXOL | I |
| 20:02.13 | *** part/#asterisk deeperror (n=deeperro@69-215-202-195.ded.ameritech.net) |
| 20:02.39 | ManxPower | no, into extensions.conf |
| 20:02.52 | FXOL | yea |
| 20:02.54 | ManxPower | anyway, you see the error |
| 20:03.03 | FXOL | yea, thanks... working now |
| 20:03.05 | *** join/#asterisk lunaphyte (n=lunaphyt@static-71-120-128-10.gdrpmi.dsl-w.verizon.net) |
| 20:08.17 | FXOL | what dir are custom recordings stored in? |
| 20:09.09 | *** join/#asterisk waKKu (n=eurulo@unaffiliated/wakku) |
| 20:10.38 | _Sam-- | ManxPower: would you be willing to fix my zaptel for pay? i'll pay 150 bucks to get it straight. |
| 20:10.41 | *** join/#asterisk KpoH (n=AID@host-89-41-66-8.moldtelecom.md) |
| 20:10.46 | _Sam-- | ive run out of knowledge, and time. |
| 20:11.02 | _Sam-- | important conference call is coming up. |
| 20:11.30 | *** join/#asterisk Boones (n=bytewalk@port-212-202-170-97.dynamic.qsc.de) |
| 20:12.06 | _Sam-- | at this point i think my biggest hurdle is the lack of specific knowledge. zaptel seems to compile fine. but i cant modprobe zaptel. |
| 20:12.13 | _Sam-- | and make install seems to work fine. |
| 20:12.17 | *** join/#asterisk h0 (n=Fakhir@unaffiliated/fakhir) |
| 20:12.23 | Echinos | < ManxPower> and make it fast, I have some outside cable plant work to finish before I leave for New Orleans |
| 20:12.27 | Echinos | he might be busy... |
| 20:12.29 | *** join/#asterisk weahzal (n=jeremy@adsl-76-230-116-17.dsl.ksc2mo.sbcglobal.net) |
| 20:14.01 | weahzal | anyone have any sugestions on why i can never get port1 to ring on an ata. even if port1 is the only one setup. 1 can dial out, just wont ring. port2 works normal. same problem on handytones and cisco 186 |
| 20:14.51 | Qwell | _Sam--: Do you have a Digium card? |
| 20:15.56 | _Sam-- | Qwell: yes |
| 20:15.56 | Qwell | save your money - call support |
| 20:15.56 | _Sam-- | time is worth more than money at this point. |
| 20:15.56 | Qwell | We offer free installation support. |
| 20:15.56 | Qwell | this would clearly be covered by that :) |
| 20:17.57 | _Sam-- | someone was nice enough to offer a helping hand, and i'll see how that works out. thank you for giving me that option. |
| 20:18.16 | *** join/#asterisk ZX81 (n=matt@202.20.97.211) |
| 20:21.01 | file | my extension is around in the documentation of Asterisk if you look... |
| 20:21.18 | Qwell | speechrec.txt! |
| 20:21.22 | Qwell | (pwnt) |
| 20:21.45 | perd | for an end user who is not computer savvy but would like the option to be able to create voicemailboxes and possibly simple menus/recordings, would you recommend trixbox, asterisknow or freepbx, or just whip up a few simple php scripts to handle that portion? |
| 20:22.09 | ZX81 | and allow=ulaw:30 doesn't seem to work |
| 20:22.34 | ZX81 | sounds like its reading frames at the wrong speed |
| 20:23.37 | ZX81 | I really want the digium card to work - but if it doesn't by the end of the day I'll have to ditch it and go with an Eicon Diva Server |
| 20:24.13 | *** join/#asterisk UCFmethod (n=UCFmetho@office.eyestreet.com) |
| 20:24.19 | ZX81 | jitterbuffer on either end doesn't seem to help |
| 20:25.16 | ZX81 | perd - I personally would whip up php scripts - but AsteriskNow looks to be making good progress |
| 20:25.36 | perd | yeah, i'm looking at trixbox right now.. man it has some pretty cool options |
| 20:25.38 | *** join/#asterisk jsmith (n=jsmith@000-190-367.area3.spcsdns.net) |
| 20:25.38 | *** mode/#asterisk [+o jsmith] by ChanServ |
| 20:25.42 | perd | the reporting and stuff looks really slick |
| 20:25.49 | ZX81 | perd: yeah but thats all third party |
| 20:25.57 | ZX81 | i.e. cdr is areski cdr stats |
| 20:26.02 | perd | ahh |
| 20:26.08 | ZX81 | operator panel is asternic flash operator panel |
| 20:28.00 | ZX81 | here goes nothing |
| 20:28.20 | perd | good luck sir |
| 20:28.27 | ZX81 | :) ty |
| 20:28.47 | *** join/#asterisk |omni| (n=rob@c-67-185-56-106.hsd1.wa.comcast.net) |
| 20:28.49 | *** join/#asterisk twisted (n=twisted@pdpc/supporter/active/twisted) |
| 20:28.49 | *** mode/#asterisk [+o twisted] by ChanServ |
| 20:28.56 | ZX81 | grrr - please press 1, please press 1 |
| 20:29.06 | ZX81 | dtmf not happening - PRI -> IAX -> Digium |
| 20:29.22 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 20:29.39 | ZX81 | yay |
| 20:29.50 | *** join/#asterisk twisted (n=root@pdpc/supporter/active/twisted) |
| 20:29.50 | *** mode/#asterisk [+o twisted] by ChanServ |
| 20:29.52 | twisted | hahah |
| 20:30.11 | ZX81 | getting there |
| 20:30.13 | ZX81 | :) |
| 20:30.13 | *** join/#asterisk Pazzo (n=ugelt@reserved-225136.rol.raiffeisen.net) |
| 20:31.09 | ZX81 | I liked the pre freeplay music better |
| 20:32.02 | _Sam-- | damn, TK-Fender left before i could show him he fixed it |
| 20:32.02 | _Sam-- | [CC] app_meetme.c -> app_meetme.o |
| 20:32.09 | ZX81 | yay got through |
| 20:32.15 | _Sam-- | he fixed me up like a million bux. |
| 20:32.24 | _x86_ | ManxPower: so what do you think? |
| 20:35.44 | *** join/#asterisk [hC] (n=hardcore@76.77.69.66) |
| 20:36.05 | *** join/#asterisk tsurko (n=tsurko@78.90.100.214) |
| 20:36.45 | riddlebox | is there a way to tell a tdm card to detect when someone hangs up quicker? |
| 20:37.25 | lirakis | night everyone |
| 20:37.28 | *** part/#asterisk lirakis (n=lirakis@65.200.191.253) |
| 20:38.08 | twisted | yay |
| 20:38.15 | twisted | now I get to integrate asterisk with a DMS10. |
| 20:38.33 | Qwell | twisted: it's...you |
| 20:38.37 | perd | yeah im about to attempt one with a callmanager 3.3 system |
| 20:38.44 | file | wow... it's twisted |
| 20:39.03 | Qwell | twisted: I'm probably gonna be bringing some family to Bumpers tomorrow night |
| 20:39.08 | twisted | perd: oh really? how so? |
| 20:39.24 | perd | so i can call to/from it |
| 20:39.27 | twisted | Qwell tomorrow night is double jeapordy league night.. won't be a good night for visitation with you guys, unforutnately :/ |
| 20:39.38 | twisted | perd: sounds sorta like what i'm going to be doing today |
| 20:39.44 | Qwell | that's cool, they wanted to check it out anyways :D |
| 20:39.50 | perd | ran out of licenses on the cm server, need something temporarily in place until i can switch over to asterisk all together |
| 20:39.54 | twisted | the dms10 is going to happen tomorrow prob. |
| 20:40.01 | *** join/#asterisk zx225 (n=joel@node49-146.ipglobal.net) |
| 20:40.21 | Qwell | twisted: any idea what time that greek place over there closes? |
| 20:40.51 | twisted | Qwell: dunno.. I want to say 6, but I know that's wrong... 8 maybe? |
| 20:40.58 | Qwell | ahh, alright |
| 20:41.34 | zx225 | anyone try the asterisk appliance |
| 20:41.37 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 20:43.09 | *** join/#asterisk clive- (n=pirch@dsl-242-174-09.telkomadsl.co.za) |
| 20:44.52 | bkruse | zx225: I have :D |
| 20:45.12 | zx225 | we are having a hard time from digium |
| 20:45.22 | zx225 | will not rma defectiitsve un |
| 20:45.23 | riddlebox | zx225, I hope I win one ;) |
| 20:45.51 | zx225 | i have 3 installed 1 of which is doa |
| 20:45.58 | Qwell | zx225: howso? |
| 20:46.03 | *** join/#asterisk jarrod (i=anon@eschatolo.gy) |
| 20:46.04 | *** join/#asterisk luisavila (n=luisavil@bl6-75-129.dsl.telepac.pt) |
| 20:46.24 | zx225 | having power issues |
| 20:46.25 | jarrod | man these asterisk appliances are the worst |
| 20:46.35 | jarrod | is anyone else having problems? |
| 20:46.56 | zx225 | it is hard to say but the support bites |
| 20:47.24 | jarrod | have you talked to that greg guy? |
| 20:47.27 | jarrod | what an idiot |
| 20:47.47 | Qwell | jsmith: ping |
| 20:47.48 | zx225 | i think he has been in the backwoods of alabama too long |
| 20:48.18 | *** join/#asterisk pfn (n=pfnguyen@hanhuy.com) |
| 20:49.01 | twisted | great |
| 20:49.26 | _x86_ | ManxPower: ? |
| 20:49.36 | twisted | i live in alabama |
| 20:49.48 | jarrod | ha.. |
| 20:49.50 | twisted | plz don't knock it until you try it. |
| 20:50.18 | jarrod | ive experienced it enough, kthx |
| 20:50.21 | _Sam-- | ManxPower : after i made the new zaptel, and configure and made a new asterisk, is all i really need the app_meetme.so? |
| 20:50.38 | _Sam-- | or do i need the new asterisk binary |
| 20:50.51 | _Sam-- | ive run into this error: http://www.pastebin.ca/703812 |
| 20:50.54 | twisted | heh |
| 20:51.22 | Qwell | _Sam--: You'll also need chan_zap |
| 20:51.25 | Corydon76-dig | <twisted> plz don't knock it until you try it. <-- I think I've said that to you, before. ;-) |
| 20:51.31 | riddlebox | is there a way to tell a tdm card to detect when someone hangs up quicker? |
| 20:51.46 | _Sam-- | it didnt make chan_zap |
| 20:51.48 | jarrod | i would stick with the free software and not use any of the digium business class crap |
| 20:52.09 | _Sam-- | it did make the app_meetme.so though. |
| 20:52.13 | zx225 | i am sorry about alabama... i know it is a grest state |
| 20:52.22 | zx225 | but the guy at digium is bonehead |
| 20:52.24 | twisted | Corydon76-dig.... i knew it wouldn't be long before you popped in |
| 20:53.05 | Corydon76-dig | twisted: just got back from the bank... ;-) |
| 20:53.52 | *** join/#asterisk apardo (n=apardo@105.37.221.87.dynamic.jazztel.es) |
| 20:54.45 | twisted | bits or gtfo |
| 20:55.08 | *** join/#asterisk apardo (n=apardo@105.37.221.87.dynamic.jazztel.es) |
| 20:55.48 | putnopvut | How you been twisted? |
| 20:56.11 | jsmith | Qwell: Pong (latency=high) |
| 20:56.22 | twisted | pretty good MM, bout you/ |
| 20:56.28 | twisted | s/\//? |
| 20:56.28 | Qwell | jsmith: hold that thought |
| 20:56.38 | putnopvut | Doing good myself. |
| 20:56.42 | _Sam-- | Qwell: all i need to fix it is chan_zap? |
| 20:56.43 | putnopvut | Headed to Astricon? |
| 20:56.46 | twisted | nope |
| 20:56.50 | putnopvut | Aw |
| 20:56.51 | jsmith | twisted: Why not? |
| 20:56.51 | twisted | i don't get to go anymore |
| 20:56.56 | twisted | only the sales people. |
| 20:57.01 | putnopvut | Lame |
| 20:57.05 | _Sam-- | i just made a new asterisk, and i saw it make chan_zap.so |
| 20:57.07 | twisted | yeah. i'm still miffed. |
| 20:58.22 | russellb | twisted: !! |
| 20:58.32 | Juggie | ahh boo.. |
| 20:58.37 | rob0 | Christian is a cheapskate. ;) |
| 20:58.57 | Qwell | jsmith: see msg (if you haven't already) |
| 20:59.02 | twisted | russellb!!! |
| 20:59.15 | twisted | is that peckerhead? |
| 20:59.31 | *** join/#asterisk apardo (n=apardo@105.37.221.87.dynamic.jazztel.es) |
| 21:00.05 | jsmith | jarrod: Got a second? |
| 21:01.09 | *** join/#asterisk apardo (n=apardo@105.37.221.87.dynamic.jazztel.es) |
| 21:01.41 | bkruse | twisted: bowling? |
| 21:02.18 | twisted | when? |
| 21:02.29 | putnopvut | I want to bowl too |
| 21:02.35 | bkruse | putnopvut: in hsv? |
| 21:02.35 | putnopvut | I'm inviting myself |
| 21:02.38 | putnopvut | I'm a leech |
| 21:02.41 | bkruse | twisted: im thinking like saturday |
| 21:02.42 | Qwell | I'll wii-bowl with you guys |
| 21:02.45 | bkruse | like the good ole days |
| 21:02.45 | putnopvut | bkruse, I'm Mark Michelson |
| 21:02.48 | bkruse | Qwell: nice |
| 21:02.49 | Qwell | real bowling is FTL |
| 21:02.54 | bkruse | putnopvut: well then in hsv :P |
| 21:02.58 | putnopvut | Yeah. |
| 21:03.00 | bkruse | Qwell: pssh |
| 21:03.03 | bkruse | ~punch Qwell |
| 21:03.04 | jbot | ACTION hits Qwell like the hot kiss and the end of a wet fist |
| 21:03.07 | putnopvut | Real bowling ownz |
| 21:03.09 | bkruse | who taught jbot that, geez... |
| 21:03.22 | bkruse | putnopvut: it does, were going, ima try to get a group, last time was hilarious |
| 21:03.50 | putnopvut | My bowl-foo is strong |
| 21:03.57 | putnopvut | And weird looking |
| 21:04.19 | bkruse | putnopvut: look forward to watching and laughing |
| 21:04.25 | bkruse | russellb went last time also |
| 21:04.37 | *** join/#asterisk apardo (n=apardo@105.37.221.87.dynamic.jazztel.es) |
| 21:04.41 | twisted | bkruse i'll prob be out of town |
| 21:04.48 | bkruse | twisted: :[ where? |
| 21:04.51 | twisted | auburn |
| 21:04.52 | bkruse | well, busy weekend for all |
| 21:04.58 | bkruse | i might try in a couple weeks, after *con |
| 21:05.04 | twisted | auburn vs. new mexico :) |
| 21:05.14 | twisted | if i'm in town, sure. |
| 21:05.35 | twisted | it'd be nice to go bowling again... it's been awhle. |
| 21:05.36 | putnopvut | twisted: Seeing if they'll make it 3 straight? |
| 21:05.38 | putnopvut | ;) |
| 21:05.53 | twisted | putnopvut: heh.. something like that :P |
| 21:06.20 | twisted | muahahaha |
| 21:06.24 | twisted | zaptel FINALLY built... |
| 21:06.30 | twisted | back to work for me for a bit |
| 21:07.51 | chemikk | hellou friend |
| 21:09.41 | *** join/#asterisk hmmhesays (n=Neg@24-119-176-74.cpe.cableone.net) |
| 21:09.58 | *** join/#asterisk apardo (n=apardo@105.37.221.87.dynamic.jazztel.es) |
| 21:11.17 | twisted | anyone know off the top of their head which version of openh323 the h323 channel drvier in 1.4.11 is based on? |
| 21:14.12 | *** join/#asterisk _ShrikE (n=ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 21:14.43 | *** join/#asterisk apardo (n=apardo@105.37.221.87.dynamic.jazztel.es) |
| 21:18.08 | MindTheGap | is there any way to do some sound recognition under asterisk? maybe spandsp? i need to hang up collect calls on a ISDN E1, for what I know zaptel cannot do this on its own, and out telco is not willing to block such calls |
| 21:18.17 | kink0 | anyone knows what is this error: !! Unknown IE 124 (cs5, Unknown Information Element) |
| 21:20.33 | perd | hrm, h.225 is supported in ooh323? |
| 21:22.23 | *** join/#asterisk Pids (i=Pids@122.sub-75-208-68.myvzw.com) |
| 21:23.10 | Pids | Anyone seen this before ? "NOTICE[12620]: chan_iax2.c:6521 socket_read: Out of idle IAX2 threads for I/O, pausing!" |
| 21:23.47 | Pids | repeats over and over. The asterisk server is not accepting calls while it happens. |
| 21:23.58 | tzafrir_laptop | MindTheGap, what sound is that exactly? |
| 21:24.44 | tzafrir_laptop | kink0, is that from Zaptel? From what connection? |
| 21:34.11 | *** join/#asterisk Avero (n=Avero@216.186.253.120) |
| 21:35.09 | *** join/#asterisk saftsack (n=saftsack@pD9E0445C.dip.t-dialin.net) |
| 21:37.53 | kink0 | tzafrir_laptop, yes, from zaptel. Is connected to E1's |
| 21:38.26 | *** join/#asterisk iCEBrkr (i=icebrkr@cyberdyne.org) |
| 21:38.27 | kink0 | span=1,1,0,ccs,hdb3,crc4 and so |
| 21:38.53 | twisted | great |
| 21:38.56 | twisted | ooh323 is also broken |
| 21:39.27 | *** part/#asterisk [T]ank (n=ckwall@206.71.78.172) |
| 21:39.57 | [hC] | Can I not do something like this with extens: _*67NXXNXXXXXX,1,NoOp _*67.,1,Authenticate(something) |
| 21:40.16 | [hC] | so that if the NXXNXXXXXX was dialed, it does nothing, but if any other pattern was, it authenticates? |
| 21:40.19 | tzafrir_laptop | Is there such an IE? |
| 21:40.19 | [hC] | it does not seem to work. |
| 21:40.40 | kink0 | tzafrir_laptop, I don't know |
| 21:43.31 | *** join/#asterisk kolian123 (n=kvirc@124.107.63.223) |
| 21:43.49 | kolian123 | Hello everybody |
| 21:44.27 | kolian123 | I have a talk off on generic Tormenta T1 card...RelaxDtmf set to no |
| 21:44.54 | riddlebox | is there a way to tell a tdm card to detect when someone hangs up quicker? |
| 21:45.07 | kolian123 | Can anyone point me to the code for DTMF receiver to change a variable to tighten a DTMF detection a bit |
| 21:45.18 | kolian123 | Is it possible |
| 21:45.37 | clive- | kolian change you features.conf to look for double ** |
| 21:45.51 | tzafrir_laptop | kolian123, main/dsp.c |
| 21:46.25 | kolian123 | tzafrir_laptop, thanks would you happen to know which variable can be adjusted up/down? |
| 21:46.40 | tzafrir_laptop | no. |
| 21:49.03 | kolian123 | tzafrir_laptop thanks for pointing this...seems like Steven Underwood is the author of original |
| 21:49.11 | kolian123 | I will try emailing him... |
| 21:49.46 | kolian123 | Russellb, hi are you around? |
| 21:50.45 | russellb | sort of .. |
| 21:50.54 | russellb | what's up |
| 21:51.03 | *** join/#asterisk holiday_42 (n=no@spike.wcta.net) |
| 21:51.04 | tzafrir_laptop | kolian123, Steve Underwood has developed that code even further, in spandsp |
| 21:51.28 | tzafrir_laptop | No point in trying to get that commited back into Asterisk |
| 21:51.32 | kolian123 | Russellb, was wondering if you know if there is a variable in dsp.c to tweak to tighten DTMF a bit |
| 21:51.38 | russellb | i have no idea |
| 21:51.58 | russellb | look at what relaxdtmf does and do the opposite, heh |
| 21:52.00 | kolian123 | Would you know who maintaining the code and can help? |
| 21:52.02 | tzafrir_laptop | There was a recent thread in asterisk-dev (?) about dtmf detection improvements |
| 21:52.11 | russellb | the best thing to do is email the asterisk-dev list |
| 21:52.17 | kolian123 | russellb, hehe good point i will try it out |
| 21:52.36 | kink0 | tzafrir_laptop, the problem with this IE have started today when I move E1's to other Telco. These E1's have been fine for long time with prior telco company |
| 21:52.57 | fujin_ | russellb: what's the url for func_devstate? |
| 21:52.59 | kolian123 | I think it just doing | relaxdtmf when passing this detection routing |
| 21:53.14 | russellb | fujin_: svncommunity.digium.com/svn/russell/func_devstate-1.4 i think |
| 21:53.25 | kolian123 | tzafrir_laptop, was there a thread let me check |
| 21:53.34 | fujin_ | ta dude |
| 21:53.38 | russellb | np |
| 21:53.48 | kolian123 | Thanks tzafrir! |
| 21:55.00 | kolian123 | a lot of threads where DTMF is not received...i guess i have an opposite problem:) |
| 21:55.12 | fujin_ | change your dtmf signaling |
| 21:56.42 | gremzoid | i have a weird DTMF problem regarding digital phones on a Siemens HiPath 3000 / HG 1500 to asterisk IAX extensions |
| 21:57.19 | gremzoid | i don't get any DTMF from phone to IAX but it works the other way around in the same call |
| 21:57.46 | gremzoid | IE i can send DTMF from the IAX phone to the digital one hanging off the hipath |
| 21:58.36 | gremzoid | yet it works in IVR menus... |
| 22:00.13 | *** join/#asterisk ltd (n=z@nox.amused.net) |
| 22:00.13 | gremzoid | the HG 1500 and asterisk are connected via a SIP trunk... (have also tried ooh323 as well to no avail) |
| 22:02.58 | *** join/#asterisk lindi- (n=lindi@kulho150.adsl.netsonic.fi) |
| 22:02.59 | *** join/#asterisk sandorp (n=sandor@firewall2.wsi.net) |
| 22:03.42 | sandorp | can someone point me to the docs that describe what buttons to push on a regular phone to have asterisk transfer a call, put someone on hold, etc? |
| 22:14.44 | *** part/#asterisk famicon (i=redz@c51447ddc.cable.wanadoo.nl) |
| 22:15.43 | wishes | anyone had the grandstream phones registering a button press 5 as a 2? |
| 22:16.22 | Qwell | wishes: wouldn't much surprise me |
| 22:16.32 | Nugget | that's a funny problem |
| 22:17.04 | wishes | just wondering if its hardware/software/firmware |
| 22:17.09 | Qwell | all of the above |
| 22:17.15 | Qwell | ~gs |
| 22:17.16 | jbot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
| 22:17.57 | Corydon76-dig | Ditto for Sucksco phones |
| 22:18.04 | Qwell | sucksco? |
| 22:18.07 | Corydon76-dig | Cisco |
| 22:18.12 | Qwell | clever |
| 22:18.14 | perd | she's a grand old suck she's a high flying suck? |
| 22:18.30 | denon | you know .. |
| 22:18.31 | kolian123 | there was a patch for dtmf code, it went to 1.4 but not into 1.2 |
| 22:18.32 | Corydon76-dig | Polycom, however, makes good phones |
| 22:18.35 | denon | people whineabout cisco |
| 22:18.37 | *** join/#asterisk marmsu (n=42cfdde2@207.250.49.24) |
| 22:18.43 | denon | but my 7960s work just fine |
| 22:18.43 | perd | cisco phones are junk |
| 22:18.48 | denon | and have worked great for years |
| 22:18.50 | Qwell | denon: the hardware is great |
| 22:18.51 | denon | tons of em |
| 22:18.54 | perd | you cant even program the buttons to do cool shit |
| 22:18.56 | *** join/#asterisk anthm (n=anthm@mb50736d0.tmodns.net) |
| 22:18.56 | *** mode/#asterisk [+o anthm] by ChanServ |
| 22:18.57 | Qwell | the sip software however...not so much |
| 22:19.00 | Qwell | erm, firmware |
| 22:19.01 | wishes | yeah well we are a cheap company |
| 22:19.05 | Qwell | skinny firmware rocks |
| 22:19.06 | Qwell | ~cheap |
| 22:19.07 | jbot | it has been said that cheap is a bad idea. If you're setting up an Asterisk system, one of the wisest pieces of advice you can take is this: DON'T BE A CHEAPSKATE. |
| 22:19.10 | wishes | grandstream was what bought |
| 22:19.17 | denon | perd: just because you can't make the buttons do "cool shit", doesn't make it a bad phone |
| 22:19.18 | wishes | i cant change it now |
| 22:19.22 | Qwell | wishes: there are other inexpensive phones that don't suck |
| 22:19.25 | perd | oh yes it does, sir. |
| 22:19.26 | wishes | like? |
| 22:19.30 | Corydon76-dig | wishes: you can get Polycom phones for the same price as GS phones |
| 22:19.30 | wishes | and can i get them in NZ ? |
| 22:19.32 | Qwell | polycom 320/330 |
| 22:19.33 | marmsu | is it possible with Asterisk to initiate a 2-way call between 2 parties? |
| 22:19.35 | denon | perd: no, it makes it a less flexible phone .. |
| 22:19.41 | Corydon76-dig | wishes: so that's no excuse |
| 22:19.54 | wishes | it is when i have to convince the powers that be to fork out for it |
| 22:19.55 | denon | but sound quality, for example, is excellent, and they go for months and months never needing to be reboot |
| 22:19.55 | Qwell | marmsu: rephrase the question? |
| 22:19.56 | perd | if i cdont have complete control over the soft buttons ... i call that horrible. |
| 22:20.01 | denon | whih makes it robust |
| 22:20.10 | Qwell | denon: you can thank polycom for the audio quality |
| 22:20.11 | marmsu | Qwell: I need to somehow trigger Asterisk to connect two numbers |
| 22:20.15 | denon | Qwell: yes, I know .. |
| 22:20.20 | Qwell | marmsu: trigger how? |
| 22:20.20 | denon | Qwell: and polycom can thank TI |
| 22:20.25 | Qwell | denon: heh |
| 22:20.39 | denon | and TI can thank a good silicon supplier, and probably indian devs |
| 22:20.49 | denon | point is, the 7960 on my desk is made by cisco, and it works well |
| 22:20.53 | marmsu | Qwell: I believe you can open a connection to asterisk and pipe it commands, no? |
| 22:21.05 | Qwell | marmsu: sure, via the manager interface |
| 22:21.06 | Corydon76-dig | Cisco made good phones when they allowed Polycom to write their firmware for them |
| 22:21.08 | marmsu | exactly |
| 22:21.10 | perd | i have a7960 on my desk that works well too.. but i cant program the soft buttons to do whatever i want |
| 22:21.13 | mcab | denon: TI is doing DSP code now? :-) |
| 22:21.18 | Qwell | Corydon76-dig: polycom used to write cisco firmware? |
| 22:21.30 | perd | and there's no templating engine so i cant change the layout or anything but the stupid 200x200 graphic |
| 22:21.32 | perd | that's just lame. |
| 22:21.36 | Corydon76-dig | Qwell: the original Cisco phones were Polycom rebrands |
| 22:21.37 | Qwell | perd: You could if it was running skinny |
| 22:21.38 | marmsu | Qwell: so, I'm wondering if I can send it the commands required to connect two people and sit in the middle .. |
| 22:21.41 | Qwell | Corydon76-dig: which? |
| 22:21.43 | Qwell | like 7910? |
| 22:21.51 | Corydon76-dig | Qwell: something like that |
| 22:21.53 | Qwell | (and obviously 7935) |
| 22:21.54 | mcab | Corydon76-dig: not really - they used Polycom DSP, but the s/w was Cisco |
| 22:21.57 | perd | well it's on CCM at the moment, but even with skinny you cant change the soft button functionality to my knowledge |
| 22:22.08 | perd | am i completely friggen wrong here? |
| 22:22.10 | Qwell | perd: yes |
| 22:22.19 | Qwell | asterisk controls the softkeys in skinny |
| 22:22.22 | perd | can you make them say something other than 'new call', 'redial' 'cfwdall' |
| 22:22.26 | Qwell | yep |
| 22:22.28 | perd | no shit... |
| 22:22.33 | Qwell | you have to code, but yeah |
| 22:22.37 | perd | i gotta get asterisk running in here |
| 22:22.46 | Corydon76-dig | Skinny is a completely different protocol, though |
| 22:22.51 | Qwell | of course, there is still a bit of functionality that it's lacking |
| 22:23.10 | perd | gotta give a little to get a little? |
| 22:23.22 | Corydon76-dig | Skinny phones are essentially implemented as dumb terminals |
| 22:23.30 | Qwell | and that's why they rock so hard |
| 22:23.43 | denon | perd: so it turns out the phones are fine, it's just your knowledge that was lacking |
| 22:23.58 | perd | i never said i'm not ignorant. |
| 22:23.58 | Corydon76-dig | All of the intelligence is in the core switch, not in the phones |
| 22:24.26 | marmsu | I suppose, to rephrase my question, is it possible to use the Manager API to initiate a call between 2 parties? |
| 22:24.39 | Qwell | marmsu: yes, via the originate action |
| 22:24.43 | Deeewayne | ~thebook |
| 22:24.43 | jbot | i guess thebook is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 22:24.52 | Qwell | ~book |
| 22:24.53 | jbot | [book] Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 or Temporarily at http://www.aocomputing.net/AsteriskTFOT.pdf |
| 22:24.56 | Qwell | has a mirror |
| 22:25.04 | denon | Only two things are infinite, the universe and human stupidity, and I'm not sure about the former. |
| 22:25.49 | Deeewayne | can the universe be infinite if it is always expanding ? |
| 22:26.03 | marmsu | Qwell: awesome, thanks for the tips. |
| 22:26.08 | marmsu | I appreciate it. |
| 22:26.13 | Corydon76-dig | Deeewayne: infinite means uncountable... |
| 22:26.19 | putnopvut | Deeewayne: isn't that what infinite means? |
| 22:26.22 | wishes | you guys lie, grandstream are $155 NZ and polycom are $455 |
| 22:26.39 | Qwell | wishes: which polycom? There are many different models |
| 22:26.42 | Qwell | look at the 320/330 |
| 22:26.44 | Corydon76-dig | wishes: Try the Polycom 320, not the 650 |
| 22:26.48 | Deeewayne | ok...I was thinking that if you somehow reached the edge of the universe, it would not be infinite |
| 22:26.50 | wishes | Polycom SoundPoint IP301 was all i could find |
| 22:27.05 | Corydon76-dig | You're comparing apples and oranges, even then |
| 22:27.05 | Qwell | 301 is pretty much useless now |
| 22:27.55 | wishes | Corydon76-dig: ahh k |
| 22:28.02 | Corydon76-dig | The 320 has 2 line appearances, while the GS only has a single line appearance |
| 22:28.23 | Corydon76-dig | 650 has 6 line appearances and wideband codec support |
| 22:29.31 | wishes | hmm cant see them for sale here |
| 22:29.44 | Corydon76-dig | http://www.google.com/products?q=polycom+320 |
| 22:29.53 | wishes | yeah i can find them, just not in NZ |
| 22:29.56 | wishes | how much are they USD? |
| 22:30.10 | Corydon76-dig | $85US |
| 22:30.25 | wishes | thats not to bad |
| 22:30.36 | Corydon76-dig | Same price as GS phones |
| 22:30.42 | syzygyBSD | :) I'll be going to NZ in a little over a month |
| 22:31.32 | twisted | yaaaaaar |
| 22:31.42 | Nugget | yay NZ |
| 22:31.48 | twisted | ubuntu, rather. |
| 22:32.06 | Nugget | http://macnugget.org/photos/nz2007/eunos_chch <-- NZ |
| 22:32.27 | Corydon76-dig | twisted: butt pirates, unite? |
| 22:34.59 | wishes | NZ is a cool place to be |
| 22:35.06 | Nugget | Indeed |
| 22:35.09 | wishes | except for the lack of cool imports :D |
| 22:36.24 | wishes | we have no major deserts, no extreme weather (apart from the odd occasion) tons of bush, no major poisonous spiders or snake .. in fact ive never seen a snake at all outside a zoo, no wild animals other than birds and pigs (never seen a wild pig either) |
| 22:37.01 | wishes | the most you have to worry about is the crap bandwidth which is improving, and otara (the south side where all the bums druggies and crims hang out) |
| 22:37.13 | Nugget | Plus lots of meat pies and the best driving roads in the known universe. |
| 22:37.18 | Nugget | what's not to love? |
| 22:37.21 | wishes | meat pies? |
| 22:37.31 | wishes | theres lots of meat pies?! |
| 22:37.35 | Nugget | indeed |
| 22:37.43 | wishes | haha whos been telling you that :) |
| 22:37.49 | Nugget | they're everywhere. |
| 22:37.57 | wishes | no more than most things |
| 22:38.00 | Nugget | I ate a kazillion of them when I was there. |
| 22:38.17 | wishes | and the roads are so-so. nice views |
| 22:38.27 | wishes | i love Big Ben pies, they are soo yum :O~~~ |
| 22:38.34 | mcab | NZ is awesome |
| 22:38.53 | Nugget | I think you are underestimating the abject meat pie shortage that exists elsewhere on the planet. :) |
| 22:39.18 | RypPn | I think you've never been to scotland then ;) |
| 22:39.42 | wishes | Nugget: possibly |
| 22:39.45 | Nugget | I have, but I stayed drunk on Guinness the whole time and had no appetite for pies. |
| 22:39.45 | wishes | meat is good |
| 22:41.41 | *** join/#asterisk BigMac (n=mike@c-71-234-95-131.hsd1.ct.comcast.net) |
| 22:42.08 | BigMac | Hey, how can I tell if my telephone is supported |
| 22:42.19 | BigMac | or how does it work exactly |
| 22:43.15 | *** join/#asterisk pepse (n=pepse@71-223-121-15.phnx.qwest.net) |
| 22:43.25 | Deeewayne | BigMac: did you try using it ? |
| 22:43.35 | pepse | is that brian igmac? |
| 22:43.43 | pepse | oh, probably not |
| 22:43.55 | BigMac | Deeewayne: I am not sure on how to use it exactly is the problem |
| 22:43.59 | BigMac | like ho does it work |
| 22:44.05 | BigMac | is it some sort of firmware |
| 22:44.09 | pepse | anyway, hey guys. how would i keep an extension from dialing parts of a dialplan? |
| 22:44.15 | pepse | (while allowing other extensions) |
| 22:44.29 | Nugget | pepse: contain it in a safe context. |
| 22:44.31 | Deeewayne | BigMac: what type of phone is it? |
| 22:45.03 | BigMac | Deeewayne: Let me check |
| 22:45.36 | pepse | Nugget: got an example or can you elaborate? |
| 22:45.44 | *** join/#asterisk PepOSX (n=pepOSX@190.72.148.251) |
| 22:46.03 | Nugget | a peer can only dial extensions that exist in the context you put it in. |
| 22:46.22 | Nugget | so put the peer inside a context that only contains (or includes) the extensions you want it to be able to dial |
| 22:46.39 | Nugget | the example extensions.conf has examples of that |
| 22:46.50 | BigMac | Deeewayne: Uniden |
| 22:46.50 | pepse | i see, so my main dialplan is in the general context? |
| 22:47.11 | pepse | err default |
| 22:47.11 | Nugget | pepse: it's however you set it up |
| 22:47.20 | Nugget | yes, probably default |
| 22:47.31 | pepse | hah, duh :) |
| 22:47.44 | pepse | thanks, dunno why i needed to ask |
| 22:47.57 | *** join/#asterisk |dennis| (n=dennis@200.32.236.10) |
| 22:48.39 | *** join/#asterisk pepse (n=pepse@71-223-121-15.phnx.qwest.net) |
| 22:48.42 | pepse | silly fingers. |
| 22:49.03 | *** part/#asterisk clive- (n=pirch@dsl-242-174-09.telkomadsl.co.za) |
| 22:49.44 | *** join/#asterisk Kirko (n=kirkalle@dsl093-224-026.slc1.dsl.speakeasy.net) |
| 22:50.08 | Kirko | I have a question about automatic voicmail detector (AMD).. when i use it my asterisk server crashes... |
| 22:50.33 | Kirko | anyone know why that might happen? |
| 22:50.57 | BigMac | Deeewayne: will it work |
| 22:51.17 | pepse | i wonder if getting latest svn/cvs/whatever will stop these annoying "WARNING: chan_sip.c:11708 handle_response_register: Got 200 OK on REGISTER that isn't a register" |
| 22:52.02 | Deeewayne | probably, but I've never used one. A quick google search of "uniden asterisk" has a lot of hits. Did you try configuring /etc/asterisk/sip.conf ? |
| 22:53.38 | pepse | oh i guess it's my stupid ata's fault. |
| 22:59.27 | BigMac | Deeewayne: I haven't installed it yet, I am trying to figure out if it will work first |
| 22:59.57 | *** join/#asterisk Strom_M (n=strom@208.127.172.112) |
| 23:00.55 | BigMac | Are there any guides for setting it up and what I should buy and such |
| 23:01.06 | BigMac | I saw sme video, but the guy talked way to fast |
| 23:01.37 | BigMac | some |
| 23:08.11 | wishes | is there a way to flush the cache of which ips to which users? |
| 23:08.41 | wishes | users/ips |
| 23:10.53 | Kirko | anyone have any experence with app_amd ? |
| 23:15.57 | davevg-btwtech | Kirko: I may be able to help, whats up? |
| 23:22.19 | wishes | arg, i changed the username on a sip phone but asterisk has cached it, now when it logs in it registeres 2 different people logging in on the same ip and sends user x calls to user y |
| 23:23.59 | *** join/#asterisk jm|laptop (n=jm|home@zen.jamiem.com) |
| 23:24.05 | *** join/#asterisk dijungal (n=kdaniel@208.0.231.76) |
| 23:24.45 | saftsack | with which settings diff files are created with diff? |
| 23:25.12 | *** join/#asterisk mindCrime (n=chatzill@cpe-065-190-188-124.nc.res.rr.com) |
| 23:25.28 | russellb | saftsack: well, "svn diff" can be used and uses good default settings |
| 23:25.36 | *** join/#asterisk Yourname`` (n=chatzill@unaffiliated/yourname/x-837320) |
| 23:25.36 | russellb | at a minimum, use "-u" |
| 23:25.43 | russellb | if using diff directly |
| 23:25.52 | dijungal | is there anyway in the Asterisk CLI to monitor only one agent? |
| 23:26.22 | Yourname`` | I'm running a dialing application at 300 channels on a box. What should I be looking at on the server to see if it's overloading the box and if channels need to be decreased? Like top or something? |
| 23:26.36 | kink0 | I get sometimes: Unknown IE 124 , even I am not ussing IAX2, I saw this error is in source code in iax2, any idea ? |
| 23:26.49 | saftsack | thx worked |
| 23:27.10 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 23:27.51 | dijungal | and i isolate all that text in the CLI to only one sip channel? |
| 23:31.00 | [TK]D-Fender | dijungal, you can enable sip deub for a single channel, but the rest of verbose output, no. |
| 23:31.39 | dijungal | k |
| 23:31.41 | *** join/#asterisk outtolunc (n=me@c-67-174-216-60.hsd1.ca.comcast.net) |
| 23:35.45 | dijungal | k |
| 23:38.51 | dijungal | any good IAX softphone recommendation? |
| 23:38.57 | denon | idefisk |
| 23:39.06 | dijungal | i'm using zoiper right now and it has too much issues |
| 23:39.09 | denon | or whatever it's called now |
| 23:39.21 | dijungal | i can't get the audio working on linux (ubuntu) |
| 23:39.29 | *** join/#asterisk blinky42 (n=me@c-71-230-47-244.hsd1.pa.comcast.net) |
| 23:39.32 | denon | oh, I dunno - Ive just used it on win32 |
| 23:39.34 | denon | works great |
| 23:39.42 | dijungal | every call that comes through to the agent that's using it drops |
| 23:39.45 | drwelby | idefisk = zopier |
| 23:39.49 | *** join/#asterisk |omni| (n=rob@c-67-185-56-106.hsd1.wa.comcast.net) |
| 23:39.50 | denon | yeah, I know |
| 23:40.21 | denon | drwelby: and that's "idefisk == zopier", you're in C land :) |
| 23:40.47 | PepOSX | http://190.72.148.251:8000/listen.m3u |
| 23:40.50 | PepOSX | :S |
| 23:40.51 | PepOSX | sorry |
| 23:41.42 | dijungal | any other softphones? |
| 23:41.43 | drwelby | if idefisk = zoiper then (you already have it). Ok? |
| 23:41.56 | dijungal | yes i've been using zoiper |
| 23:43.15 | *** part/#asterisk Siya (n=djerk@217-195-248-251.dsl.easynet.nl) |
| 23:44.04 | drwelby | How about Kiax? |
| 23:45.02 | *** join/#asterisk Keltus (i=Keltus@about/cooking/nakedchef/beefstew/Keltus) |
| 23:51.05 | *** join/#asterisk slakware (n=slak@201.53.76.85) |
| 23:52.04 | dijungal | how so i enable debugging on one iax channel? |
| 23:52.45 | slakware | has anyone used the g726 codec? I've tried it and the sound quality is horrible, running SVN-branch-1.4-r81832. ulaw is wonderfull. i understand g726 is half the bit rate of ulaw, however the quality is horrible. i can hardly make out the voicemail prompts... |
| 23:53.00 | fujin_ | so use ulaw? |
| 23:53.09 | JT | slakware: lan? |
| 23:53.12 | Yourname`` | There was a document that said how to use sox for asterisk.. can someone point me to it please? |
| 23:53.35 | fujin_ | sox for what in particular? |
| 23:53.47 | slakware | its over the internet. when i set to ulaw all is great. i'm using a linksys WRTP54G |
| 23:54.07 | JT | slakware: packet loss, jitter? |
| 23:54.13 | slakware | ulaw is great, however i'd like g726 because of bw |
| 23:54.41 | JT | G.726 is a poor choice over poor bandwidth |
| 23:54.45 | fujin_ | It's worrying to me that you don't have the spare bandwidth for ulaw |
| 23:54.48 | slakware | i've tried setting a jitter buffer, and all. i dont think its packet loss, decent ping rates... 120ms |
| 23:54.53 | gremzoid | gsm? |
| 23:55.07 | slakware | the linksys i have doesnt support gsm |
| 23:55.23 | gremzoid | bugger... it works well for low bandwidth |
| 23:55.39 | slakware | the quality sounds really noisey. however, again ulaq is crystal clear |
| 23:55.45 | gremzoid | not to great audio quality, but at least it dosn't jitter |
| 23:56.21 | slakware | its not that the bw is poor. its that i'm conserving bw, its 1,000mbps upload |
| 23:56.35 | JT | 120ms is hardly "decent" |
| 23:56.46 | dijungal | is there anyway to set iax debugging on one iax channels? |
| 23:56.49 | JT | G.726 requires very consistant network connectivity |
| 23:56.49 | dijungal | channel |
| 23:56.52 | slakware | 120ms is decent enough for a crystal clean ulaw connection. so thats decent |
| 23:57.06 | *** join/#asterisk AJaymn (i=AJaymn@71-82-218-158.dhcp.mdsn.wi.charter.com) |
| 23:57.09 | JT | G.726 is a delta codec |
| 23:57.20 | JT | 120ms isn't decent though |
| 23:57.24 | JT | "mildly ok" |
| 23:57.32 | JT | that's not the issue anyway |
| 23:59.17 | waltj | Is there any way for an AGI script to find out (without unparking it) whether a parked call still exists or hung up? |
