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| 00:02.40 | *** mode/#asterisk [+o jtodd] by ChanServ |
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| 00:28.23 | Wired_Life | how i enable sip acc monitoring? i can see unmonitored in sip show peers |
| 00:39.58 | k-man | is there a way to get asterisk to map a phone number to a name and put that in the email when it sends you an email of a voicemail message? |
| 00:47.50 | rob0 | um, voicemail.conf has names, where have you looked? |
| 00:48.27 | lanning | you can load mappings into astdb and modify the caller ID name using astdb lookups. |
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| 01:05.13 | phix | :D |
| 01:05.28 | phix | should I use .ael files? |
| 01:06.19 | phix | eg, extensions.ael instead of / as well as extensions.conf? |
| 01:06.29 | phix | what is the point of two different sytanx conf files/ |
| 01:10.59 | *** join/#asterisk knarfly (n=vtserije@c-75-74-113-9.hsd1.fl.comcast.net) |
| 01:12.21 | knarfly | just switched to Linux and *-1.6.0.9 with dahdi instead of zaptel...having touble with getting the X101P to dialout |
| 01:12.36 | knarfly | dahdi_scan shows the card is there with no alarms |
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| 01:14.40 | knarfly | Unable to create channel of type 'DAHDI' (cause 0 - Unknown) |
| 01:14.47 | knarfly | any advice out there? |
| 01:15.19 | termas_ | is the module loaded...? |
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| 01:16.09 | knarfly | if you mean wcfxo, yes it's loaded |
| 01:16.58 | termas_ | chan_dahdi |
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| 01:17.55 | termas_ | i should have been more clear, the asterisk module, not the kernel module. |
| 01:19.04 | knarfly | I'm not sure I follow you....I added what was in my zapata.conf file to the bottom of the chan_dahdi file |
| 01:19.35 | knarfly | everything above it was commented out...at least that's what I think...double checking now |
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| 01:22.07 | termas_ | modules show |
| 01:22.13 | termas_ | in the asterisk console. |
| 01:22.24 | knarfly | stand by |
| 01:22.47 | termas_ | show modules, i should say. |
| 01:23.05 | termas_ | or module show |
| 01:24.12 | _ShrikE | Wired_Life: set qualify=yes in the peer |
| 01:24.33 | orangeservice | hey guys - I seem to be stuck in a WaitExten loop... the first iteration is s@incoming-context, then next time it repeats it is dialed-extention@incoming-context - any ideas |
| 01:24.45 | orangeservice | basically I just get the same prompt over and over again |
| 01:24.56 | knarfly | module show is a long list but chan_dahdi.so is in there |
| 01:25.42 | orangeservice | I think I need to somehow tell waitexten that I do not want to use the "incoming" context along with the dialed number... how would I do that? |
| 01:36.00 | termas_ | karfly, well that's all the easy answers covered... |
| 01:36.38 | knarfly | yeah, I'm gonna sleep on it and try again tomorrow night...it's a bitch |
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| 01:44.40 | thehar | ~install |
| 01:44.41 | infobot | methinks install is http://linuxmafia.com/faq/Debian/installers.html or the official guide is at http://www.debian.org/releases/woody - see the tips at http://www.linuxmafia.com/debian/tips, or http://www.debian.org/releases/stable/installmanual |
| 01:45.01 | thehar | russellb: what's your fast svn download of trunk and install? |
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| 02:11.08 | kc8pxy | Wired_Life1: heya |
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| 02:24.26 | suma | can someone help in getting my musiconhold running |
| 02:24.45 | suma | When i play MusiconHold() it is playing indefinitely |
| 02:24.53 | suma | but it id not playing with DIal |
| 02:25.03 | thehar | use the right flag |
| 02:25.21 | suma | I used the option m(default) which is the one exists in musiconhold.conf |
| 02:25.38 | suma | thehar: flag ? where ? |
| 02:26.54 | suma | thehar: Am I missing anything ? |
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| 02:27.58 | mchou | hah, skype IPO. like maybe 5 years too late |
| 02:28.28 | thehar | old |
| 02:30.13 | suma | thehar: not getting you |
| 02:48.13 | *** join/#asterisk axscode (n=axscode@axscode.skytel.com.ph) |
| 02:48.46 | axscode | hi any trixbox user? is trixbox comes with web admin for adding user? or it should be adding it via sip.conf ? |
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| 02:52.40 | murdock_ut | ~trixbox |
| 02:52.41 | infobot | from memory, trixbox is a full linux distro that includes FreePBX and other 3rd party add-ons. It is all this extra stuff which makes trixbox VERY difficult to support, and is not supported in #asterisk. Try asking in #trixbox or on their forums & wiki at http://www.trixbox.org/. We do not recommend using it. |
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| 03:25.33 | axscode | can someone point me, how trixbox will say, "if you know the extension number, please dial now" |
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| 03:53.51 | *** join/#asterisk martyn-dev (n=andres_m@190.27.175.33) |
| 03:54.04 | martyn-dev | Hi |
| 03:54.37 | martyn-dev | I try use video with ekiga softphone with sip, i enabled videosupport on sip.conf but video is not passed, only voice.. |
| 03:54.44 | martyn-dev | No error in cli, do know why ? |
| 03:55.20 | drmessano | Did you enable the video codecs? |
| 03:55.33 | tessier | What's the dialplan function that lets you specify a password and then put someone into another context? |
| 03:55.49 | tessier | Like if you want people to be able to authenticate through an ivr and then get into another context such as for dialing out? |
| 03:55.52 | tessier | I can't recall... |
| 03:56.06 | martyn-dev | yeah: i disallow=all ; First disallow all codecs |
| 03:56.06 | martyn-dev | allow=ulaw |
| 03:56.06 | martyn-dev | allow=alaw |
| 03:56.06 | martyn-dev | allow=ilbc |
| 03:56.06 | martyn-dev | allow=gsm |
| 03:56.07 | martyn-dev | allow=h261 |
| 03:56.22 | martyn-dev | drmessano: :) |
| 03:56.23 | drmessano | ~pb |
| 03:56.24 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines 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 , or , http://bin.cakephp.org/ |
| 03:56.39 | martyn-dev | infobot: "sorry" :P |
| 03:57.20 | martyn-dev | drmessano: look http://pastebin.ca/1392467 |
| 03:57.38 | martyn-dev | But not message errors in cli is show :S |
| 03:58.52 | drmessano | Are you using h261? |
| 03:59.42 | martyn-dev | drmessano: in my ekiga preferences is h261 enable |
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| 04:00.04 | drmessano | Do the sip extensions have allow and disallow lines? |
| 04:00.16 | tessier | Ah, disa! |
| 04:00.47 | martyn-dev | i dont understand you drmessano |
| 04:01.02 | drmessano | pastebin your sip.conf pls |
| 04:01.06 | martyn-dev | let me |
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| 04:05.33 | martyn-dev | <PROTECTED> |
| 04:05.40 | martyn-dev | sorry, is solved :) |
| 04:05.43 | martyn-dev | just syntax :D |
| 04:05.45 | martyn-dev | Thank you |
| 04:05.48 | martyn-dev | video its working |
| 04:05.48 | martyn-dev | :D |
| 04:05.52 | drmessano | ok cool |
| 04:05.54 | martyn-dev | one last. |
| 04:06.07 | martyn-dev | How much bandwitdh is to video ? |
| 04:06.20 | drmessano | Depends on codec |
| 04:10.04 | martyn-dev | UHm, ok, so i'm going to read about h261 ... |
| 04:10.38 | martyn-dev | drmessano: do you know if i have video use the same ports that SIP .. or have i to listen some additional ports ? |
| 04:11.16 | drmessano | Its over SIP |
| 04:11.30 | xpot | Question: is there a way to use the directory and read the name? ie: "Susan Smith" instead of "S, U, S, A, N, 'space', S, M, I, T, H" ?? |
| 04:11.46 | [TK]D-Fender | xpot: go RECORD the name |
| 04:12.10 | xpot | [TK]D-Fender: I thought festival was supposed to take care of that? |
| 04:12.20 | martyn-dev | ok thank you drmessano |
| 04:12.28 | [TK]D-Fender | xpot: You clearly haven't looked through VoiceMailMain :) |
| 04:13.17 | [TK]D-Fender | xpot: (0) "voicemail options" (3) "Record your name for the directory" |
| 04:13.32 | xpot | [TK]D-Fender: :| obviously not.. I will go check it out, thanks |
| 04:13.35 | [TK]D-Fender | xpot: they hid it in the BIG PRINT :p |
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| 04:16.05 | martyn-dev | one last. quickly question. sip port 5060 is udp or tcp ? |
| 04:16.06 | martyn-dev | :D |
| 04:16.13 | xpot | udp |
| 04:16.40 | suma | how can I make an audio to play simultaneously to the caller and calee |
| 04:16.47 | drmessano | Thats not accurate |
| 04:16.51 | martyn-dev | thank you xpot |
| 04:16.54 | suma | when i use the DIal Application it plays only to once channel at a time |
| 04:17.05 | suma | it plays to caller and then it plays to the callee |
| 04:17.12 | drmessano | SIP on 1.6 can be TCP or UDP |
| 04:17.49 | martyn-dev | os interesting drmessano ;) |
| 04:18.12 | martyn-dev | I have my server in my home and i have a adsl with router ... i can configure ports with nat to send traffic to my server in my bedroom. IF I do nat to 5060 port to my server home, is sufficient to have SIP communication ? |
| 04:18.34 | martyn-dev | or HAve i that listen some other port ? |
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| 04:19.14 | drmessano | no |
| 04:19.19 | drmessano | ~sipnat |
| 04:19.20 | infobot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 04:19.24 | drmessano | Follow that |
| 04:21.08 | martyn-dev | let me see drmessano ;) |
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| 04:33.30 | martyn-dev | drmessano: do you know if externip param can be a domain ? |
| 04:34.26 | [TK]D-Fender | martyn-dev: Go read it again |
| 04:36.22 | martyn-dev | ok.. again.. |
| 04:36.38 | martyn-dev | :P YEah. it can :) |
| 04:36.42 | martyn-dev | Thank you |
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| 04:45.25 | martyn-dev | i understand that if my asterik server is in behind nat and my phones outside.. with port forwarding works... |
| 04:45.40 | martyn-dev | But, my router isn't port forwarding :( |
| 04:46.02 | martyn-dev | really not have port range forwarding , just nat port by port. |
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| 04:46.28 | [TK]D-Fender | martyn-dev: get another router |
| 04:48.04 | martyn-dev | [TK]D-Fender: but I can NAT 5006 port.. just i cant 10000-20000 ports range redirect to my server.. |
| 04:48.58 | Juggie | upgrade the router? |
| 04:49.07 | Juggie | it doesnt do triggered port forwarding? |
| 04:49.48 | martyn-dev | Juggie: in my admin configuration exits a Tab link that say "Port Trigger" |
| 04:50.32 | martyn-dev | but i dont know if with this can i do the redirect to my server |
| 04:51.06 | [TK]D-Fender | martyn-dev: Got a DMZ option? |
| 04:51.13 | martyn-dev | let me see.. |
| 04:51.23 | martyn-dev | say "DMZ Host Setup" |
| 04:51.30 | martyn-dev | to put a IP address.. |
| 04:51.39 | [TK]D-Fender | martyn-dev: Do taht an lock down other services manually on your server |
| 04:52.07 | martyn-dev | with iptable do you mean ? or what ? |
| 04:52.18 | [TK]D-Fender | martyn-dev: Yes |
| 04:52.35 | martyn-dev | But, dmz, ? |
| 04:52.39 | martyn-dev | [TK]D-Fender: |
| 04:53.35 | martyn-dev | if necesary rtpstart and rtpend in 10.000 and 20.000 respectly.. or can i use 1 port ? 10.000 |
| 04:53.50 | [TK]D-Fender | martyn-dev: No, you can't use *1* port. |
| 04:54.11 | Juggie | martyn-dev, no one will not work, port trigger would be what you need or you could just forward 10,000->20,000 |
| 04:54.17 | [TK]D-Fender | Asterisk uses 1 port per diretion per call, and you can't restrict the ones you want to use for WAN or that * will receive against |
| 04:54.32 | Juggie | you dont need that many though, its configurable in rtp.conf |
| 04:54.41 | [TK]D-Fender | martyn-dev: So get rid of ryour port forwarding on the router, and just DMZ it and firewall your router |
| 04:54.43 | martyn-dev | çuhm. i see. |
| 04:55.39 | martyn-dev | I will try with iptables.. |
| 04:55.43 | martyn-dev | thank you .. |
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| 06:01.28 | kc8pxy | so far i've been playing with 1.6.0.x with purely sip channels. i plan to move to hardware channels soon. from what I'm reading, i need to ./configure --with-dahdi=path, and says it needs to point to my dahdi files. which files from dahdi is it refering to? |
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| 06:12.22 | drmessano | You dont need the path unless the install is custom, IIRC.. ive never used a path |
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| 06:42.37 | ultrav1olet | How can I say to asterisk that my DAHDI channel allows only one outgoing call at a time? |
| 06:43.49 | ultrav1olet | When someone is calling outside and someone else tries to call the second person only hears eternal silence |
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| 07:38.31 | sacoetzee | hey hey everyone, irc first-timer here, i need help on asterisk. i have a voip system running on ulaw and i suffer from interruptions (for lack of a better word) internal phonecalls are crystal, but incoming calls are clear, except if someone interrupts on the other end it cuts out for a second or two.. any ideas? |
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| 07:43.59 | axscode | <PROTECTED> |
| 07:44.06 | sacoetzee | is scratching his head |
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| 07:45.22 | jblack | sacoetzee: Echo cancellation might be getting in the way? |
| 07:45.24 | remalit | hello guys , |
| 07:45.31 | remalit | i try to read cat /sys/module/wctdm/dtmf |
| 07:45.39 | ultrav1olet | wow, everyone's suddenly alive |
| 07:45.41 | remalit | cat: /sys/module/wctdm/dtmf: No such file or directory |
| 07:45.46 | remalit | when i try to create it |
| 07:45.53 | remalit | it said premssion denied |
| 07:45.54 | mosty | remalit, what are you trying to do? |
| 07:45.59 | remalit | i logged as root |
| 07:46.06 | jblack | remalit: A kernel module would create that, not you, I believe. |
| 07:46.21 | ultrav1olet | remalit: you CANNOT create that file |
| 07:46.21 | jblack | goes off to read his new book "Pride & Prejudice & Zombies" |
| 07:46.25 | remalit | i wana do echo "1" > /sys/module/wctdm/dtmf |
| 07:46.33 | ultrav1olet | linux kernel module is what creates that file |
| 07:46.34 | BlackRat | hi, have a strange question, .. have te220 + dahdi-linux + tools + asterisk (compiled with "--with-dahdi") .. dahdi_tool shows line ok .. but PSNT provider says line isnt fully up .. any clues ? |
| 07:46.39 | remalit | then how i can do this :) |
| 07:46.40 | mosty | remalit, why do you want to do that- ie what are you trying to do? |
| 07:46.48 | jblack | remalit: Load the kenrel module that creates that file. |
| 07:46.52 | remalit | to get caller id working in my countery |
| 07:47.00 | remalit | jblack how to load it |
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| 07:47.34 | jblack | Contact digium support for instructions on how to build kernel modules. |
| 07:47.55 | remalit | i dd so many times no answers :( |
| 07:48.04 | ultrav1olet | BlackRat: that means you have't set up dahdi properly |
| 07:48.29 | BlackRat | ultrav1olet: seems im missing chan_dahdi.so |
| 07:58.17 | jblack | Ok. This is gonna be a good book. |
| 07:58.20 | jblack | later all. |
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| 08:01.44 | sacoetzee | hmmm.. can anyone give me the rundown on hardware gain on echo cancellation vs software gain, maybe if i had the software gain right i could use that instead of the hardware and check if the card isn't maybe faulty |
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| 08:07.28 | BlackRat | ll try to rephrase my question ... got asterisk 1.4.24.1 -> compiled with ./configure --with-dahdi=/usr/include ... maked .. but seems to be missing chan_dahdi.so .. any clues what im doing wrong ? (am I suppoused to get that chan_dahdi.so at all ? ) |
| 08:07.31 | *** join/#asterisk tmjb (n=tmjb@212.200.239.230) |
| 08:07.46 | *** join/#asterisk senyai232 (n=Miranda@62.141.75.44) |
| 08:08.28 | mosty | BlackRat, is dahdi installed? is it enabled in "make menuselect" ? |
| 08:08.54 | BlackRat | its installed through kmp and modules are loaded .. dahdi_tool shows ok on line |
| 08:09.43 | mosty | in the asterisk source dir, when you do "make menuselect", is dahdi enabled? |
| 08:10.39 | BlackRat | yep |
| 08:11.04 | BlackRat | if X means selected :) |
| 08:13.08 | BlackRat | ahh it explains .. dependencies |
| 08:14.45 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 08:18.25 | senyai232 | hi! Has anyone a working example of IAXclient with attened transfer working? |
| 08:30.16 | *** join/#asterisk axarob (n=ebash@cpc3-barn8-0-0-cust288.brnt.cable.ntl.com) |
| 08:31.26 | *** part/#asterisk sacoetzee (n=s-a-coet@dsl-247-74-35.telkomadsl.co.za) |
| 08:35.54 | *** join/#asterisk creativx (n=creadure@226.62-97-205.bkkb.no) |
| 08:36.42 | creativx | what are my options when i have 15 "stuck" iax2 channels? |
| 08:36.50 | creativx | in "ring" state.. i wish to purge them |
| 08:40.17 | senyai232 | Does anyone familar with IAXclient? I can not make iax_setup_transfer function work :( |
| 08:45.08 | *** join/#asterisk j_kroon (n=jkroon@dsl-247-74-35.telkomadsl.co.za) |
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| 08:54.25 | j_kroon | hi all, with asterisk 1.6.0.3 and dahdi-2.1.0.3, how can i test the effects of echo cancellation. with echocancel=no there is no breaks in voice whatsoever, but with echocancel=yes (default=128, also tried 32 and 64) I get nasty "cut outs" or "fading" (depending on who the person is you're asking). |
| 08:54.35 | j_kroon | how would the four rx/tx and soft/hard gains affect this? |
| 08:57.23 | *** join/#asterisk j_kroon (n=jkroon@196.35.70.126) |
| 08:58.42 | j_kroon | hi all, with asterisk 1.6.0.3 and dahdi-2.1.0.3, how can i test the effects of echo cancellation. with echocancel=no there is no breaks in voice whatsoever, but with echocancel=yes (default=128, also tried 32 and 64) I get nasty "cut outs" or "fading" (depending on who the person is you're asking). |
| 08:58.44 | j_kroon | how would the four rx/tx and soft/hard gains affect this? |
| 08:58.56 | j_kroon | sorry for double posting ... my connection is really bad. |
| 09:00.55 | defsdoor | j_kroon: can see gain affecting it at all |
| 09:00.58 | defsdoor | cant* |
| 09:01.20 | defsdoor | you got h/w echo canc ? |
| 09:01.27 | j_kroon | hw |
| 09:01.42 | defsdoor | what card |
| 09:02.02 | j_kroon | Digium 24xxp with 5 x Quad FXOs + echo canceller (all digium) |
| 09:02.44 | *** part/#asterisk jstoker (n=jstoker@82.31.47.238) |
| 09:03.15 | defsdoor | dunno about digium cards - but on sangoma the echo cancellation comes on during a call - so the first .3s maybe sound different until it comes on - you can tell it to be on permanently instead - maybe similar on digium ? |
| 09:03.41 | *** join/#asterisk guenter65 (n=pit@158.64.156.100) |
| 09:03.56 | *** join/#asterisk jstoker (n=jstoker@82.31.47.238) |
| 09:04.19 | defsdoor | I had a site with 6fxo and they got echo for the first snippet of a call - then the h/w took it out |
| 09:04.19 | *** part/#asterisk jstoker (n=jstoker@82.31.47.238) |
| 09:04.21 | j_kroon | defsdoor, don't think so. |
| 09:04.28 | j_kroon | echo isn't my problem, it's breakup. |
| 09:04.38 | j_kroon | on the inbound audio _from_ the device. |
| 09:04.58 | defsdoor | throughout a call ? |
| 09:05.02 | j_kroon | jip. |
| 09:05.14 | j_kroon | and it's only when both parties speak at the same time. |
| 09:05.23 | ultrav1olet | How can I say to asterisk that my DAHDI channel allows only one outgoing call at a time? |
| 09:05.37 | defsdoor | but only with echo on ? |
| 09:05.39 | ultrav1olet | When one person is calling outside and someone else tries to call the second person only hears eternal silence |
| 09:06.00 | j_kroon | defsdoor, yes. |
| 09:06.12 | j_kroon | echocancel=no and I get echo, but no breakups. |
| 09:06.26 | defsdoor | maybe your echo is too bad |
| 09:06.37 | j_kroon | ultrav1olet, put the unused channels in another group. |
| 09:07.08 | ultrav1olet | j_kroon: I have the only channel! |
| 09:07.26 | j_kroon | ultrav1olet, pastebin your chan_dahdi.conf file |
| 09:08.29 | ultrav1olet | wait a sec |
| 09:08.45 | guenter65 | hi, I've got a strange connection problem with my sip provider... it can't register. |
| 09:08.56 | guenter65 | My firewall blocks packs originiating from my provider but on a strange port: SPT=5060 DPT=59117 |
| 09:09.21 | guenter65 | Why 59117??? why does my provider contact my on 59117? |
| 09:09.23 | j_kroon | defsdoor, how would i know if the echo is too bad, or could i maybe make the echo can less agressive? |
| 09:09.28 | j_kroon | ie, set it to let some through? |
| 09:09.33 | j_kroon | up the taps, down it? |
| 09:09.51 | defsdoor | j_kroon: sorry - dunno - I've never really suffered with it so not had to solve it :) |
| 09:10.03 | j_kroon | screwed echo can card maybe? |
| 09:10.04 | ultrav1olet | http://pastebin.ca/1392593?nomobile=1 |
| 09:10.28 | defsdoor | j_kroon: I'd doubt it |
| 09:10.35 | ultrav1olet | j_kroon: seems like you are right - I have four lines enabled ... or maybe not |
| 09:10.46 | j_kroon | channel => 1-4 <-- change this to channel => 1 |
| 09:10.53 | *** join/#asterisk sacoetzee (n=s-a-coet@dsl-247-74-35.telkomadsl.co.za) |
| 09:11.24 | ultrav1olet | one more question - even with this config asterisk doesn't detect BUSY signal |
| 09:11.47 | defsdoor | analog sucks :) |
| 09:12.08 | ultrav1olet | indI know |
| 09:12.11 | ultrav1olet | I know |
| 09:12.28 | ultrav1olet | indications.conf is also set to the right country :( |
| 09:12.46 | defsdoor | ultrav1olet: does it need to detect busy on outgoing ? |
| 09:12.56 | defsdoor | ultrav1olet: the user knows a busy tone when they hear it |
| 09:12.59 | ultrav1olet | certainly |
| 09:13.16 | ultrav1olet | e.g. if I call someone and he/she ends the call asterisk will never disconnect us |
| 09:13.23 | defsdoor | thats not busy |
| 09:13.25 | defsdoor | that's hangup |
| 09:13.34 | ultrav1olet | in Russia hangup = busy |
| 09:13.40 | ultrav1olet | for analogue lines |
| 09:13.49 | defsdoor | you need to check with your telco for something called disconnect supervision or clear disconnect |
| 09:13.52 | guenter65 | I have set bindport=5060, there is nowhere a mention in my config of port 59177! :( |
| 09:13.54 | ultrav1olet | the audio signal is absolutely the same |
| 09:14.07 | defsdoor | hanguponpolarityreversal=yes |
| 09:14.10 | ultrav1olet | defsdoor: if only they knew themselves |
| 09:14.23 | ultrav1olet | to a dahdi_channel? |
| 09:14.49 | defsdoor | ultrav1olet: that's in my fxo site's zapata.conf |
| 09:15.08 | defsdoor | but that will only work if your telco does that signalling |
| 09:15.50 | *** join/#asterisk mikkel (n=mikkel@130.226.37.63) |
| 09:15.55 | defsdoor | I had a knightmare with it all though - I'm not touching analog again |
| 09:16.29 | ultrav1olet | defsdoor: I quite agree with you |
| 09:18.24 | guenter65 | I don't think it's a firewall issue |
| 09:18.26 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
| 09:18.28 | ultrav1olet | defsdoor: is it like hanguponpolarityswitch? |
| 09:18.38 | defsdoor | yup |
| 09:18.39 | ultrav1olet | cause chan_dahdi only lists this option |
| 09:18.42 | *** join/#asterisk BlargMaN00 (n=blarg@12.234.16.130) |
| 09:18.43 | guenter65 | I've deactivated it, and registration still didn't work |
| 09:18.57 | *** join/#asterisk jstoker (n=jstoker@82.31.47.238) |
| 09:19.02 | defsdoor | but - in the uk at least - you have to get the telco to add that |
| 09:19.36 | ultrav1olet | it seems like it's on by default but anyway I'll put this option there |
| 09:20.04 | *** part/#asterisk jstoker (n=jstoker@82.31.47.238) |
| 09:21.09 | ultrav1olet | as expected it didn't work |
| 09:21.37 | ultrav1olet | i have restarted asterisk after putting that option |
| 09:22.38 | *** join/#asterisk oej_ (n=olle@ns.webway.se) |
| 09:24.16 | j_kroon | defsdoor, i hate analog with a passion. |
| 09:24.44 | j_kroon | hanguponpolarityswitch is handy. just a shame it's not supported in ZA. |
| 09:25.46 | guenter65 | question: rtp ports are not used during the registration, only during calls right? |
| 09:25.59 | j_kroon | correct. |
| 09:32.03 | guenter65 | I don't get it! |
| 09:33.52 | guenter65 | I constantly get dropped packs on port 59177 |
| 09:34.27 | j_kroon | NAT port remapping? |
| 09:34.27 | guenter65 | and it is not rtp because it occurs not during a call |
| 09:34.39 | guenter65 | no remapping |
| 09:39.45 | BlackRat | yet another question bugs me, got dahdi + asterisk 1.4.24 working busy works "strangly" .. it doesnt send busy code back .. hangup works but busy .. nope |
| 09:39.55 | BlackRat | any clues ? |
| 09:46.46 | j_kroon | defsdoor, echocancel=1024 fixed it. |
| 09:46.51 | j_kroon | wtf?!? |
| 09:47.02 | j_kroon | anyway ... much happier now. |
| 09:49.46 | *** join/#asterisk Aurs (n=Ove_Aurs@apb9hb.ip.ssc.net) |
| 09:51.32 | j_kroon | now i just need to fxotune -i the whole lot again :) |
| 09:51.54 | Aurs | does anyone here have any experience with sip clients behind a ms ISA server connecting to asterisk? |
| 09:53.04 | mosty | aurs: i've never used ISA, but set a unique rtp port on each of your sip clients, and add a port forwarding rule for each on the firewall |
| 09:53.47 | Aurs | it works, but the clients get UNREACHABLE now and then, seemingly random |
| 09:54.40 | mosty | aurs: do you have port forwarding rules like i suggested already? |
| 09:55.33 | mosty | because that sounds like the symptoms that occur when you dont. (correction: it's not an RTP port, but a "SIP identity port" or similar) |
| 09:55.49 | Aurs | mosty: don't know, it's not my isa server. just very strange that my googling of this have found nothing |
| 09:56.38 | Aurs | usually, these UNREACHABLE symptoms are very... whats the word... they come with a constant interval |
| 09:56.52 | Aurs | because of nat routers killing the udp connection |
| 09:57.19 | mosty | multiple sip clients behind a single NAT causes these sorts of problems- i suggest you contact your firewall admin and ask them to setup those port forwarding rules for you |
| 09:57.50 | Aurs | now you're just simplifying things mosty :) |
| 09:57.52 | tzafrir_laptop | j_kroon, what EC do you use? |
| 09:58.17 | Aurs | nat is usually not a problem if the client use a proper nat router, imho :) |
| 09:59.12 | mosty | aurs: it's basically due to SIP not coping well with NAT, not usually a problem with the NAT router |
| 09:59.48 | Aurs | I have many installations with more sip clients than this case, also behind nat, with no issues (after some configuring of the nat routers) |
| 10:00.29 | j_kroon | tzafrir_laptop, hw. |
| 10:00.44 | Aurs | usually, the UNREACHABLE problem, is because routers kill "idle" connections prematurely (in my experience) |
| 10:01.33 | Aurs | and that is done by the nat router, and not the sip protocol. so I still blame the routers ;) |
| 10:02.46 | mosty | well either way, port forwarding works, just have to make sure to keep your settings in sync |
| 10:03.13 | ultrav1olet | is it a bug? iax2 show peers shows a lot of online peers but at the end of the list I see: 37 iax2 peers [0 online, 0 offline, 37 unmonitored] |
| 10:03.42 | Aurs | ultrav1olet: add "qualify=yes" to your peers |
| 10:03.55 | mosty | ultrav1olet, that's not a bug. it's just saying that you're not monitoring the peers |
| 10:03.56 | *** join/#asterisk smbrienz (n=smbrienz@host202-204-dynamic.1-87-r.retail.telecomitalia.it) |
| 10:04.08 | Aurs | *lunch* |
| 10:04.20 | j_kroon | tzafrir_laptop, does it make sense at all? |
| 10:04.26 | j_kroon | either way, gotta run. |
| 10:05.11 | ultrav1olet | fine |
| 10:07.23 | *** join/#asterisk JT (n=j@unaffiliated/jt) |
| 10:08.02 | ultrav1olet | I have some unpleasant problem. |
| 10:08.37 | ultrav1olet | We have two offices and if someone from IAX phone calls our SIP phone, then audio goes only in one direction |
| 10:08.58 | ultrav1olet | iax phone -> office1 asterisk -> office2 asterisk -> SIP phone |
| 10:09.12 | ultrav1olet | if we call in reverse direction then everything is fine |
| 10:09.35 | ultrav1olet | there are no errors reported |
| 10:09.45 | mosty | on the office2 asterisk server, make sure you have localnet and externip set correctly in sip.conf |
| 10:10.17 | ultrav1olet | and there are some troubles with NAT :( |
| 10:12.15 | BlackRat | if anyone cares .. busy didnt work on pri isdn .. had to use hangup(17) |
| 10:12.22 | BlackRat | thnx for help guys |
| 10:12.25 | ultrav1olet | office1 asterisk 'sees' _all_ clients from office2 LAN as office2_asterisk_server_IP |
| 10:12.59 | mosty | ultrav1olet, are you using iax between the two asterisk servers? |
| 10:13.26 | ultrav1olet | mosty: yes |
| 10:13.57 | mosty | and you have sip clients in office2 registering to the office1 asterisk server? |
| 10:14.38 | *** part/#asterisk BlackRat (n=ulmik@balticom-17-8.balticom.lv) |
| 10:14.52 | ultrav1olet | no, all office2 clients bind to office2_asterisk and the same situation is in office1 |
| 10:15.31 | ultrav1olet | it looks like native bridging is to be blamed - can I disable it completely? |
| 10:18.42 | *** join/#asterisk plantseeker (n=chatzill@77.240.56.22) |
| 10:19.57 | mosty | ultrav1olet, the situation you are describing was between an IAX client and a SIP client- there is no native bridging in this case |
| 10:20.16 | ultrav1olet | that's more than weird |
| 10:20.35 | ultrav1olet | is there a way to see the state of audio channels for a call? |
| 10:21.13 | defsdoor | damn - cursed by j_kroon |
| 10:21.22 | defsdoor | "we have a problem, the incoming calls all seem to have an echo any ideas?" |
| 10:26.56 | *** join/#asterisk visioconf (i=3ea03b25@gateway/web/ajax/mibbit.com/x-e3644deef3457cd7) |
| 10:27.09 | visioconf | hello |
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| 10:28.41 | *** part/#asterisk smbrienz (n=smbrienz@host202-204-dynamic.1-87-r.retail.telecomitalia.it) |
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| 10:30.06 | visioconf | i'm now managing a contact center technical affairs , i want to use the best CRM that can fit Asterisk , any suggestions |
| 10:30.11 | visioconf | please ?? |
| 10:30.44 | ultrav1olet | mosty: http://img220.imageshack.us/img220/2971/scheme.png |
| 10:32.38 | visioconf | hello anybody there ? |
| 10:33.12 | mosty | ultrav1olet, can you run packet tracers on each of the asterisk machines? |
| 10:34.31 | ultrav1olet | yes, what kind of? |
| 10:34.35 | ultrav1olet | tcpdump? :) |
| 10:35.35 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 10:36.08 | mosty | whichever you know best. see if you can figure out where the lost audio packets are going |
| 10:45.27 | ultrav1olet | seems like I've traced the culprit: "Channel 'IAX2/in_ikar-11844' unable to transfer" |
| 10:45.48 | ultrav1olet | or maybe not |
| 10:46.05 | mosty | i don't think that's related |
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| 10:48.43 | ultrav1olet | SIP account has "canreinvite=no" - is it OK? |
| 10:49.13 | mosty | that's what you want |
| 10:49.31 | mosty | have you tried the sip settings i mentioned earlier? |
| 10:49.32 | ultrav1olet | I thought it would be better not to allow native connections |
| 10:49.55 | mosty | you can't do "native connections" between iax devices and sip devices |
| 10:50.08 | ultrav1olet | externip and localnet? |
| 10:50.52 | mosty | yes |
| 10:50.53 | ultrav1olet | so that option is not necessary? |
| 10:51.08 | mosty | have you tried them? |
| 10:51.11 | ultrav1olet | on which server - can you remind me? by looking at the pic I've sent |
| 10:51.26 | mosty | on the asterisk server that has sip clients |
| 10:51.45 | ultrav1olet | that server has the only network interface |
| 10:51.50 | ultrav1olet | it's in the LAN |
| 10:52.11 | *** join/#asterisk sacoetzee (n=s-a-coet@dsl-247-74-35.telkomadsl.co.za) |
| 10:52.15 | ultrav1olet | I cannot believe setting that options will have any positive effect |
| 10:53.12 | mosty | try it |
| 10:53.31 | ultrav1olet | ok |
| 10:53.48 | mosty | externip is the public ip of that internet connection |
| 10:55.30 | ultrav1olet | mosty: I don't wanna seem annoying but those two options are related to functioning behind NAT. We don't have NAT between these two servers :( |
| 10:55.58 | ultrav1olet | they are in the same VPN network |
| 10:56.12 | ultrav1olet | however they have different networks |
| 10:57.44 | mosty | there is a chance that your VPN gateway is using a form of NAT |
| 10:57.54 | ultrav1olet | I've got an idea |
| 10:57.56 | ultrav1olet | a bad idea |
| 10:58.04 | mosty | those settings are still worth testing- it will only take a couple of minutes to try |
| 10:58.16 | ultrav1olet | on asterisk2 server we have some iptables modules related to SIP |
| 10:58.22 | ultrav1olet | I'll try to disable them |
| 10:59.32 | mosty | <ultrav1olet> office1 asterisk 'sees' _all_ clients from office2 LAN as office2_asterisk_server_IP |
| 10:59.40 | mosty | that sounds like NAT to me |
| 11:01.36 | ultrav1olet | I have remedied that situation :) |
| 11:01.51 | ultrav1olet | now all packets are walking normally :) |
| 11:03.39 | ultrav1olet | mosty: how can that happen that audio signal can depend on a call direction? |
| 11:20.44 | *** join/#asterisk nighty^ (n=nighty@x122091.ppp.asahi-net.or.jp) |
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| 11:37.25 | *** join/#asterisk mort_gib (n=mjensen@adsl-2-234.gibnet.gi) |
| 11:40.00 | mosty | ultrav1olet, because NAT is not symmetrical, typically |
| 11:42.01 | ultrav1olet | there's no NAT right now |
| 11:42.08 | ultrav1olet | anywhere :) |
| 11:42.23 | ultrav1olet | what about private messages I've sent you? |
| 11:42.59 | mosty | i prefer to talk in the channel, use pastebin |
| 11:44.24 | ultrav1olet | you haven't even read them? ;) |
| 11:45.46 | mosty | i went out to get some lunch, came back to see a bunch of messages (half scrolled away due to other traffic) |
| 11:46.21 | mosty | but anyway, i'm not really willing to help any further if you refuse to try the settings i have suggested |
| 11:50.03 | *** join/#asterisk pif (n=ldm@zenon.apartia.fr) |
| 11:50.48 | mort_gib | mosty: That HAS to be the kindest ever refusal to help I have EVER seen in here :-) |
| 11:50.53 | ultrav1olet | mosty: http://pastebin.ca/1392699?nomobile=1 |
| 11:51.35 | ultrav1olet | It looks like I need to disable native bridging between Asterisk2 server clients and Asterisk1 server. |
| 11:51.52 | ultrav1olet | as you can see there's no NAT |
| 11:56.06 | mosty | mort_gib, sometimes people just make it too hard to help them |
| 11:57.10 | *** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim) |
| 11:57.11 | mort_gib | True |
| 11:58.21 | ultrav1olet | mosty: I have enabled both settings - everything is the same. |
| 11:59.36 | ultrav1olet | From the very beginning I thought they wouldn't help ;) |
| 12:00.58 | ultrav1olet | ok, what about my last question: How I can disable native bridging between Asterisk2 server clients and Asterisk1 server? |
| 12:01.34 | ultrav1olet | BTW, both servers have "transfer=no |
| 12:01.39 | ultrav1olet | " for each other |
| 12:02.01 | ultrav1olet | so I cannot understand how native bridging can occur in this situation |
| 12:02.25 | lbt | I have a Wildcard X100P. Last night we had lightning :( and I got lots of "PCI Master Aborts". I rebooted and the card comes up just as before - nothing in the kernel messages indicate a problem - but it's not making or receiving calls.... should I just buy a new one or is there any chance it's OK? |
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| 12:04.35 | *** part/#asterisk jstoker (n=jstoker@80-41-183-84.dynamic.dsl.as9105.com) |
| 12:06.16 | ultrav1olet | RMA it :) |
| 12:06.32 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 12:11.49 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 12:17.40 | *** join/#asterisk maxagaz (n=maxagaz@soho2.i-xanadu.com) |
| 12:18.31 | ultrav1olet | I could have thanked you, but I have solved the problem on my own :( |
| 12:19.07 | ultrav1olet | transfer=no |
| 12:19.52 | ultrav1olet | this is what I have added to the [genera] section of iax.conf on asterisk 2 server |
| 12:20.09 | *** join/#asterisk maxagaz (n=maxagaz@soho2.i-xanadu.com) |
| 12:23.07 | *** join/#asterisk M1s3ry (n=M1s3ry@boromir.api-digital.com) |
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| 12:32.48 | *** part/#asterisk jstoker (n=jstoker@80-41-183-84.dynamic.dsl.as9105.com) |
| 12:36.41 | *** join/#asterisk leifmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 12:36.41 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 12:42.59 | *** join/#asterisk macros73 (n=cs@dsl093-063-232.pit1.dsl.speakeasy.net) |
| 12:43.41 | *** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
| 12:50.50 | *** join/#asterisk pif (n=ldm@zenon.apartia.fr) |
| 12:51.41 | *** part/#asterisk ultrav1olet (n=ultrav1o@94.180.11.169) |
| 12:56.11 | *** join/#asterisk coppice (n=chatzill@46.166.17.210.dyn.pacific.net.hk) |
| 13:00.58 | *** join/#asterisk Katty (n=asterisk@mail.copi-rite.com) |
| 13:04.09 | [TK]D-Fender | Katty: Mew. |
| 13:04.34 | Katty | [TK]D-Fender: morning |
| 13:04.46 | Katty | i think. |
| 13:05.31 | leifmadsen | morning all! |
| 13:07.23 | leifmadsen | has anyone had an issue with polycom phones that temporarily lose a network connection, but then will not attempt to re-register? |
| 13:07.26 | *** join/#asterisk MaliutaLap (n=biteme@kiev.lusan.id.au) |
| 13:08.05 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 13:09.07 | *** join/#asterisk tcseke (n=chatzill@217.20.134.239) |
| 13:14.06 | *** join/#asterisk esperegu (n=esperegu@145.116.15.244) |
| 13:15.07 | esperegu | I am trying to connect a cisco 7970 Sip phone to a register directly but I only get unauthorised back and it does not sent an autorization request. anyone an idea? |
| 13:16.42 | jaytee | leifmadsen, I've had problems with a few of the 80 some odd 330's I have rebooting and then coming up with duplicate IP address so they can't even attempt to register but we're using DHCP from a Microsoft Windows domain server that passes the FTP boot server address as part of it's lease options. |
| 13:17.26 | eppigy | UND KEINER EIR |
| 13:19.10 | *** join/#asterisk theHub (n=theHub@69.177.93.21) |
| 13:21.56 | eppigy | theHub: when your useage approaches 30% |
| 13:22.05 | eppigy | network performance degrades exponentialy |
| 13:22.42 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
| 13:23.10 | *** join/#asterisk putnopvut (n=putnopvu@asterisk/master-of-queues/mmichelson) |
| 13:23.10 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 13:25.36 | *** join/#asterisk czindy (n=czindy@91.120.30.42) |
| 13:32.33 | *** join/#asterisk Skarmeth (n=Skarmeth@201.57.179.27) |
| 13:34.15 | czindy | Hello, I have a problem (http://pastebin.com/d34306ea6) with poke_noanswer, I got notices: Peer '203' is now UNREACHABLE! ..... Peer '203' is now REACHABLE! with 10 seconds between them. Please help me to find out why these messages comeing. Thank you. |
| 13:35.05 | czindy | And these messages are alwayse comeing on all peers. |
| 13:36.24 | *** join/#asterisk pmhaddad-work (n=pmhaddad@141.219.87.43) |
| 13:38.29 | Katty | i have a solution |
| 13:38.31 | Katty | the solution is Jager |
| 13:38.34 | Katty | and redbull, of course. |
| 13:39.41 | *** join/#asterisk telnettech (i=telnette@gw.percipia.com) |
| 13:39.57 | Katty | telnettech: ohai |
| 13:40.02 | Katty | eppigy: what was for dinner? |
| 13:40.15 | telnettech | good morning Katty |
| 13:40.44 | telnettech | TK: good morning |
| 13:40.46 | Corydon76-dig | czindy: turn on qualifysmoothing |
| 13:41.01 | telnettech | jaytee: good morning |
| 13:41.33 | Katty | hugs Corydon76-dig |
| 13:41.41 | Corydon76-dig | hugs Katty |
| 13:42.23 | czindy | @Corydon76-dig: ok I'm trying, thank you |
| 13:42.51 | *** join/#asterisk omer (n=_omer@119.152.8.208) |
| 13:42.57 | omer | hello |
| 13:43.33 | omer | "All circuits are busy now" .... how to play this message without answering the call? |
| 13:44.54 | czindy | @Corydon76-dig: is it possible to define this parameter under realtime database? |
| 13:44.56 | guax | do not execute Answer() before Playback() |
| 13:45.02 | guax | omer, ^ |
| 13:45.17 | pmhaddad-work | omer, just do a Playback() then a Hangup() |
| 13:45.18 | guax | but that of course could bring problems to some softphones and ata's |
| 13:45.27 | jaytee | morning brian |
| 13:45.33 | Katty | hey jaytee |
| 13:45.38 | *** join/#asterisk _omer (n=_omer@119.152.10.22) |
| 13:45.41 | jaytee | morning Katty *hugs* |
| 13:45.49 | Katty | hugs jaytee |
| 13:45.58 | eppigy | Katty: wendys double cheese |
| 13:45.59 | _omer | "All circuits are busy now" .... how to play this message without answering the call? |
| 13:46.03 | eppigy | YES |
| 13:46.04 | eppigy | man |
| 13:46.07 | guax | -_- |
| 13:46.11 | eppigy | I am already in a bad mood |
| 13:46.16 | guax | _omer, [10:45] <guax> do not execute Answer() before Playback() |
| 13:46.47 | _omer | can I use background() instead of playback() ? |
| 13:47.24 | guax | why would you do that? |
| 13:48.32 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 13:48.32 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 13:48.37 | Katty | mister madsen |
| 13:49.15 | _omer | I never used playback since the day playback gave me alot of problems...anyhow i will try |
| 13:50.25 | eppigy | TRABAJO |
| 13:51.09 | czindy | is it possible to define qualifysmoothing parameter under realtime database? Do I need to define a row with this name in the database? |
| 13:52.33 | Katty | dave |
| 13:52.42 | *** join/#asterisk jmacz (n=jmacz@190.144.75.22) |
| 13:53.34 | *** join/#asterisk djMax (n=chatzill@66.92.91.132) |
| 13:53.37 | jaytee | NO TRABAJO!!! VACACIONES AHORA!!!! |
| 13:54.38 | Katty | hawaii? |
| 13:54.40 | Katty | i wanna go |
| 13:54.43 | Katty | let me get my towel! |
| 13:54.53 | Katty | we can buy everything else THERE |
| 13:55.09 | djMax | (back on this tbct thing) so if I want to transfer a call generally, I just Dial and then drop off the dialplan? |
| 13:55.10 | Katty | frozen pineapple with vodka in it |
| 13:55.12 | Katty | mmmmm |
| 13:56.22 | Katty | idk, if i go to hawaii i might not come back |
| 13:56.34 | [TK]D-Fender | djMax: Transfer(), not Dial() |
| 13:56.42 | eppigy | whiskey |
| 13:56.45 | eppigy | straight |
| 13:56.48 | eppigy | warm |
| 13:56.50 | djMax | you sure about that? Because Transfer doesn't work seemingly. |
| 13:57.03 | Katty | eppigy: did you have wendy's last night? |
| 13:57.09 | eppigy | I surely did |
| 13:57.13 | eppigy | it was delicious |
| 13:57.29 | eppigy | i can feel its remains coursing through my body |
| 13:57.30 | czindy | when I create a databse for IAX2 clients for asterisk realtime, is it possibe to define: qualifysmoothing, qualifyfreqok, qualifyfreqnotok ?? |
| 13:57.34 | djMax | The TBCT part supposedly is irrespective of how it got there. Basically * says "oh, I see two channels taken up talking to each other, let me toss this to the other end of the circuit) |
| 13:57.34 | Katty | i put the makings of chicken stroganoff in the crockpot this morning. |
| 13:57.35 | eppigy | fueling my insane rage |
| 13:57.41 | eppigy | oh dang |
| 13:57.44 | eppigy | DANG |
| 13:58.02 | Katty | 4 ingredients :P |
| 13:58.05 | [TK]D-Fender | djMax: Also depends on carrier support, and the only 2 protocols I'm aware that cover this are DMS100 & 5ESS to date. This may have changed, but I don't have anything I can confirm |
| 13:58.22 | djMax | the list claims it's now supported for national as well (which I have) |
| 13:58.37 | [TK]D-Fender | djMax: "The TBCT part supposedly is irrespective of how it got there." <- huh? |
| 13:58.45 | [TK]D-Fender | djMax: and what list? |
| 13:58.57 | djMax | but the confusing response was this: Would I âtriggerâ it via a Dial command or commands, or via Transfer? Neither... it happens automagically! Some time after the second leg of the call has answered, Asterisk will send a facility message to the CO switch |
| 13:59.01 | djMax | asterisk-users |
| 13:59.25 | djMax | "neither" obviously makes no sense, so I assumed he meant "doesn't matter" |
| 13:59.39 | [TK]D-Fender | djMax: vendor does not just "know", YOU have to tell them. |
| 13:59.43 | eppigy | I like stroganoff |
| 13:59.47 | Katty | eppigy: you're welcome to come over for dinner. |
| 13:59.47 | eppigy | of all types |
| 13:59.51 | [TK]D-Fender | djMax: the only tool * offers is a telco redirect. |
| 13:59.53 | Katty | eppigy: there's like... 8 servings in there. |
| 13:59.57 | eppigy | :D |
| 14:00.05 | [TK]D-Fender | djMax: there is no mid-stram hand-off |
| 14:00.07 | eppigy | I would def swing by |
| 14:00.16 | eppigy | if it wouldnt involve a commmercial airplane |
| 14:00.31 | djMax | TBCT is inherently midstream as I read it. |
| 14:00.35 | djMax | in Avaya manuals as well |
| 14:00.45 | Katty | eppigy: yeah you'd be better off buying the 4 ingredients and following recipe directions |
| 14:00.53 | Katty | eppigy: do you have a crockpot? |
| 14:01.01 | bpgoldsb | Anyone familiar with call parking, and how to setup hints for it? |
| 14:01.02 | eppigy | but I wouldnt have such good comapny ^_____^ |
| 14:01.10 | eppigy | I do not have a corck pot |
| 14:01.17 | [TK]D-Fender | djMax: Transfer() is it. All we've got. |
| 14:01.19 | Katty | you should get one. |
| 14:01.24 | eppigy | I agree |
| 14:01.28 | Katty | they're not expensive...they're sold everywhere. |
| 14:01.33 | Katty | you can make /so/ much stuff in them too |
| 14:01.35 | Katty | easy dinner. |
| 14:01.45 | Katty | you can even get really really lazy |
| 14:01.53 | [TK]D-Fender | Katty: LIES :p |
| 14:01.53 | Katty | canned chicken, drained, and bbq sauce |
| 14:01.55 | Katty | simmer all day |
| 14:02.01 | Katty | deposit on bun |
| 14:02.01 | [TK]D-Fender | Katty: Still takes more than 2 steps! |
| 14:02.22 | czindy | when I create a databse for IAX2 clients for asterisk realtime, is it possibe to define: qualifysmoothing, qualifyfreqok, qualifyfreqnotok columns? |
| 14:02.38 | Katty | [TK]D-Fender: it takes minimal effort compared to making dinner for an hour |
| 14:02.45 | eppigy | DANG SON |
| 14:02.55 | djMax | should "dahdi show channels" tell me which PRI channels are in use? Because when I have a bridged call up it only shows 1 with an extension |
| 14:03.13 | Katty | [TK]D-Fender: drive through is still easier, i guess. |
| 14:03.16 | *** join/#asterisk KavanS (n=KavanS@static-71-117-242-28.ptldor.dsl-w.verizon.net) |
| 14:03.52 | [TK]D-Fender | Katty: for me its "grab item from freezer, put in microwave. |
| 14:03.54 | Katty | [TK]D-Fender: and more expensive. |
| 14:04.11 | [TK]D-Fender | Katty: Or "open tin of soup, apply heat" |
| 14:04.13 | eppigy | boston market makes some great tv dinners |
| 14:04.24 | Katty | [TK]D-Fender: soup )= |
| 14:04.25 | djMax | so apparently TBCT is not compiled into * by default, can anyone else confirm/deny? |
| 14:04.32 | [TK]D-Fender | Katty: Expensive? I got a crate of Campbell's @ 1$ ea |
| 14:04.33 | Katty | unless it's chowder, i don't want it. |
| 14:04.44 | eppigy | MRE'S |
| 14:04.46 | [TK]D-Fender | djMax: Where do you see this? |
| 14:04.55 | Katty | colby corn chowder, in crockpot! |
| 14:04.58 | djMax | list: Also someone should have told you that the 2BCT code is by default not compiling and you could enable it by editing chan_dahdi.c |
| 14:05.05 | Katty | cream corn, potatos, milk, lots of colby cheese... |
| 14:05.08 | Katty | and smooshed up |
| 14:05.39 | [TK]D-Fender | djMax: Link please |
| 14:05.55 | djMax | it was in email, not sure it has a link yet, but I'll dig. |
| 14:06.27 | pmhaddad-work | :o |
| 14:06.33 | djMax | http://lists.digium.com/pipermail/asterisk-users/2009-April/230467.html |
| 14:06.39 | pmhaddad-work | why did i lose dahdi in asterisk :| |
| 14:06.40 | pmhaddad-work | weird |
| 14:06.52 | pmhaddad-work | recompiles |
| 14:07.15 | Katty | phix: DOOM |
| 14:07.17 | Katty | oh |
| 14:07.26 | Katty | s/phix/pmhaddad-work/ |
| 14:07.54 | Katty | i wonder if spaghetti can be made in the crockpot |
| 14:08.02 | eppigy | D: |
| 14:08.11 | Katty | that'd be like..what, 8 hours of simmering? |
| 14:08.13 | *** join/#asterisk axisys (n=axisys@117.18.231.108) |
| 14:08.17 | [TK]D-Fender | djMax: I'm not aware of anywhere to set whether a call is to be handed off, and I'm sure that it needs to be a choice. |
| 14:08.30 | djMax | now, to induce everybody's pointing an laughing, I'm using * straight from yum, not compiled by me, so I guess I have to recompile dahdi at least. |
| 14:08.33 | [TK]D-Fender | djMax: Just because they COULD be 2BCT-d, doesn't mean they should. |
| 14:08.57 | Katty | i've never yummed/apt-getted asterisk before. |
| 14:09.05 | Katty | so not sure :< |
| 14:09.07 | pmhaddad-work | Katty, i know. idk what i did lol |
| 14:09.13 | djMax | I hear you, I wondered that too. A previous message said that you could set options to STOP it, but it would work by default: http://lists.digium.com/pipermail/asterisk-users/2009-April/230442.html |
| 14:09.32 | djMax | "Asterisk and zaptel(Dahdi) can handle it and will always attempt to do 2BCT on ALL native bridging of channels on the same trunk group" |
| 14:10.19 | mweichert | hello guys and girls |
| 14:10.21 | pmhaddad-work | Katty, OOOH i was stupid |
| 14:10.29 | Katty | pmhaddad-work: oh? |
| 14:10.38 | pmhaddad-work | chan_dahdi.c:8558 mkintf: Signalling requested on channel 1 is FXO Kewlstart but line is in FXS Kewlstart signalling |
| 14:10.42 | [TK]D-Fender | djMax: "I have set up 2BCT for two different call center clients before, and neither implementation went smoothly as far as the carrier's part of it was concerned. Asterisk and zaptel(Dahdi) can handle it and will always attempt to do 2BCT on ALL native bridging of channels on the same trunk group if you have Transfer=yes in zapata.conf(or the Dahdi equivelent ile). I have never configured... |
| 14:10.44 | [TK]D-Fender | ...2BCT on an Asterisk 1.6 system, only 1.2 and 1.4 (using zaptel 1.4 for both), although I can't see any reason why it wouldn't work." - from post |
| 14:10.44 | pmhaddad-work | kicks self |
| 14:10.50 | Katty | pmhaddad-work: ha! |
| 14:11.17 | djMax | yep. I've beaten on the carrier, they assure me it's turned on. |
| 14:11.46 | mweichert | in a .call file, can I specify a macro to dial instead of a channel? |
| 14:12.02 | djMax | Clearly I don't believe that, but at least I need to be sure my side is setup properly as far as I can tell... I don't even see "attempts" at 2BCT in the PRI debug output, but need to look more |
| 14:12.21 | [TK]D-Fender | mweichert: Go read the list of Asterisk channel types a few dozen times and consider asking again :) |
| 14:12.47 | mweichert | [TK]D-Fender, ok... do you know off hand where I might find such a list? |
| 14:13.21 | mweichert | [TK]D-Fender, n/m, I think I found something. Thanks |
| 14:13.35 | [TK]D-Fender | djMax: And from the "transfer=yes" + "Native Bridging", it sounds like it should "jsut work" with a regular dial... UNLESS * has a reason to stay in the path. Check that you aren't setting and crazy dial options. |
| 14:13.54 | djMax | straight Dial(DAHDI/g0/number) |
| 14:14.18 | djMax | even without 2bct, what's confusing me is why dahdi show channels only shows one taken |
| 14:14.28 | mweichert | [TK]D-Fender, the syntax for SIP is: Syntax: SIP/extension[@hostname[:port]] |
| 14:14.40 | mweichert | [TK]D-Fender, can hostname be the name of a trunk defined in FreePBX? |
| 14:16.18 | [TK]D-Fender | mweichert: ..... Go look at what FreePBX does when you pleace a call. THAT is what you can put in "channel" |
| 14:16.33 | [TK]D-Fender | djMax: Can't say |
| 14:17.37 | *** join/#asterisk omer (n=_omer@119.152.10.22) |
| 14:17.41 | omer | hello |
| 14:17.54 | Katty | ohaider |
| 14:18.06 | omer | PLAYBACK answers the call and then play "all circuits are busy now" |
| 14:18.26 | djMax | so slipstreaming a new dahdi into a yum'ed asterisk: easy/hard? |
| 14:18.31 | djMax | does yum drop the source somewhere? |
| 14:19.44 | [TK]D-Fender | djMax: Should be fine, though I'd remove the yum version first. that's how it got in, that's how it should leave. |
| 14:20.13 | omer | how to play "All circuits are busy now" without answering the call ? |
| 14:20.28 | [TK]D-Fender | omer: What is the call coming in on? |
| 14:22.42 | bpgoldsb | Anyone familiar with call parking, and how to setup hints for it? Specifically using AEL/Ast 1.6 |
| 14:22.57 | djMax | I'll file that under "things not to do with 10 active calls" |
| 14:23.33 | omer | i am getting calls over the SIP and dialing them out |
| 14:23.39 | mweichert | can a .call file include comments? |
| 14:24.01 | [TK]D-Fender | omer: Describe the flow of this call. |
| 14:24.30 | [TK]D-Fender | mweichert: Comments? What for? The file is generally grabbed instantly by *, what use are comments? |
| 14:24.51 | omer | SOFTPHONe (ASTERISK) Asterisk dialing out using a Route ..... only 2 channels are available.....if this is a third call then play CIRCUIT BUSY but dont anwer the call. |
| 14:25.04 | mweichert | [TK]D-Fender, when testing, I'd like to 'comment out' a linke |
| 14:25.06 | mweichert | *line |
| 14:25.17 | omer | i can do it but i dont know how to play a file without answering the call |
| 14:25.34 | [TK]D-Fender | omer: its a SOFT-PHONE. What is the point of not playing it as standard audio and answering? |
| 14:25.41 | codefreeze-lap | bpgold: see the voipinfo wiki http://www.voip-info.org/wiki/view/Asterisk+AEL2#Extensions |
| 14:25.58 | omer | he will get charged if call is answered... |
| 14:26.15 | [TK]D-Fender | omer: How does a soft-phone get "chaged"? |
| 14:26.23 | [TK]D-Fender | charged* |
| 14:26.30 | *** join/#asterisk jmacz (n=jmacz@190.144.75.22) |
| 14:26.57 | codefreeze-lap | uh, bpgoldsb: see the voipinfo wiki http://www.voip-info.org/wiki/view/Asterisk+AEL2#Extensions |
| 14:27.01 | omer | well. I want to play an audio without answering.... |
| 14:27.19 | [TK]D-Fender | omer: "core show application playback" |
| 14:27.32 | *** join/#asterisk blebleble (n=godie@216.222.194.7) |
| 14:27.52 | omer | I have just tried playback ... NO AUDIO ...... background gives audio... |
| 14:27.53 | *** join/#asterisk seanmh (n=johndoe@198.59.129.24) |
| 14:28.09 | [TK]D-Fender | omer: Show me. |
| 14:28.26 | bpgoldsb | codefreeze-lap, thanks, will do |
| 14:28.26 | [TK]D-Fender | omer: and your softphone has to support this feature as well. |
| 14:28.32 | [TK]D-Fender | omer: Which is not *'s problem |
| 14:30.28 | eppigy | 8[] |
| 14:30.59 | omer | ok let me try |
| 14:31.20 | omer | I got it .... why there is not any audio..... let me check playback details. |
| 14:33.48 | mweichert | if I'm dialing a number over a SIP ITSP channel using a .call file, do I have to send a username, secret, or registration attempt? |
| 14:34.50 | mosty | mweichert, normally that would be in sip.conf |
| 14:35.45 | mweichert | mosty, ok - I wasn't sure if sip.conf was used at all when using a .call file |
| 14:36.30 | [TK]D-Fender | mweichert: you dial it EXACTLY the way you would is a normal Dial() command |
| 14:36.32 | mosty | mweichert, typically you put all your sip accounts in sip.conf, and give them each a name. then you use those names in extensions.conf and call files and other places |
| 14:37.32 | mweichert | [TK]D-Fender, hmm, I don't know why my ITSP trunk won't pick up the call then: Channel SIP/link2voip-sw1-0d8cf000 was never answered. |
| 14:38.06 | [TK]D-Fender | mweichert: and I don't SEE what you tried putting in, or the debug of a failed call in a pastebin either |
| 14:38.11 | [TK]D-Fender | ~pb |
| 14:38.12 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines 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 , or , http://bin.cakephp.org/ |
| 14:38.33 | mweichert | [TK]D-Fender, fair enough. :) |
| 14:40.02 | *** join/#asterisk jtodd (n=jtodd@71.sub-75-254-113.myvzw.com) |
| 14:40.02 | *** mode/#asterisk [+o jtodd] by ChanServ |
| 14:40.14 | mweichert | do priorities have to be in the same order? for example, can I define priority 4 first, and then priority 1? |
| 14:41.15 | *** join/#asterisk telnettech (n=telnette@gw.percipia.com) |
| 14:41.22 | Katty | telnettech: ohaiagain |
| 14:41.40 | telnettech | question.......can 1.4.24 play mp3 formatted files for moh |
| 14:42.58 | telnettech | im being told that it cant....so i want to verify....reason for the question |
| 14:44.59 | *** join/#asterisk oej (n=olle@80.251.192.3) |
| 14:46.22 | Katty | telnettech: idk, i've always used sox to convert them to gsm. |
| 14:48.08 | mweichert | [TK]D-Fender, could I get you to take a look here if you don't mind, please, http://pastie.org/447329 |
| 14:48.43 | mweichert | [TK]D-Fender, what I'm trying to do is dial extension 1111 which then places a call to 15193456789 and plays back a soud |
| 14:48.45 | mweichert | *sound |
| 14:49.05 | *** join/#asterisk UQlev (n=yuriy@nb11-125.static.cytanet.com.cy) |
| 14:49.52 | [TK]D-Fender | mweichert: it "looks" like they ITSP says the person you are calling is "busy". this would be jsuts fine. What is definitely not fine however is that I don's see any SIP DEBUG in that PB |
| 14:50.43 | *** join/#asterisk tfrew (n=tfrew@office.neteasyinc.com) |
| 14:50.48 | tfrew | hi |
| 14:51.40 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) |
| 14:52.25 | telnettech | TK: can asterisk ver 1.4.24 play mp3 files for music on hold |
| 14:53.05 | [TK]D-Fender | telnettech: Yes |
| 14:53.14 | telnettech | thats what i thought.....thanks |
| 14:57.42 | *** join/#asterisk ccesario (n=ccesario@189-19-9-100.dsl.telesp.net.br) |
| 14:57.58 | *** join/#asterisk SkywaIker (n=pirch@58.147.17.166) |
| 15:00.06 | *** join/#asterisk mog (n=mog@c-68-62-169-246.hsd1.al.comcast.net) |
| 15:00.06 | *** mode/#asterisk [+o mog] by ChanServ |
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| 15:00.22 | *** join/#asterisk TuKao (n=tuk@189-54-251-22-nd.cpe.vivax.com.br) |
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| 15:01.18 | *** part/#asterisk TuKao (n=tuk@189-54-251-22-nd.cpe.vivax.com.br) |
| 15:01.27 | *** join/#asterisk axarob (n=ebash@cpc3-barn8-0-0-cust288.brnt.cable.ntl.com) |
| 15:01.32 | mweichert | [TK]D-Fender, this is the abnormality I get in the log: http://pastie.org/447358 |
| 15:02.08 | [TK]D-Fender | mweichert: Thats a common response to codec mismatch. |
| 15:02.27 | [TK]D-Fender | mweichert: Could be something else, but then again it'd be nice to see the FULL dialog |
| 15:05.04 | mweichert | [TK]D-Fender, okay, sorry - thought I was helping by getting rid of the other stuff for you.. http://pastie.org/447365 |
| 15:07.49 | *** join/#asterisk grEvenX (n=even@cB78D5AC1.dhcp.bluecom.no) |
| 15:09.36 | mweichert | [TK]D-Fender, here is the log of a successful call using the same trunk: http://pastie.org/447373 |
| 15:09.40 | [TK]D-Fender | mweichert: https://www.link2voip.com/faq/index.html#faq28 |
| 15:11.10 | [TK]D-Fender | mweichert: Your call-file shows taht for whatever eason only ALAW is added and they don't support it |
| 15:24.57 | Katty | quick! |
| 15:25.06 | Katty | i need a dish. |
| 15:25.11 | Katty | name something yummy! |
| 15:25.34 | *** join/#asterisk angryuser (n=angryuse@LPuteaux-151-42-35-99.w193-251.abo.wanadoo.fr) |
| 15:25.46 | [TK]D-Fender | Katty: Eliza Dushku :p |
| 15:26.11 | Katty | [TK]D-Fender: aren't you just phoney |
| 15:26.23 | *** join/#asterisk knarfly (n=vladt@72-17-192-21.ftl.fdn.com) |
| 15:26.36 | [TK]D-Fender | isn't "that bad", he's actually far worse, its just the limited perception of others :p |
| 15:26.37 | *** join/#asterisk jstoker (n=jstoker@80-41-183-84.dynamic.dsl.as9105.com) |
| 15:26.46 | knarfly | can anyone offer some advice on getting dahdi configured? |
| 15:26.48 | Katty | mhmm, sure it is. |
| 15:27.43 | [TK]D-Fender | Katty: http://www.huffingtonpost.com/2009/04/13/naked-padma-lakshmi-chels_n_186090.html |
| 15:28.04 | knarfly | Unable to create channel of type 'DAHDI' keeps appearing when I try to dial out |
| 15:28.24 | [TK]D-Fender | Katty: And no... not phony at all, though some airbrushing(s/$/Photoshop) is as always, expected... |
| 15:29.21 | beek | [TK]D-Fender: nice... |
| 15:29.42 | *** part/#asterisk jstoker (n=jstoker@80-41-183-84.dynamic.dsl.as9105.com) |
| 15:30.32 | [TK]D-Fender | beek: People can't handle my kind of honesty :p |
| 15:30.47 | tfrew | i have a problem with cdrreport |
| 15:31.01 | tfrew | when somone calls in, hits ring group a, and a line picks up |
| 15:31.09 | tfrew | the call recording shows up fine |
| 15:31.23 | [TK]D-Fender | tfrew: cdrreport? Doesn't sound like an * problem to me... |
| 15:31.34 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 15:31.48 | *** join/#asterisk blitzrage (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 15:31.48 | *** mode/#asterisk [+o blitzrage] by ChanServ |
| 15:31.54 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 15:32.19 | Katty | hugs blitzrage |
| 15:32.25 | [TK]D-Fender | blitzrage: NA NA NA NA NA! |
| 15:32.48 | [TK]D-Fender | LEIF IS LEIF! |
| 15:33.13 | tfrew | what group creates cdrreport |
| 15:33.14 | Katty | mister madsen does not appreciate your capslock. |
| 15:33.54 | *** join/#asterisk Assimilate (n=Assimila@72.22.242.66) |
| 15:34.16 | *** join/#asterisk tmjb (n=tmjb@212.200.239.230) |
| 15:34.54 | blitzrage | lol |
| 15:34.55 | djMax | would the pastebin'ing of pri debug span have secure info in it other than phone numbers? |
| 15:35.19 | [TK]D-Fender | Katty: I'm singing, and this SM58 has no gain on it! |
| 15:35.30 | [TK]D-Fender | kicks Shure in the nads |
| 15:35.32 | blitzrage | does not appreciate his linksys doing a route flap |
| 15:35.54 | [TK]D-Fender | djMax: No, and nobody would care either way. |
| 15:36.00 | Katty | harsh :< |
| 15:36.14 | djMax | clearly true, but the web is forever. :) |
| 15:36.20 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 15:36.44 | [TK]D-Fender | djMax: We juts usually want people to shut up about their whining. "Help" is the preferred medium :p |
| 15:36.59 | [TK]D-Fender | djMax: Set an expirey on your PB's |
| 15:36.59 | tfrew | when a call comes in to an extension, and this is transfered to another extension, cdrreport shows the call recordings for the original call only |
| 15:37.21 | tfrew | (like it shows three calls, but points to the wav file from the first call only) |
| 15:37.32 | djMax | did that, but was going to send to the list too so figured maybe the next person that wants 2bct working might want to see it |
| 15:38.12 | djMax | here's the transfer convo for anyone who is still with me. http://pastebin.com/m3fa5b291 |
| 15:38.25 | *** join/#asterisk lesouvage (n=lesouvag@82.73.69.76) |
| 15:39.22 | *** join/#asterisk ghento (n=ghento@CPE0014bfaf7d7a-CM00159a03312c.cpe.net.cable.rogers.com) |
| 15:40.01 | *** join/#asterisk MindTheGap (n=MindTheG@mail.lpj.com.br) |
| 15:40.48 | *** join/#asterisk marv[work] (n=timr@router.asteriasgi.com) |
| 15:40.49 | [TK]D-Fender | tfrew: "cdrreport" is not supported here. Go check to see what they offer for it |
| 15:41.43 | blebleble | is there a single overall stats application (cdr reports, queue stats etc) commercial or free that is any good? |
| 15:43.12 | [TK]D-Fender | djMax: configs please. |
| 15:43.34 | Katty | [TK]D-Fender: do you think shallots would do okay in a crockpot? |
| 15:43.42 | Katty | [TK]D-Fender: REF: http://www.recipezaar.com/Crock-Pot-Special-209398 |
| 15:43.42 | [TK]D-Fender | Katty: Sure |
| 15:44.12 | [TK]D-Fender | Katty: "1 (11 ounce) can cheddar cheese soup" <- wtf |
| 15:44.29 | djMax | dahdi, anything else? |
| 15:44.31 | Katty | yeah i'm not sure. i'm using celery soup instead. |
| 15:44.42 | [TK]D-Fender | djMax: that'll do |
| 15:44.44 | *** join/#asterisk zchaos (n=none@CPE001d7ef0ba9d-CM001ceab63f9a.cpe.net.cable.rogers.com) |
| 15:44.50 | Katty | [TK]D-Fender: putting a carrot, shallots, and maybe some other veggies in there. not sure. |
| 15:44.53 | *** join/#asterisk ttl- (n=patrick@d5153AE82.access.telenet.be) |
| 15:45.21 | *** join/#asterisk james4765 (n=james476@office.neteasyinc.com) |
| 15:45.34 | djMax | http://pastebin.com/d641404b7 |
| 15:45.59 | *** join/#asterisk digitalirony (i=digitali@my.grandma.uses.shellium.org) |
| 15:46.44 | [TK]D-Fender | djMax: Your configs are broken, everything after "channel =" is IGNORED |
| 15:47.08 | james4765 | trying to debug a cdr php script, and have a question about the uniqueid it puts out |
| 15:47.55 | djMax | ok, i'll reorder. Does that group=63 line even make sense? |
| 15:48.07 | djMax | I think freepbx made this, but could've been me in a drunken stupor. |
| 15:48.14 | *** join/#asterisk tobias (n=tobias@cpe-071-070-219-040.nc.res.rr.com) |
| 15:48.39 | james4765 | some of the recordings end up with the second part of the unique id different than the unique id redorded in the database |
| 15:48.49 | djMax | asterisk restart will pick up this config fix right? or need a reboot? |
| 15:49.22 | djMax | hey! much better error. |
| 15:49.38 | djMax | Unable to handle return result on switchtype 1. |
| 15:52.39 | djMax | somebody mentioned 5ess switches could handle this, worth treating it as one even though it's national? |
| 15:52.56 | knarfly | djMax: I'm getting Unable to create channel of type 'DAHDI' even though just about everything else shows dadhi is ok...any advice? |
| 15:53.14 | djMax | wow, somebody targeting a question at me. Trouble for sure. |
| 15:53.28 | *** join/#asterisk Urthwhyte (n=urthwhyt@0x5da320aa.cpe.ge-1-1-0-1101.oebrnqu2.customer.tele.dk) |
| 15:53.29 | *** join/#asterisk seb- (n=seb@li30-51.members.linode.com) |
| 15:53.30 | djMax | what does dahdi show channels say? |
| 15:53.48 | seb- | [TK]D-Fender: are you there? |
| 15:53.59 | [TK]D-Fender | seb-: yup |
| 15:54.01 | knarfly | <PROTECTED> |
| 15:54.08 | knarfly | but nothing else...! |
| 15:54.09 | seb- | [TK]D-Fender: saw my email? |
| 15:54.12 | [TK]D-Fender | seb-: may be avail for a short time tonight |
| 15:54.26 | [TK]D-Fender | seb-: Yeah, very late, I didn't make it home last night |
| 15:54.55 | [TK]D-Fender | knarfly: And the reason we should trust your configs are right is....? |
| 15:55.03 | seb- | [TK]D-Fender: may not need to both be present at same time if email works for you....that may be more convenient for you if that works |
| 15:55.23 | [TK]D-Fender | seb-: ok, I'll test when I'm home if its up. |
| 15:55.24 | djMax | here's the new pri debug output http://pastebin.com/d74343e8 |
| 15:55.32 | knarfly | [TK]D-Fender: I copied over my zaptel.conf and zapata.conf to dahdi config files |
| 15:55.40 | seb- | [TK]D-Fender: in that case i'll turn it back on |
| 15:55.41 | knarfly | it worked with zaptel just fine |
| 15:55.54 | djMax | knarfly, pastebin them |
| 15:56.02 | knarfly | standy |
| 15:56.21 | *** join/#asterisk axisys (n=axisys@117.18.231.188) |
| 15:56.44 | [TK]D-Fender | knarfly: Do you go putting Honda parts in your Ford too? |
| 15:57.00 | [TK]D-Fender | knarfly: And no, that doesn't make me trust them any more. |
| 15:59.53 | Qwell | [TK]D-Fender: that doesn't work? |
| 15:59.53 | djMax | Doesn't Honda own Ford now? :) |
| 15:59.55 | [TK]D-Fender | djMax: Ask again next week ;) |
| 16:00.14 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 16:00.19 | djMax | I suppose Ford is the only one doing ok. |
| 16:01.51 | *** join/#asterisk Victor_Yure (n=victor@unaffiliated/victoryure/x-837844) |
| 16:03.34 | *** join/#asterisk bgmarete (n=marebri_@196.201.210.130) |
| 16:03.54 | knarfly | ok can someone tell me where I find the link once I pastebin.com? |
| 16:04.04 | knarfly | I don't see where it assigned a number or anything |
| 16:04.14 | djMax | browser bar |
| 16:04.26 | djMax | it just redirects to that page |
| 16:04.39 | [TK]D-Fender | it IS that page |
| 16:04.40 | knarfly | http://pastebin.com/d9077d51 |
| 16:04.54 | *** join/#asterisk jstoker (n=jstoker@80-41-183-84.dynamic.dsl.as9105.com) |
| 16:04.58 | djMax | (do they allocate the id before you post?) |
| 16:04.59 | knarfly | thanks....it works differnt that what I was used to |
| 16:05.07 | *** join/#asterisk imranimran (i=7c6d3119@gateway/web/ajax/mibbit.com/x-b1332883dc78b1c9) |
| 16:05.31 | imranimran | hi how to setup trunk in pri line |
| 16:05.52 | [TK]D-Fender | knarfly: Where is the dump from CLI? results of dahdi_cfg -vvvv? verification that the module is loaded? |
| 16:06.06 | [TK]D-Fender | knarfly: and everything under your "channel=>1" gets IGNOREd |
| 16:06.08 | *** part/#asterisk lbt (n=david@78.32.229.233) |
| 16:08.03 | imranimran | ? |
| 16:08.54 | knarfly | [TK]D-Fender: http://pastebin.com/m76cc6fae |
| 16:09.24 | knarfly | I wasn't aware they everything below "channel=>1" is ignored so I should place that last??? |
| 16:11.05 | *** part/#asterisk jstoker (n=jstoker@80-41-183-84.dynamic.dsl.as9105.com) |
| 16:14.24 | knarfly | has to attend a meeting... bbl |
| 16:15.31 | blitzrage | [TK]D-Fender: actually it should be better stated to say that everything above channel => 1 is assigned/configured for that channel |
| 16:17.16 | *** join/#asterisk grEvenX (n=even@cB78D5AC1.dhcp.bluecom.no) |
| 16:17.52 | *** join/#asterisk jameswf (n=james@unaffiliated/jameswf-home) |
| 16:19.24 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 16:21.53 | james4765 | anyone clueful about the mysql cdr recording about? |
| 16:22.12 | kc8pxy | which files is ./configure --with-dahdi=path saying i should point it at? I'm trying to move from working purely with SIP channels, to some hardware channels. i'm using 1.6.0.9. |
| 16:23.15 | james4765 | trying to tie the recoreded file to the database record, and have a question about the call unique id |
| 16:26.42 | Qwell | kc8pxy: Where did you install dahdi? |
| 16:27.54 | mosty | james4765, how are you initiating call recording? |
| 16:28.43 | james4765 | it's part of the trixbox system - I'm debugging the broken CDR |
| 16:28.52 | Qwell | james4765: That's why it's broken. |
| 16:29.07 | james4765 | I know - my boss tasked me with unbreaking it |
| 16:29.12 | Qwell | #trixbox |
| 16:29.29 | james4765 | this is a general asterisk question though |
| 16:29.39 | Qwell | No it isn't |
| 16:29.46 | *** join/#asterisk Victor_Yure (n=victor@unaffiliated/victoryure/x-837844) |
| 16:30.01 | james4765 | the unique id is generated from asterisk, right? |
| 16:30.18 | Qwell | It's generated via the Asterisk-based application that trixbox uses. |
| 16:30.34 | james4765 | so asterisk does not generate a unique id by default? |
| 16:30.50 | Qwell | mysql? by default, no |
| 16:31.01 | Qwell | at least not in 1.4.. 1.6 may be different |
| 16:31.34 | james4765 | ah. Well, that answers my question, at least... |
| 16:31.36 | kc8pxy | Qwell: yes, via "package", but being that i'm on gentoo, and ESPECIALLY with dahdi being a new package int the tree, it's compiled from source. i have the source in the filesystem. |
| 16:31.37 | james4765 | thanks |
| 16:31.41 | edoceo | I have a sip device registered, context = sip-inbound But the only items in that part of the dial-plan are s,1,Answer(); s,n,Playback(hellow) |
| 16:31.58 | Qwell | kc8pxy: you should be able to leave off the --with-dahdi, and it should pick it up |
| 16:32.00 | edoceo | When I try to dial from SIP I get a 404 extension not found - do I need a pattern to match anything dialed? |
| 16:32.20 | Qwell | james4765: but, we can't tell how they build that cdr app, so... #trixbox |
| 16:32.49 | kc8pxy | Qwell: ./configure and it should see it, then my make menuslect should let me ass dadhi-based options? |
| 16:33.05 | Qwell | blinks |
| 16:33.32 | Qwell | It would not let you ass them, no. :) It should let you see them, however. |
| 16:34.48 | kc8pxy | add |
| 16:34.57 | kc8pxy | gar.. sorry typo |
| 16:35.22 | Qwell | yes |
| 16:36.39 | *** join/#asterisk Kisu (n=k@daniel1117.broker.freenet6.net) |
| 16:41.04 | kc8pxy | Qwell: nice.. now i just need a dahdi card :) |
| 16:41.40 | *** join/#asterisk knarfly (n=vladt@72-17-192-21.ftl.fdn.com) |
| 16:42.37 | *** join/#asterisk jstoker (n=jstoker@80-41-183-84.dynamic.dsl.as9105.com) |
| 16:42.41 | knarfly | [TK]D-Fender: thanks, moving the channel=>1 to the end of the file seemed to have got it working. I'm at my office but via ssh I was able to console dial my cellphone and the darn thing worked. |
| 16:44.36 | imranimran | hmmm |
| 16:45.09 | *** part/#asterisk jstoker (n=jstoker@80-41-183-84.dynamic.dsl.as9105.com) |
| 16:45.50 | imranimran | i neeed help how i configure trunk and outbound routing i am using pri line |
| 16:46.59 | *** join/#asterisk propellerhead (n=yogurt2u@host110.190-136-239.telecom.net.ar) |
| 16:47.07 | knarfly | now to move over my sound files and I'm in business...no more chains from FreeBSD's porters forcing me to use their version of Asterisk |
| 16:47.46 | kc8pxy | knarfly: you have a freebsd asterisk box? |
| 16:48.08 | imranimran | some one help me:( |
| 16:48.28 | *** join/#asterisk deadpigeon (n=deadpige@office.xpressamerica.net) |
| 16:49.51 | imranimran | i m using TE122P pri card and configured it at trixbox machine card light is green its single span card but i dont know how to make possible to call outside |
| 16:50.36 | *** join/#asterisk naxxfish (n=fish@raptor.ukc.ac.uk) |
| 16:50.47 | naxxfish | evening all |
| 16:51.31 | naxxfish | question: asterisk-gui - what manager permissions does it need to give access to everything? |
| 16:51.36 | knarfly | kc8pxy: yes, but not anymore....I had to transfer over to Linux...too many strings attached to use the ports on FreeBSD |
| 16:51.43 | naxxfish | i've got write: system, call, log, verbose, command, agent, user, config |
| 16:51.49 | naxxfish | but it's complaingina still |
| 16:52.26 | imranimran | :( |
| 16:53.52 | blitzrage | naxxfish: see #asterisk-gui |
| 16:54.04 | imranimran | what about me |
| 16:55.22 | [TK]D-Fender | blitzrage: I might agree if those parms applied to another channel :) |
| 16:56.26 | Qwell | blitzrage: see msg! |
| 16:56.36 | Qwell | from like...an hour ago :D |
| 16:56.37 | blitzrage | Qwell: ya I saw it... and didn't really get it |
| 16:56.46 | blitzrage | I think russellb would think it was funnier |
| 16:56.55 | Qwell | You didn't get it? lame! |
| 16:57.00 | Qwell | ~!!! |
| 16:57.01 | infobot | it has been said that !!! is BANG BANG BANG at http://www.starterupsteve.com/swf/Group_X_video.html |
| 16:57.17 | Qwell | newb :p |
| 16:58.18 | imranimran | hello |
| 16:59.27 | hardwire | I need hawaii DID.. stat |
| 16:59.34 | hardwire | anybody wanna go in on a PRI sharing situation :) |
| 16:59.53 | imranimran | what i doo:( |
| 17:00.07 | imranimran | anybody tell me how to configure trunk and ooutgoing route |
| 17:00.29 | *** join/#asterisk jstoker (n=jstoker@80-41-183-84.dynamic.dsl.as9105.com) |
| 17:01.05 | UQlev | imranimran: *now? |
| 17:01.24 | *** part/#asterisk jstoker (n=jstoker@80-41-183-84.dynamic.dsl.as9105.com) |
| 17:01.26 | naxxfish | blitzrage: cheerz |
| 17:01.54 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 17:03.27 | Katty | [TK]D-Fender: well, i ended up getting 1lb of cubed beef stew meat, and 1lb of cubed pork stew meat. |
| 17:03.36 | Katty | [TK]D-Fender: the roast looked plain /awful/ |
| 17:03.55 | [TK]D-Fender | Katty: Stroganoff time! |
| 17:04.01 | Katty | [TK]D-Fender: no. |
| 17:04.08 | Katty | [TK]D-Fender: i'm having chicken stroganoff for dinner. so i guess yeah. |
| 17:04.18 | *** join/#asterisk nicoAMG (i=asgalt@201.203.96.42) |
| 17:04.38 | Katty | [TK]D-Fender: but i got a bag of shredded carrots, and some shallots to go with the cubed meats. and a packet of onion soup mix, mushroom soup and celery soup |
| 17:04.41 | [TK]D-Fender | imranimran: FreePBX and Trixbox are NOT supported here, Go ask in their channels |
| 17:04.55 | [TK]D-Fender | Katty: Ok, that sounds like a winner. |
| 17:05.09 | Katty | [TK]D-Fender: they had whole wheat pasta on sale too. so i'm going to serve it over that. |
| 17:05.11 | imranimran | but i want to know about trunk and outbound route configuration |
| 17:06.05 | [TK]D-Fender | imranimran>but i want to know about trunk and outbound route configuration <- this IS done via the GUI and nothing we want anything to do with |
| 17:08.52 | imranimran | ok:( |
| 17:08.55 | Katty | imranimran: how about some lemonade instead? |
| 17:09.13 | Katty | oh man. |
| 17:09.16 | eppigy | WHAT IS DO |
| 17:09.18 | Katty | i should make homemade limonade |
| 17:09.24 | Katty | hi dave. |
| 17:09.26 | eppigy | hi |
| 17:09.28 | eppigy | oh man |
| 17:09.31 | Katty | you missed a great shopping experience. |
| 17:09.35 | eppigy | :[ |
| 17:09.48 | eppigy | were guaps spent |
| 17:09.55 | eppigy | on cool stuffs |
| 17:09.57 | Katty | gaups? |
| 17:10.06 | eppigy | large amounts of disposable income |
| 17:10.08 | Katty | gaups does not parse. please try again. |
| 17:10.14 | Katty | i spent about 60 bucks. |
| 17:10.25 | eppigy | that is recession guap |
| 17:10.28 | Katty | not a whole lot. |
| 17:10.34 | Katty | it's about what i spend usually. |
| 17:10.38 | eppigy | yes |
| 17:10.39 | eppigy | good |
| 17:10.44 | eppigy | do not change your spedning habits |
| 17:10.50 | Katty | the recession isn't affecting me. |
| 17:10.53 | eppigy | YES |
| 17:11.03 | Katty | i even got crystal light candys. |
| 17:11.07 | eppigy | haha |
| 17:11.09 | Katty | economy is booming in kattyland. |
| 17:11.09 | eppigy | nice |
| 17:11.23 | eppigy | I am glad to hear it |
| 17:11.29 | Katty | but all the ingredients i was going to get for my roast thingy were on sale! |
| 17:11.40 | eppigy | I am also spending money like a I am trying to regain control of a region |
| 17:11.46 | eppigy | with a dug in resistance force |
| 17:11.52 | eppigy | with the support of the local populus |
| 17:12.07 | ghento | Hi all. For incoming calls, is there a variable that will contain the dialed number? |
| 17:12.17 | eppigy | ghento: are you serious? |
| 17:12.21 | Katty | umm. |
| 17:12.25 | eppigy | $EXTEN |
| 17:12.28 | Katty | that would be ${EXTEN} |
| 17:12.38 | eppigy | HOLLEA |
| 17:12.40 | eppigy | HOLLA |
| 17:12.41 | Katty | it goes next ot the ABC blocks. |
| 17:12.50 | Katty | in dave's crib. |
| 17:12.53 | eppigy | yes |
| 17:13.05 | eppigy | with the hanging duckies |
| 17:13.08 | Katty | i wonder if people feed their children pasta. |
| 17:13.13 | Katty | like... at the age where cheerios is all the rage |
| 17:13.27 | eppigy | these are things I do not know |
| 17:13.32 | Katty | i don't know either. |
| 17:13.37 | Katty | who has a kid? |
| 17:13.42 | [TK]D-Fender | Katty>that would be ${EXTEN} <- questionable, but I'll let it slide :p |
| 17:13.52 | Katty | [TK]D-Fender: i'll let YOU slide here in a minute. |
| 17:13.53 | Qwell | Katty: like a baby goat? |
| 17:14.01 | Katty | Qwell: no, like a real kid. |
| 17:14.11 | Katty | Qwell: otherwise known as screaming monster. |
| 17:15.07 | [TK]D-Fender | Katty>[TK]D-Fender: i'll let YOU slide here in a minute. <- you know full well what I can do to statements like that! |
| 17:15.23 | [TK]D-Fender | Katty: Little minx :p |
| 17:15.39 | Katty | oh sod off. |
| 17:16.00 | Katty | doesn't you have someone to be sarcastic at, besides me? :P |
| 17:16.06 | [TK]D-Fender | Katty: Misery-girl speaks brit aye? |
| 17:16.23 | Katty | irc has its influences. |
| 17:16.51 | eppigy | ALLO GOVNA |
| 17:16.54 | Katty | i didn't mean to start the flame war... |
| 17:17.22 | Katty | isn't that ello govna? |
| 17:17.37 | Katty | the redcoats are here! |
| 17:17.54 | Katty | jolly good! |
| 17:18.29 | eppigy | NOW BEHAVE YASELF WHAT YA GONNA DO |
| 17:19.09 | eppigy | youve gotta get in to this business proper mate |
| 17:19.23 | *** join/#asterisk astassistant (n=skip@65-126-63-1.dia.static.qwest.net) |
| 17:19.47 | Katty | i could use an assistant |
| 17:19.51 | Katty | to assist me. |
| 17:19.58 | Katty | [TK]D-Fender: no comments. |
| 17:21.29 | kc8pxy | knarfly: I've seen that lots of people say my native gentoo linux is similar to bsd, and i have no issues with asterisk. welcome to linux :) what distro, if you don't mind? |
| 17:23.34 | eppigy | Katty: what uh |
| 17:23.41 | eppigy | what do you need assistance with |
| 17:23.50 | Katty | idk |
| 17:23.56 | Katty | finding my sanity? |
| 17:23.59 | eppigy | yes |
| 17:24.04 | eppigy | I need a request buffer |
| 17:24.11 | eppigy | and sny remark maker |
| 17:24.41 | Katty | you should see [TK]D-Fender about that. |
| 17:25.52 | *** join/#asterisk grEvenX (n=even@cB78D5AC1.dhcp.bluecom.no) |
| 17:26.39 | *** join/#asterisk zchaos (n=none@CPE001d7ef0ba9d-CM001ceab63f9a.cpe.net.cable.rogers.com) |
| 17:27.10 | Katty | hmm. made me think nachos. |
| 17:27.34 | kc8pxy | is looking for the next thing to test.. i have it working for sip peer to call, and get a test playback, and i got 2 peers to call each other. what should i try next? |
| 17:27.51 | Katty | meetme! |
| 17:27.55 | *** join/#asterisk Defraz (n=T0tal@fw-poky.fuzecore.com) |
| 17:28.31 | kc8pxy | is that in the oreilly book? |
| 17:28.39 | Katty | i would think so |
| 17:28.42 | Katty | infobot: meetme? |
| 17:29.26 | Katty | infobot: Meetme is a conference bridge. More info at http://www.voip-info.org/wiki-Asterisk+cmd+MeetMe |
| 17:29.26 | infobot | Katty: okay |
| 17:29.31 | Katty | infobot: Meetme? |
| 17:29.32 | infobot | it has been said that meetme is a conference bridge. More info at http://www.voip-info.org/wiki-Asterisk+cmd+MeetMe |
| 17:32.30 | knarfly | kc8pxy: I'm using Fedora 10 |
| 17:32.42 | *** join/#asterisk puzzled (n=foo@puzzled.xs4all.nl) |
| 17:33.42 | knarfly | I'm still a big fan of FreeBSD but the port for now includes a version of zaptel that won't work with my anolog X101P card. They say they had to do it and just suggested I upgrade my hardware. So I did...to Linux |
| 17:34.13 | knarfly | hey is there a reason the new Asterisk-1.6.0.9 would not recognize the WAIT command in my extensions.conf |
| 17:34.17 | *** join/#asterisk lanning (n=lanning@nat/yahoo/x-89349a1b17fffd37) |
| 17:35.12 | Katty | knarfly: maybe you mean WaitExten? |
| 17:35.17 | knarfly | I tried to call the asterisk box from an outside line. The CLI showed the call coming in but said No Application "Wait,20" for .... |
| 17:35.47 | knarfly | This is the same extensions.conf file I was using under *-1.4.21 |
| 17:36.21 | knarfly | [incoming] |
| 17:36.21 | knarfly | exten => s,1,Wait,20 ; Wait 20 seconds |
| 17:36.29 | Qwell | knarfly: old syntax. |
| 17:36.35 | Qwell | exten => s,1,Wait(20) |
| 17:36.49 | knarfly | no kidding, thanks... |
| 17:38.53 | *** join/#asterisk Victor_Yure (n=victor@unaffiliated/victoryure/x-837844) |
| 17:39.05 | *** join/#asterisk jstoker (n=jstoker@80-41-183-84.dynamic.dsl.as9105.com) |
| 17:39.54 | knarfly | Qwell: Thanks dude, that did the trick. Now I'm heading on without the FreeBSD limitations on Asterisk |
| 17:44.40 | *** join/#asterisk legis_ (n=JJ@unaffiliated/legis) |
| 17:46.27 | *** join/#asterisk dkdkd (n=vbvb@c-68-40-109-189.hsd1.mi.comcast.net) |
| 17:47.55 | dkdkd | hello, i have a simple question on asterisk -- |
| 17:48.20 | rob0 | 42 |
| 17:48.36 | dkdkd | i'm trying to ascertain if what i want to do is easy to setup |
| 17:48.53 | dkdkd | i have a piece of software that claims to integrate with asterisk for callouts |
| 17:48.56 | *** join/#asterisk jeffgus (n=jeffgus@green.zimage.com) |
| 17:49.24 | dkdkd | i think all i have to do is configure that software to use asterisk, and then configure asterisk with a VOIP<->PSTN provider so that it can ring real phones |
| 17:49.28 | dkdkd | is that correct? |
| 17:50.06 | [intra]lanman | dkdkd: what is the software? |
| 17:50.14 | dkdkd | http://www.alarmpoint.com |
| 17:51.07 | dkdkd | the end goal is that when an alarm is triggered in alarm point, it uses asterisk as its VOIP/SIP provider |
| 17:51.27 | dkdkd | but i need to configure that so that it can call a real phone. |
| 17:51.32 | [TK]D-Fender | dkdkd: "configure that software to use asterisk" <- with a lack of detail like this we can't answer |
| 17:51.55 | dkdkd | that part "configure that software to use askterisk" i don't need help with |
| 17:52.15 | dkdkd | the part i'm not sure about is how to configure asterisk so that it can callout into the PSTN |
| 17:52.25 | [TK]D-Fender | dkdkd: well we can't tell if * is capable of talking to it |
| 17:53.16 | [TK]D-Fender | dkdkd: Every call is jsut a call. that isn't the complicated part |
| 17:53.17 | dkdkd | D-Fender: i got that...i can set that up myself (config between alarmpoint and *). i just have never configured * so that is the part i'm unsure of |
| 17:53.41 | [TK]D-Fender | dkdkd: how do you intend on getting to the PSTN? |
| 17:54.02 | dkdkd | the question i have i guess, is that i think i need to sign up with some kind of provider who can switch into PSTN for me... |
| 17:54.59 | dkdkd | i'm guessing once i sign up with that provider, i provide asterisk with those details, and it forwards on to that, is that correct? |
| 17:55.26 | rob0 | Either start with the book or hire a consultant. |
| 17:55.50 | dkdkd | rob0 - its a pretty basic question, no? |
| 17:55.57 | ttl- | Hey |
| 17:55.58 | [TK]D-Fender | dkdkd: * takes calls from devices you set up with it (like your other software), nd processes that call in the dialplan. In your dialplan you can dial out to any other resource you have configured |
| 17:56.32 | [TK]D-Fender | dkdkd: Go download the book, and take a look at this guide as a SAMPLE of a relatively simple system |
| 17:56.33 | [TK]D-Fender | ~book |
| 17:56.34 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 17:56.37 | [TK]D-Fender | ~jerjerguide |
| 17:56.37 | infobot | [~jerjerguide ] Jeremy McNamara's quick sample guide to setting up * : http://www.jeremy-mcnamara.com/2007/02/26/how-to-configure-asterisk-your-first-installation/ |
| 17:56.45 | rob0 | Yes, it's too basic to be answered with more than a "yes". Your followups indicate that you have a huge learning curve ahead. Enjoy. |
| 17:57.58 | dkdkd | D-Fender, thanks for the links. I guess all I was asking for was a simple "yes", if you want to switch out to PSTN, one way to do it is to sign up with a provider who offers this service, and configure asterisk to use that. |
| 17:59.01 | dkdkd | does anyone recommend any providers of such a service. i know there is a huge list of them, but any recommendations are helpful |
| 17:59.24 | Qwell | ~itsplist-us |
| 17:59.25 | infobot | [~itsplist-us] Here are some popular ITSPs (USA) starting with the more respected ones : http://www.teliax.com , http://connect.voicepulse.com , http://www.nufone.net , http://www.broadvoice.com, http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net |
| 17:59.35 | dkdkd | thx |
| 17:59.54 | ttl- | I managed to get the SPA3102 ATA to work... The solution was to connect my LAN (on which my Asterisk box resides) to the WAN-port instead of the LAN-port and everything worked... Linksys logic??? |
| 18:00.13 | Qwell | ttl-: it's a switch. the WAN port is correct |
| 18:00.27 | ttl- | omg |
| 18:00.38 | [TK]D-Fender | ttl-: indeed |
| 18:00.40 | ttl- | Qwell: yeah but didn't find it logical |
| 18:00.42 | Qwell | the LAN port is for another device hanging off of it (a desktop for example) |
| 18:00.49 | Qwell | it's perfectly logical |
| 18:01.10 | [TK]D-Fender | ttl-: the SPA-3102 is considered CPE and expected to talk to a remote SIP provider. that means it only speaks SIP on the WAN port |
| 18:01.14 | ttl- | Qwell: it wasn't form me but ok :-)) |
| 18:01.31 | Qwell | ttl-: I would suggest reading up on some networking theory |
| 18:01.42 | ttl- | [TK]D-Fender: indeed, and i will ,ever forget :) |
| 18:01.43 | Qwell | it's fairly important if you're doing this kind of stuff |
| 18:01.52 | [TK]D-Fender | Qwell: Networking theory won't quite cover this, but close enough :) |
| 18:02.08 | [TK]D-Fender | ttl->[TK]D-Fender: indeed, and i will ,ever forget :) - Ever indeed |
| 18:02.09 | Qwell | [TK]D-Fender: that was nicer than saying "basic networking" |
| 18:02.25 | eppigy | fundamentals |
| 18:02.31 | Qwell | eppigy: there you go |
| 18:02.34 | [TK]D-Fender | Qwell: this is a lot more "device specific". Can't fault a lack of book-smarts on this one. |
| 18:02.43 | Qwell | [TK]D-Fender: well, any switch would behave the same |
| 18:03.03 | rob0 | I asked him the other day if he had ever done this before, and that question went unanswered. |
| 18:03.18 | [TK]D-Fender | eppigy: Yes, clearly the "fundamentals of his networking" are strong. Just ask Sen. McCain |
| 18:03.34 | eppigy | I am a tubes expert |
| 18:03.36 | [TK]D-Fender | Qwell: Switches have WAN ports? ;) |
| 18:03.41 | ttl- | [TK]D-Fender: I know enough about networking and stuff to manage but didn't understood the logic of the LAN- and WAN port |
| 18:03.49 | eppigy | integrated devices |
| 18:03.56 | Qwell | [TK]D-Fender: touche sir |
| 18:03.59 | eppigy | which perform switching packet forwarding |
| 18:04.04 | eppigy | and every thing do |
| 18:04.06 | ttl- | I will remember now |
| 18:04.24 | Qwell | so it's a router |
| 18:04.25 | [TK]D-Fender | eppigy: Yes... and my bicycle has air-condition. And by bicycle I mean car. And by air-conditioning I mean a cigarette lighter |
| 18:04.27 | pfn | ttl-, apparently, WAN is an easier term to understand than "uplink" |
| 18:04.43 | Qwell | pfn: uplink ports haven't existed since like 1998 |
| 18:04.47 | [TK]D-Fender | Qwell: Yes the SPA is abasic NAT router as well... |
| 18:04.53 | Qwell | we've got auto-sensing now :p |
| 18:04.55 | pfn | and it's a common thing across simple end-user/consumer electronics |
| 18:04.57 | ttl- | Anyway thanks for all the help from you guys |
| 18:05.05 | pfn | Qwell, for switches and hubs, yes |
| 18:05.06 | ttl- | pfn: yeah indeed |
| 18:05.12 | pfn | Qwell, but not for routers and such devices |
| 18:05.19 | [TK]D-Fender | Qwell: Completely meant for home provisioned CPE use by chumps. |
| 18:05.28 | pfn | Qwell, I suppose they call them all WAN |
| 18:05.30 | pfn | now, though |
| 18:06.03 | [TK]D-Fender | pfn: Nope, its all the same network segment. WAN implies a different one |
| 18:06.08 | ttl- | the SPA3102 did speak sip on the LAN port but only for the |
| 18:06.13 | ttl- | FXS port |
| 18:06.22 | [TK]D-Fender | ttl-: WAN, not FXS |
| 18:06.28 | [TK]D-Fender | 8sign* |
| 18:06.31 | eppigy | now that that TRABAJO |
| 18:08.38 | ttl- | [TK]D-Fender: the device has a FXS and FXO port, i mean i could connect to my phone (FXS port) but not to the PSTN (FXO port) with my asterisk box connected to the LAN port of the ATA device... |
| 18:08.57 | ttl- | now i can use both |
| 18:09.21 | [TK]D-Fender | .... |
| 18:09.53 | *** join/#asterisk Victor_Yure (n=victor@unaffiliated/victoryure/x-837844) |
| 18:10.06 | eppigy | u trippin |
| 18:10.13 | [TK]D-Fender | BOOYA |
| 18:10.44 | eppigy | DONDE ESTBOOYA |
| 18:11.29 | *** join/#asterisk w9sh (n=sph@adsl-068-209-117-205.sip.asm.bellsouth.net) |
| 18:11.33 | ttl- | [TK]D-Fender: Now with my asterisk box connected to the WAN (uplink) port i can control both the FXS and FXO port of the ATA with asterisk and i'm very happy now :) |
| 18:12.05 | *** join/#asterisk ocnarf (n=chatzill@125.252.90.5) |
| 18:12.59 | *** join/#asterisk SilentGreen (n=chatzill@201.217.56.16) |
| 18:15.42 | ocnarf | I need help in setting up my queue. What i need is when the caller is waiting he will hear moh and when there is an avail agent, the caller will hear a ring after the moh. here is my config http://pastebin.com/d1009bd26 |
| 18:16.15 | *** join/#asterisk flujan (n=flujan@189.111.254.251) |
| 18:16.27 | *** part/#asterisk flujan (n=flujan@189.111.254.251) |
| 18:16.37 | ocnarf | i've been trying but all i hear is moh, i tried using r option on queue but i wont work cause it will change moh to ring |
| 18:17.38 | [TK]D-Fender | ocnarf: Wow, and no queue configs... impressive |
| 18:19.47 | ocnarf | @D-Fender: oops, sorry.. here is the config @queues.conf http://pastebin.com/d2763b626 |
| 18:22.30 | *** part/#asterisk jstoker (n=jstoker@80-41-183-84.dynamic.dsl.as9105.com) |
| 18:25.06 | *** join/#asterisk codefreeze-lap (n=murf@72.21.67.40) |
| 18:27.23 | *** join/#asterisk nima0102 (n=nima@91.98.221.75) |
| 18:32.36 | *** join/#asterisk lppitsfwnyuphtjm (n=bell@dsl-kpobrasgw1-fe54df00-180.dhcp.inet.fi) |
| 18:32.47 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 18:33.29 | lppitsfwnyuphtjm | why I don't get any debug info about chan_mobile? I start asterisk with -ddddddddddddddddddddddddvvvvvvvvvvvvvvvvvvvvvvvvvvvvvr |
| 18:34.08 | *** join/#asterisk CheBuzz_Home (n=CheBuzz_@213-170-247.netrunf.cytanet.com.cy) |
| 18:34.11 | CheBuzz_Home | If I am getting "SIP/2.0 404 Not Found" when trying to make a call, what would be the problem? |
| 18:34.37 | _Steve_ | so i call out to my cell phone from my x-lite through my asterisk. answer call on cell. talk, etc. then hang up on cell. x-lite still thinks the call is connected. why? is this a problem with asterisk or x-lite? or something else? |
| 18:37.16 | *** join/#asterisk jstoker (n=jstoker@80-41-183-84.dynamic.dsl.as9105.com) |
| 18:37.55 | *** part/#asterisk jstoker (n=jstoker@80-41-183-84.dynamic.dsl.as9105.com) |
| 18:39.47 | *** join/#asterisk twitchnln (n=twitchnl@loopback-dktc.atl.oneringnetworks.net) |
| 18:40.02 | twitchnln | afternoon everyone |
| 18:40.40 | lppitsfwnyuphtjm | also, "mobile show devices" says SMS is not supported, but five different Nokia phones have been tested to work with SMS when using PCSuite |
| 18:41.43 | Qwell | lppitsfwnyuphtjm: PCSuite != Asterisk |
| 18:41.51 | twitchnln | i'm having a problem on one of my boxen in the field that is rejecting calls that are coming in number:npdi@server has anyone else seen this and if so how do i get it to pass calls to correct context? |
| 18:42.07 | Qwell | drmessano: stalker |
| 18:43.07 | *** join/#asterisk wierdo (n=jimmy@wifi-traf5.networx-bg.com) |
| 18:43.24 | *** join/#asterisk juanIMP (n=Juancho@200.71.41.22) |
| 18:43.42 | lppitsfwnyuphtjm | Qwell: I had a point. where does asterisk get the idea that sms is not supported? |
| 18:44.37 | [TK]D-Fender | CheBuzz_Home: Means it can't matcht he # dialed. |
| 18:44.48 | [TK]D-Fender | CheBuzz_Home: wHO is responding 404? |
| 18:46.37 | *** join/#asterisk fskrotzki_ (n=fskrotzk@cpe-74-74-245-250.rochester.res.rr.com) |
| 18:47.07 | CheBuzz_Home | Hm... it works... but I had to remove some lines from my extensions.conf: |
| 18:47.14 | CheBuzz_Home | exten => _1877.,3,Dial(SIP/+${EXTEN}@gizmo5) |
| 18:47.14 | CheBuzz_Home | exten => _1888.,4,Dial(SIP/+${EXTEN}@gizmo5) |
| 18:47.14 | CheBuzz_Home | exten => _X.,5,Dial(SIP/+${EXTEN}@voipdiscount) |
| 18:47.47 | CheBuzz_Home | If I comment out the gizmo5 stuff, and make voipdiscount priority 1, it seems to work |
| 18:48.52 | [TK]D-Fender | CheBuzz_Home: You clearly have no concept of dialplan priorities |
| 18:49.03 | CheBuzz_Home | You are absolutely correct |
| 18:49.06 | *** join/#asterisk ingenius (n=alektro@69.90.72.173) |
| 18:49.26 | [TK]D-Fender | [14:47]<CheBuzz_Home>exten => _1888.,4,Dial(SIP/+${EXTEN}@gizmo5) <- this is STEP 4 for the pattern _1888. |
| 18:49.39 | [TK]D-Fender | CheBuzz_Home: This does not have a step 1 |
| 18:49.54 | CheBuzz_Home | Ah... so the X matches all? |
| 18:49.57 | [TK]D-Fender | CheBuzz_Home: This is not an order to relate to other completely different patterns. |
| 18:50.07 | [TK]D-Fender | CheBuzz_Home: Steps for a given pattern. |
| 18:50.09 | [TK]D-Fender | ^^^^ |
| 18:50.18 | kc8pxy | CheBuzz_Home: i recommend the oreilly book for you. |
| 18:50.19 | [TK]D-Fender | CheBuzz_Home: go read chapter 5 intensley |
| 18:50.21 | [TK]D-Fender | ~book |
| 18:50.22 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 18:50.25 | [TK]D-Fender | ^^^^^6 |
| 18:50.28 | *** join/#asterisk AndyML (n=AndyML@209.17.170.166) |
| 18:50.52 | CheBuzz_Home | Definitely need to. This is day 3 with asterisk |
| 18:51.25 | kc8pxy | CheBuzz_Home: nice. |
| 18:51.26 | AndyML | does anyone have further information regarding the bug in 1.4.24's app_voicemail that sets the permissions on the .txt file such that it is inaccessible to the asterisk user? |
| 18:52.38 | CheBuzz_Home | One last question, do I need a Hangup after each of those? |
| 18:52.40 | *** join/#asterisk voxter (n=voxter@76.77.95.2) |
| 18:55.33 | blebleble | how can i change the default queue recording filenames? I see fname_base listed in the docs but not sure what my options are to set it. I would like the $date-$time-$extension (currently its $date-$time-8numbers that dont seem to be anything) |
| 18:57.22 | lppitsfwnyuphtjm | [Apr 15 21:55:11] DEBUG[11384] chan_mobile.c: rfcomm_write() (19) [AT+CNMI=3,1,0,0,0^M] |
| 18:57.25 | lppitsfwnyuphtjm | [Apr 15 21:55:11] DEBUG[11384] chan_mobile.c: [Nuuka] ERROR |
| 18:57.27 | lppitsfwnyuphtjm | [Apr 15 21:55:11] DEBUG[11384] chan_mobile.c: [Nuuka] error setting CNMI |
| 18:57.28 | *** join/#asterisk fskrotzki_ (n=fskrotzk@cpe-74-74-245-250.rochester.res.rr.com) |
| 18:57.29 | lppitsfwnyuphtjm | [Apr 15 21:55:11] DEBUG[11384] chan_mobile.c: [Nuuka] no SMS support |
| 18:57.55 | lppitsfwnyuphtjm | so, how'd I go about the SMS support now? |
| 18:58.45 | *** join/#asterisk Carlos_PHX (n=Carlos@ip68-3-162-244.ph.ph.cox.net) |
| 19:03.01 | *** join/#asterisk theHub (n=theHub@69.177.93.21) |
| 19:03.42 | *** join/#asterisk knarfly (n=vladt@72-17-192-21.ftl.fdn.com) |
| 19:07.59 | *** join/#asterisk Victor_Yure (n=victor@unaffiliated/victoryure/x-837844) |
| 19:08.50 | *** join/#asterisk jstoker (n=jstoker@80-41-183-84.dynamic.dsl.as9105.com) |
| 19:09.26 | Qwell | lppitsfwnyuphtjm: fix the SMS handling in chan_mobile for Nokia phones. |
| 19:09.50 | lppitsfwnyuphtjm | what util can I use to send "AT+CNMI=?" to the phone, and read the reply? |
| 19:09.50 | Qwell | without one, and without a "correct" trace, it isn't something we can ever fix |
| 19:10.02 | Qwell | minicom can connect via rfcomm I do believe |
| 19:10.18 | Qwell | or, actually, I think you can 'rfcomm send' from Asterisk |
| 19:10.27 | Qwell | mnicholson: ping? is that correct? |
| 19:10.28 | lppitsfwnyuphtjm | <4>[292179.084575] sysfs: duplicate filename 'hci0:46' can not be created |
| 19:10.38 | Qwell | lppitsfwnyuphtjm: try to do it from Asterisk |
| 19:11.27 | lppitsfwnyuphtjm | No such command 'rfcomm'. |
| 19:11.28 | lppitsfwnyuphtjm | safari*CLI> module load rfcomm |
| 19:11.28 | lppitsfwnyuphtjm | safari*CLI> help rfcomm |
| 19:11.29 | lppitsfwnyuphtjm | safari*CLI> |
| 19:11.35 | *** join/#asterisk freakazoid0223 (n=matt@pool-71-246-17-74.phlapa.fios.verizon.net) |
| 19:11.55 | Qwell | lppitsfwnyuphtjm: it's part of chan_mobile IIRC.. one sec |
| 19:11.58 | Qwell | 1.6? |
| 19:12.31 | Qwell | mobile rfcomm <device ID> <command> |
| 19:12.43 | *** join/#asterisk tamiel (n=tamiel@ip-176.net-89-2-117.rev.numericable.fr) |
| 19:12.49 | Qwell | so something like: mobile rfcomm bobsphone AT+CNMI=? |
| 19:12.50 | lppitsfwnyuphtjm | after "module load rfcomm" nothing seemed to work, I have 1.6.1 rc1 from fedora |
| 19:13.18 | Qwell | try that ^^ |
| 19:14.02 | Qwell | you'll need debugging turned on to see the output though I think |
| 19:14.18 | lppitsfwnyuphtjm | I can't type ? char..! ;( |
| 19:14.34 | lppitsfwnyuphtjm | Qwell: yes, it took a while to find I have to uncomment some lines in logger.conf |
| 19:14.42 | Qwell | d'oh |
| 19:14.46 | Qwell | umm |
| 19:15.02 | lppitsfwnyuphtjm | why is ? forbidden |
| 19:15.09 | Qwell | ? is tab completion |
| 19:15.26 | Qwell | you can try typing ? elsewhere and pasting it in.. not sure how that'll work |
| 19:15.36 | lppitsfwnyuphtjm | does not |
| 19:15.44 | Qwell | well, shit, heh |
| 19:16.00 | Qwell | I don't know what to tell you.. I've seen this before though |
| 19:16.52 | lppitsfwnyuphtjm | maybe compile without libedit etc? |
| 19:17.04 | Qwell | not possible |
| 19:17.06 | *** join/#asterisk ccesario (n=ccesario@189-19-9-100.dsl.telesp.net.br) |
| 19:17.17 | Qwell | can you stop asterisk and connect to it with miniicom? |
| 19:17.20 | Qwell | minicom* |
| 19:20.13 | lppitsfwnyuphtjm | ok do I need to bind the device to /dev/something with some application |
| 19:21.30 | Qwell | it's probably creating a tty |
| 19:22.31 | Qwell | oh, you have to rfcomm connect first, from the CLI |
| 19:22.38 | Qwell | (Linux CLI) |
| 19:22.43 | Qwell | http://pratyeka.org/rfcomm/ |
| 19:22.48 | *** join/#asterisk magronez (n=eusei@unaffiliated/magrao/x-2903) |
| 19:22.52 | *** join/#asterisk jstoker (n=jstoker@80-41-183-84.dynamic.dsl.as9105.com) |
| 19:23.08 | knarfly | so if I can console dial a number from the console, how can I converse with the caller once the call is connected? |
| 19:23.20 | *** join/#asterisk gswain (n=Administ@67-60-195-60.cpe.cableone.net) |
| 19:23.35 | gswain | where is a good place to buy sip trunks from for my asterisk pbx |
| 19:23.37 | knarfly | will this work through the sound card with my microphone and speakers? |
| 19:24.29 | *** part/#asterisk jstoker (n=jstoker@80-41-183-84.dynamic.dsl.as9105.com) |
| 19:26.04 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 19:28.39 | [TK]D-Fender | gswain: Depends on your needs & locations |
| 19:28.58 | [TK]D-Fender | knarfly: Yes, but do not go using * like a softphone. Install a proper one |
| 19:29.08 | lppitsfwnyuphtjm | Qwell: OK I did "rfcomm connect 0 00:... 1", then minicom for device /dev/rfcomm0, then: |
| 19:29.22 | lppitsfwnyuphtjm | AT+CNMI=? |
| 19:29.23 | lppitsfwnyuphtjm | ERROR |
| 19:29.59 | Qwell | do other commands work? |
| 19:30.00 | *** join/#asterisk nullable_type (n=nullable@hq.verbx.net) |
| 19:30.54 | lppitsfwnyuphtjm | ATZ |
| 19:30.55 | lppitsfwnyuphtjm | OK |
| 19:31.42 | Katty | deposits choclate chip muffin on jaytee's desk. |
| 19:32.33 | [TK]D-Fender | +++ATH0 |
| 19:32.37 | *** part/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 19:32.42 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 19:32.45 | [TK]D-Fender | OMFG |
| 19:32.46 | *** part/#asterisk twitchnln (n=twitchnl@loopback-dktc.atl.oneringnetworks.net) |
| 19:32.47 | *** kick/#asterisk [[TK]D-Fender!i=north@pdpc/sponsor/digium/Qwell] by Qwell (1994 hax) |
| 19:32.50 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 19:32.51 | Qwell | pwnt |
| 19:33.25 | Katty | he told you :P |
| 19:33.28 | [TK]D-Fender | Qwell: You were hoping to do that before I bounced on my own :) |
| 19:33.46 | Qwell | no I wasn't |
| 19:34.24 | nullable_type | Hey guys anyone know if there is any difference between 1.4 and 1.6 on Sip Reinvite?! I am testing SIP Reinvite and getting SIP/2.0 488 ReInvite Rejected by the sip provider |
| 19:35.19 | *** join/#asterisk jstoker (n=jstoker@80-41-183-84.dynamic.dsl.as9105.com) |
| 19:37.04 | nullable_type | Anyone can help? |
| 19:37.12 | [TK]D-Fender | nullable_type: 488 is typically a codec issue |
| 19:37.16 | *** part/#asterisk jstoker (n=jstoker@80-41-183-84.dynamic.dsl.as9105.com) |
| 19:37.26 | nullable_type | ohh..... |
| 19:37.36 | [TK]D-Fender | nullable_type: which is understandable if * was forced to transcode something 1 end can't speak |
| 19:38.10 | gswain | [TK]D-Fender: just want to make local and ld calls. It would seem like one of the big guys like att or sprint would offer sip call termination |
| 19:38.45 | [TK]D-Fender | gswain: depends to WHERE. how many simultaneous calls. # of muintes/month, etc |
| 19:39.52 | gswain | [TK]D-Fender: all us domestic calling, would how many simultaneous call and minutes and such basically amount to different plans, a termination provider ight offer? |
| 19:39.55 | eppigy | hi |
| 19:40.21 | [TK]D-Fender | gswain: Clearly |
| 19:40.26 | lppitsfwnyuphtjm | Qwell: thanks for the help so far. do I have to sniff what pcsuite sends, or what next? |
| 19:40.27 | [TK]D-Fender | eppigy: you are dave |
| 19:40.28 | nullable_type | D-Fender >> Thanks, I will try a different codec for the Reinvite |
| 19:40.37 | eppigy | FACT |
| 19:41.45 | gswain | [TK]D-Fender: so Im trying to find out who might offer these plans... |
| 19:44.01 | [TK]D-Fender | gswain: And you haven't answered what your expected needs are |
| 19:44.09 | AndyML | does anyone have further information regarding the bug in 1.4.24's app_voicemail that sets the permissions on the .txt file such that it is inaccessible to the asterisk user? |
| 19:44.25 | [TK]D-Fender | AndyML: What bug? |
| 19:44.55 | *** join/#asterisk ccesario (n=ccesario@189-19-9-100.dsl.telesp.net.br) |
| 19:45.14 | AndyML | well, maybe i used the wrong explaination... that is the behavior i'm experiencing on an asterisk install from source in ubuntu 8.10, and have seen it before on 8.04. never could figure out what it was. |
| 19:45.47 | bpgoldsb | Can someone help me convert this line: 'exten => 701,hint,park:701@parkedcalls' to AEL? I'm trying 'hint(701) 701 => park:701@parkedcalls' but it fails aelparse. |
| 19:45.52 | gswain | [TK]D-Fender: I dont see why its so complicated I need to make local calls, i need to make long distance calls, I cant possibly know where every client will want to call |
| 19:45.58 | AndyML | i seem to recall someone telling me it was a known bug but they didn't have any further details, and i couldn't find it. |
| 19:47.19 | [TK]D-Fender | gswain: And that still leaves # of minutes, simultaneous channels, etc |
| 19:47.39 | codefreeze-lap | bpgoldsb: hint(/sip/1) 701 => { dialplan code;} |
| 19:47.46 | _Steve_ | how can i tell if i've got my QoS setup right in my router for prioritizing VoIP traffic (SIP and RTP)? |
| 19:48.24 | codefreeze-lap | uh, hint(sip/1) 701 => .... whatever, had a slash in there that shouldn't have been |
| 19:48.35 | bpgoldsb | I was about to ask ;) |
| 19:50.17 | codefreeze-lap | bpgoldsb: now, there's some substructure in what you can put in the hint() parens... I'm not as versed on what to put there, but your park: stuff might fit there. |
| 19:50.42 | bpgoldsb | codefreeze-lap, ya, thats what I'm trying to figure out |
| 19:50.58 | bpgoldsb | I'm trying to get my Snom BLF's working with Ast 1.6 |
| 19:51.47 | bpgoldsb | And it's really kicking my arse :( |
| 19:52.09 | gswain | [TK]D-Fender: how on earth could that possible matter? I'm selling phone systems id like to work with a provider that can make the phone systems work, i dont know from person to person those things, only that a company can service me no matter how many minutes or simultaneous channels or one. |
| 19:52.14 | gswain | they would amount to plans |
| 19:52.45 | gswain | [TK]D-Fender: dont worry i just found bandwidth.com, broadvox, and broadvoice |
| 19:53.37 | [TK]D-Fender | gswain: And you think Broadvoice allows you to RESELL their service? let alone provision DID's? |
| 19:53.50 | KavanS | lol |
| 19:53.57 | [TK]D-Fender | gswain: and the diffrerence can be night & day on the PRICE scale. |
| 19:54.23 | KavanS | gswain, just resell it...and make sure your passport/method of fleeing your appropriate jurisdiction is current |
| 19:54.24 | gswain | I dont think they let my provison anything |
| 19:54.29 | [TK]D-Fender | gswain: this stuff matters because its the kind of things that put people out of business |
| 19:55.32 | Katty | oh boy. |
| 19:55.36 | Katty | we got a live one |
| 19:55.42 | codefreeze-lap | bpgoldsb: I do that, too... I have a hint for all my "real" extensions... hint(DAHDI/1) 21 => Dial((Dahdi/1,20,T); ... and then, in the phone config, I make the light dep on ext 21 in that context. |
| 19:55.53 | gswain | I wasnt going to just go out and resell blindly or provision DID's |
| 19:56.10 | gswain | but could i not provide the person with a couple of people he could call to get service? |
| 19:56.20 | *** join/#asterisk twanny796 (n=twanny@195.158.85.248) |
| 19:56.23 | *** join/#asterisk jstoker (n=jstoker@80-41-183-84.dynamic.dsl.as9105.com) |
| 19:56.41 | gswain | the point was all along that i couldnt possible know all those things until I actually asked the client |
| 19:56.59 | gswain | and i was only looking for one or two names of companies, that people are using reliably to call |
| 19:57.06 | gswain | and ask them about their plans |
| 19:57.11 | *** join/#asterisk [I]nfecT (n=Inf@bas12-montreal02-1242553868.dsl.bell.ca) |
| 19:57.33 | bpgoldsb | codefreeze-lap, I think it's more of a snom problem than a asterisk issue |
| 19:57.51 | bpgoldsb | the call parking stuff is already creating the hint without me doing it in the dialplan |
| 19:58.04 | bpgoldsb | 601@parkedcalls : park:601@parkedcalls State:Idle Watchers 0 |
| 19:58.07 | _Steve_ | hmm, i've noticed that all my VoIP calls continue if the other end hangs up first |
| 19:58.20 | bpgoldsb | I just can't seem to get my phone to subscribe to it, and I don't know why exactly |
| 19:59.18 | *** part/#asterisk jstoker (n=jstoker@80-41-183-84.dynamic.dsl.as9105.com) |
| 20:00.26 | _Steve_ | what could cause that? |
| 20:00.57 | [TK]D-Fender | gswain: If your clients are small and want to use ITSp services directly tahts another matter. |
| 20:01.01 | [TK]D-Fender | ~itsplist-us |
| 20:01.01 | infobot | [~itsplist-us] Here are some popular ITSPs (USA) starting with the more respected ones : http://www.teliax.com , http://connect.voicepulse.com , http://www.nufone.net , http://www.broadvoice.com, http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net |
| 20:01.03 | Katty | _Steve_: improper disconnect via the voice providor? |
| 20:01.10 | Katty | _Steve_: might try using the h on the dial |
| 20:01.21 | Katty | _Steve_: and you could manually hang up your end |
| 20:01.52 | gswain | [TK]D-Fender: thats all i asked for |
| 20:01.58 | _Steve_ | i do, but if the other end hangs up, the call should be disconnected. |
| 20:02.09 | Katty | _Steve_: who's your voip provider? |
| 20:02.16 | _Steve_ | bandwidth.com |
| 20:02.20 | Katty | ah yes |
| 20:02.23 | Katty | they can help you |
| 20:02.29 | _Steve_ | oh? |
| 20:02.30 | Katty | and look at your edgebox to tell why it's not disconnecting right |
| 20:02.32 | Katty | give them a call |
| 20:02.48 | _Steve_ | ok. are they going to tell me it's something with my equipment? or change some setting in my system or router? |
| 20:02.56 | Katty | they will tell you why |
| 20:03.01 | Katty | which will help you figure out what to do |
| 20:03.30 | eppigy | SE ME GUSTA |
| 20:03.42 | Katty | your face. |
| 20:04.19 | eppigy | your face ia augusta |
| 20:04.22 | eppigy | is |
| 20:04.26 | Katty | your mom's face is augusta. |
| 20:05.02 | eppigy | my mom's face being augusta is directly proportional to your mom's face being somewhere even less desireable to live |
| 20:05.22 | Katty | you go too far. |
| 20:05.26 | eppigy | :< |
| 20:05.35 | Katty | i still love you tho |
| 20:05.37 | blebleble | for "MONITOR_FILENAME=/var/spool/asterisk/monitor/${CALLERID(name)}-q${EXTEN}-${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)}-${UNIQUEID}) (queue recording) is there a way to add in the extension that took the call so i can associate it with an agent? |
| 20:05.40 | eppigy | ilu2 |
| 20:05.45 | _Steve_ | calls.... |
| 20:06.02 | *** join/#asterisk cr0n (n=asgasg@dsl-145-69-176.telkomadsl.co.za) |
| 20:06.20 | cr0n | hi, quick question - im playing around with incoming sip calls. If I dial '<extention>@hostname' from another sip service - am I suppose to get through to the extention located at that particular network? I can see the incoming call in my logs but I'm getting a "noservice" message however from the local network if i dial the extention, it goes through 100% - what am i missing and where? |
| 20:08.36 | cr0n | ah no worries, fixed :) |
| 20:08.38 | *** part/#asterisk cr0n (n=asgasg@dsl-145-69-176.telkomadsl.co.za) |
| 20:08.53 | *** join/#asterisk oej (n=olle@79.135.0.94) |
| 20:09.54 | Katty | well that was easy. |
| 20:10.41 | pmhaddad-work | is the only one in the office |
| 20:10.51 | Katty | :< |
| 20:10.54 | pmhaddad-work | my coworkers are slackers! |
| 20:11.20 | _Steve_ | put another way, it's easy to look like the hardest worker in the office. :) |
| 20:11.21 | Katty | at leas tyou have coworkers |
| 20:11.22 | Katty | and a job. |
| 20:11.25 | *** join/#asterisk vader-- (n=me@c-68-36-9-8.hsd1.nj.comcast.net) |
| 20:11.36 | pmhaddad-work | Katty, true true |
| 20:12.03 | *** join/#asterisk jstoker (n=jstoker@80-41-183-84.dynamic.dsl.as9105.com) |
| 20:12.06 | *** part/#asterisk gswain (n=Administ@67-60-195-60.cpe.cableone.net) |
| 20:13.24 | *** part/#asterisk jstoker (n=jstoker@80-41-183-84.dynamic.dsl.as9105.com) |
| 20:17.46 | _Steve_ | oh bandwidth.com... your customer support is, so far, not impressive. |
| 20:19.07 | *** join/#asterisk putnopvut (n=putnopvu@asterisk/master-of-queues/mmichelson) |
| 20:19.07 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 20:20.25 | *** join/#asterisk seanmh (n=johndoe@198.59.129.24) |
| 20:25.07 | *** join/#asterisk Rico29 (n=Rico@lns-bzn-48f-81-56-220-28.adsl.proxad.net) |
| 20:25.42 | Katty | baked cheetos, or popcorn? |
| 20:25.56 | eppigy | popcorn |
| 20:25.57 | eppigy | well |
| 20:25.58 | eppigy | man |
| 20:26.03 | eppigy | I used to makie my own popcorn |
| 20:26.06 | eppigy | make |
| 20:26.09 | eppigy | it was awesome |
| 20:26.22 | Katty | these are the orville 100 calorie baggies |
| 20:26.27 | Katty | mini bags |
| 20:26.38 | eppigy | :[ |
| 20:27.03 | *** join/#asterisk jstoker (n=jstoker@80-41-183-84.dynamic.dsl.as9105.com) |
| 20:28.47 | *** part/#asterisk jstoker (n=jstoker@80-41-183-84.dynamic.dsl.as9105.com) |
| 20:29.11 | *** join/#asterisk j_kroon (n=jkroon@dsl-240-190-12.telkomadsl.co.za) |
| 20:29.28 | j_kroon | what could possibly cause "Unable to get a clear outside line" when running fxotune -i ? |
| 20:29.57 | j_kroon | using dahdi 2.1.0.4 with -tools 2.1.0.2. |
| 20:32.26 | bpgoldsb | Using Ast1.6/AEL, is there a way to check the hint value of hint object? |
| 20:32.56 | *** part/#asterisk SilentGreen (n=chatzill@201.217.56.16) |
| 20:36.53 | *** join/#asterisk hfb (n=hfb@pool-96-247-109-136.lsanca.dsl-w.verizon.net) |
| 20:37.34 | *** join/#asterisk oej (n=olle@79.135.0.94) |
| 20:37.53 | eppigy | yeah its rollin up on dinner time |
| 20:42.11 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 20:45.49 | *** join/#asterisk ruben23 (n=AGENT@124.107.3.178) |
| 20:51.47 | *** join/#asterisk umpc (n=Justin@unaffiliated/umpc) |
| 20:52.11 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 20:53.46 | *** join/#asterisk DanyWalker (n=elmo@201.230.182.67) |
| 20:55.07 | DanyWalker | hi people , i have a question |
| 20:55.33 | DanyWalker | i'm using phpagi() |
| 20:55.58 | *** join/#asterisk jstoker (n=jstoker@80-41-183-84.dynamic.dsl.as9105.com) |
| 20:56.01 | DanyWalker | is possible reboot the service asterisk using phpagi() ? |
| 20:56.16 | Katty | eppigy: what's furr dinner |
| 20:58.34 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194) |
| 20:58.51 | *** join/#asterisk Rico29 (n=Rico@lns-bzn-48f-81-56-220-28.adsl.proxad.net) |
| 21:00.23 | *** part/#asterisk jstoker (n=jstoker@80-41-183-84.dynamic.dsl.as9105.com) |
| 21:01.22 | eppigy | Katty: I am trying to decide :[ |
| 21:01.39 | Katty | what options are you weighing |
| 21:01.47 | [TK]D-Fender | eppigy: If you choose not to decide, you still have made a choice ;) |
| 21:03.03 | eppigy | hrmmm |
| 21:03.09 | eppigy | I am thinking cracker barrel |
| 21:03.42 | Katty | i should see about getting a ham and hashbrown casserole recipe |
| 21:03.53 | eppigy | YES |
| 21:05.01 | Katty | wif crumb topping |
| 21:07.55 | eppigy | oh man |
| 21:07.57 | eppigy | yeah dude |
| 21:08.02 | eppigy | I think cracker barrel it is |
| 21:14.53 | *** join/#asterisk nicoAMG (i=asgalt@201.203.96.42) |
| 21:17.24 | _Steve_ | i'm not sure if that was tweedledee or tweedledumb that called me back from bandwidth.com tech support. |
| 21:17.49 | Katty | victor? |
| 21:18.04 | _Steve_ | i have no idea |
| 21:18.16 | Katty | call josh |
| 21:18.19 | _Steve_ | but no real answer on the "why calls take 30 seconds to hang up" question. |
| 21:18.23 | Katty | ext 1924 |
| 21:18.53 | Katty | be sure to tell him you weren't happy with the tech that called you back |
| 21:19.00 | *** join/#asterisk knarfly (n=vtserije@c-75-74-113-9.hsd1.fl.comcast.net) |
| 21:19.01 | mnicholson | Qwell, poing |
| 21:19.12 | Qwell | mnicholson: nm :D |
| 21:19.31 | mnicholson | k |
| 21:19.37 | Qwell | somebody was having chan_mobile issues. think he figured out to get what he needs |
| 21:19.49 | _Steve_ | i mean, he's right, it's not a huge deal, but having calls last 30 seconds longer than they should is annoying. |
| 21:19.53 | knarfly | does the TFOT 2nd Edition have the newest commands for *-1.6.x.x? |
| 21:20.05 | [TK]D-Fender | knarfly: No |
| 21:20.36 | *** join/#asterisk oglynn (n=me@173.9.102.81) |
| 21:20.48 | knarfly | is it going to be Digium's practice to keep changing to syntax, it's making a moving target out of the commands |
| 21:21.02 | Katty | knarfly: it is their intent to make things better. |
| 21:21.08 | Katty | knarfly: so you better believe it. |
| 21:21.21 | Katty | knarfly: and you're welcome. |
| 21:21.21 | oglynn | i am having trouble with a new install with dahdi |
| 21:21.39 | oglynn | asterisk v 1.6.0.9 |
| 21:22.32 | knarfly | oglynn: I just did a build and got mine working with dahdi...thanks to the help here |
| 21:22.38 | knarfly | what's the issue |
| 21:22.45 | *** join/#asterisk mnicholson (n=mnichols@nat/digium/x-bd00373750618062) |
| 21:22.45 | oglynn | when i try and start dahdi service i get FATAL: Module dahdi not found |
| 21:23.05 | Katty | load module chan_dadhi.so |
| 21:23.36 | knarfly | Katty: shouldn't asterisk do this automatically at startup |
| 21:24.00 | oglynn | my problem seems to be without even trying asterisk |
| 21:24.09 | Katty | knarfly: if everything magically worked all the time, there wouldn't be nearly 100 people in here--now would there? espeically you. |
| 21:24.17 | oglynn | dahdi should run at startup i believe |
| 21:24.26 | Katty | oglynn: yes, it should. |
| 21:24.30 | knarfly | Katty: like your job ??? 8-) |
| 21:24.40 | eppigy | oh boi |
| 21:24.48 | oglynn | and if it was running i would see /dev/dahdi and i dont |
| 21:24.49 | Katty | knarfly: why yes, i adore my job (= |
| 21:25.01 | Katty | knarfly: thanks for asking. so nice of you. |
| 21:25.08 | knarfly | then go do it and leave your attitude at the door |
| 21:25.10 | [TK]D-Fender | Katty: Sure there would |
| 21:25.18 | oglynn | i have 1 module enabled wct4xxp |
| 21:25.28 | Katty | knarfly: sorry, attitude comes with my volunteering here. |
| 21:25.31 | Katty | knarfly: take it or leave it |
| 21:25.45 | Katty | knarfly: if you want nice and polite, go hire a consultant |
| 21:26.04 | Katty | [TK]D-Fender: well not everyone is here to socialize ;) |
| 21:26.05 | knarfly | Katty: sorry to hear that |
| 21:26.19 | eppigy | knarfly: calm down guy |
| 21:26.30 | _Steve_ | i'm not your guy, buddy! |
| 21:26.39 | Katty | you're not my buddy, man! |
| 21:26.52 | _Steve_ | i'm not your man, friend! |
| 21:27.09 | Katty | idk what comes after that part |
| 21:27.16 | Katty | i'm not your friend....dude? |
| 21:27.21 | _Steve_ | it's "i'm not your friend, guy! |
| 21:27.24 | _Steve_ | so that it loops around |
| 21:27.28 | oglynn | after dahdi make config i see I think that the DAHDI hardware you have on your system is: |
| 21:27.28 | oglynn | pci:0000:06:08.0 wct4xxp- d161:0420 Wildcard TE420 (4th Gen) |
| 21:27.36 | Katty | oh, right ;) |
| 21:27.45 | _Steve_ | and there's 3, buddy, guy, and friend. then it loops |
| 21:27.54 | Katty | i so smrt now |
| 21:27.57 | knarfly | oglynn: then dahdi at least sees your hardware |
| 21:28.10 | oglynn | yep |
| 21:28.12 | *** join/#asterisk DarkRift (n=dark@bas1-sthubert21-1096611921.dsl.bell.ca) |
| 21:28.21 | lppitsfwnyuphtjm | Qwell: AT+CNMI=? gives ERROR, so I don't know how to use SMS with chan_mobile.. phone is supposed to work with pcsuite, maybe it does something differently? |
| 21:28.26 | _Steve_ | anyone here using sip trunking? or am i crazy for using it? |
| 21:28.29 | oglynn | but when i try to do service dahdi start i get |
| 21:28.31 | Katty | i do |
| 21:28.31 | knarfly | my config files required a bit of tweaking but outside of that dahdi took control and I was dialing and calling |
| 21:28.36 | Katty | _Steve_: we go thorugh bandwidth.com |
| 21:28.40 | _Steve_ | perhaps i should pick up a T1 card or something |
| 21:28.45 | oglynn | FATAL: Module dahdi not found. |
| 21:28.45 | oglynn | <PROTECTED> |
| 21:28.51 | Katty | _Steve_: it's not too bad. |
| 21:28.58 | _Steve_ | or perhaps i should just ignore the slow disconnect and tiny bit of stutter i get |
| 21:29.05 | Katty | _Steve_: just make sure you have enough bandwidth :P |
| 21:29.09 | knarfly | oglynn: what does lsmod show you from the command line (not the CLI) |
| 21:29.21 | Katty | _Steve_: well it /is/ going across the internets |
| 21:29.27 | _Steve_ | Katty: i have 5mb both ways... it's fiber... i can crank it up both ways... |
| 21:29.30 | eppigy | knarfly: loaded modules |
| 21:29.42 | eppigy | oh god |
| 21:29.44 | eppigy | nm |
| 21:29.48 | _Steve_ | they'll sell me 100mb if i want... |
| 21:29.49 | eppigy | I am switchign between windows |
| 21:29.55 | oglynn | no mention of dahdi on lsmod |
| 21:29.56 | Katty | oh god |
| 21:29.56 | Katty | oh god |
| 21:30.08 | Katty | sorry, god's not in right now |
| 21:30.14 | knarfly | oglynn: no you'll be looking for wct4xxp |
| 21:30.15 | Katty | please leave a Beep at the message |
| 21:30.17 | Katty | MESSAGE |
| 21:30.31 | oglynn | no mention of wct4xxp either |
| 21:30.32 | *** join/#asterisk SwK (n=SwK@freeswitch/developer/swk) |
| 21:30.37 | Katty | well |
| 21:30.39 | Katty | looky there |
| 21:30.41 | Katty | it's mister rice. |
| 21:30.41 | _Steve_ | it's fiber to a media converter into my firewall... |
| 21:30.45 | [TK]D-Fender | BEEP |
| 21:30.53 | oglynn | locate wct4xxp gives me /lib/modules/2.6.18-128.1.6.el5/dahdi/wct4xxp/wct4xxp.ko |
| 21:31.05 | Katty | [TK]D-Fender: well done, sir, well done. |
| 21:31.05 | eppigy | max_hungry |
| 21:31.23 | *** part/#asterisk nullable_type (n=nullable@hq.verbx.net) |
| 21:31.35 | Katty | you should call cracker barrel and place your order early so it's ready when you get there ;) |
| 21:31.44 | knarfly | oglynn: I'm still new to linux...but you'll need to load the module, even though dahdi should take care of this at startup |
| 21:31.58 | knarfly | in freebsd we use kldload...what is the linux command? |
| 21:32.00 | eppigy | Katty: oh it'll be ready |
| 21:32.09 | eppigy | they only take like 10 minutes for breakfast stuff |
| 21:32.12 | knarfly | anyone, anyone, anyone....Bueller |
| 21:32.17 | eppigy | I call when I am almost home |
| 21:32.20 | _Steve_ | insmod |
| 21:32.22 | Katty | eppigy: jolly good. i'll just swing by and pick it up for you. |
| 21:32.27 | DanyWalker | guys, is possible reboot the service asterisk, using phpagi() ? |
| 21:32.30 | eppigy | yes plz |
| 21:32.38 | Katty | and eat it for you. |
| 21:32.39 | lppitsfwnyuphtjm | modprobe |
| 21:32.43 | eppigy | :{ |
| 21:32.48 | knarfly | thank you modprobe |
| 21:32.52 | DanyWalker | guys, is possible restart the service asterisk, using phpagi() ? |
| 21:32.58 | _Steve_ | insmod doesn't work? |
| 21:33.10 | lppitsfwnyuphtjm | _Steve_: module might have other dependencies |
| 21:33.38 | knarfly | oglynn: use modprobe to load the wct4xxp module |
| 21:33.46 | _Steve_ | oh, right |
| 21:34.02 | Katty | eppigy: 20min! |
| 21:34.28 | eppigy | YES |
| 21:34.44 | knarfly | oglynn: lsmod | grep wct |
| 21:34.59 | _Steve_ | http://imnotyourbuddyguy.ytmnd.com/ |
| 21:35.01 | knarfly | mine is showing all the wctxxxxxx modules are loaded |
| 21:35.55 | Katty | eppigy: what did you order |
| 21:35.58 | oglynn | when i do modprobe -l |grep wct4 it returns nothing |
| 21:36.49 | knarfly | oglynn: what happens when you modprobe wct4xxp |
| 21:36.55 | blebleble | in asterisk -rx "show queues" does anyone know what the queue holdtime is defined upon (like holdtime per the last 5 minutes, last hour , today ?) |
| 21:37.11 | oglynn | FATAL: Module wct4xxp not found. |
| 21:37.18 | rob0 | I ate it. |
| 21:37.30 | lppitsfwnyuphtjm | oglynn: where is the matching wct4xxp.ko for your kernel? |
| 21:37.33 | Katty | rob0: i don't want to be around you later. |
| 21:37.43 | oglynn | i think its not in the correct directory? |
| 21:37.47 | tzafrir_laptop | modinfo wct4xxp |
| 21:37.47 | knarfly | oglynn: I know you hate to hear this but have you tried rebooting...the dahdi setup on my machine just found the hardware and started working with it |
| 21:37.48 | rob0 | wct4xxp is mild, don't worry |
| 21:38.04 | oglynn | its in /lib/modules/2.6.18-128.1.6.el5/dahdi/wct4xxp/wct4xxp.ko |
| 21:38.19 | oglynn | i think modprobe is not looking in there? |
| 21:38.20 | rob0 | Now wcfxo ... a different story, ugh, I was sick for 3 days the last time I ate one of those. |
| 21:38.32 | tzafrir_laptop | oglynn, what's the output of: uname -r |
| 21:38.40 | lppitsfwnyuphtjm | oglynn: does "uname -r" print 2.6.18-128.1.6.el5 , and did you run depmod -a after copying the .ko there? |
| 21:38.48 | *** join/#asterisk _pepo_ (n=pepo@200.55.224.2) |
| 21:38.55 | oglynn | uname -r 2.6.18-128.el5 |
| 21:38.58 | knarfly | rob0: I use wcfxo with an X101P and it works fine |
| 21:39.02 | _pepo_ | hi friends |
| 21:39.15 | lppitsfwnyuphtjm | oglynn: not quite the same |
| 21:39.18 | tzafrir_laptop | oglynn, that's a different kernel version |
| 21:39.49 | oglynn | i seem to have a dir /lib/modules/2.6.18-128.el5/ |
| 21:39.56 | *** join/#asterisk telecos (n=sergio@19.167.219.87.dynamic.jazztel.es) |
| 21:39.58 | knarfly | okay, what tzafrir_laptop said....this is new to me because in FreeBSD we don't deal with all the differnt kernels |
| 21:40.02 | rob0 | 2.6.18, yikes, that's old |
| 21:40.20 | oglynn | and a /lib/modules/2.6.18-128.1.6.el5/ |
| 21:40.30 | tzafrir_laptop | centos keeps changing it |
| 21:40.35 | oglynn | the only thing in /lib/modules/2.6.18-128.1.6.el5/ is dahdi and wct4xxp |
| 21:40.43 | oglynn | it is a fresh centos 5.3 |
| 21:40.52 | eppigy | Katty: uncle herschels favorite |
| 21:40.58 | eppigy | with hashed brown casserole |
| 21:41.01 | eppigy | and country ham |
| 21:41.03 | oglynn | should i make some sym link to the other folder? |
| 21:41.05 | lppitsfwnyuphtjm | oglynn: where did you get those .1.6. modules from? |
| 21:41.06 | eppigy | eggs over medium |
| 21:41.31 | oglynn | when i build dahdi it put them there... |
| 21:41.40 | rob0 | throws in a spare wct4xxp.ko for eppigy's feast |
| 21:41.50 | rob0 | enjoy |
| 21:41.57 | oglynn | originally when i build dahdi i had to fight with it as it could not find the kernel source |
| 21:42.13 | rob0 | And as a side benefit, Katty won't want to be around you later! ;) |
| 21:42.29 | Katty | eppigy: you lucky ducky |
| 21:42.49 | lppitsfwnyuphtjm | oglynn: they need to be of matching kernel version, but you can try. it probably complains about version mismatch or something. |
| 21:43.12 | eppigy | haha |
| 21:43.13 | lppitsfwnyuphtjm | oglynn: you have kernel sources from newer kernel than the currently running kernel? |
| 21:43.23 | _pepo_ | please, I am trying to use DISA in mi extensions.conf but when I hear the tone and pulse some key, inmediatly I receive busy sounds |
| 21:43.30 | _pepo_ | what is wrong? |
| 21:44.02 | oglynn | when i build dahdi /lib/modules/2.6.18-128.el5/ did not have valid source i kept complaining so i sym linked /lib/modules/2.6.18-128.el5/build to /usr/src/kernels/2.6.18-128.1.6.el5-i686/ |
| 21:44.11 | Katty | _pepo_: i'd say... it's not setup right... |
| 21:44.16 | Qwell | oglynn: don't do that |
| 21:44.17 | eppigy | haha |
| 21:44.25 | Qwell | oglynn: yum install kernel-devel-`uname -r` kernel-headers |
| 21:45.04 | Katty | [TK]D-Fender: i didn't mean to start the flame war... |
| 21:45.21 | eppigy | why do you hate america? |
| 21:45.40 | oglynn | @Qwell: downloading now |
| 21:46.16 | Katty | eppigy: do you see the time?! |
| 21:46.25 | oglynn | it errored package kernel-devel-2.6.18-128.1.6.el5.i686 (which is newer than kernel-devel-2.6.18-128.el5.i686) is already installed |
| 21:46.29 | eppigy | I do |
| 21:46.30 | eppigy | I DO |
| 21:46.36 | Katty | eppigy: you're late! you're late! |
| 21:46.41 | Qwell | oglynn: reboot into the new kernel that was installed |
| 21:46.45 | Katty | eppigy: for a VERY important date! |
| 21:46.53 | eppigy | I am nevear late |
| 21:47.09 | eppigy | I havent called them yet |
| 21:47.11 | eppigy | amn |
| 21:47.13 | Katty | eppigy: no time to say hello, goodbye! don't forget to clean off your plate! |
| 21:47.17 | eppigy | I am about to leave though |
| 21:47.31 | eppigy | i need to study hard core tongiht |
| 21:47.39 | Katty | eppigy, member of the clean plate society since 1974. |
| 21:47.46 | eppigy | yesh |
| 21:48.26 | jameswf | study hardcore? you get paid for that? |
| 21:49.06 | eppigy | no |
| 21:49.19 | eppigy | study at my house |
| 21:49.55 | eppigy | I mean I do it at work too |
| 21:50.01 | eppigy | and then I guess I do get paid to |
| 21:50.45 | oglynn | uname -r still returns 2.6.18-128.el5 |
| 21:51.04 | eppigy | then yum install kernel |
| 21:51.20 | Katty | i thinks i shall leave now. |
| 21:51.22 | Katty | buhbye |
| 21:51.23 | oglynn | just did yum update kernel somehow my headers and kernel are not the same.... |
| 21:51.41 | Katty | ctrl ad |
| 21:51.44 | oglynn | reboot part 2 |
| 21:51.48 | eppigy | no dude |
| 21:51.54 | eppigy | check /etc/groff/ |
| 21:51.57 | eppigy | whoops |
| 21:52.01 | eppigy | /etc/grub.conf |
| 21:52.23 | eppigy | by Katty |
| 21:52.26 | eppigy | be safe |
| 21:53.06 | DanyWalker | guys, is possible restart the service asterisk, using phpagi() ? |
| 21:55.31 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 21:55.39 | Qwell | DanyWalker: AGI is only active during a call. You want to call an extension and have it restart Asterisk? |
| 21:55.44 | oglynn | that was the ticket sorry for being a newbie and thanks for the help! |
| 21:56.05 | oglynn | it came right up on it's own now! |
| 21:59.38 | *** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim) |
| 22:08.40 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 22:10.38 | *** join/#asterisk hi365 (n=hi365@94.159.144.212) |
| 22:14.06 | DanyWalker | @Qwell , not i want use like example : if(changes in database){$agi->exec("some words fors restartasteirks from agi");} |
| 22:20.15 | [TK]D-Fender | DanyWalker: Silly approach |
| 22:21.24 | [TK]D-Fender | DanyWalker: You're already in an external process. there is no dialplan app made to manipulate * like that |
| 22:21.59 | kc8pxy | DanyWalker: what happens when your asterisk goes down, because someone injected something wonky to yoru db? |
| 22:21.59 | [TK]D-Fender | DanyWalker: You COULD use AGI to call System to call * from NIX CLI to restart, but you're already in an external process and could do that directly without placing load on * |
| 22:22.16 | [TK]D-Fender | DanyWalker: You could of course use AMI as the signalling method for this |
| 22:24.57 | DanyWalker | becaus i nedd calculate a minuts fron users and assign minutos to users |
| 22:25.37 | [TK]D-Fender | DanyWalker: Make sure your AGI is finished doing everything else it has to do because it itself will die almost instantly |
| 22:25.40 | Qwell | and...why would you need to restart for that? |
| 22:27.28 | seb- | [TK]D-Fender: want to make a time tonight for * stuff? |
| 22:27.42 | seb- | [TK]D-Fender: tell me when you're free |
| 22:27.57 | knarfly | my new setup is working well, but I just noticed that it did not send an e-mail to my user account when a message was left. |
| 22:27.58 | knarfly | voicemail.conf is set to do this and it worked on my previous 1.4 setup |
| 22:28.04 | Qwell | DanyWalker: ? |
| 22:30.10 | [TK]D-Fender | seb-: NIght is about to be taken up and I have no idea when I'll be back |
| 22:31.45 | seb- | [TK]D-Fender: ok....feel free to reply to my email if you have time...my * server is up for you |
| 22:32.09 | seb- | [TK]D-Fender: it should be up 'til around 11pm PST |
| 22:32.11 | knarfly | \book |
| 22:33.59 | *** join/#asterisk Steve_J-obs (n=Chris123@pool-68-239-37-146.bos.east.verizon.net) |
| 22:34.40 | Steve_J-obs | hello everybody! |
| 22:35.03 | knarfly | yo Jobs...how's the apple |
| 22:35.36 | *** part/#asterisk oglynn (n=me@173.9.102.81) |
| 22:36.09 | suma | How can I simultaneously broadcast a file in a call to the callee and to the caller |
| 22:39.32 | Steve_J-obs | knarfly: what's up |
| 22:41.10 | knarfly | just tweaking my new Fedora server with Asterisk and doing some work in AutoCAD |
| 22:46.28 | *** join/#asterisk hi365 (n=hi365@94.159.144.212) |
| 22:47.32 | *** join/#asterisk smbrienz (n=guest@host71-247-dynamic.1-87-r.retail.telecomitalia.it) |
| 23:00.37 | _pepo_ | I receive a message (in debug mode) that the call is placed on hold, How do I enable that call? |
| 23:11.43 | *** part/#asterisk ruben23 (n=AGENT@124.107.3.178) |
| 23:14.48 | *** join/#asterisk jmacz (n=jmacz@190.144.75.22) |
| 23:21.40 | *** join/#asterisk Carlos_PHX (n=Carlos@ip68-3-163-241.ph.ph.cox.net) |
| 23:21.51 | Steve_J-obs | <Steve_J-obs> hi, can anybody give me an opinion what can I do to hear ringback on a sip call? |
| 23:21.51 | Steve_J-obs | <Steve_J-obs> it is as if asterisk can't handle ringback on their rtp packets |
| 23:27.00 | *** join/#asterisk sah-work (n=Bawbatos@65-119-47-100.dia.static.qwest.net) |
| 23:27.49 | *** join/#asterisk jmacz (n=jmacz@190.144.75.22) |
| 23:27.51 | sah-work | dumb question. how do i dump a stuck call from the cli |
| 23:27.57 | *** join/#asterisk hi365 (n=hi365@94.159.144.212) |
| 23:28.01 | sah-work | i have a call that shows it has been up for 148 hours :) |
| 23:29.46 | *** join/#asterisk pif (n=ldm@zenon.apartia.fr) |
| 23:29.57 | *** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194) |
| 23:36.01 | *** part/#asterisk knarfly (n=vtserije@c-75-74-113-9.hsd1.fl.comcast.net) |
| 23:41.44 | hardwire | has mixmonitor ever made your asterisk go crazy crazy? |
| 23:41.48 | hardwire | I thought it was more stable than that. |
| 23:42.11 | hardwire | three times today asterisk has taken the load on my system to 30 :) |
| 23:42.11 | hardwire | heh |
| 23:42.47 | hardwire | but it may be another issue as well.. they had me do something evil after a Dial command that seemed to hork things up a bit earlier. |
| 23:44.05 | *** join/#asterisk juanIMP (n=Juancho@200.26.152.222) |
| 23:44.15 | *** join/#asterisk GreenCult (n=greencul@200.48.85.18) |
| 23:45.04 | hardwire | consider that thinking out loud |
| 23:47.55 | DanyWalker | sorry people |
| 23:48.01 | DanyWalker | i want to reload asterisk |
| 23:48.05 | DanyWalker | not restart |
| 23:48.11 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 23:48.18 | DanyWalker | with exec() |
| 23:50.04 | *** join/#asterisk Vec (n=Vec@196-210-141-165-rrba-esr-3.dynamic.isadsl.co.za) |
| 23:53.49 | drmessano | @qwell moi? |
| 23:54.56 | Vec | I installed a Digium PRI card TE110P in my server purely for timing, is there a way I can check that asterisk is using it for timing, the module for that card is loaded (lsmod | grep zap|) ? |
| 23:56.39 | Vec | its actually a TE121 |
| 23:56.49 | Vec | dahdi show status shows the card is installed and unconfigured |
| 23:57.49 | suma | How can I simultaneously broadcast a file in a call to the callee and to the caller? |
| 23:59.03 | DanyWalker | is possible reload with exec , asterisk ? |
| 23:59.41 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 23:59.50 | drmessano | DanyWalker: yep |
