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| 00:35.05 | doctorray | What's the trick to defaulting a verbosity level when running 'asterisk -r' ? my asterisk was started with -vvv and I have "verbose" on the console line in logger.conf. latest 11.1.2 |
| 00:36.03 | doctorray | or do I have to run asterisk -rvvv ? |
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| 00:41.20 | doctorray | found it in the docs, thanks and nevermind! |
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| 04:11.10 | deo_ | hello guys.. may i ask for what is the command to check if your sip provider is online? or where to check it??? |
| 04:11.58 | WIMPy | sip show peer[s| <name>] or sip show registry |
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| 04:13.58 | deo_ | hi WIMPy thanks. will try it.. |
| 04:14.48 | deo_ | yes it works.. thanks WIMPy ... our sip provider is online but we cant go through :( |
| 04:14.58 | volga629 | Hello Everyone, trying troubleshoot [2013-01-13 15:44:59] WARNING[31598]: res_rtp_asterisk.c:2143 ast_rtp_read: RTP Read too short |
| 04:15.02 | deo_ | checking on this... |
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| 04:51.28 | BeeBuu | is anyone try sipml5 with asterisk 11? |
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| 06:27.23 | BeeBuu | my sipml5 client can't get audio with asterisk 11,anyone help me please? |
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| 09:23.46 | jzaw | morning peeps |
| 09:23.56 | jzaw | re sip + ipv6 |
| 09:24.22 | jzaw | i put bindaddr=:: in sip.conf and of course i get both ipv4 and ipv6 sip connectivity |
| 09:24.47 | jzaw | but i want to put a specific ipv4 and specific ipv6 ip in the conf (multiple ips on the interface) |
| 09:24.58 | jzaw | but when i do that ... and sip reload |
| 09:25.04 | jzaw | i get a whole raft of errors and no sip |
| 09:25.11 | jzaw | any pointers? |
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| 09:31.13 | Wiretap | jzaw, those addresses must be assigned to an interface |
| 09:31.40 | jzaw | Wiretap: ifconfig gives me the ips |
| 09:31.46 | jzaw | for eth0 for instance |
| 09:32.03 | jzaw | i have native ipv6 |
| 09:32.37 | Wiretap | I'll admit I use :: and use a firewall to manage connectivity |
| 09:33.16 | jzaw | if i use bindaddr=w.x.y.z for a specific ipv4 ip |
| 09:33.20 | jzaw | that works |
| 09:33.24 | jzaw | no ipv6 of course |
| 09:33.50 | Wiretap | have you tried putting the IPv6 address in square brackets? |
| 09:33.59 | jzaw | but if i us a specific bindaddr=actualipv6ip blammo |
| 09:34.09 | jzaw | i thought of that just as you typed it |
| 09:34.12 | jzaw | will try now |
| 09:34.15 | jzaw | thanks for the prompt |
| 09:34.16 | Wiretap | some stuff prefers it that way |
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| 09:34.32 | Wiretap | the other thing to try would be the full IPv6 address, all zeroes included] |
| 09:36.42 | jzaw | wiretap .. correction to my description of the error ... if i have two bindaddr lines one with ipv4 and a second with ipv6 |
| 09:36.49 | jzaw | i get onlhy iv6 onnectivity |
| 09:36.58 | jzaw | my ipv4 peers become unreachable |
| 09:37.00 | Wiretap | seperate the addresses with a comma |
| 09:37.06 | jzaw | with streams of [Jan 14 09:36:58] WARNING[602]: chan_sip.c:3871 __sip_xmit: sip_xmit of 0x9d7a8d8 (len 444) to (null) returned -1: Invalid argument |
| 09:37.08 | Wiretap | rather than two bindaddr statements |
| 09:37.32 | jzaw | ah .... x.x.x.x , ipv6 |
| 09:37.39 | jzaw | or x.x.x.x, ipv6 |
| 09:37.56 | Wiretap | no |
| 09:37.58 | Wiretap | no space |
| 09:37.59 | Wiretap | just comma |
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| 09:38.43 | jzaw | done ...now i have ipv4 no ipv6 hehe |
| 09:38.49 | jzaw | [Jan 14 09:38:36] WARNING[647]: chan_sip.c:3871 __sip_xmit: sip_xmit of 0xa443790 (len 748) to [2001:470:70:1b7::2]:5060 returned -1: Address family not supported by protocol |
| 09:39.54 | Wiretap | with it in square brackets? |
| 09:39.59 | Wiretap | I'm afraid I'm now stumped |
| 09:40.31 | jzaw | trying ... like you say you have to eliminate all the obvious combinations |
| 09:42.30 | jzaw | nope not that either |
| 09:42.36 | jzaw | back to bindaddr=:: |
| 09:42.39 | jzaw | :/ |
| 09:44.40 | jzaw | ta for your help Wiretap |
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| 10:16.37 | R1ck | hi. I have a question about the Read() application. I want customers to be able to skip entering input by just hitting #, but this does not seem to work (they have to input *something*), how can I make this work? |
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| 10:25.08 | wdoekes | R1ck: set attempts to 1 ? |
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| 10:25.47 | wdoekes | I believe you can check ${READSTATUS} (or similar) for what happened |
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| 10:27.19 | R1ck | isn't 1 attempt the default? |
| 10:27.32 | R1ck | ah, guess not |
| 10:27.36 | R1ck | setting to 1 worked :) |
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| 12:08.02 | krotos | hi all |
| 12:09.29 | krotos | i'm playing with hint and Custom devstate. I'm encountering some difficult for configure a BLF ( used for showing day/night state) |
| 12:10.06 | krotos | on the phone i can't see the BLF change |
| 12:11.16 | krotos | my extensions.conf is here http://pastebin.com/qzv3xFmu |
| 12:18.50 | WIMPy | devstate list |
| 12:18.55 | WIMPy | core show hints |
| 12:20.08 | krotos | on the devstate list i can see the change |
| 12:20.39 | krotos | on core show hints, watchers = 0 |
| 12:20.48 | krotos | on the phone i've to set,in my case, 60 |
| 12:20.50 | krotos | right= |
| 12:20.56 | krotos | right? |
| 12:21.22 | WIMPy | Hmm. |
| 12:21.26 | WIMPy | yes |
| 12:21.36 | WIMPy | But I'm not sure the /callerID thing works for hints. |
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| 12:21.57 | krotos | mmmmmmmmmmm, let me try without this |
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| 12:23.12 | WIMPy | I didn;t have that idea, yet. But I think it might be usefull. |
| 12:23.19 | krotos | i confirm you |
| 12:23.45 | krotos | 3 day working on:) |
| 12:23.54 | krotos | the callerid on hints does not work |
| 12:26.09 | WIMPy | Bad luck. I liked the idea. |
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| 13:30.36 | alsuren | Hey guys: Any ideas why agi STREAM FILE might hang forever? |
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| 13:44.20 | krotos | WIMPy: you know if there are some way to notify two hint wit only one exten=? Something like exten=> 60,hint,Custom:blf1&Custom:blf2 |
| 13:44.27 | krotos | as i write before, not work :( |
| 13:45.33 | WIMPy | Should work exactely that way. |
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| 14:04.12 | krotos | WIMPy: i think it work when i specify SIP/user1&SIP/user2 |
| 14:04.23 | krotos | but when i use Custom:blf1&Custom:blf2 is not working |
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| 14:05.38 | WIMPy | It might be becuase of the states you're using. They have certain priorities. |
| 14:06.54 | krotos | do you know the right way for notify multple hint on same extension? |
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| 14:10.04 | LGPhoenix | Greetings! Does anyone have any information as to when and where AstriCon 2013 will be held? |
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| 14:19.42 | raden | morning alll |
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| 14:21.54 | krotos | morning |
| 14:24.20 | [TK]D-Fender | krotos, That is the syntax for regular devices. I'm betting they simply didn't account for people using anything after the first "Custom". This is somewhere between a bug to fix and a feature request |
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| 14:56.17 | Danerdo | Is this an open channel for anyone inquiring about Asterisk? |
| 14:56.36 | WIMPy | ~ask |
| 14:56.36 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 14:58.26 | Danerdo | Received an error: "WARNING[4786] app_dial.c:2341 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown)." Are there any recommendations that I can research that may shed some light on this error you can recommend? |
| 14:59.06 | leifmadsen | Danerdo: means the channel you're trying to dial isn't registered/online |
| 14:59.14 | WIMPy | The peer isn't reachable. |
| 14:59.14 | leifmadsen | or, that chan_sip isn't loaded |
| 15:00.17 | WIMPy | That should give another error. |
| 15:00.22 | leifmadsen | ah I couldn't remember |
| 15:00.30 | leifmadsen | because I never load asterisk without chan_sip anymore :D |
| 15:00.40 | Danerdo | My network admin assigned this this task to me. I'm just starting the long learning-curve. I appreciate the information. |
| 15:01.29 | kaldemar | Danerdo: you'll see with "sip show peers" that asterisk does not have an ip address for the peer that was dialed when that warning showed up. |
| 15:01.32 | Danerdo | First thank you for the insight. Can you advise a course of action I can read up on to delve into the error. |
| 15:02.15 | kaldemar | or did not have at the time, it may now if there was a temporary network issue for example. |
| 15:03.10 | Danerdo | The remote site is not receiving incoming CIDs on Friday. I'll admit I have to look at the server this morning, but wanted to educate myself beforehand. Google was of no help. |
| 15:03.34 | kaldemar | resolving it depends on the circumstances. is the peer a phone? has it worked before? is the peer configured to register to asterisk? |
| 15:07.31 | Danerdo | It is a phone, worked before and is registered. Howeve,r I didn't perform the "sip show peers" I will investigate the logs today to see if the error is present again. |
| 15:09.17 | kaldemar | is the phone behind a wireless network? |
| 15:10.44 | Danerdo | They are cabled. |
| 15:11.12 | Danerdo | It sounds like this is fully unrelated as to why the end phone recipients are unable to see caller ID. I appreciate the information. |
| 15:11.43 | WIMPy | Definitely unrelated. |
| 15:11.46 | *** join/#asterisk PbxMan (c335d959@gateway/web/freenode/ip.195.53.217.89) |
| 15:12.00 | WIMPy | caller ID from where to where? |
| 15:12.11 | PbxMan | hello |
| 15:12.18 | Danerdo | The 4786 error occured twice on Friday afternoon and fortunately isn't consistent. |
| 15:12.38 | Danerdo | All incoming calls were not showing CIDs. |
| 15:12.57 | WIMPy | Comming from where? |
| 15:12.59 | Danerdo | "dialparties.agi: Caller ID name is 'unknown' number is 'unknown'" |
| 15:13.00 | [TK]D-Fender | Danerdo, Sounds like you're just hitting a qualify timeout or something similar. increase it. |
| 15:13.45 | WIMPy | "dialparties" smells liek freepbx. |
| 15:14.02 | [TK]D-Fender | It is. |
| 15:14.18 | Danerdo | Yes it is. |
| 15:14.25 | Danerdo | Good confirmation. |
| 15:14.25 | *** join/#asterisk qakhan (~qakhan@208.253.91.58) |
| 15:14.45 | WIMPy | Then you need to debug that part in #freepbx. |
| 15:14.53 | *** join/#asterisk NOT_guru (~chatzilla@24-241-103-142.static.stls.mo.charter.com) |
| 15:15.00 | qakhan | i am using page app with d and q option. when i page there is beep in call |
| 15:15.45 | qakhan | it should not be beep in call while i am using q option, anyone can tell me how i can resolve this problem |
| 15:16.14 | WIMPy | Are you sure it's from Asterisk and not from the phone(s)? |
| 15:16.18 | kaldemar | qakhan: i've never had it play a beep with the q option. show a CLI output. |
| 15:17.34 | qakhan | kaldemar 1 min |
| 15:18.54 | *** join/#asterisk nantou (~phonetic@gateway/tor-sasl/martinphone) |
| 15:21.55 | Danerdo | WIMPy I'm not fully certain, but I believe it's on the asterisk side as it has occurred with two different extensions |
| 15:22.42 | [TK]D-Fender | Danerdo, There is no issue with "asterisk". Only the devices and services sending it calls. |
| 15:25.35 | Danerdo | gentleman, thank you much for your insight. I will look at my logs at this time. D-Fender and WIMPy I appreciate very much. |
| 15:27.12 | qakhan | kaldemar here is http://pastebin.com/EXn6EqWd |
| 15:27.59 | [TK]D-Fender | qakhan, pastebin your actual dialplan |
| 15:29.05 | [TK]D-Fender | qakhan, This is with Polycom phones isn't it? |
| 15:29.11 | qakhan | yes |
| 15:29.14 | [TK]D-Fender | qakhan, If so pastebin your PHONES configs as well\ |
| 15:29.15 | [TK]D-Fender | ^ |
| 15:29.37 | qakhan | you mean phone1.cfg? |
| 15:29.57 | [TK]D-Fender | I mean whatever your phones are using. |
| 15:30.07 | [TK]D-Fender | qakhan, They're your phones. You should know which ones |
| 15:30.46 | kaldemar | qakhan: what version of asterisk is this? |
| 15:31.23 | qakhan | kaldemar 1.4.38 |
| 15:31.39 | kaldemar | are you getting full duplex audio? |
| 15:31.48 | qakhan | yes |
| 15:32.20 | kaldemar | ok, then it's not the pipe. didn't remember when the pipe was removed as an argument delimiter. |
| 15:33.31 | qakhan | [TK]D-Fender here is my dialplan and phone conf |
| 15:33.55 | qakhan | kaldemar i am not using pipe and i am use comma " , " |
| 15:34.07 | qakhan | http://pastebin.com/08cUVEaZ |
| 15:41.29 | *** join/#asterisk Azrael808 (~peter@212.161.9.162) |
| 15:42.16 | *** join/#asterisk jrose_atDigium (~jrose_atD@nat/digium/x-yhujdnqgkbqlxkrt) |
| 15:44.23 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
| 15:46.51 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 15:46.51 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 15:46.56 | *** join/#asterisk Pegasus_RPG (~chatzilla@p5081F4C7.dip.t-dialin.net) |
| 15:47.40 | Pegasus_RPG | Hello. I just updated to the latest security update from Debian for * (v1.6.2.9) and now I get a segfault whenever it tries to open any SIP channels |
| 15:54.13 | mjordan | Pegasus_RPG: 1.6.2.x is no longer a supported version, even for security releases. |
| 15:54.36 | Pegasus_RPG | Okay. Guess it's time to dist-upgrade to Wheezy then |
| 15:56.49 | *** join/#asterisk elico (~Thunderbi@bzq-79-180-187-53.red.bezeqint.net) |
| 16:01.04 | [TK]D-Fender | qakhan, <OVERRIDES se.rt.custom1.name="Paging" se.rt.custom1.ringer="ringer15" se.rt.custom1.timeout="800" se.rt.custom1.type="ring-answer" se.rt.default.timeout="5000" up.idleBrowser.enabled="1" up.localClockEnabled="0" voIpProt.SIP.alertInfo.1.class="custom1" voIpProt.SIP.alertInfo.1.value="Paging" /> |
| 16:01.18 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.23.20) |
| 16:01.22 | [TK]D-Fender | qakhan, You POLYCOM phones are generating the ringing because that is what YOU told it to do |
| 16:04.16 | *** join/#asterisk jmetro (~nickserv@75-150-221-198-Illinois.hfc.comcastbusiness.net) |
| 16:05.17 | qakhan | where i told them? |
| 16:05.45 | [TK]D-Fender | qakhan, Was I unclear in any way? |
| 16:05.53 | qakhan | yes |
| 16:06.08 | [TK]D-Fender | qakhan, How? Your configs did it. |
| 16:06.25 | [TK]D-Fender | qakhan, You are responsible for them. It is not an Asterisk issue at all. |
| 16:06.41 | qakhan | can u highlight where i made that mistake |
| 16:06.42 | [TK]D-Fender | qakhan, Go read the Polycom Admin Guide. |
| 16:06.51 | [TK]D-Fender | qakhan, I did.. I pasted the entire line. |
| 16:09.43 | *** join/#asterisk DoSJustin (~justin@vpn.bctconsulting.com) |
| 16:16.55 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 16:17.27 | *** join/#asterisk rsd (~rsd@200.146.78.150.static.gvt.net.br) |
| 16:18.02 | rsd | has anyone used GOIP gsm VOIP gateways? are they good enough? |
| 16:18.18 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 16:19.39 | *** part/#asterisk Pegasus_RPG (~chatzilla@p5081F4C7.dip.t-dialin.net) |
| 16:37.57 | *** join/#asterisk Azrael808 (~peter@212.161.9.162) |
| 16:49.07 | *** join/#asterisk vinhdizzo (~vinh@vqn-portege.ics.uci.edu) |
| 16:53.49 | *** join/#asterisk gg608f (~Adium@187.207.6.250) |
| 17:01.55 | joris2 | Hi all, I've a question about asterisk monitoring. |
| 17:02.04 | joris2 | I'll explain: |
| 17:02.26 | joris2 | my asterisk acts as a client to an upstream sip provider |
| 17:02.44 | joris2 | in sip.conf is a register line and this all works fine |
| 17:03.14 | joris2 | I am willing to monitor the registration status of the sip subscription to my upstream provider |
| 17:03.59 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 17:03.59 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 17:04.00 | joris2 | this can be done by wrapping a shell script around the 'sip show registery' command |
| 17:04.31 | joris2 | but i would like to known what you would suggest... there's not funtion in the asterisk manager as far as I known |
| 17:04.47 | ghost75 | i use nagios for this |
| 17:05.11 | joris2 | my monitoring is also nagios (icinga) |
| 17:05.24 | ghost75 | there are plugins available for asterisk |
| 17:05.45 | joris2 | but they are all shell script around the astrisk CLI |
| 17:06.05 | joris2 | isn;t there an other way? |
| 17:06.20 | ghost75 | where is again path for nagios plugins? |
| 17:07.12 | *** join/#asterisk Katty (~Katty@97-91-121-70.static.stls.mo.charter.com) |
| 17:07.29 | joris2 | /usr/lib/nagios/plugins/ |
| 17:07.41 | Katty | hello my asterisk does not work at all how to fix?? plz answer is urgent thx. |
| 17:08.06 | *** join/#asterisk _Corey_ (~chatzilla@173-161-229-46-Philadelphia.hfc.comcastbusiness.net) |
| 17:08.35 | joris2 | Katty: what does /etc/init.d/asterisk restart do? |
| 17:08.56 | jmetro | Katty: did you try wrapping the phone cord going out of your PC in aluminum foil? just make sure to seal it with CAULK or the water will degrade your console |
| 17:09.24 | ghost75 | joris2: yes is using asterisk -rx .... |
| 17:09.24 | Katty | you know i did recaulk the shower last weekend. |
| 17:09.38 | Katty | it was easier than i thought it would be. |
| 17:09.47 | jmetro | ^ most house repair |
| 17:09.47 | Katty | joris2: that would restart the asterisk service. |
| 17:10.07 | Katty | jmetro: yes. we tore out a cabinent in the basement with a pry bar sunday. |
| 17:10.13 | Katty | jmetro: most fun House Repair ever. |
| 17:10.32 | ghost75 | other way is to use AMI |
| 17:10.39 | Katty | jmetro: there are two cracks to fix in the basement. then we can start building walls ^____^ |
| 17:10.57 | Katty | also! squirrels this morning! |
| 17:10.59 | joris2 | ghost75: I'm trying to find another way... more clean in my opinion |
| 17:10.59 | Katty | infobot: crittercam |
| 17:11.00 | infobot | it has been said that crittercam is Katty's Critter Cam http://tinyurl.com/b5k3lt4 |
| 17:11.07 | jmetro | i re-stained an old oak cabinet in the bathroom, and then put a frame up around the giant mirror above it stained the same color. Didnt take either peice off the wall. |
| 17:11.14 | ghost75 | there is nothing cleaner than ami |
| 17:11.15 | Katty | what a cute squirrel. |
| 17:11.30 | Katty | jmetro: i hope you were careful about it. |
| 17:11.42 | joris2 | true... but ami does not have an implementation for this (afaik) |
| 17:11.43 | Katty | jmetro: i wouldn't want my wallpaper to be Oak Stain color. |
| 17:11.55 | jmetro | Yep. It looks beautiful too. The most nerve-wracking part was using gorilla-glue to put the wood on the mirror to make the frame. |
| 17:12.06 | Katty | good lord. |
| 17:12.11 | Katty | you glued trim to the mirror? |
| 17:12.23 | Katty | well i guess that's one way of doing it ^_- |
| 17:12.25 | jmetro | its cheap, looks great, and super easy. |
| 17:12.30 | jmetro | works really really well. |
| 17:12.35 | Katty | well that's all that matters i suppose. |
| 17:12.38 | Qwell | until the mirror breaks |
| 17:12.43 | Katty | yeah. |
| 17:12.47 | Katty | then 10 years of bad luck for you! |
| 17:12.57 | Katty | ...mister squirrel. what /are/ you doing |
| 17:13.01 | chuckf | Katty: it looks like there's a bunch of small white animals imprisoned in your feeder |
| 17:13.03 | jmetro | well it was already mounted on the wall, so if the mirror breaks its the same amount of work to get it off. |
| 17:13.19 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 17:13.33 | Katty | chuckf: i don't know what that squirrel is doing. apparently the sunflowers in the bird feeder is more appetizing than on the sunflowers on the tray |
| 17:13.45 | Katty | chuckf: but i'm very amused, either way |
| 17:14.00 | Katty | chuckf: he's up to no good. and i like that. |
| 17:14.04 | chuckf | he's an active one |
| 17:14.10 | Katty | or at least curious. |
| 17:14.11 | *** join/#asterisk navaismo (~navaismo@189.191.2.44) |
| 17:16.01 | Katty | jmetro: i'd be interested in seeing how it turns out. |
| 17:16.08 | Katty | jmetro: take a picture next time you think about it |
| 17:16.18 | *** join/#asterisk JoeyJoeJo (~brian@pool-173-72-191-174.clppva.fios.verizon.net) |
| 17:16.50 | chuckf | So, to ask an asterisk question, when I get a call from an outside line in, there is about a 1 second delay in the voice. What might be an easy way to overcome that? Its happening on an * 11.1 box and, my wife tells me, the previous 1.8.x box I had up. Vitelity is the provider. |
| 17:17.49 | Katty | i used to have that problem here. |
| 17:18.07 | Katty | then i found that people were talking at the same exact instant they were hitting the answer button on the polycom |
| 17:18.10 | chuckf | This is a home system, so there isn't much load on it |
| 17:18.28 | Katty | there wasn't much i could do about it other than tell them wait for the call to connect |
| 17:18.34 | Katty | or pick up the headset |
| 17:18.56 | Katty | course you might have a different issue. |
| 17:18.59 | JoeyJoeJo | My company has two offices in different cities and they want to use asterisk so users in one office can easily call users in the other office. I'm new to asterisk, so does anyone know of a good diagram or howto guide for this type of setup? |
| 17:19.11 | chuckf | I'll give that a shot Katty, thanks |
| 17:19.13 | Katty | JoeyJoeJo: tunnel. |
| 17:19.17 | Katty | JoeyJoeJo: vpn tunnel. |
| 17:19.26 | Katty | JoeyJoeJo: put everyting on the same 'lan' |
| 17:19.31 | Katty | well. |
| 17:19.38 | Katty | 192.168.1.1 and 0.1 |
| 17:19.41 | Katty | still the same lan |
| 17:19.43 | Katty | but you know what i mean |
| 17:19.47 | JoeyJoeJo | Katty: That's what I thought. I already have openvpn setup between the two offices |
| 17:19.56 | Katty | eggg celent. |
| 17:20.12 | ghost75 | doesnt work over routers? |
| 17:20.28 | JoeyJoeJo | So I don't need an asterisk server at both locations, right? |
| 17:20.52 | ghost75 | right |
| 17:22.12 | Katty | JoeyJoeJo: newp. |
| 17:22.20 | Katty | JoeyJoeJo: as far as the asterisk box things, everyone's on the same network |
| 17:22.26 | Katty | s/things/thinks/ |
| 17:22.37 | Katty | JoeyJoeJo: just keep an eye on your bandwidth. |
| 17:22.43 | Katty | JoeyJoeJo: wouldn't want any pesky packet loss... |
| 17:22.47 | ghost75 | why it should matter in which network? |
| 17:22.49 | Qwell | if they're in different cities, and you want local PSTN connectivity, you really should have 2 boxes. |
| 17:23.09 | Qwell | ghost75: Because dealing with NAT when you don't have to is utterly silly. |
| 17:23.36 | ghost75 | if its only for connecting phones from outside, why not |
| 17:24.23 | JoeyJoeJo | If I can, I want to set it up so all calls originate from the main office. So if a call comes from the branch office, it uses the other office's PTSN |
| 17:24.49 | ghost75 | u want to use sip? |
| 17:25.07 | JoeyJoeJo | I don't know. If SIP can accomplish this, then yes |
| 17:25.52 | Katty | SIP can make toast. |
| 17:26.11 | Qwell | I'll toast *you*. |
| 17:26.33 | *** join/#asterisk kikohnl (~keith@udp278022uds.hawaiiantel.net) |
| 17:26.52 | Katty | in a minute? |
| 17:27.01 | Qwell | On medium. |
| 17:27.15 | Katty | mister squirrel would like the mostly frozen bowl of peanuts to be toasted on medium. |
| 17:27.24 | Qwell | peanuts can freeze? |
| 17:27.35 | Katty | well it rained some. and then it sleeted. |
| 17:27.41 | ghost75 | u need a reliable and fast wan for this |
| 17:27.42 | Katty | and so there was water between the peanuts. in the bowl. |
| 17:27.55 | Katty | then night happened. and it froze. |
| 17:29.11 | Katty | poor dear. |
| 17:29.18 | Katty | he's trying so hard. |
| 17:32.50 | ghost75 | poor one |
| 17:33.39 | Katty | oh boy, two squirrels on the same feeder. |
| 17:33.52 | Katty | seems like a recipe for disaster. they usually fight. |
| 17:33.57 | Katty | maybe they're related. |
| 17:34.25 | ghost75 | JoeyJoeJo how many users? |
| 17:35.10 | JoeyJoeJo | Close to 100 |
| 17:35.27 | ghost75 | in each office? |
| 17:35.32 | JoeyJoeJo | No, total |
| 17:35.37 | [TK]D-Fender | JoeyJoeJo, how many remote? |
| 17:35.50 | JoeyJoeJo | I'd guess it's about 70/30 |
| 17:36.02 | [TK]D-Fender | JoeyJoeJo, I'd recommend a small * box at the remote then |
| 17:36.47 | [TK]D-Fender | JoeyJoeJo, that way their traffic doesn't waste your bandwidth going from remote>head>remote and that they aren't screwed internally if your connection fails |
| 17:37.12 | [TK]D-Fender | JoeyJoeJo, And you can deal with a failover. Almost certainly a 911 requirement, etc. |
| 17:37.32 | JoeyJoeJo | Good points. Thanks! |
| 17:39.22 | ghost75 | is it difficult to configure iax ? |
| 17:41.08 | Katty | are there any laws about 911? |
| 17:41.30 | [TK]D-Fender | Katty, Plenty.\ |
| 17:41.43 | Katty | link? |
| 17:41.46 | [TK]D-Fender | ghost75, Less than SIP typically |
| 17:41.55 | [TK]D-Fender | Katty, www.google.com |
| 17:41.57 | [TK]D-Fender | :p |
| 17:42.02 | Katty | pats [TK]D-Fender |
| 17:42.09 | Katty | you're just /so/ helpful dear. |
| 17:42.11 | [TK]D-Fender | Katty, varies by country, state, etc.... |
| 17:42.11 | Qwell | Katty: it's going to be different down to the city level |
| 17:42.12 | ghost75 | how this works with dialplan when you have 2 servers |
| 17:42.17 | [TK]D-Fender | Even city |
| 17:42.23 | Katty | interesting. i had no idea. |
| 17:42.24 | [TK]D-Fender | ghost75, Same as anything else |
| 17:43.06 | [TK]D-Fender | ghost75, using a single peer to send calls to another server is no different that setting up for an ITSP, etc/. |
| 17:43.48 | *** join/#asterisk Penguin (~xwQ5kwYl6@cobalt.esxi.hosts.a2infotech.com) |
| 17:43.52 | ghost75 | then the other server is just a peer |
| 17:44.18 | jmetro | katty: i should setup a u-stream of the hummingbird feeder outside. we get a family of hummingbirds every year |
| 17:44.28 | Katty | jmetro: that'd be wonderful to watch. |
| 17:44.53 | ghost75 | and is there such thing as cluster? |
| 17:45.40 | ghost75 | active/passive ? |
| 17:48.33 | *** join/#asterisk BCrookAtRA (~bcrook@BillyCrook-4-pt.tunnel.tserv9.chi1.ipv6.he.net) |
| 17:48.49 | [TK]D-Fender | ghost75, No. |
| 17:49.08 | [TK]D-Fender | ghost75, * is every bit as flat and boring as it looks. |
| 17:49.36 | BCrookAtRA | My boss wants to pay a company to build and install an asterisk system for us. Any recommendations? (I've already looked at switchvox and fonality) |
| 17:49.38 | ghost75 | k then still virtualizing could be used |
| 17:50.02 | jmetro | bcrookatra build your own, become familiar with it, do your own support of it. |
| 17:54.10 | [TK]D-Fender | BCrookAtRA, What do you actually want to have & do with it? |
| 17:57.48 | BCrookAtRA | jmetro: I would like to do exactly that. It is clearly the best decision. Nontheless, My boss wants to pay a company to build and install an asterisk system for us. Any recommendations? (I've already looked at switchvox and fonality) |
| 17:58.40 | BCrookAtRA | [TK]D-Fender: receive calls through our t1 and through a new sip provider, answer on phones in the office, or in our remote workers' houses, or on sip clients on our smartphones |
| 17:59.07 | [TK]D-Fender | BCrookAtRA, How is it "clearly the best"? |
| 17:59.08 | BCrookAtRA | place calls outbound preferentially through T1, but through sip if T1 is full. transcribe voicemails to email |
| 17:59.35 | [TK]D-Fender | BCrookAtRA, And you can almost forget transcribing voicemail to e-mail |
| 17:59.45 | BCrookAtRA | why's that? |
| 17:59.54 | [TK]D-Fender | BCrookAtRA, Almost all ASR sucks far too bad for that and is a big hassle to try to integrate |
| 18:00.12 | BCrookAtRA | darn didelly darn |
| 18:01.19 | BCrookAtRA | we want a support queue for customers to reach the first person who can answer in a group, and to be able to initiate calls via https url like vonage offers |
| 18:01.25 | [TK]D-Fender | BCrookAtRA, So far everything you've stated (aside from ASR) can be done with a quick-install ISO using FreePBX |
| 18:01.43 | BCrookAtRA | I realise |
| 18:01.48 | [TK]D-Fender | Which doesn't require learning much Asterisk to get off the ground. |
| 18:02.18 | BCrookAtRA | boss wants to blow shit tonnes and lock ourselves in to a support contract so we can remain ignorant of the technology we depend on to do business |
| 18:02.36 | [TK]D-Fender | BCrookAtRA, Not necessarily |
| 18:02.46 | Katty | ha |
| 18:02.51 | Katty | BCrookAtRA: i deal with that every day. |
| 18:02.58 | [TK]D-Fender | BCrookAtRA, Just because someone else builds your setup doesn't mean it has to be difficult for you take ownership of or maintain. |
| 18:03.00 | Katty | BCrookAtRA: the only way the boss learns is if it bites him in the rear. |
| 18:03.06 | BCrookAtRA | i'm asking for recommendations on the type of rope we should use to hang ourselves |
| 18:03.10 | Katty | BCrookAtRA: so let it bite him the rear. |
| 18:03.17 | [TK]D-Fender | BCrookAtRA, You are drawing rather wild conclusions. |
| 18:03.18 | Katty | BCrookAtRA: i /completely/ understand. |
| 18:03.46 | BCrookAtRA | Katty: at least we can re-use the phones when he comes to his senses |
| 18:04.05 | Katty | BCrookAtRA: http://www.voip-info.org/wiki/view/Asterisk+consultants+USA <- try that |
| 18:04.22 | [TK]D-Fender | BCrookAtRA, Again you are painting a picture of the end result without a basis of comparison. |
| 18:04.40 | Katty | BCrookAtRA: learn what you can and try to maintain it yourself as much as possible afterwards. |
| 18:04.49 | Katty | BCrookAtRA: you'll find it's easier than contacting support most times. |
| 18:04.58 | Katty | BCrookAtRA: and if the boss doesn't like it...not your problem. |
| 18:05.05 | Katty | BCrookAtRA: not your software, not your responsibility. |
| 18:05.19 | Katty | BCrookAtRA: maybe when he gets bit in the rear end he'll listen to you. |
| 18:05.25 | Katty | BCrookAtRA: but don't count on it ;) |
| 18:05.58 | [TK]D-Fender | It's always your fault. Especially when it isn't. |
| 18:06.11 | Katty | BCrookAtRA: i'd get a couple quotes from the companies nearest to you. |
| 18:06.18 | Katty | BCrookAtRA: let the boss-man pick the company it comes from. |
| 18:06.25 | Katty | BCrookAtRA: that way that can't bite you in the rear either |
| 18:07.04 | *** join/#asterisk Janos (~Janos@186.4.6.239) |
| 18:07.19 | [TK]D-Fender | I'd amend that with "present him choices YOU can live with and let him pick from those". That way you aren't screwed right from the start and if you are in the end... then it's on him :) |
| 18:08.17 | BCrookAtRA | it amuses me to no end when i try and call an asterisk reseller and their phone sounds like shit |
| 18:08.27 | BCrookAtRA | this guy from fonality had about 800ms latency |
| 18:08.39 | Qwell | BCrookAtRA: take Fonality off your list entirely. |
| 18:08.42 | Qwell | Don't even bother. |
| 18:08.50 | BCrookAtRA | that's like a fire station that catches fire |
| 18:08.52 | Katty | BCrookAtRA: that's a bad sign. |
| 18:08.57 | Katty | BCrookAtRA: just don't go there. |
| 18:09.02 | jmetro | well youre also an inbound caller, and that determines a lot of things about the connection |
| 18:09.04 | Katty | BCrookAtRA: if they don't act professional, don't consider them |
| 18:09.44 | rrittgarn | I've had really good luck with NexVortex as of late. Very attentive, good service thus far. They only handle the sip setup, media is mostly processed by Level3 |
| 18:09.44 | Janos | tzafrir_laptop, hello, sorry to bother, i just updated asterisk in my debian squeeze box with this http://bugs.debian.org/cgi-bin/bugreport.cgi?bug=697230 and i´m getting a segmentation fault, do you have any other report about this ? if this is not the right channel for this let me know |
| 18:09.49 | Katty | BCrookAtRA:common sense sort of stuff. |
| 18:09.50 | Qwell | bchia: They aren't even worth the toll charges you spent, calling their toll-free number. |
| 18:09.54 | Qwell | BCrookAtRA too |
| 18:10.35 | tzafrir_laptop | Janos, right. See http://people.debian.org/~tzafrir/ast_squeeze10/ for a fix |
| 18:10.41 | tzafrir_laptop | Hope to upload it shortly |
| 18:10.44 | Janos | tzafrir_laptop, thanks a lot, checking |
| 18:16.17 | BCrookAtRA | What's the dirt on Fonality? |
| 18:16.42 | Katty | seems like we had them here once. |
| 18:16.45 | Katty | super expensive. |
| 18:16.52 | BCrookAtRA | I can't help but wonder now. If their mention drew such vitrol from the crowd, it must be juicy. |
| 18:16.55 | Katty | seemed like they were pawning trixbox at the time, i think |
| 18:17.27 | Katty | i've not had much /real/ interaction from them. |
| 18:17.33 | Katty | s/from/with/ |
| 18:18.15 | BCrookAtRA | why thank you infobot. I'm using IRC, but I've never heard of sed |
| 18:19.06 | Janos | tzafrir_laptop, that fixed it, thanks a lot |
| 18:20.37 | ghost75 | a fix for a fix oO |
| 18:21.20 | BCrookAtRA | it would be cool if my irc client could let people ammend their own messages in-place with sed syntax |
| 18:22.32 | ghost75 | seds dead |
| 18:25.27 | BCrookAtRA | I use it every day |
| 18:25.34 | fubada | rrittgarn: i use nexvortex here, not bad |
| 18:25.56 | BCrookAtRA | ghost75: it is undoubtedly in execution a dozen times right now by various scripts I have written |
| 18:26.16 | ghost75 | dont u know pulp fiction? |
| 18:27.04 | BCrookAtRA | ghost75: ooooooooooh |
| 18:27.19 | ghost75 | actually was zed :) |
| 18:27.44 | BCrookAtRA | yeah i remember |
| 18:28.05 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
| 18:32.12 | *** join/#asterisk moos3 (~moos3@pool-72-73-92-118.ptldme.east.myfairpoint.net) |
| 18:33.31 | moos3 | is there a way to create a function that i can past a sound file and have it plug my list of audio files to play |
| 18:45.35 | *** join/#asterisk luckman212 (~luckman21@unaffiliated/luckman212) |
| 18:46.33 | *** join/#asterisk fakhir_ (~fakhir@unaffiliated/fakhir) |
| 18:47.00 | [TK]D-Fender | moos3, that makes no sense. What is this list you're talking about? * doesn't have a "list". What's this about "pasting"? |
| 18:49.30 | moos3 | i have a macro that looks like this http://hastebin.com/qotiyivoro.coffee |
| 18:50.07 | moos3 | if i tell the dialplan to go to that how can I return to the same spot in the various menus that might get called form |
| 18:50.21 | Qwell | why don't you just pass in the files as args? almost like a macro or something. |
| 18:51.48 | [TK]D-Fender | Or like ... use priorities... or something.... |
| 18:52.02 | [TK]D-Fender | And use a GOSUB |
| 18:52.06 | [TK]D-Fender | ~book |
| 18:52.06 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 18:52.09 | moos3 | GOSUB ? |
| 18:52.09 | [TK]D-Fender | ^ |
| 18:52.15 | Qwell | ~gosub |
| 18:52.21 | Qwell | infobot: silly bot |
| 18:52.21 | infobot | :) |
| 18:52.25 | [TK]D-Fender | moos3, "core show application gosub: |
| 18:52.28 | [TK]D-Fender | moos3, "core show application gosub" |
| 18:52.36 | moos3 | [TK]D-Fender thanks |
| 18:52.39 | moos3 | Qwell thanks |
| 18:57.50 | *** join/#asterisk vlad_starkov (~vlad_star@178.176.227.140) |
| 18:58.15 | ghost75 | [TK]D-Fender: how was that module for freepbx called again to configure sip phones? |
| 18:58.56 | tm1000 | ?epm |
| 18:59.02 | ghost75 | yes |
| 18:59.14 | ghost75 | is this from you? |
| 18:59.15 | tm1000 | what are you asking. |
| 18:59.32 | tm1000 | is the module from me? |
| 18:59.36 | ghost75 | nothing right now, want to try it |
| 18:59.46 | tm1000 | use your words. what are you asking |
| 19:00.03 | ghost75 | nothing right now :) |
| 19:00.59 | jmetro | ~motto |
| 19:01.10 | jmetro | aw... our motto should be "use your words, what are you asking" |
| 19:07.34 | *** join/#asterisk luckman212 (~luckman21@unaffiliated/luckman212) |
| 19:09.11 | jmetro | Katty: You should add a water dish to the crittercam. Either that, or turn the squirrel feeder into one of those spinners. |
| 19:12.06 | jmetro | ala http://www.youtube.com/watch?v=F9H-HtBJ2fw |
| 19:14.30 | *** join/#asterisk coreyf1513 (~cfarrell@ws2.cfware.com) |
| 19:14.34 | [TK]D-Fender | jmetro, http://www.youtube.com/watch?v=KIp7V7VcCX8 |
| 19:14.39 | [TK]D-Fender | jmetro, 5 years ago.... |
| 19:14.45 | [TK]D-Fender | jmetro, That's old news around here ;) |
| 19:16.13 | jmetro | The yankee flipper looks like it gets higher velocity =p |
| 19:16.42 | [TK]D-Fender | jmetro, Check out the Binford 5000 ;) |
| 19:19.07 | jmetro | snowblower? |
| 19:21.52 | *** join/#asterisk timahvo1 (~rogue@41.212.120.45) |
| 19:23.19 | *** join/#asterisk vlad_starkov (~vlad_star@178.176.227.140) |
| 19:24.06 | *** join/#asterisk DoSJustin (~justin@vpn.bctconsulting.com) |
| 19:25.55 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 19:33.38 | *** join/#asterisk bn-7bc (~bjarne-im@macbook-pro.lan-sx.noare-1.holmedal.net) |
| 19:41.18 | jmls | is there any way of resetting a voicemail pin that a user has forgotten short of using either realtime or vi ? |
| 19:41.34 | [TK]D-Fender | jmls, "vi voicemail.conf" |
| 19:42.16 | jmls | short of using either realtime or vi ? |
| 19:42.19 | jmls | short of using either realtime or vi ? |
| 19:42.22 | jmls | short of using either realtime or vi ? |
| 19:42.23 | jmls | oops |
| 19:42.29 | [TK]D-Fender | jmls, Voicemailmain([box]@context,s) |
| 19:42.33 | jmls | bloodty irc client. sorry |
| 19:42.35 | [TK]D-Fender | jmetro, take your pick |
| 19:42.43 | [TK]D-Fender | jmls, ^ |
| 19:43.37 | jmls | yeah, I'll add an admin "password" to make sure that only admins can access the mailbox. |
| 19:43.45 | jmls | was hoping that there was an ami command |
| 19:43.52 | jmls | thanks |
| 19:49.00 | *** join/#asterisk vedic (~V@117.235.106.15) |
| 19:49.28 | *** join/#asterisk thatOneGuyHere (~thatOneGu@74.115.41.6) |
| 19:57.09 | Katty | looks in |
| 19:57.29 | jmetro | Squirrel spin cam \o/ |
| 20:11.21 | *** join/#asterisk hariom (~hariom@117.235.106.15) |
| 20:11.54 | ghost75 | Version 8.5(3) was released October 8, 2009. This release appears to have a new problem where the phone will continue to indicate that an inbound call is "ringing", even after asterisk has stop ringing the extension. |
| 20:11.55 | ghost75 | oO |
| 20:13.09 | hariom | Hello Friends, I am getting error while loading chan_alsa.so . I want to use 'dial' command from CLI |
| 20:13.51 | hariom | 65 modules will be loaded. *** Failed to load module chan_alsa.so - Required |
| 20:15.58 | hariom | When I manually try to load chan_alsa.so, it gives this msg: Unable to read ALSA configuration file alsa.conf. Aborting Unable to load module chan_alsa.so |
| 20:16.09 | hariom | From where is it looking for alsa.conf? |
| 20:19.14 | hariom | ok, I am restarting from system. Will be back in 1 min. Read in some forum that there is a bug so restarting help when loading chan_alsa.so |
| 20:22.06 | *** join/#asterisk hariom (~hariom@117.235.106.15) |
| 20:25.43 | *** part/#asterisk volga629 (~volga629@host7.pythian.com) |
| 20:26.25 | jzaw | any users of * sip and ipv6? |
| 20:26.57 | jzaw | re bindaddr=:: to eneable ipv6 (and ipv4) in sip.conf |
| 20:26.59 | Katty | hides |
| 20:27.08 | jzaw | is there anyway to bind actual ips |
| 20:27.25 | jzaw | i used to bind just one ipv4 ip |
| 20:27.29 | Katty | melted marshmallows are wonderious and binding things. |
| 20:27.35 | jzaw | as ive many ips on the interfaces |
| 20:27.41 | jzaw | same with ipv6 ips |
| 20:27.50 | Katty | you ever try to get a batch of rice crispy treats out of a pan? that stuff is better than gorilla glue! |
| 20:27.56 | jzaw | but it doesnt seem to work with ipv6 |
| 20:28.17 | jzaw | or oatflakes out of a bowl if you leave them 6 hours |
| 20:28.32 | Katty | never had that before. |
| 20:28.55 | jzaw | cornflakes too ... needs a pneumatic drill to remove those |
| 20:29.00 | Katty | lol |
| 20:29.10 | jzaw | but ... sip ipv6 ? |
| 20:29.15 | Katty | i'm guessing this comes from someone who doesn't load the dish washer every night. |
| 20:29.39 | jzaw | i used to be a single person ;) we'll say no more |
| 20:29.42 | Katty | i've not used ipv6, so i can't help. |
| 20:29.47 | Katty | ah. that explains it! |
| 20:29.50 | jzaw | np |
| 20:29.57 | *** join/#asterisk ujjain (ujjain@unaffiliated/ujjain) |
| 20:31.14 | *** join/#asterisk tamiel (~tamiel@208.66.27.62) |
| 20:33.03 | *** join/#asterisk ghost75 (~trechber@dslb-088-066-175-182.pools.arcor-ip.net) |
| 20:34.31 | ghost75 | do you use atftpd or tftpd ? |
| 20:34.59 | [TK]D-Fender | No |
| 20:36.12 | ghost75 | dnsmasq ? |
| 20:36.46 | [TK]D-Fender | nope |
| 20:37.00 | ghost75 | what else? |
| 20:37.22 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
| 20:38.33 | [TK]D-Fender | ghost75, You should rethink the nature of your questions. |
| 20:39.10 | ghost75 | motto: use your words, what are you asking |
| 20:39.21 | [TK]D-Fender | Exactly. |
| 20:39.24 | [TK]D-Fender | Your words suck :p |
| 20:39.35 | ghost75 | knows |
| 20:40.10 | [TK]D-Fender | You asked a vauge question about 2 specific tools with no specified target demographic and with no stated intent as to what it would be used for. |
| 20:40.13 | *** join/#asterisk mokmeister (~mokmeiste@93.107.26.110) |
| 20:40.31 | [TK]D-Fender | You gavfe no details so ANYONE can answer and it won't mean that it's pertinent to you |
| 20:41.13 | ghost75 | as we are in #asterisk obviously for asterisk |
| 20:41.30 | [TK]D-Fender | Asterisk doesn't use TFTP. For anything. |
| 20:41.40 | [TK]D-Fender | Again too vague. |
| 20:41.48 | ghost75 | phones (what else?) |
| 20:41.53 | [TK]D-Fender | You seem to have difficulty in being specific.]\ |
| 20:41.59 | [TK]D-Fender | WHICH phones? |
| 20:42.07 | ghost75 | this matters? |
| 20:42.11 | [TK]D-Fender | YES |
| 20:42.35 | [TK]D-Fender | because my phones support a LOT of different provisioning protocols AND can be configured via a web interface on them |
| 20:42.36 | ghost75 | why |
| 20:42.38 | [TK]D-Fender | ^ |
| 20:42.48 | *** join/#asterisk nuken (~nuken@open.integrada.coop.br) |
| 20:42.57 | [TK]D-Fender | Maybe I don't NEED any provisioning server at all. My response is still valid. |
| 20:43.11 | [TK]D-Fender | But FYI, I provision mine using VSFTPD. |
| 20:43.29 | ghost75 | cisco 79xx |
| 20:43.33 | [TK]D-Fender | Which is neither one of your 3 items |
| 20:43.58 | [TK]D-Fender | ghost75, And your question was a .... |
| 20:44.00 | [TK]D-Fender | ~poll |
| 20:44.00 | infobot | Script for automating Fidonet polls. URL: http://www.drmach.demon.co.uk/vashti/software/index.html |
| 20:44.07 | [TK]D-Fender | ~polls |
| 20:44.07 | infobot | "Does anyone have X or use Y?" is taking a poll, not asking a valid question. Don't do it or our army of rabid weasels will hurt you. Usually, people other than those with the exact same set up can help you and those who have sometimes will not be able to help you. Also see <ask> |
| 20:44.09 | [TK]D-Fender | ^^^^ |
| 20:45.08 | *** join/#asterisk nantou (~phonetic@gateway/tor-sasl/martinphone) |
| 20:46.47 | [TK]D-Fender | So ... rather then just getting an X users use Y... got a real question behind there? |
| 20:46.53 | nuken | does anyone already work with avaya pabx ? |
| 20:47.04 | *** join/#asterisk FLeiXiuS (~FLeiXiuS@unaffiliated/fleixius) |
| 20:47.10 | nuken | there is a feature called coverage path |
| 20:47.23 | nuken | i'm trying to do something like that... any suggestion ? |
| 20:47.38 | [TK]D-Fender | nuken, And what does this "coverage path" do? |
| 20:48.07 | nuken | Ok, we have group of extensions |
| 20:48.19 | nuken | and something like ring groups |
| 20:48.35 | *** join/#asterisk blee (~blee@67.8.206.215) |
| 20:49.47 | [TK]D-Fender | grabs some popcorn |
| 20:49.50 | [TK]D-Fender | ok, and? |
| 20:49.59 | nuken | for example, a sales department, four extensions, 200-204, imagine that all of them will be not answered |
| 20:50.30 | nuken | the 200 rings and 201 after,.. 202. ... 203 |
| 20:50.35 | *** join/#asterisk BarthezZ (~bart@monitoring.deheij-ict.nl) |
| 20:50.39 | nuken | after 20 seconds.. |
| 20:50.45 | [TK]D-Fender | nuken, So far this sounds like a Queue. |
| 20:50.57 | [TK]D-Fender | nuken, Nice and boring. Anything special after this? |
| 20:51.20 | nuken | i've try with queues |
| 20:51.34 | nuken | the problem is that i can't specify a order to extensions rings |
| 20:51.43 | [TK]D-Fender | nuken, Add memebers to your queue. Pick your distribution strategy. Can dial them in sequence. |
| 20:52.13 | nuken | i can work with sequeces in queues ? |
| 20:52.17 | nuken | can I * |
| 20:52.30 | [TK]D-Fender | nuken, Should be able to put the members in the order you want... |
| 20:52.47 | nuken | ok [TK]D-Fender. i will try ! |
| 20:52.53 | nuken | thanks |
| 20:53.08 | [TK]D-Fender | nuken, Worst case is if not you can just do up a little dialplan for them to do it. |
| 20:53.28 | [TK]D-Fender | That alone should take 5 mins tops. |
| 20:53.33 | nuken | i will have a lot of dialplan;... |
| 20:53.39 | nuken | i have about 100 extensions |
| 20:53.48 | [TK]D-Fender | 5 people in a group? 5 minutes work. |
| 20:53.57 | [TK]D-Fender | Let to build up for other groups |
| 20:54.09 | [TK]D-Fender | less* |
| 20:54.39 | nuken | but i will have to do a special dialplan for each exten rigth ? |
| 20:54.51 | nuken | to forward the call when have no answer |
| 20:54.53 | nuken | or busy |
| 20:55.02 | *** join/#asterisk DoSJustin (~justin@vpn.bctconsulting.com) |
| 20:55.38 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 20:56.19 | [TK]D-Fender | There is no "special". It's all just dialplan. Dial exits on time-out, busy, etc. |
| 21:16.54 | *** join/#asterisk twanny796 (~twanny@195.158.64.25) |
| 21:24.41 | [TK]D-Fender | checkout time, BBL |
| 21:25.13 | *** join/#asterisk kikohnl (~keith@udp278022uds.hawaiiantel.net) |
| 21:26.47 | twanny796 | I have a client on DMZ, what ports do I need to pinhole? |
| 21:27.42 | *** part/#asterisk twanny796 (~twanny@195.158.64.25) |
| 21:31.40 | *** join/#asterisk jmls (~somefake@77.107.171.82) |
| 21:37.47 | jzaw | i see twanny796 waited all of 54 seconds for an answer! |
| 21:43.33 | *** topic/#asterisk by mjordan -> #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 11.2.0 (2013/01/14), 10.12.0 (2013/01/14), 1.8.20.0 (2013/01/14), DAHDI-linux 2.6.1 (2012/04/20), DAHDI-tools 2.6.1 (2012/04/20), libpri 1.4.14 (2012/12/20) -=- Visit the official Asterisk wiki: wiki.asterisk.org |
| 21:56.47 | *** join/#asterisk TimeRider (~steve@timerider.plus.com) |
| 22:06.57 | *** join/#asterisk phix (~threat@123-243-44-131.static.tpgi.com.au) |
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| 22:23.55 | *** join/#asterisk gusto (~gusto@2001:470:1f0b:a42:224:1dff:fecd:234c) |
| 22:31.04 | gusto | hi, what's up? |
| 22:35.14 | *** join/#asterisk gbit (~gbit@unaffiliated/gbit) |
| 22:35.14 | *** join/#asterisk slidesinger (~slidesing@c-69-141-208-250.hsd1.nj.comcast.net) |
| 22:37.13 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 22:38.50 | *** join/#asterisk gg608f (~Adium@189.242.149.77) |
| 22:41.32 | gbit | Does anyone knows how to configure the Cisco SRP 521W FXO port? I got FXS working with asterisk, but can't figure out how to set up the FXO port to receive and make calls from asterisk. |
| 22:45.27 | *** join/#asterisk felipealmeida (~user@mvx-187-16-79-187.mundivox.com) |
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| 23:02.13 | *** join/#asterisk sampdc (~sam@static-50-47-42-82.sttl.wa.frontiernet.net) |
| 23:02.25 | sampdc | Hey all, I'm running Asterisk 11.1.2 and trying to turn sip debugging on. Looks like 'sip set debug on' isn't taking, has this changed with asterisk 11? |
| 23:03.16 | [TK]D-Fender | shouldn't have. Show us. |
| 23:03.19 | [TK]D-Fender | ~pb |
| 23:03.19 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 23:04.32 | sampdc | pastebin.com/WRCDQSGC |
| 23:04.35 | jmetro | you might not have chan_sip on if you cant do simple sip commands |
| 23:04.38 | jmetro | from modules. |
| 23:04.42 | *** part/#asterisk jmetro (~nickserv@75-150-221-198-Illinois.hfc.comcastbusiness.net) |
| 23:05.02 | [TK]D-Fender | sampdc: "sip show peers". Does that error out as well? |
| 23:05.53 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 23:05.53 | sampdc | [TK]D-Fender: Yep |
| 23:06.43 | [TK]D-Fender | sampdc: looking like cha_sip isn't eve loaded |
| 23:06.50 | [TK]D-Fender | module load chan_sip.so |
| 23:08.04 | sampdc | pastebin.com/cKfAzCxY |
| 23:08.45 | [TK]D-Fender | help sip <tab> |
| 23:09.55 | sampdc | pastebin.com/qA7QTuDC |
| 23:11.29 | gusto | that sip debug is still ugly |
| 23:12.23 | [TK]D-Fender | sampdc: I'm a little mystified there... |
| 23:13.17 | [TK]D-Fender | sampdc: module unload chan_sip.so |
| 23:13.21 | [TK]D-Fender | sampdc: module load chan_sip.so |
| 23:13.35 | [TK]D-Fender | the check the help and other commands. |
| 23:13.39 | [TK]D-Fender | PB all if failed |
| 23:14.02 | mjordan | hm. Most likely the module load failed on initial load but registered the SIP API. |
| 23:15.04 | mjordan | the SIP API shouldn't be registered until much later in the load sequence. It's currently registered quite early, so a failure in something else has a good chance of creating this situation |
| 23:15.25 | sampdc | pastebin.com/tfmhnhj6 |
| 23:15.30 | mjordan | you may want to look at your log file when you first loaded Asterisk - it should have the failure reason in it as to why chan_sip didn't get loaded properly. In the meantime however, this is a bug |
| 23:16.08 | mjordan | sampdc: correct, at this point you're hosed until you restart. When you start, it's going to attempt to load chan_sip and fail, but put you back into this position |
| 23:16.23 | mjordan | sampdc: I'd put a noload => chan_sip.so in your modules.conf so Asterisk starts up completely |
| 23:16.35 | mjordan | then perform a 'module load chan_sip.so' to manually load it |
| 23:16.43 | mjordan | it will still fail, but we should get the error reason as to why it failed |
| 23:17.09 | sampdc | mjordan: Interesting, so loading chan sip fails on startup but not on restart? |
| 23:17.27 | mjordan | not really |
| 23:17.49 | mjordan | there's lot of things it checks when it loads/reloads. One of the first is whether or not an API is exposes is registered with the Asterisk core |
| 23:18.01 | mjordan | if it's already registered, it bails. This is why you aren't able to unload/load the module. |
| 23:18.51 | mjordan | However, something else during startup failed - I'm not sure what. However, because it successfully registered the API - and it didn't de-register it when it failed - a part of it is sticking around in the Asterisk core |
| 23:19.19 | mjordan | that is preventing you from loading it now - you're asking it to load a module, but some piece of it is still sticking around saying "I'm already loaded" even though there is no module loaded |
| 23:19.32 | sampdc | Ahhh makes sense, I'll track that down. Thanks for the help! |
| 23:19.42 | mjordan | np - please do file a bug report |
| 23:19.45 | mjordan | that shouldn't happen. |
| 23:20.23 | sampdc | Will do, where again can I find the asterisk startup log file? |
| 23:20.32 | mjordan | /var/log/asterisk |
| 23:20.37 | mjordan | or whatever your logger.conf specified |
| 23:21.06 | sampdc | Thanks |
| 23:22.15 | jzaw | sorry to ask again .... is it possible to bindaddr specific ipv4 AND ipv6 addresses rather than just bindaddr=:: in sip.conf |
| 23:22.37 | jzaw | and q #2 .... does or how does iax ipv6 work at all? |
| 23:23.27 | jzaw | my experience with svn branch 11 is that if i bindaddr any ipv6 ip ... i dont get ipv4 connectivity |
| 23:23.52 | jzaw | and in any case i see messages like ...warning bindaddr ignored |
| 23:25.08 | mjordan | jzaw: ipv6 is not implemented in chan_iax2. |
| 23:25.28 | jzaw | i suspected as much mjordan ... shame |
| 23:25.40 | jzaw | ive had native ipv6 for near 8 years |
| 23:25.56 | mjordan | jzaw: would you like to contribute towards a solution? |
| 23:26.14 | jzaw | gosh if only i had the skill!! i certainly would |
| 23:26.19 | mjordan | kk. |
| 23:26.41 | jzaw | i can test and contribute in other collaborative ways |
| 23:27.13 | *** join/#asterisk yoyolala (~yoyolala@unaffiliated/yoyolala) |
| 23:27.14 | mjordan | if someone picks up the development of it, I'll let them know folks are interested in testing :-) |
| 23:27.30 | jzaw | cool ... count me in |
| 23:28.56 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-qkrsmlrsshnjsokl) |
| 23:29.33 | jzaw | btw for sip.conf whats the difference between udpbindaddr and bindaddr? |
| 23:59.34 | *** join/#asterisk minotaur01 (~minotaur0@S01060018e7f9c7df.hm.shawcable.net) |