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| 00:57.25 | SeRi | p3nguin: you in? |
| 00:57.26 | p3nguin | SeRi, Leave a message after the beep. *BEEP* |
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| 01:46.23 | WIMPy | Hmm. Do you think the Astribank is really bad? |
| 01:46.37 | WIMPy | I'm pretty excited to get my hands on it. |
| 01:56.43 | *** join/#asterisk lorsungcu (~anonymous@209-173-236-18.usfamily.net) |
| 02:02.29 | leifmadsen | astribank seems to have a lot of driver work done for it, so I can only suspect it must integrate fairly well with Asterisk |
| 02:02.42 | leifmadsen | tzafrir is pretty heavily involved with development |
| 02:02.55 | leifmadsen | I've never had an opportunity to try one, but I've always been interested |
| 02:06.03 | WIMPy | I know. but the comments earlier on were... interesting. |
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| 02:07.36 | jpsharp | I just can't bring myself to use USB like that. |
| 02:09.33 | WIMPy | I have to admit that I'm a little on the hardcore side as well. Usually servers I set up don't have USB support in the first place. |
| 02:10.21 | WIMPy | Nor do I have udev on a server usually. Had to change that for Digium hardware. |
| 02:13.19 | WIMPy | I tink the Astribank might be quite handy as a mobile debugging tool. Not that small, but still better to carry a laptop and an Astribank than a full PC. |
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| 02:38.16 | ChannelZ | "<WIMPy> I'm pretty excited to get my hands on it." Wait, I thought you shot it? |
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| 04:58.19 | FuriousGeorge | hey all. looking for a good asterisk init script for centos 6.2 |
| 04:58.52 | FuriousGeorge | what i find on voipinfo crashes asterisk... perhaps because i didn't uinstall dhadi, libpri etc |
| 04:59.06 | FuriousGeorge | i can start asterisk without the scripts, so i know it's working |
| 05:01.45 | ChannelZ | There are all kinds of scripts in the contrib directory |
| 05:04.10 | FuriousGeorge | ChannelZ: there are a bunch of scripts there, but i don't see any that look like init scripts |
| 05:04.25 | ChannelZ | nothing like the init.d directory? |
| 05:05.55 | FuriousGeorge | my bad i was in /contrib/scripts |
| 05:07.30 | kaldemar | FuriousGeorge: "make config" will install an init script for your system. |
| 05:07.39 | ChannelZ | Did your reinstall come back working? |
| 05:08.46 | FuriousGeorge | nah, im doing it now. same problem only from source i don't have working init scripts like using yum... eg now I get ERROR: __ASTERISK_SBIN_DIR__/asterisk not found when starting the service. still only one phone rings on inbound |
| 05:09.20 | FuriousGeorge | the only thing it could be as to the latter is my sip.conf, so im gonna try to rebuild that from scratch before I give up |
| 05:11.25 | FuriousGeorge | kaldemar: unless the init script is called something other than asterisk, it did not |
| 05:17.37 | ChannelZ | pastebin your sip.conf if you want. I don't remember if we saw it before or not.. |
| 05:19.49 | FuriousGeorge | ChannelZ: k, lemme take my credentials out of it frist |
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| 05:25.46 | FuriousGeorge | ChannelZ: in the process of redacting my credentials i see a comment that wrapped and was not re-commented |
| 05:25.51 | FuriousGeorge | lemme see if this makes the difference |
| 05:28.17 | FuriousGeorge | nm, that's just pastebin wrapping for me |
| 05:36.11 | FuriousGeorge | ChannelZ: http://pastebin.ca/2201330 |
| 05:36.34 | FuriousGeorge | i basically took the sample sip.cond and painstakingly merged my v1.2 sip.conf |
| 05:36.52 | FuriousGeorge | errr 1.4 |
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| 05:40.08 | [TK]D-Fender | a thousand lines of commented out junk |
| 05:40.09 | [TK]D-Fender | trash it |
| 05:41.49 | [TK]D-Fender | And your init script is not pointing to the proper asterisk.conf or similar... |
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| 05:42.52 | FuriousGeorge | [TK]D-Fender: thanks i got this one i think |
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| 06:19.02 | kaldemar | FuriousGeorge: sed < sip.conf 's/^\s*// ; /^;/d' |
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| 08:28.42 | bulkorok | hey... is it possible to use Wait() in the h-extension ?! |
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| 10:24.02 | WIMPy | ChannelZ: I shot it on ebay :-) |
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| 10:47.15 | hariom | I have experience in installing asterisk on remote server in non root execution mode. But trying FreePBX first time. How would one access FreePBX remotely then? |
| 10:49.44 | kaldemar | ~freepbx |
| 10:49.45 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 10:51.33 | hariom | Are there any drawbacks in running Asterisk on a server inside a virtual machine? |
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| 10:52.29 | hariom | I have a 8 core server with 8 GB RAM. In my host system, I am running webserver and few other scripts. I want to keep Asterisk which will connect to E1 line inside a virtual machine (as I can't afford another server). |
| 10:53.11 | hariom | What could be problems with running Asterisk with E1 line inside a virtualmachine like VirtualBox |
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| 11:00.16 | hariom | I have tried and succeeded connecting Asterisk over GPRS connection when used with FreePBX. But I can't connect Asterisk over GPRS without FreePBX. What is missing in Asterisk? |
| 11:02.37 | WIMPy | How to you plan to get that E1 in to your server? |
| 11:04.34 | hariom | WIMPy: Via E1 card in a PCI slot. |
| 11:04.57 | WIMPy | As far as I know you can't access them from a virtual machine. |
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| 11:24.52 | hariom | Can anybody give pointers that can explain about what extra is installed by FreePBX compared to what asterisk already has? I want to note if there are any feature that are not part of default Asterisk but only available in FreePBX |
| 11:25.33 | WIMPy | You should ask that in #freepbx |
| 11:26.10 | WIMPy | But the question really is: What features are pre-configured and which you have to add yourself. |
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| 11:28.20 | bulkorok | hey... if I issue an AGI script via AGI() in a h-extension and in that agi is an sleep(60)... the channel in asterisk is kept open as long as the sleep takes... |
| 11:28.28 | bulkorok | after a while I get |
| 11:28.43 | bulkorok | [Sep 5 13:25:30] WARNING[14618]: chan_sip.c:3872 __sip_autodestruct: Autodestruct on dialog 'xxx' with owner in place (Method: BYE). Rescheduling destruction for 10000 ms |
| 11:29.00 | bulkorok | bug or feature or working as expected!? |
| 11:29.06 | bulkorok | how can I avoid this message?! |
| 11:30.07 | WIMPy | Why do you sleep while tearing down the channel? If you need a sleep, it might be a good idea to start that script in the background. |
| 11:30.35 | bulkorok | it's faxing.... |
| 11:30.50 | bulkorok | I want to resend the fax after x seconds if sending has failed before... |
| 11:31.18 | bulkorok | how do I start "in background" ?! |
| 11:31.31 | WIMPy | Well, yes, don't do it in the old call then. |
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| 11:32.03 | bulkorok | another idea?! |
| 11:32.10 | bulkorok | with a variable?! |
| 11:32.12 | WIMPy | Does it have to be an AGI? Otherwise you can just use &. If it has to be an AGI you have to do it from within the script. |
| 11:32.27 | bulkorok | mmh |
| 11:33.53 | bulkorok | how to send the & with pearl exec ?! |
| 11:34.01 | bulkorok | perl... |
| 11:36.47 | kaldemar | why are you using exec()? why not system()? |
| 11:37.23 | bulkorok | don't know anymore... but I know I did that choice for a reason... |
| 11:38.03 | bulkorok | use system instead of exec if you want it to return |
| 11:38.14 | bulkorok | I don't have to return... |
| 11:40.31 | kaldemar | bulkorok: use the single argument version. |
| 11:40.43 | bulkorok | !? |
| 11:42.23 | kaldemar | bulkorok: exec('/path/to/cmd arg arg arg &') |
| 11:42.34 | bulkorok | I'll try... |
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| 11:49.24 | mirela666 | MixMonitor doesn't support mp3, damn shame :) |
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| 11:53.06 | nunne | using debian squueze + asterisk from apt. I'm having weird problems with listening to voicemail from cellphones coming in via a sip trunk. Im also having problems with language is set wrongly on these channels as well. even though language is set correctly in sip.conf. anyone else experiencec this?? weird is that it clipps the audio dialing into the voicemail from these cellphones, at random, but dialing from a regular sip ph |
| 11:53.06 | nunne | one both language is set correctly and also no problem with audio clipping. And I have set the channel to answer etc before going into voicemail. even the old playback(silence/1) trick as well. to no use :( |
| 11:54.32 | kaldemar | nunne: use CHANNEL(language) in dialplan instead of setting the language in sip.conf. |
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| 11:56.13 | TheCompWiz | nunne: First... take a deep breath and take one issue at a time. Second, Are you sure the clipping audio isn't due to the cell-phone cutting out? Third, What is different about your "regular sip phone" and the cell-phone? |
| 11:59.03 | kaldemar | the clipping might be due to network issues or because of your provider, if the audio works ok locally. |
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| 12:14.40 | nunne | kaldemar, that is what I'm doing. But I just find it extremley weird that on calls comming in on the sip-trunk is set as english instead of swedish. |
| 12:16.02 | WIMPy | If you configured that peer to use swedish, it's obviousely not using it, i.e. comming in as guest. Maybe you need an insecure=. |
| 12:16.23 | nunne | TheCompWiz, thats what I'm wondering as well! The only real difference I can think of is that using this provider the call comes in as ulaw. and the rest of the system is using alaw as default. but all settings allow both alaw and ulaw. and no. we are talking about two different sites one with 60+ devices and one eith 20+ devices. and on one site they just recently changed PBX.. same config.. but comming from centos + asteris |
| 12:16.23 | nunne | k to debian + asterisk. |
| 12:16.49 | nunne | WIMPy, good call. I'll take a look |
| 12:17.36 | nunne | the funny part is that it cuts the audio. just at random. either on the voicemail it self. or the menu. but it doesnt terminate the call. so if i press next for example it will start playing the next message.. and cut that one at random as well. |
| 12:17.56 | TheCompWiz | nunne: Have you installed the ulaw/alaw specific voice-prompts? |
| 12:18.12 | WIMPy | That sounds rather strange. |
| 12:20.01 | TheCompWiz | I suspect it's got XX/voiceprompt.alaw ... but not XX/voiceprompt.ulaw ... so it defaults to non-language specific |
| 12:20.05 | nunne | TheCompWiz, no. they are only avaible in .gsm unfortunatley. but that shouldnt mean it should break the actualy voicemessage someone has left |
| 12:20.31 | WIMPy | right |
| 12:20.40 | TheCompWiz | ... what does language have to do with voicemail? |
| 12:20.45 | nunne | WIMPy, they are all set to insecure = invite,port. so it's not comming in as guest. because it ends up in the right context |
| 12:21.05 | WIMPy | Sounds like some RTP (timing) issue |
| 12:21.24 | TheCompWiz | check dahdi_test |
| 12:21.36 | nunne | TheCompWiz, it uses the voice prompt language of the channel variable? |
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| 12:21.43 | WIMPy | But it doesn't use the configured language? |
| 12:22.01 | nunne | so it's not so nice that they end up with english prompts from their cellphones instead of english. but i have solved this with setting the channel variable |
| 12:22.05 | TheCompWiz | nunne: if there's not a voice-prompt it can use in the specified language selection... it reverts back to using a non-language specific prompt. |
| 12:22.37 | TheCompWiz | i.e. /var/lib/asterisk/sounds/voiceprompt.gsm instead of /var/lib/asterisk/sounds/language/voiceprompt.gsm |
| 12:22.43 | nunne | TheCompWiz, it works.. if i set the channel variable. but i want to know why it ignores language in sip.conf for these trunks |
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| 12:23.11 | TheCompWiz | nunne: perhaps you set it incorrectly someplace else in your dialplan? |
| 12:23.22 | nunne | well. debian has screwed up the location of the voice prompts recently.. but this is not the real problem. I have no problem playing the prompts. just that it cuts the prompts at random |
| 12:23.42 | nunne | it starts playing it... and then at random (and sometimes not at all). clips it |
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| 12:24.14 | nunne | its on two different installations. no setting language anywhere in the dialplan, until i put in this "fix". |
| 12:24.18 | TheCompWiz | 1 more reason not to use the repo. |
| 12:24.24 | kaldemar | nunne: enable verbosity (core set verbose 10) and sip debug and make a call. you'll see which peer it matches if it does. |
| 12:24.39 | nunne | but never mind the language.. i can live with that.. but i just find it a weird behaviour.. that maybe asteriskversion in apt is broken |
| 12:25.53 | nunne | kaldemar, i know it matches a peer that exists. because i have somehing like 200 sip peers to the same host. but it always enters the top most sip peer in sip.conf.. which is the behaviour i have always seen.. which is okay. |
| 12:26.10 | nunne | i have not set language variable on the peer it self. just the global one in sip.conf |
| 12:27.06 | nunne | and none of the boxes are behind NAT as well. So I have no idea why it clips my audio :/ and why it works using a fixed SIP-phone on their local net.. |
| 12:27.48 | WIMPy | I put my bets on silence suppression. |
| 12:28.26 | WIMPy | At least until last time I tried, Asterisk did not like that at all. |
| 12:28.55 | WIMPy | But that last time was probably way back in 1.8. |
| 12:31.25 | nunne | WIMPy, it's a provider. so it doesnt use silence suppresion (the provider uses asterisk as well;)). |
| 12:32.09 | TheCompWiz | nunne: what is the result of dahdi_test? |
| 12:32.10 | *** join/#asterisk dtcrshr (~datacrush@unaffiliated/datacrusher) |
| 12:33.00 | nunne | Unable to open dahdi interface: No such file or directory |
| 12:33.14 | nunne | TheCompWiz, that doesnt sound to "healthy" :P |
| 12:33.51 | kaldemar | nunne: that's ok if you don't use DAHDI. |
| 12:34.11 | TheCompWiz | what is your timing source set to? |
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| 12:34.42 | nunne | kaldemar, not using dahdi |
| 12:37.22 | nunne | TheCompWiz, res_timing_timerfd.so Timerfd Timing Interface 26 |
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| 12:38.15 | *** mode/#asterisk [+o mjordan] by ChanServ |
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| 12:38.33 | TheCompWiz | weeeeeeeeee... upgrading drivers is fun! |
| 12:38.47 | IsUp | TheCompWiz: always |
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| 12:39.26 | nunne | a little google on that timing gave me: http://us.generation-nt.com/answer/bug-557888-solution-help-196865711.html |
| 12:39.30 | TheCompWiz | nunne: did you try "timing test" in asterisk? |
| 12:39.32 | nunne | might be something... hmm. gonna try it |
| 12:40.03 | nunne | Attempting to test a timer with 50 ticks per second. |
| 12:40.03 | nunne | Using the 'timerfd' timing module for this test. |
| 12:40.03 | nunne | It has been 1000 milliseconds, and we got 50 timer ticks |
| 12:40.14 | TheCompWiz | try "timing test 10000" |
| 12:40.34 | nunne | Attempting to test a timer with 10000 ticks per second. |
| 12:40.34 | nunne | Using the 'timerfd' timing module for this test. |
| 12:40.34 | nunne | It has been 1000 milliseconds, and we got 9982 timer ticks |
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| 12:40.41 | TheCompWiz | is that thing in a VM? |
| 12:40.49 | nunne | TheCompWiz, nope |
| 12:41.07 | TheCompWiz | 99.8% isn't terrible... but not flawless. |
| 12:41.18 | *** join/#asterisk l2trace99 (~l2trace@70-119-141-5.res.bhn.net) |
| 12:41.19 | TheCompWiz | run it a few more times & lemme know the avg you get. |
| 12:41.55 | TheCompWiz | (high/low/average...) |
| 12:41.55 | nunne | on the other box, which is faaaar better then this, i got 9992 |
| 12:42.25 | TheCompWiz | 3 9s is usually the minimum I am "happy with" |
| 12:42.58 | nunne | the better box has 9991-9997 |
| 12:43.13 | TheCompWiz | that's pretty good. |
| 12:43.18 | TheCompWiz | what about this box? |
| 12:44.04 | nunne | and the "crappy" got a bunch of 10000 and average 9994.. but one was down to 9964 even. |
| 12:44.34 | TheCompWiz | I've seen worse... so it should be *ok* for most things... |
| 12:44.45 | mandrei | hello guys. I have a problem with Asterisk 1.8.15.0. I use tls transport. My softphones can register but in Asterisk CLI i see that they have status=Unrecheable and "Unable to connect SIP socket to ip:port : Connection refused" permanently appear. when i use udp transport everything is just fine. What should i do to monitor all sips with tls encryption? Thak you in advance. |
| 12:45.50 | TheCompWiz | nunne: you could try switching to pthread timing instead of timerfd... maybe you'll get better results. |
| 12:46.39 | *** join/#asterisk ks3_ (~ks3@74.115.41.6) |
| 12:48.17 | *** join/#asterisk bchia (~Adium@user-24-236-95-16.knology.net) |
| 12:48.36 | nunne | another "funny" thing is that when the audio cuts.. and it plays the next "option" file etc. it will not be heard either. but if i for example press 6 for next etc. it will start playing againm |
| 12:50.44 | nunne | TheCompWiz, I will try to change timing interface.. since other people seems to have problems with this in debian squueze as well |
| 12:51.06 | TheCompWiz | honestly ... I have no issues with timing in debian. but all my stuff is purpose built. |
| 12:51.48 | TheCompWiz | and I refuse to use the deb packages. it's just easier to build it myself. |
| 12:52.59 | nunne | TheCompWiz, I think so to. Had some other issues with the deb packs as well.. both permission and file location. so will probably rebuild myself. but i need a quick fix at the moment :P |
| 12:53.13 | nunne | apparently people need to hear their voicemails, haha. |
| 12:53.25 | TheCompWiz | pfft... just email it to 'em :D |
| 12:53.53 | *** join/#asterisk darksk1ez (~mhb@fsf/member/darkskiez) |
| 12:53.55 | mandrei | guys can you help me? |
| 12:55.47 | TheCompWiz | mandrei: Sorry, I'm not a licensed Psychoanalyst. Perhaps you should consult an Ouija board. |
| 12:55.49 | nunne | TheCompWiz, apparently "push mail" kills their poor little iphone batteries :P |
| 12:56.05 | TheCompWiz | nunne: pfft... iphone. |
| 12:56.09 | mandrei | TheCompWiz I have a problem with Asterisk 1.8.15.0. I use tls transport. My softphones can register but in Asterisk CLI i see that they have status=Unrecheable and "Unable to connect SIP socket to ip:port : Connection refused" permanently appear. when i use udp transport everything is just fine. What should i do to monitor all sips with tls encryption? Thak you in advance. |
| 12:56.13 | *** join/#asterisk kriger (~norge@68-190-90-80.dhcp.mdsn.wi.charter.com) |
| 12:56.47 | TheCompWiz | mandrei: is there a firewall in the middle someplace? |
| 12:56.59 | mandrei | no |
| 12:57.08 | TheCompWiz | all on the lan? |
| 12:57.15 | mandrei | yes |
| 12:57.25 | mandrei | for udp as i said everything is working |
| 12:57.43 | nunne | TheCompWiz, I know.. Then you understand what users I'm dealing with her.. they complained about their iphone softphone "drained" battery as well.. and they have no reason to use that app, unless they are abroad.. but it's an APP! very cool stuff for them :P |
| 12:59.02 | TheCompWiz | who buys a smartphone and doesn't use it for receiving emails? |
| 13:00.13 | *** join/#asterisk serafie (~erin@nat/digium/x-qrbcojviwrtuefoj) |
| 13:00.32 | TheCompWiz | mandrei: I can only guess at this point... but it looks like your phones are rejecting connections from the asterisk box. |
| 13:01.29 | mandrei | TheCompWiz i also get in /var/log/asterisk/messages this error "Sep 5 03:57:10 localhost avahi-daemon[1911]: Invalid query packet.", could this happen because of avahi-deamon error? |
| 13:02.07 | TheCompWiz | what does avahi have to do with the price of tea in china? |
| 13:05.34 | IsUp | mandrei: you can try capturing packets with 'tcpdump -i eth0 -s 65000 -w test.pcap port 5060' and see whats going on |
| 13:06.10 | IsUp | mandrei: i dont know anything about TLS so much but its using 5060 port too as far i know |
| 13:07.37 | *** join/#asterisk evil-man (~evil-man@insider-mail.icf.org.ru) |
| 13:08.28 | TheCompWiz | mandrei: did you set transport=tls in your extensions' config? ... or did you leave it unset? |
| 13:08.35 | mandrei | IsUp it is using 5061 port |
| 13:08.39 | mandrei | yes |
| 13:08.49 | mandrei | and encryption= yes |
| 13:08.54 | TheCompWiz | if you didn't set tls as the transport... I think asterisk will still try udp. |
| 13:09.14 | TheCompWiz | is there a NAT? |
| 13:09.17 | mandrei | no |
| 13:09.38 | TheCompWiz | from the asterisk box... can you telnet to the phone on port 5061? |
| 13:09.38 | Nugget | telnet is eeeeeeevil! |
| 13:09.56 | TheCompWiz | (no arguments... but it's a quick & dirty test) |
| 13:10.22 | *** join/#asterisk gusto (~gusto@2001:a60:11ff:1200::42:4) |
| 13:10.41 | bulkorok | kaldemar: exec with & or \& didn't help |
| 13:10.59 | *** join/#asterisk dms (~dms@nat/digium/x-wcratghatccrmpoi) |
| 13:11.18 | kaldemar | bulkorok: & makes it run a command in the background. |
| 13:11.22 | TheCompWiz | of course not... the ampersand is a reserved linux command line variable. |
| 13:11.34 | TheCompWiz | s/variable/parameter/ |
| 13:11.37 | bulkorok | kaldemar: yeah, but the channel is still open... |
| 13:11.55 | TheCompWiz | bulkorok: what are you trying to accomplish? |
| 13:11.58 | kaldemar | bulkorok: well, i have no idea what your setup does. |
| 13:12.08 | bulkorok | me neither ;-) |
| 13:12.34 | bulkorok | so... I have script that makes an originate |
| 13:12.37 | bulkorok | vi manager... |
| 13:12.46 | bulkorok | several vars... |
| 13:12.55 | TheCompWiz | and what does that have to do with exec? |
| 13:12.58 | bulkorok | and if originate is ok SendFax() is called |
| 13:13.08 | TheCompWiz | ... still waiting for the exec bit. |
| 13:13.26 | bulkorok | and... if the fax failed for any reason it sohould be resend after one minute... |
| 13:13.36 | TheCompWiz | and exec is used to do what? |
| 13:13.39 | bulkorok | for that I call a per-script in the h-extension |
| 13:13.46 | TheCompWiz | ... and why exec? |
| 13:13.51 | bulkorok | with exec that calls the initial script again |
| 13:13.56 | TheCompWiz | nope it doesn't. |
| 13:14.57 | *** join/#asterisk darksk1ez (~mhb@fsf/member/darkskiez) |
| 13:15.11 | bulkorok | so... sendfax-script => failed-fax => AGI(faxsent.pl) => exec sendfax-script and wait there for 1 minute |
| 13:15.40 | bulkorok | but the initial channel remains up until the second sendfax-script is executed... |
| 13:16.00 | TheCompWiz | as it should do. |
| 13:16.09 | Katty | stretches |
| 13:16.09 | kaldemar | bulkorok: sure if the hangup extension does not fall through. |
| 13:16.31 | TheCompWiz | snuggles up to a warm Katty. |
| 13:16.32 | bulkorok | what do you mean!? |
| 13:16.52 | TheCompWiz | bulkorok: why would the channel terminate? |
| 13:17.20 | Katty | TheCompWiz: morning |
| 13:17.35 | bulkorok | i'm not sure if I'm connected if asterisk is in the h-extension... |
| 13:17.52 | bulkorok | connected with the called one |
| 13:17.56 | *** join/#asterisk hariom (~V@117.195.170.54) |
| 13:17.59 | TheCompWiz | mornin' Katty! |
| 13:18.06 | kaldemar | bulkorok: maybe you should pastebin your dialplan, the AGI and CLI output of a call. |
| 13:18.21 | bulkorok | maybe ;-) |
| 13:18.26 | TheCompWiz | or maybe you should try & explain what you *want* to happen. |
| 13:19.11 | bulkorok | I would like... If I send a fax with SendFax and the transmit fails, that it will be resend 3 times with a break of 1 minute between the next tries... |
| 13:20.26 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 13:20.27 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 13:20.32 | TheCompWiz | bulkorok: what initiates the "sendfax" |
| 13:21.22 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 13:21.22 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 13:21.52 | bulkorok | it's an originate to the destination with sending to a context/extension/prio |
| 13:22.00 | bulkorok | and there is the sendfax... |
| 13:22.32 | *** join/#asterisk bchia (~Adium@nat/digium/x-ecuvhazeymfnojxb) |
| 13:22.41 | bulkorok | mom |
| 13:23.27 | TheCompWiz | bulkorok: you could originate another call on failure. |
| 13:23.43 | bulkorok | in the h-exten!? |
| 13:23.57 | TheCompWiz | sure... why not? |
| 13:24.23 | TheCompWiz | you can do it anywhere actually... the "h" is just as good a place as any. |
| 13:24.52 | bulkorok | mmh... how do I send vars via dialplan originate!? |
| 13:25.18 | bulkorok | I'll show you my actual script for originate |
| 13:27.41 | TheCompWiz | twiddles thumbs. |
| 13:28.02 | bulkorok | pastes & copies as hell... |
| 13:28.43 | TheCompWiz | I hope you're doing that the other way around. pasting then copying ... doesn't usually work. |
| 13:29.15 | bulkorok | seems like... I have some text ^^ |
| 13:29.28 | bulkorok | http://pastebin.com/DEaWBbFv |
| 13:32.24 | nunne | TheCompWiz, changing timing source worked btw.. So seems it's broken in debian squeeze |
| 13:34.19 | TheCompWiz | nunne: glad to hear it |
| 13:35.19 | bulkorok | is back in 30 minutes |
| 13:37.15 | Kobaz | speaking of timing source |
| 13:37.40 | *** join/#asterisk _Corey_ (~chatzilla@64.215.11.114) |
| 13:37.43 | Kobaz | i've found that in certain kernel versions and hardware, asterisk's use of timerfd conflicts with the network driver |
| 13:38.29 | *** join/#asterisk qakhan (~qakhan@70-88-142-141-regency-cab-md.hfc.comcastbusiness.net) |
| 13:38.33 | qakhan | hi all |
| 13:38.48 | leifmadsen | Kobaz: that's why I use nothing but res_timing_dahdi |
| 13:38.52 | qakhan | can i use Avaya ip phone with asterisk exts? |
| 13:42.51 | [TK]D-Fender | qakhan, Does it speak a protocol that * speaks? |
| 13:43.38 | [TK]D-Fender | <bulkorok> mmh... how do I send vars via dialplan originate!? <- you can't. use another method |
| 13:44.27 | [TK]D-Fender | AMI would do it... |
| 13:45.09 | Kobaz | leifmadsen: yeah |
| 13:45.24 | Kobaz | leifmadsen: well i usually do but i didnt build a new one for the new kernel i was using |
| 13:45.27 | Kobaz | so it used fd |
| 13:45.42 | Kobaz | and after 1-2 minutes all my phoes went unreachable |
| 13:47.29 | leifmadsen | neat. |
| 13:47.44 | Kobaz | yeap |
| 13:47.54 | Kobaz | no more traffic on eth1 |
| 13:48.01 | Kobaz | and i loaded dahdi and it fixed that |
| 13:48.10 | Kobaz | but it took me about two weeks to figure that out |
| 13:49.26 | leifmadsen | odd |
| 13:50.02 | Kobaz | and it was strange |
| 13:50.08 | Kobaz | all the phones went unreachable one by one |
| 13:50.15 | Kobaz | like i could still ping ones at the bottom of the list |
| 13:50.21 | Kobaz | while the ones at the top were totally blocked |
| 13:51.44 | *** join/#asterisk vlad_starkov (~vlad_star@213.79.102.163) |
| 13:54.33 | qakhan | [TK]D-Fender yes. i want to use avaya phones with asterisk exts |
| 13:57.37 | mandrei | guys what can be wrong with my asterisk? phones are registered, i can make encrypted calls using tls transport but they are not displayed as registered in CLI when executing sip show peers? |
| 13:59.24 | [TK]D-Fender | qakhan, that is not what I asked |
| 13:59.40 | [TK]D-Fender | qakhan, I did not ask if you wanted to USE them. You already TOLD us you did |
| 14:01.59 | bulkorok | [TK]D-Fender: I call an perl agi in the h-exten already... but the channel is up until the next originate is done... |
| 14:02.55 | qakhan | [TK]D-Fender what are u asking? |
| 14:03.37 | *** join/#asterisk zerohalo (~zerohalo@74.61.196.236) |
| 14:04.06 | [TK]D-Fender | <[TK]D-Fender> qakhan, Does it speak a protocol that * speaks? |
| 14:04.11 | [TK]D-Fender | qakhan, What is unclear about that? |
| 14:05.11 | _Corey_ | qakhan: If you're looking for help on HOW to use an Avaya phone with Asterisk, you're probably not going to find much help here or from Avaya. If it talks SIP, it may work fine but you're going to have to do the research and try it out. (I had bad results a few years ago with a batch of Avaya IP phones...) |
| 14:05.56 | qakhan | i just want to know can i use avaya phone with asterisk * |
| 14:06.10 | _Corey_ | You can use just about any SIP phone with Asterisk... |
| 14:06.57 | _Corey_ | There are a lot of different Avaya phones though, so we have not enough information to answer your question. |
| 14:07.16 | [TK]D-Fender | qakhan, DO THEY SPEAKT HE SAME PROTOCOLS? |
| 14:07.52 | _Corey_ | If you're talking about some digital Avaya handset, call Citel. They make some adapters... |
| 14:08.10 | qakhan | thats i my question. |
| 14:08.43 | [TK]D-Fender | qakhan, Go look at the documentation for the exact model you have |
| 14:09.03 | qakhan | does * support H.323 |
| 14:09.04 | qakhan | ? |
| 14:09.08 | [TK]D-Fender | badly. |
| 14:09.13 | Kobaz | heh |
| 14:10.35 | italorossi | Is there any way to map columns of queue_log to database columns using odbc? (callid -> unique_id) |
| 14:13.38 | mandrei | [TK]D-Fender i can see all the history of asterisk operations in /var/log/asterisk/messages but i can`t see in real time from CLI nothing. Tell me if you knowhow can i set my asterisk to see all the actions (except debug) in CLI |
| 14:14.45 | [TK]D-Fender | core set verbose 10 |
| 14:15.36 | mandrei | i used this command |
| 14:15.39 | mandrei | now i have |
| 14:15.44 | mandrei | Connected to Asterisk 1.8.15.0 currently running on localhost (pid = 6357) |
| 14:15.44 | mandrei | Verbosity is at least 999 |
| 14:15.50 | mandrei | and also nothing |
| 14:16.03 | _Corey_ | mandrei: Check your logger.conf and make sure you have console => verbose |
| 14:16.12 | [TK]D-Fender | Something else is wrong then |
| 14:16.42 | mandrei | i have there console => notice,warning,error,verbose,dtmf |
| 14:17.49 | _Corey_ | do "logger show channels" to make sure it's being read |
| 14:18.05 | _Corey_ | (you should see a Console enabled) line showing all that |
| 14:18.29 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 14:19.49 | mandrei | var/log/asterisk/messages File Enabled - NOTICE WARNING ERROR VERBOSE DTMF |
| 14:19.49 | mandrei | <PROTECTED> |
| 14:20.51 | _Corey_ | sounds like it should be working... |
| 14:21.13 | mandrei | this shoul be but it doesn`t |
| 14:21.40 | mandrei | anyway this is not so important for me at this time |
| 14:22.42 | *** join/#asterisk StaRetji (~LittleAll@178.79.11.166) |
| 14:23.03 | mandrei | _Corey_ do you used tls support in asterisk? |
| 14:23.25 | _Corey_ | not much, sorry |
| 14:23.25 | mandrei | *tls encryption |
| 14:23.36 | *** join/#asterisk miztic (~miztic@75-149-203-105-Illinois.hfc.comcastbusiness.net) |
| 14:23.40 | mandrei | it`s ok |
| 14:23.44 | StaRetji | Hi folks, I know this channel is for asterisk only, but I thought some of you are using a2billing for billing. I just need to know if I want to sell wholesale that I am doing it right if I setup number_try=1 |
| 14:24.16 | StaRetji | I look asterisk cli and it seems I receive call 1 at the time, so I'm affraid if I'm maybe limiting to 1 call my wholesale client |
| 14:24.21 | StaRetji | thx for any help |
| 14:25.02 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 14:25.04 | mandrei | _Corey_ qualify check over udp the clients? |
| 14:25.50 | mandrei | i mean asterisk if have qualify=yes |
| 14:51.44 | *** join/#asterisk amessina (~amessina@2001:470:c1dc:7779:d6be:d9ff:fe8d:7c1e) |
| 14:57.28 | *** join/#asterisk timahvo1 (~rogue@196.200.32.36) |
| 14:57.48 | *** join/#asterisk anonymouz666 (~anonymouz@189-25-92-167.user.veloxzone.com.br) |
| 15:00.50 | mandrei | guys why this could be? in asterisk logs i see it conects a peer with one port [Sep 5 05:48:55] VERBOSE[8602] chan_sip.c: -- Registered SIP '1222' at 10.187.2.1:56180 and after 1 minute he want to monitor this peer on other port [Sep 5 05:49:45] ERROR[8586] tcptls.c: Unable to connect SIP socket to 10.187.2.1:56168: Connection timed out |
| 15:01.26 | *** join/#asterisk fergus (~fergus@178.124.149.113) |
| 15:04.23 | *** join/#asterisk akshayms (~akshayms@ec2-23-21-125-169.compute-1.amazonaws.com) |
| 15:04.49 | *** part/#asterisk akshayms (~akshayms@ec2-23-21-125-169.compute-1.amazonaws.com) |
| 15:04.55 | *** join/#asterisk akshayms (~akshayms@ec2-23-21-125-169.compute-1.amazonaws.com) |
| 15:05.13 | mandrei | ani ideeas? |
| 15:05.20 | mandrei | *any |
| 15:06.31 | akshayms | Hey guys.. Just installed asterisk. Its not listening on any port. Changed bindaddr to 0.0.0.0:5060 in sip.conf and its not working.. |
| 15:10.49 | leifmadsen | in 1.8, bindaddr and bindport are separate, but your format should also work. |
| 15:10.55 | leifmadsen | akshayms: CentOS/RHEL server? |
| 15:11.19 | akshayms | leifmadsen, currently on an ubuntu AMI |
| 15:11.33 | leifmadsen | akshayms: oh then not sure -- only seen that happen when SELinux is enabled |
| 15:11.34 | [TK]D-Fender | akshayms, "sip show settings", from * CLI, and then show us how you checked if * was listening. |
| 15:11.35 | [TK]D-Fender | ~pb |
| 15:11.36 | 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. |
| 15:11.37 | leifmadsen | otherwise should work fine |
| 15:11.37 | [TK]D-Fender | ^^^ |
| 15:11.52 | leifmadsen | works fine on my AMI's on AWS |
| 15:13.36 | mandrei | leifmadsen why this could be? in asterisk logs i see a peer is connecting with one port [Sep 5 05:48:55] VERBOSE[8602] chan_sip.c: -- Registered SIP '1222' at 10.187.2.1:56180 and after 1 minute he want to monitor this peer on other port [Sep 5 05:49:45] ERROR[8586] tcptls.c: Unable to connect SIP socket to 10.187.2.1:56168: Connection timed out |
| 15:14.16 | akshayms | leifmadsen, [TK]D-Fender , http://pastebin.com/zAFJ8cxh |
| 15:15.15 | leifmadsen | mandrei: no idea |
| 15:15.25 | [TK]D-Fender | akshayms, Sure looks like * is bound to all.. and now your backup for how you were verifying that it is not doing what it is saying it is...? |
| 15:15.43 | leifmadsen | akshayms: what makes you think it isn't binding to a port? |
| 15:16.03 | akshayms | unable to connect to it via twinkle |
| 15:16.15 | akshayms | and lsof doesn't show the port either |
| 15:16.16 | leifmadsen | is your security group open to port 5060? doesn't look like you've configured asterisk to bind to an interface (just all), and haven't configured externip or localnet |
| 15:16.47 | leifmadsen | on AWS Asterisk is essentially behind NAT, so you need to configure it as such |
| 15:16.50 | mandrei | [TK]D-Fender you have idee why asterisk make this? |
| 15:17.04 | mandrei | making* |
| 15:17.19 | mandrei | it isn`t because of firewall |
| 15:17.47 | mandrei | it conects with one port but check for if peer is avaible on other port |
| 15:17.51 | akshayms | leifmadsen, I opened up the port and I'm using an elastic IP to connect |
| 15:18.02 | leifmadsen | it is still behind NAT |
| 15:18.17 | leifmadsen | your local IP on the machine is likely a 10.x.x.x address |
| 15:18.21 | akshayms | yes |
| 15:18.23 | leifmadsen | it won't be the elastic IP |
| 15:18.28 | [TK]D-Fender | akshayms, "netstat -an|grep 5060" |
| 15:18.30 | [TK]D-Fender | ^ |
| 15:18.34 | leifmadsen | which means you've got to configure asterisk as if it is behind NAT |
| 15:18.34 | akshayms | Nope, my elastic IP is 21.X.X.X |
| 15:18.38 | leifmadsen | exactly |
| 15:18.43 | [TK]D-Fender | If it's there then your other networking is at fault |
| 15:18.50 | leifmadsen | bindaddr=10.x.x.x |
| 15:18.54 | leifmadsen | bindport=5060 |
| 15:18.59 | akshayms | Anyway, I have another machine at home where I'm replicating everything |
| 15:19.02 | leifmadsen | externip=<elastic ip> |
| 15:19.08 | akshayms | I'm facing the same issue |
| 15:19.09 | leifmadsen | localnet=<internal ip>/32 |
| 15:19.24 | leifmadsen | beyond that, could be OS related |
| 15:19.26 | _Corey_ | I'd leave the bindaddr as 0.0.0.0... it's going to change if you power up/down the AMI |
| 15:19.35 | leifmadsen | _Corey_: oh true, forgot about that |
| 15:19.53 | _Corey_ | They work reliably otherwise... I've got a bunch in production ;) |
| 15:19.59 | leifmadsen | _Corey_: same |
| 15:20.04 | leifmadsen | configured via Chef:) |
| 15:20.06 | [TK]D-Fender | akshayms, Do you see it listed or not? |
| 15:20.15 | akshayms | [TK]D-Fender, listed |
| 15:20.26 | [TK]D-Fender | akshayms, then other network settings aare wrong. |
| 15:20.28 | mandrei | thanks for help, got to go |
| 15:20.45 | [TK]D-Fender | akshayms, "sip set debug on" go see if the packets from twinkle are even arriving |
| 15:20.47 | akshayms | leifmadsen, Any idea what could be wrong on my local setup? |
| 15:20.51 | _Corey_ | leifmadsen: I've been doing more with Puppet lately... have you tried both? |
| 15:20.59 | akshayms | [TK]D-Fender, ill give that a try, hold on |
| 15:21.05 | leifmadsen | akshayms: if I knew I'd have already suggested it |
| 15:21.06 | leifmadsen | _Corey_: I have yes |
| 15:21.11 | leifmadsen | _Corey_: started with Puppet actually |
| 15:21.27 | leifmadsen | _Corey_: really, same thing other than one uses ruby and the other uses it's own DSL |
| 15:21.43 | *** part/#asterisk hariom (~V@117.195.170.54) |
| 15:22.03 | leifmadsen | knife integration with chef is really well done -- harder to do with puppet I found (need to use marionette) |
| 15:22.14 | _Corey_ | I'm planning to rework some of our automation this weekend... we're implementing a cluster of Eucalyptus machines in our colo |
| 15:22.29 | leifmadsen | basically went with Chef because my current employer was already using it, and I didn't have enough invested in Puppet to argue for a change :) |
| 15:22.41 | _Corey_ | I'm going to have to poke around with Chef a little more |
| 15:22.41 | leifmadsen | _Corey_: you'll likely be interested in the Kickstand Project |
| 15:22.50 | leifmadsen | _Corey_: http://github.com/kickstandproject |
| 15:22.56 | _Corey_ | yeah, I've got it bookmarked :) |
| 15:23.03 | leifmadsen | nice |
| 15:23.14 | *** join/#asterisk gusto (~gusto@2001:a60:11ff:1200::42:4) |
| 15:23.33 | gusto | hi folks |
| 15:24.28 | _Corey_ | leifmadsen: I hate hardware with a passion... this stuff sort-of makes it fun again :) |
| 15:24.36 | leifmadsen | indeed :) |
| 15:24.52 | leifmadsen | this weekend having a geek out weekend with a friend and we're learning openstack with about 10 physically servers |
| 15:24.58 | leifmadsen | half rack of equipment to play on |
| 15:25.23 | _Corey_ | lol, I have 16 dell 1850s sitting next to my desk on a pile |
| 15:25.31 | _Corey_ | hello lab |
| 15:25.50 | leifmadsen | neat :) |
| 15:26.05 | _Corey_ | I decided to go Eucalyptus vs. OpenStack because of the AWS API compatibility |
| 15:26.14 | leifmadsen | ah gotcha |
| 15:26.22 | _Corey_ | it's all basically the same |
| 15:26.41 | leifmadsen | I have no allegiance and hadn't really thought about eucalyptus :) |
| 15:27.19 | _Corey_ | I forget whether their CEO spoke at Astricon or ITEXPO or some other show I was at, but he made a good impression |
| 15:29.19 | leifmadsen | gotcha |
| 15:29.40 | leifmadsen | not sure it was AstriCon, but someone in the distributed computing arena likely isn't a bad guest speaker |
| 15:29.45 | akshayms | [TK]D-Fender, debug didn't show anything |
| 15:30.04 | akshayms | and twinkle says "fetching registrations failed: 404 not found" |
| 15:30.08 | _Corey_ | leifmadsen: he was the CEO of MySQL and sold it to Oracle... |
| 15:30.41 | leifmadsen | _Corey_: that helps the bank acocunt |
| 15:30.44 | leifmadsen | account* |
| 15:31.02 | soulslayer | hey guys.. how can i execute script when call is not answered.. ? |
| 15:31.22 | soulslayer | i talk about any event accept answer ... i tried to execute it right after the hangup application |
| 15:31.29 | soulslayer | and right before it ... but without any luck |
| 15:32.22 | gusto | what is openstack? |
| 15:33.00 | soulslayer | OpenStack is an Infrastructure as a Service (IaaS) cloud computing project started by Rackspace Cloud and NASA |
| 15:33.01 | soulslayer | :D |
| 15:33.09 | pigpen | Hi all. I am having issues with the DPMA (asterisk 10.7), and having a phone register with a realtime sip account. For that matter, I have the same issue with an "external_line" (ie: remote secondary asterisk sip account) |
| 15:33.51 | pigpen | soulslayer, use the "h" extension |
| 15:34.06 | gusto | well |
| 15:34.10 | soulslayer | the h extension ? |
| 15:34.15 | soulslayer | must google |
| 15:34.31 | pigpen | yeah, it is a "standard" extension. invoked when a call is hung up. |
| 15:34.46 | pigpen | or you create a fall through when the timer expires on the dial, to "go do this" |
| 15:34.55 | Qwell | _Corey_: Martin Mikos |
| 15:34.58 | gusto | i have this http://en.wikipedia.org/wiki/OpenStack page open, but i still do not understand what is it good for |
| 15:34.59 | Qwell | Marten* |
| 15:35.00 | pigpen | no DPMA genisus in irc today? |
| 15:35.25 | soulslayer | pigpen, but .. this will be executed when the call ends right ? |
| 15:35.26 | soulslayer | or no ? |
| 15:35.29 | Qwell | pigpen: You've given zero information about your problem. |
| 15:35.42 | _Corey_ | Qwell: Yeah, I remembered his name just not the spelling... ;) I've been on the phone and too lazy to look it up. |
| 15:36.08 | gusto | and who should be that guy? |
| 15:36.39 | pigpen | Qwell, heh. Yeah, you are right. Sorry. Seems the config is being pushed out to the phone with the incorrect secret. |
| 15:36.57 | pigpen | what? IRC doesn't read my mind? |
| 15:37.07 | gusto | CEO of MySQL? |
| 15:37.28 | [TK]D-Fender | akshayms, 404 says that * did respond with "nothing to see here" for it's request |
| 15:37.45 | [TK]D-Fender | akshayms, make sure sip debug is enabeld and restart twinkle |
| 15:38.00 | gusto | well, nothing to see here is an often response i get |
| 15:38.04 | akshayms | [TK]D-Fender, SIP debug is enabled. restarting twinkle now |
| 15:38.28 | pigpen | So a bit more info: DPMA works fine with a sip account that exists in the sip.conf file. However, if it exists in realtime (or "external_line" as digium calls it) it does not authenticate, due to an incorrect secret. |
| 15:38.51 | Qwell | pigpen: realtime and external lines have absolutely nothing to do with each other. |
| 15:38.53 | soulslayer | ok the h extension do the trick |
| 15:39.04 | soulslayer | but .... it is executed when a call is ended after it's answered |
| 15:39.09 | pigpen | For realtime, 3 additional items are added to the line: secret, mailbox and context, all are set. |
| 15:39.12 | soulslayer | i want to trigger it only when call i sno accepted ? |
| 15:39.17 | gusto | well, that twinkle does not seem to be as good as linphone |
| 15:39.39 | pigpen | Qwell, as I noted, they are having the same issue. Not that they are the same. |
| 15:39.45 | *** join/#asterisk amessina (~amessina@2001:470:c1dc:7779:d6be:d9ff:fe8d:7c1e) |
| 15:39.57 | pigpen | ie: both saying fail due to incorrect secret. |
| 15:40.09 | akshayms | [TK]D-Fender, still the smae |
| 15:40.12 | pigpen | as they both are set with similar config settings. |
| 15:40.24 | akshayms | I'm trying this only on my local setup now btw |
| 15:40.27 | [TK]D-Fender | akshayms, sounds like your networking is bad between them |
| 15:40.39 | [TK]D-Fender | akshayms, no debug = no packets arriving |
| 15:40.39 | akshayms | I'm able to ssh into the other machine |
| 15:40.40 | Qwell | pigpen: They are completely different. Trying to relate them is useless. Pick one to debug. |
| 15:40.45 | [TK]D-Fender | akshayms, irrelevant. |
| 15:40.48 | pigpen | Realtime. |
| 15:40.51 | [TK]D-Fender | akshayms, SIP packets aren't making it over |
| 15:41.05 | Qwell | Great. Show me the SIP debug with realtime. |
| 15:41.44 | gusto | is "realtime" some extension, or just real time? |
| 15:41.48 | *** join/#asterisk Nemus (~Nemus@c-76-27-99-15.hsd1.ut.comcast.net) |
| 15:42.01 | pigpen | realtime db. |
| 15:42.07 | pigpen | sip accounts live in a db. |
| 15:42.10 | gusto | DBMS? |
| 15:42.23 | pigpen | in this case, postgres |
| 15:42.37 | gusto | well, i have my accounts in sip.conf, i do not have many |
| 15:42.47 | pigpen | yeah, that works fine. |
| 15:42.55 | gusto | but postgres is something i am trying out now as well |
| 15:43.04 | gusto | but i still do not understand the roles system |
| 15:43.18 | gusto | however, i am going to figure that out soon |
| 15:43.37 | gusto | how is it connected to asterisk? with that standard ODBC module? |
| 15:45.05 | soulslayer | hmm so the h extension is the only thing that is execute after call reject ... and it works like a charm but is there a way to filter the answered and then ended calls |
| 15:45.22 | soulslayer | i want to tell that i need only the calls that are not answered |
| 15:45.55 | gusto | hm, interesting, i am lucky when calls are not answered ;-) |
| 15:46.15 | soulslayer | :D |
| 15:46.26 | soulslayer | yep i need to execute something if they are not :D |
| 15:46.36 | soulslayer | and h executes at any call |
| 15:46.38 | Qwell | pigpen: pastebin any SIP and DPMA configs that are relevant for that account/phone |
| 15:46.56 | pigpen | k |
| 15:47.28 | *** part/#asterisk StaRetji (~LittleAll@178.79.11.166) |
| 15:47.49 | *** join/#asterisk brdude (~brdude@c-24-7-76-160.hsd1.ca.comcast.net) |
| 15:48.37 | soulslayer | or the only way is to handle that ... outside asterisk ? |
| 15:49.26 | _Corey_ | soulslayer: look at DIALSTATUS |
| 15:49.48 | soulslayer | yep that's the other thing that i think will work |
| 15:49.58 | soulslayer | but want to keep it simple as possible |
| 15:50.22 | soulslayer | because idk in what enviroment i will deploy there can be some .. giant dial plans that i want to change ... little as possible |
| 15:50.42 | soulslayer | i think outside of asterisk will be better then goto .... if dialstatus |
| 15:51.16 | soulslayer | okay thx again :D |
| 15:52.02 | pigpen | Qwell, http://pastebin.com/dnXd9yUH |
| 15:53.49 | Qwell | pigpen: and what is the secret set to in realtime? |
| 15:53.55 | Qwell | for peer 200 |
| 15:55.30 | pigpen | the same. |
| 15:55.49 | pigpen | letmein, but I just reset both just in case. |
| 15:56.17 | pigpen | getting: [Sep 5 10:55:41] NOTICE[7554]: chan_sip.c:26128 handle_request_register: Registration from '"200" <sip:200@192.168.100.1>' failed for '192.168.100.13:5060' - Wrong password |
| 15:56.28 | drmessano | ha |
| 15:56.31 | drmessano | lmi for the win |
| 15:56.47 | pigpen | but this time I set to a magical: letmein! << super secure password |
| 15:57.04 | drmessano | Wait... I thought super secret was 1235 |
| 15:57.10 | drmessano | No, thats for PINs |
| 16:00.15 | Qwell | pigpen: and you know that Asterisk has the correct secret from realtime? It's not caching an old one or anything? |
| 16:00.44 | pigpen | good point. I'll stop all to have it re-read just in case. |
| 16:01.21 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v012-133.mobile.uci.edu) |
| 16:02.25 | pigpen | same thing, wrong password. |
| 16:02.54 | Qwell | And you've reconfigured the phone so that it has the correct secret? |
| 16:03.13 | pigpen | verifying that now, again, just in case. |
| 16:03.42 | pigpen | secret=lemtein! (copy/paste) |
| 16:05.09 | pigpen | from psql: 1 | 200 | 200-Office | | | Office | yes | from-sip | | rfc2833 | | | dynamic | | | 200@from-sip | | no | | | | | | yes | | | | letmein! | friend | 200 | all | ulaw | | 0 | | yes | 100 | 100 | | | |
| 16:05.09 | pigpen | <PROTECTED> |
| 16:05.19 | pigpen | dam that looks like shit. but you can see the entry. |
| 16:06.30 | leifmadsen | pigpen: that's what pastebin is for... |
| 16:07.08 | pigpen | leifmadsen, yeah, it is just one line, but a bit long for sure, and surly looks like shit. |
| 16:07.19 | leifmadsen | :) |
| 16:07.20 | leifmadsen | indeed |
| 16:07.31 | leifmadsen | I only have my screen about 300x400 px |
| 16:07.41 | pigpen | hah, yeah, no kidding . |
| 16:08.42 | pigpen | Qwell, to me it seems that the dpma is not packaging the entry "secret=password" correctly. Although not "related" using the secret= entry on a secondary line to a remote asterisk box has the same issue with a wrong password. |
| 16:09.05 | pigpen | Which if it is a "handling" issue, could affect both. |
| 16:09.43 | Qwell | pigpen: It would help if your secrets matched. |
| 16:10.10 | Qwell | "letmein!" is not the same as "letmein" |
| 16:10.23 | pigpen | well, like I said, I went back and re-entered to make sure they were. |
| 16:10.36 | pigpen | as I found a typo. |
| 16:11.20 | pigpen | never the less, the phone that I was originally testing with was correct, but with this issue. Sure, not good, but the issue remains. |
| 16:11.56 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 16:12.08 | Qwell | and you're able to register another phone with that 200 peer? |
| 16:12.51 | pigpen | yes. I have had this same D70 registered with XML Provisioning since April, then before that, a Polycom 650 for 3 years, before that a 601 for 4 years. |
| 16:13.23 | pigpen | Qwell, yeah. It's goofy. |
| 16:13.45 | pigpen | I would love an easy one, like the password. But I'm not that lucky. |
| 16:14.25 | Qwell | turn off caching, make sure the passwords actually match, restart Asterisk. |
| 16:14.58 | Katty | waves at Qwell |
| 16:15.05 | Qwell | Katty: ohai |
| 16:15.37 | Katty | how be? |
| 16:15.54 | Qwell | I am well, and you know why. |
| 16:16.00 | Katty | :>>>> |
| 16:16.06 | Katty | fist bumps Qwell |
| 16:16.12 | Katty | yay for having an awesome day |
| 16:16.28 | jaytee | ~softphone |
| 16:16.28 | infobot | [~softphone] A soft-phone is a program that lets your computer use a compatible audio interface (usually a speaker&mic, or headset), and optionally a webcam, to place VoIP calls with a given protocol (SIP/IAX/H.323/MGCP/etc). For links enter ~zoiper , ~xlite , ~twinkle , ~bria , ~eyebeam , ~ekiga |
| 16:18.00 | Qwell | pigpen: and if you're working with support, TELL SOMEBODY HERE FIRST |
| 16:18.38 | Qwell | I don't like seeing multiple people dealing with the same problem, asking questions in internal channels, while I'm helping that person as well. |
| 16:19.34 | *** join/#asterisk ctaloi (~ctaloi@50-56-202-179.static.cloud-ips.com) |
| 16:20.11 | adeel|work | i'm having trouble getting directrtpsetup to work for inbound calls....it seems that asterisk still provides its IP on the initial invite, and then re-invites out of the path....it works fine for outbound calls.... |
| 16:20.38 | pigpen | Qwell, sorry. I thought I did. |
| 16:20.44 | adeel|work | if directrtpsetup is defined in the sip.conf global section, and not redefined for peers/extensions, does it still default to no? |
| 16:21.06 | pigpen | Qwell, and I didn't know you were with them. Otherwise, I would have made sure you knew. |
| 16:21.33 | Katty | who wants to help me throw marshmallows at cisco. |
| 16:21.43 | Katty | loads marshmallow shooter |
| 16:23.18 | pigpen | Qwell, caching off, passwords verifed, asterisk stopped/started, same issue. |
| 16:23.36 | pigpen | I can deal with support via the open ticket if you wish. |
| 16:34.40 | *** join/#asterisk Galen (~Galen@rrcs-24-43-20-118.west.biz.rr.com) |
| 16:41.32 | bmoraca_work | Qwell: feature request for DPMA: a setting to hide the dialplan executions from the console/logs...man, it's really ballooning my logs...just two phones running for a day is 20mb of logs |
| 16:42.39 | *** join/#asterisk deo (~dnepangue@112.198.90.143) |
| 16:43.49 | *** join/#asterisk fergus_ (~fergus@178.124.149.113) |
| 16:45.13 | *** part/#asterisk kresp0 (~kresp0@213.37.150.59.dyn.user.ono.com) |
| 16:45.21 | *** join/#asterisk Defraz (~Defraz@mail.pocatellochildren.com) |
| 16:46.35 | *** join/#asterisk Adyn (~adyn@pdpc/supporter/active/adyn) |
| 16:46.48 | leifmadsen | bmoraca_work: you should file that to Digium directly through the website as Qwell isn't responsible for the DPMA stuff at all |
| 16:47.04 | *** join/#asterisk wtfitsme (~WTFitsME@asams.mserve.com) |
| 16:47.50 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 16:49.10 | wtfitsme | hello I am using the REGEX function and I am trying to determine if an extension starts with 718,212,347. What is the bet way to form the REGEX, I thought that something like this would work: exten => s,n,GotoIf($[${REGEX("[^1212|^1718|^1646|^1347|^1917|^1929|^1800|^1888|^1866|^1877|^1855]" ${DIALED_NUMBER:4})}]?no_account_codes:account_codes)….but my regex fu has left me. |
| 16:49.42 | wtfitsme | This is more of a regular expression question i guess. |
| 16:49.54 | leifmadsen | I'd test it outside of asterisk first |
| 16:50.11 | leifmadsen | because the REGEX stuff in Asterisk can be kind of picky |
| 16:50.15 | wtfitsme | leifmadsen: asterisk uses PCRE? |
| 16:50.22 | leifmadsen | I don't think so |
| 16:50.27 | wtfitsme | hmm. |
| 16:50.29 | leifmadsen | it's some weird common denominator of regex |
| 16:51.17 | *** join/#asterisk amessina (~amessina@2001:470:c1dc:7779:d6be:d9ff:fe8d:7c1e) |
| 16:52.56 | bmoraca_work | leifmadsen: right...he was just helping me with it recently |
| 16:53.28 | leifmadsen | bmoraca_work: right, I'm just saying, if you want that suggestion to be passed to someone who can do something with it, you need to use the right channels |
| 16:53.38 | leifmadsen | this is not the correct place for a feature suggestion |
| 16:54.31 | *** join/#asterisk navaismo (~navaismo@189.144.249.74) |
| 17:00.53 | *** join/#asterisk barbosa2 (~juliano.b@177.43.91.206) |
| 17:07.45 | *** join/#asterisk learath (47f6db23@gateway/web/freenode/ip.71.246.219.35) |
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| 17:19.13 | *** join/#asterisk imox (~imox@91-64-185-199-dynip.superkabel.de) |
| 17:20.13 | *** join/#asterisk TuxBlackEdo (~TuxBlackE@unaffiliated/tuxblackedo) |
| 17:21.07 | TuxBlackEdo | Hello, I am a complete newb. I am trying to set up some phones for my sales people in my office (about 10 phones) and I would like to use VoIP.. So what kind of provider should I use that is reliable? |
| 17:22.20 | paulc | ~itsp |
| 17:22.20 | infobot | [~itsp] An ITSP is an Internet Telephony Service Provider (or VoIP telephone company). They allow you to either SEND calls to the PSTN (this is called termination), RECEIVE calls from the PSTN (called origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs. |
| 17:22.50 | TuxBlackEdo | ~itsplist-us |
| 17:22.50 | infobot | Here are some popular ITSPs (USA) starting with the more respected ones: http://www.teliax.com , http://www.voicepulse.com/connect/ , http://www.broadvoice.com , http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net , http://voip.ms and http://flowroute.com |
| 17:23.06 | pigpen | TuxBlackEdo, flat lines are typically very reliable. Just run them via an FXO of some sort. |
| 17:23.10 | paulc | TuxBlackEdo: Depends if you want an ITSP to send you traffic, and you have an Asterisk box that those 10 people connect to.. or if you're looking for some kind of "hosted PBX" offering |
| 17:25.18 | *** join/#asterisk darkdrgn2k (~darkdrgn@209.90.253.66) |
| 17:25.20 | darkdrgn2k | hey guys |
| 17:25.46 | darkdrgn2k | i just lost access to my spa2102... i set it to bridge mode.. an di know it has an ip ( can isee it in the dhcp table) but i cant access the web insterace on it.. |
| 17:26.10 | akshayms | [TK]D-Fender, this just showed up in the debug - [Sep 5 21:28:25] WARNING[2830]: chan_unistim.c:3541 parsing: 192.168.1.6 Read error |
| 17:27.10 | [TK]D-Fender | akshayms, Please tell more about this parallel dimension where Twinkle speaks UniSTIM... |
| 17:28.41 | *** join/#asterisk deo (~dnepangue@112.198.90.11) |
| 17:28.54 | *** part/#asterisk deo (~dnepangue@112.198.90.11) |
| 17:29.20 | akshayms | [TK]D-Fender, did not follow you there |
| 17:29.39 | [TK]D-Fender | akshayms, that has nothing to do with your SIP issues |
| 17:30.23 | akshayms | any other softphone that I should try? Probably some issue with Twinkle |
| 17:31.03 | TuxBlackEdo | what is a good hardware pbx? |
| 17:31.24 | TuxBlackEdo | for like <10 |
| 17:31.53 | darkdrgn2k | depends if your looking for asterisk or not :-P |
| 17:32.57 | TuxBlackEdo | oh.. i am just looking for something everyone uses so when something goes wrong i can easily google the problem and get like 100s of forums posts of people experiencing that same problem |
| 17:35.28 | *** join/#asterisk brdude (~brdude@12.155.183.30) |
| 17:36.33 | *** join/#asterisk ChkDigit (~u388mw@74.3.144.66) |
| 17:38.11 | *** join/#asterisk italorossi (~textual@189.124.200.92) |
| 17:42.04 | *** join/#asterisk bandroidx (~bandroidx@2607:f358:1:fed5:4:0:414:11) |
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| 17:43.33 | *** join/#asterisk vinhdizzo (~vinh@vqn-portege.ics.uci.edu) |
| 17:45.12 | paulc | TuxBlackEdo: Get a Nortel Meridian Norstar off eBay - cheap and cheerful, key system working, ideal for <10 people.. or go the Asterisk route and either buy an appliance (Digium have some) or build a box yourself (or pay someone to) (hi! I do that!).. then buy a bunch of SIP phones.. |
| 17:45.21 | *** join/#asterisk brdude (~brdude@12.155.183.30) |
| 17:46.10 | [TK]D-Fender | akshayms, Packets aren't making it to your server. You haev screwed up your networking. It isn't twinkle's fault. |
| 17:48.36 | *** join/#asterisk corretico (~luis@190.211.93.11) |
| 17:50.30 | *** join/#asterisk timahvo1 (~rogue@196.200.32.36) |
| 17:52.42 | *** join/#asterisk d0pe_ (d0pe_@31-16-49-111-dynip.superkabel.de) |
| 17:55.16 | *** join/#asterisk Trinitel (5eaa6147@gateway/web/freenode/ip.94.170.97.71) |
| 17:56.51 | *** join/#asterisk timahvo1 (~rogue@196.200.32.36) |
| 18:01.35 | Trinitel | Hello guys, is somebody using iLBC codecs? Do you have problems with robotic voice if transcoded by asterisk to other codecs? |
| 18:04.13 | *** join/#asterisk j-fish (~hhkkhkj@unaffiliated/j-fish) |
| 18:07.04 | jpsharp | ilbc has a robotic tone to it to begin with. |
| 18:08.59 | TuxBlackEdo | thanks paulc |
| 18:10.06 | *** join/#asterisk bn-7bc (~bjarne-im@cl-1271.sto-01.se.sixxs.net) |
| 18:10.56 | Trinitel | jpsharp: I know but the quality of iLBC to iLBC call is much better than iLBC to G711 call. |
| 18:11.46 | leifmadsen | lpc10 is my fave |
| 18:13.24 | Trinitel | I'm using polycom phones which only support G711, G722, G729 and iLBC. iLBC is the only free low bitrate codec here... |
| 18:13.46 | Qwell | bmoraca_work: fwiw, phonefeatures@digium.com |
| 18:15.39 | *** join/#asterisk bipul (~vipul@unaffiliated/bipul/x-4918593) |
| 18:16.50 | Qwell | bmoraca_work: oh, looks like you found it. nm! |
| 18:19.38 | _Corey_ | Anyone know what the # of reactivations is on a DPMA key? |
| 18:20.11 | Qwell | _Corey_: I doubt it's limited |
| 18:20.17 | Qwell | actually, maybe. |
| 18:20.30 | _Corey_ | seems to be |
| 18:20.33 | Qwell | ponders whether he can check that |
| 18:21.20 | Qwell | 30, unless your key is special somehow O.o |
| 18:21.29 | _Corey_ | it's, um, old |
| 18:21.34 | Qwell | so, probably special :p |
| 18:21.40 | _Corey_ | they sent a new one, so all good |
| 18:23.53 | Qwell | has a super magic awesome key |
| 18:24.15 | _Corey_ | lol, mine was super too until it stopped working :) |
| 18:27.05 | *** join/#asterisk YoMomma (~YoMomma@71.95.164.90) |
| 18:27.53 | *** join/#asterisk kresp0 (~kresp0@213.37.150.59.dyn.user.ono.com) |
| 18:28.14 | *** join/#asterisk hariom (~V@117.195.170.54) |
| 18:31.19 | hariom | I am installing asterisk in non root user (I use ubuntu server 10.04 so it means non sudoer user). I have created a user asteriskpbx as a normal user which is not in sudoer list. How to install it for asteriskpbx user? I have default sudo user as 'myuser'. |
| 18:33.40 | hariom | Should I login into asterisbpbx user and untar the source do ./configure and make as the 'asteriskpbx' user and then change to sudo user i.e. 'myuser' as 'su - myuser' and do sudo make install? Later change the owner ship of directories like /var/lib/asterisk etc to asteriskpbx:asteriskpbx ? Is this the correct way? |
| 18:34.21 | hariom | I see the similar procedure here: http://ofps.oreilly.com/titles/9780596517342/asterisk-Install.html |
| 18:34.43 | hariom | but the confusion is that in the link author assumes that asteriskpbx is the sudoer user |
| 18:35.05 | Qwell | hariom: ./configure && make && sudo make install |
| 18:35.11 | hariom | I think it reduces the security of the system |
| 18:35.45 | Qwell | It doesn't matter what name you use. Pick one, and just use that name for everything. |
| 18:35.46 | hariom | Qwell: How to ensure that it works with the user who is not in the sudoer list |
| 18:35.57 | Qwell | By following the steps that you linked. :) |
| 18:36.12 | leifmadsen | :) |
| 18:36.44 | *** join/#asterisk timahvo1 (~rogue@196.200.32.36) |
| 18:37.36 | hariom | Qwell: I think it makes the difference. Lets say I follow the link as it is and install into 'asteriskpbx' user (which is also a sudoer user i.e. root user). If password of the 'asteriskpbx' user is cracked or some how entry is made into the shell of asteriskpbx user, the person can control the entire system. |
| 18:38.01 | leifmadsen | hariom: you can always remove sudo access after you've installed.... |
| 18:38.04 | Qwell | You don't need SSH enabled on that account. |
| 18:38.09 | leifmadsen | and that too |
| 18:38.18 | Qwell | or just rename it like I said. The name is completely unimportant. |
| 18:38.20 | leifmadsen | there are ways to solve your problems |
| 18:38.45 | hariom | Qwell: System is going to be remotely managed. So ssh will remain. I can't have benefit of that. |
| 18:38.52 | *** join/#asterisk Hive (~Hive@173-165-205-1-jacksonville.hfc.comcastbusiness.net) |
| 18:39.06 | Qwell | hariom: You can SSH to your system management user. |
| 18:39.09 | leifmadsen | then better control your access via ssh as well via keys and firewall |
| 18:39.21 | Qwell | The user in the guide is just like the apache, mysql, or postfix users. |
| 18:39.39 | Qwell | Hell, run it as nobody. It does not matter. |
| 18:39.41 | leifmadsen | create a separate group for ssh access, and give those people sudo access, and control the maintenance users via ssh keys, etc. |
| 18:39.50 | hariom | leifmadsen, Qwell: The user in the guide is the user who is installing Ubuntu |
| 18:40.04 | leifmadsen | so change the user who is installing |
| 18:40.11 | leifmadsen | you can adduser after for asteriskpbx |
| 18:40.28 | hariom | leifmadsen: yea, thats what I was mentioning perviously |
| 18:40.34 | leifmadsen | no one is stopping you from doing it |
| 18:40.40 | leifmadsen | I'm not sure what we're discussing here... |
| 18:40.46 | hariom | leifmadsen: Have a user which is not in sudoer |
| 18:41.00 | leifmadsen | ok, soooooo... don't have that user be a sudo user |
| 18:41.05 | leifmadsen | you have the power |
| 18:41.53 | hariom | leifmadsen: I guess you have ignored the actual query. Let me repeat for you to know what I am looking for: I am installing asterisk in non root user (I use ubuntu server 10.04 so it means non sudoer user). I have created a user asteriskpbx as a normal user which is not in sudoer list. How to install it for asteriskpbx user? I have default sudo user as 'myuser'. Should I login into asterisbpbx user and untar the source do ./configure and make as the 'asteriskpb |
| 18:42.28 | leifmadsen | ./configure and make don't require sudo access |
| 18:42.35 | leifmadsen | you can use another user to run 'make install' if you want |
| 18:42.58 | leifmadsen | same thing with chmod and chown |
| 18:42.59 | hariom | leifmadsen: I also didn't say ./configure and make require sudo |
| 18:43.09 | leifmadsen | hariom: I agree -- they don't require it |
| 18:43.21 | leifmadsen | you don't *have* to run make install as asteriskpbx |
| 18:43.22 | Trinitel | Hello, is somebody using iLBC codecs? Do you have problems with robotic voice if transcoded by asterisk to other codecs? iLBC to iLBC calls are ok but iLBC to G711 is very robotic, |
| 18:43.24 | leifmadsen | use another user |
| 18:43.37 | hariom | leifmadsen: ok |
| 18:43.40 | leifmadsen | Trinitel: known issue and open on the issue tracker |
| 18:44.57 | Trinitel | leifmadsen: so there is no workaround at the moment? I will check the issue tracker |
| 18:45.18 | Hive | when i run 'core show channels' in the CLI, is there a way to extend the channel name section so that some chanel names are not truncated? |
| 18:45.28 | leifmadsen | Trinitel: whatever information is known about the issue will be on the issue in the tracker, so check if there is a work around there (I'm not aware of one) |
| 18:45.43 | Trinitel | leifmadsen: thanks |
| 18:45.46 | leifmadsen | Hive: you could try 'core show channels verbose' or modify the field side in the code |
| 18:46.42 | Hive | thanks for the suggestions leifmadsen |
| 18:54.25 | Nephfl | so, how many channels are the rasberry pi boards able to support? |
| 18:55.01 | *** join/#asterisk Tu13es (~brandon@vervet.isomerica.net) |
| 18:57.07 | navaismo | Nephfl, so far i was used 5 sip channels without transcoding ulaw-ulaw and no recordings |
| 18:57.14 | navaismo | so dependes on your needs |
| 18:57.55 | navaismo | but im using realtime and mysql and php so maybe deleting that and use a complete vanilla asterisk you can get more |
| 18:57.59 | Nephfl | wondering if it would work as a freepbx server for a few phones |
| 18:59.11 | navaismo | talking in simultaneous calls how much? |
| 18:59.37 | Tu13es | hm, i have an astrisk server running freepbx....its set up to support video and we can make ptp calls using video. should it also "just work" for conference calls? |
| 18:59.38 | navaismo | and yes you can install freepbx on it, actually there are a distro with FPBX 2.10 ready to donwloand and copy to thje SDcard |
| 18:59.40 | Tu13es | (it doesn't) |
| 19:01.40 | navaismo | using confbridge with asterisk 10 maybe |
| 19:02.46 | Tu13es | hmmm |
| 19:02.53 | Tu13es | any pointers to upgrading a 1.8 to 10? |
| 19:02.58 | Tu13es | i've searched but couldn't find much |
| 19:04.10 | Tu13es | oh, i'm running asterisknow, if it matters |
| 19:04.54 | navaismo | Nope i dont have guides to do that i preffer everything from sources compile and install, if I want to upgrade uninstall and compile again |
| 19:05.21 | *** join/#asterisk timahvo1 (~rogue@196.200.32.36) |
| 19:06.35 | kresp0 | Hi all, |
| 19:06.35 | kresp0 | Would you say that this number: "29004100" will match this "_[012345]XXXXZ!0." ? |
| 19:10.32 | navaismo | hmm 8 digits against 9 dialpatern digit I bet for NO |
| 19:10.49 | navaismo | but maybe easier if I try it on asterisk |
| 19:11.48 | kresp0 | According to The Book (ejem leifmadsen), I understand it should. The "!" character here should match 0 or more characters. But no. |
| 19:11.51 | kresp0 | navaismo |
| 19:12.02 | kresp0 | I need to use that expression instead: _[012345]XXXXZ0. The problem is, that in the future, I will need also to mach numbers like "29004990666". |
| 19:12.15 | kresp0 | s/that/this |
| 19:12.42 | leifmadsen | ! certainly works |
| 19:12.53 | leifmadsen | kresp0: but you can't put anything after the ! |
| 19:12.57 | leifmadsen | that's the problem you have there |
| 19:13.03 | kresp0 | ohhh thank you leif |
| 19:13.11 | leifmadsen | once you use ! or ., nothing else can be added after |
| 19:13.28 | kresp0 | maybe The Book should say it |
| 19:13.43 | *** join/#asterisk tstyle (~tstyle@96.53.116.78) |
| 19:14.48 | kresp0 | and it is possible to add several "!" ? |
| 19:15.20 | kresp0 | like _XXX!! |
| 19:15.42 | kresp0 | to match 3, 4 or 5 digits |
| 19:15.45 | Qwell | kresp0: no, ! or . must be at the end, and only used once. |
| 19:15.54 | kresp0 | thank you Qwell |
| 19:16.00 | Qwell | also !! makes no sense. |
| 19:16.11 | kresp0 | sure Qwell? |
| 19:16.15 | Qwell | !! would mean zero or zero or more characters |
| 19:16.24 | kresp0 | zero or one |
| 19:16.29 | Qwell | no, zero or more. |
| 19:16.38 | kresp0 | right Qwell |
| 19:16.51 | Qwell | If you want zero or one, use 2 extensions |
| 19:16.58 | Qwell | _123, and _123X |
| 19:17.05 | kresp0 | ok |
| 19:17.17 | kresp0 | I was using it wrong |
| 19:17.18 | Qwell | leifmadsen: I'm surprised there is no ? |
| 19:17.36 | Qwell | maybe it's not that useful though |
| 19:19.00 | leifmadsen | it'd be nice if you could use something like ? etc -- there was a discussion about extending the dialplan a couple years ago |
| 19:19.07 | leifmadsen | I was supposed to document it, but didn't get that far |
| 19:19.37 | Qwell | oh, well I'll blame you for its lack of existence then |
| 19:22.11 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 19:25.54 | kresp0 | leifmadsen: you document things that didnt even exist yet? |
| 19:26.06 | kresp0 | sorry about my poor english |
| 19:27.36 | leifmadsen | kresp0: yes sometimes |
| 19:31.35 | *** join/#asterisk ks3_ (ks3@2600:3c03::f03c:91ff:feae:97e3) |
| 19:32.48 | TheCompWiz | anyone play with a specralink 8440? |
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| 19:47.19 | Nephfl | what is a quick way to test if meetme was compiled in? |
| 19:47.54 | Nephfl | I guess it is in help, so it is there? |
| 19:48.16 | Nephfl | think it might have something to do with meetme? |
| 19:48.37 | Nephfl | since asterisk keeps parsing meetme.conf when monsast is running |
| 19:48.59 | Nephfl | wrong window |
| 19:49.02 | Nephfl | hmm |
| 19:49.03 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 19:49.04 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 19:51.16 | navaismo | i think if app_meetme.so exist |
| 19:54.02 | Nephfl | anybody here use the monast script much? |
| 19:54.56 | *** join/#asterisk ks3 (ks3@2600:3c03::f03c:91ff:feae:97e3) |
| 19:56.31 | navaismo | what exactly? |
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| 20:02.58 | Tu13es | navaismo: hm, do you have any experience with confbridge and video? |
| 20:03.24 | Tu13es | i upgraded to asterisk 10 and am using confbridge, and i can call into a conf and turn on video, and the asterisk server gets video from the clients, but never sends any video to the clients |
| 20:04.33 | navaismo | only in testing environments, do you configure the confbridge for the video |
| 20:04.49 | Tu13es | I set video_mode=follow_talker |
| 20:04.53 | Tu13es | is there something else i need to enable? |
| 20:05.31 | navaismo | and asterisk dont show multivideo. Like a grid array |
| 20:05.48 | Tu13es | nope, i don't get any video coming from asterisk |
| 20:05.57 | navaismo | don't remember but you can see in the wiki |
| 20:06.19 | Tu13es | yeah, this is what I was looking at: https://wiki.asterisk.org/wiki/display/AST/ConfBridge+10 |
| 20:07.54 | *** join/#asterisk timahvo1 (~rogue@196.200.32.36) |
| 20:11.35 | navaismo | yep, you will see the video source in this case the talker and you can switch the video source, you dont see it? |
| 20:14.06 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 20:14.48 | mjordan | Tu13es: are your sip peers configured with video support? |
| 20:15.04 | Tu13es | mjordan: yep, both peers are using jitsi and the same codec |
| 20:15.11 | Tu13es | and video works fine during a normal call |
| 20:15.21 | Tu13es | navaismo: nope, i don't see any video |
| 20:15.43 | Tu13es | I did notice that if I do "confbridge show profile bridge default_bridge" |
| 20:15.50 | Tu13es | it says "Video Mode: no video" |
| 20:16.41 | mjordan | If you don't set a video source in the profile, you have to set one programmatically either using DTMF or an AMI action |
| 20:17.01 | Tu13es | hmmm |
| 20:17.20 | mjordan | what is the video_mode setting in your bridge profile? |
| 20:18.48 | Tu13es | [default_bridge] |
| 20:18.48 | Tu13es | type = bridge |
| 20:18.49 | Tu13es | video_mode = follow_talker |
| 20:19.21 | Tu13es | oh, huh, whatever I did changed it so the command above now shows Video Mode: follow_talker |
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| 20:20.02 | Tu13es | mjordan: hm, ok, so I guess I may need to do what you said |
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| 20:22.30 | Tu13es | mjordan: hm, any pointers on that? |
| 20:23.05 | Tu13es | oh, heh, these are your mailing list posts i'm reading |
| 20:23.52 | Tu13es | oh, were you just talking about video_mode |
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| 20:47.39 | Tu13es | hmm, i tried enabling "Talk Detect Events" no no avail |
| 20:53.57 | Tu13es | also tried last_marked mode and changed default_user to marked=yes, to no avail |
| 20:56.07 | *** part/#asterisk kresp0 (~kresp0@213.37.150.59.dyn.user.ono.com) |
| 20:57.25 | Tu13es | goes insane |
| 20:58.12 | pabelanger | possible it is broken |
| 20:58.15 | pabelanger | try using 10.0.0 |
| 20:58.26 | pabelanger | last time I tested with jitsu, it worked with that |
| 20:58.44 | Tu13es | i'm using 10.7.1, i believe |
| 20:58.50 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 20:58.54 | Tu13es | maybe that's too new? :| |
| 20:59.01 | Tu13es | and yeah, i'm using jitsi |
| 20:59.05 | Tu13es | and video works fine p2p |
| 21:05.36 | Tu13es | bah, have to head out for the day. any suggestions feel free to PM me, otherwise expect me back tomorrow :P |
| 21:06.50 | TheCompWiz | YEAH FOR POLYCOM... /whimper. |
| 21:07.25 | TheCompWiz | Just incase anyone has a polycom spectralink 8440... and uses an open wifi for their voip links... you'll NEVER get DHCP to work until you update the firmware. |
| 21:08.10 | TheCompWiz | you'd think that'd be one thing they would have checked before leaving the factory. (wifi phone that can't get DHCP? ... SPIFFY!) |
| 21:09.54 | pabelanger | Now you know |
| 21:10.01 | pabelanger | and knowing is half the battle |
| 21:11.33 | jpsharp | red & blue lasers are the other half. |
| 21:11.54 | leifmadsen | sharks! |
| 21:13.18 | TheCompWiz | sharks & frickin' laser beams! |
| 21:13.43 | leifmadsen | it's all I really want |
| 21:13.53 | TheCompWiz | additionally... the default "Regulatory Domain" ... of "none" is invalid. You must set the option to 1... even though... it's the only option. |
| 21:14.10 | leifmadsen | hopes TheCompWiz is writing this down in a wiki somewhere |
| 21:14.20 | pabelanger | leifmadsen: IKR |
| 21:14.21 | TheCompWiz | (unless you're lucky enough not to live in US/Canada) |
| 21:14.42 | TheCompWiz | wishes Polycom would document some of this too... |
| 21:15.20 | TheCompWiz | I spent all day trying to badger this damn thing into working properly... and I even spent time on the phone w/ Polycom... and they still drew a blank. |
| 21:15.53 | TheCompWiz | they blamed it on my APs... because everyone knows... enterprise-grade APs are always to blame. |
| 21:16.03 | leifmadsen | sounds right |
| 21:16.10 | leifmadsen | if you can pass the buck, it's not your prolem |
| 21:16.37 | TheCompWiz | ... too bad their older phones (8030s and 8020s) worked perfectly. |
| 21:17.49 | TheCompWiz | grrr... and now I get to figure out why their NTP is b0rked. |
| 21:17.58 | TheCompWiz | or rather... their understanding of NTP. |
| 21:18.32 | TheCompWiz | (I set the offset to -18000 {-5gmt} and pointed it at my ntp server... and it says 10PM} |
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| 21:39.19 | Qwell | TheCompWiz: Enterprise-grade == Cisco + java, so yeah, I'm right there with them on that. |
| 21:41.14 | Qwell | Point: Qwell |
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| 22:20.34 | SeRi | p3nguin: you around? |
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| 23:15.24 | devilsoulblack | hi i need backup the last day of queue_log i think something like that SELECT * FROM `queue_log` WHERE time_id >= UNIX_TIMESTAMP("INTERVAL −1 DAY") |
| 23:15.26 | devilsoulblack | ani idea ? |
| 23:34.47 | *** part/#asterisk kresp0 (~kresp0@213.37.150.59.dyn.user.ono.com) |
| 23:36.09 | leifmadsen | devilsoulblack: 302 Redirect #mysql |
| 23:36.29 | devilsoulblack | thanks leifmadsen |
| 23:36.30 | leifmadsen | or rather, #sql |
| 23:41.09 | SeRi | has any body here played wih a flukso devise before? |
| 23:50.14 | *** join/#asterisk qakhan (~qakhan@pool-71-163-79-89.washdc.fios.verizon.net) |
| 23:50.29 | qakhan | can someone tell me is there any library or plugin in PHP which connects to astersik |
| 23:55.56 | navaismo | phpagi |