| 00:01.52 | *** join/#asterisk lucasb (~lucasb@S0106000c42710923.ok.shawcable.net) |
| 00:20.27 | pdtpatrick | anyone? |
| 00:20.52 | pdtpatrick | im seeing Festival and also VMXL app for asterisk .. |
| 00:31.56 | pabelanger | pdtpatrick: asterisk does not support VXML |
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| 01:06.20 | tzanger | 'evening coppice |
| 01:06.39 | coppice | 'morning tzanger |
| 01:07.18 | *** join/#asterisk johnnyasterisk (~johnnyast@89.18.71.45) |
| 01:08.27 | tzanger | what's new in the world of host based signal processing? Or rather what've you been up to lately |
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| 02:23.21 | dijib | anybody alive? |
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| 02:24.40 | dijib | am i alive? |
| 02:24.43 | dijib | ok i am. |
| 02:24.47 | dijib | anybody else alive? |
| 02:25.04 | p3nguin | ~ask |
| 02:25.04 | 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. |
| 02:25.19 | dijib | oh p3nguin god how are you tonight |
| 02:25.21 | dijib | ? |
| 02:25.35 | dijib | outside client, connections established but no audo. |
| 02:25.41 | dijib | ports forwarded. |
| 02:25.49 | dijib | my side |
| 02:26.06 | dijib | firewall on box was the problem. |
| 02:26.14 | dijib | for connections anyways. |
| 02:26.56 | dijib | not audio. ports 10000-20000 for audio ? |
| 02:26.59 | p3nguin | You probably don't need to have a firewall on the server that's on the LAN. |
| 02:27.15 | p3nguin | Check rtp.conf for the correct UDP port range to forward. |
| 02:27.25 | dijib | yeah i # /etc/init.d/firewall and rebooted |
| 02:27.35 | dijib | its default. |
| 02:33.41 | p3nguin | Do you have no audio at all, or one-way audio? |
| 02:33.52 | dijib | none |
| 02:34.04 | dijib | http://pastebin.com/m3J6bfes |
| 02:34.09 | dijib | here is sip.conf |
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| 02:34.23 | dijib | as present... and yes externalhost...... is intentional |
| 02:37.18 | p3nguin | That looks okay to me, so it must be the rtp stuff that needs attention. |
| 02:37.25 | p3nguin | Did you forward your port range? |
| 02:38.40 | dijib | yep. |
| 02:39.07 | dijib | actually one sec on that i thinki i have rad & ports bot forwarding 5060 |
| 02:41.08 | dijib | sip show peers doesnt see him. |
| 02:41.20 | p3nguin | Then he didn't register. |
| 02:42.07 | dijib | hmmm |
| 02:42.14 | dijib | then i dont know |
| 02:42.17 | dijib | im stumped |
| 02:42.26 | dijib | im going to dmz it. |
| 02:42.29 | dijib | see what it do. |
| 02:42.29 | p3nguin | no |
| 02:42.32 | dijib | y? |
| 02:42.46 | *** join/#asterisk X-Rob (~Rob@eth2083.qld.adsl.internode.on.net) |
| 02:42.59 | p3nguin | Just forward the friggin' ports. |
| 02:43.18 | *** join/#asterisk X-Rob (~Rob@eth2083.qld.adsl.internode.on.net) |
| 02:43.18 | dijib | bro they are, you want exteral admin to my router? |
| 02:43.21 | dijib | ssh you in? |
| 02:43.45 | p3nguin | I don't need it. Just forward the ports and make sure there's no firewall blocking stuff where it shouldn't be blocked. |
| 02:43.55 | p3nguin | Turn off any SIP ALG crap you may have. |
| 02:44.06 | p3nguin | On every piece of equipment that has it, that is. |
| 02:44.21 | dijib | whats alg? |
| 02:44.42 | p3nguin | What it is, or what does it do? |
| 02:45.26 | p3nguin | I'm sure google can tell you what it is, but what it does is breaks asterisk's ability to compensate for NAT. |
| 02:46.50 | dijib | ok then. im reading now. thanks for your tolorance of my ignorance |
| 02:46.59 | p3nguin | If your friend has a Belkin router, you'll probably never get a SIP phone working behind it. |
| 02:47.22 | p3nguin | Also Cisco SOHO stuff is a bother. |
| 02:48.05 | p3nguin | Those are the only two plastic routers I have experience with. |
| 02:48.10 | dijib | he doesnt. |
| 02:48.15 | p3nguin | What's he using? |
| 02:48.26 | dijib | finding out. but he has much better taste then that |
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| 02:48.51 | p3nguin | I can't remember if D-Link works with SIP phones or not. |
| 02:48.52 | dijib | ill check into the soho but i doubt. |
| 02:49.02 | dijib | dlink run 3rd party? |
| 02:52.09 | p3nguin | If you'll get him to register, we can start debugging to see if we can find out where the audio is stopping. |
| 02:56.30 | dijib | well maybe registering is the problem cuz the prts are good. |
| 02:57.41 | dijib | im running DD-WRT v24-sp2 (04/23/10) micro on my router, using QoS is that causing me a problem or do you know of a sip alg diable option in this? or should i keep looking here |
| 02:59.43 | p3nguin | You said earlier there was no audio. That tells me that you were able to make a call, but you just couldn't hear any voice over the line. What happened to that? |
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| 03:26.28 | dijib | wow that was hardcord |
| 03:27.50 | p3nguin | Is that anything like a lamp cord? |
| 03:28.07 | p3nguin | or a rip cord? |
| 03:28.37 | p3nguin | umbilical cord? |
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| 03:37.05 | dijib | lol |
| 03:37.11 | dijib | i definitly ment hardcore |
| 03:37.27 | dijib | ok, i had my router completely crap out and corrupt. |
| 03:37.46 | dijib | so i grabbed another, and restored my config |
| 03:38.00 | dijib | ow sip alg.. setting in ddwrt known? |
| 03:38.03 | dijib | cuz i cant find it |
| 03:38.14 | p3nguin | No clue. |
| 03:41.37 | dijib | reading. http://www.dd-wrt.com/wiki/index.php/Siproxd |
| 03:41.48 | dijib | indicated my version of ddwrt isnt friendly |
| 03:41.54 | p3nguin | You certainly don't need a SIP proxy. |
| 03:42.38 | dijib | so you dont think that applies? |
| 03:43.19 | p3nguin | I don't care to read it, but I'm telling you to disable ANYTHING that is related to SIP and/or RTP. |
| 03:44.01 | dijib | let me keep looking |
| 03:44.12 | dijib | what about my QOS? |
| 03:44.46 | p3nguin | It's probably okay, but I'd turn it off until you everything else working 100%. |
| 03:44.59 | p3nguin | It's not going to do any good anyway. |
| 03:45.23 | p3nguin | Your ISP isn't going to obey your QoS, so they'll put voice in with other bulk traffic. |
| 03:50.13 | dijib | why dont you think my sip will work? |
| 03:50.19 | dijib | i can make and take calls just fine |
| 03:50.58 | p3nguin | I never said your SIP won't work. |
| 03:52.11 | dijib | you said qos wont work, and why not? |
| 03:52.24 | dijib | and i cant find anything rtp or sip in ddwrt |
| 03:52.48 | dijib | but im probably blind |
| 03:52.56 | p3nguin | For QoS to work over a given link, both sides of the link must have the QoS policy. Your ISP couldn't care less about your personal QoS policies. |
| 03:54.08 | p3nguin | You can do traffic shaping all you want, but they will never do it after the traffic leaves your equipment and they will never do it on traffic they send to you. |
| 03:55.36 | p3nguin | Very few ISPs actually provide QoS to customer links. They'll normally consider all your traffic to be the same, so your http traffic will have the same weight as your voip and your ssh. |
| 03:55.48 | p3nguin | I mean residential services. |
| 03:56.07 | p3nguin | Commercial circuits often have QoS available. |
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| 04:20.13 | carlp | Greetings - I'm having problems using leave and join sounds with ConfBridge(). Basically, if two connections use different codecs then asterisk will crash when the callers hangup |
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| 06:15.58 | syntaxx | i have a question.. i have a asterisk running on a vm under bridge.. and my client running on a vm under nat.. i can hear him but he cant hear me using a bridge client.. any idea how to fix it? |
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| 06:41.52 | Maliuta | syntaxx: sounds like an rtp issue resulting from NAT |
| 06:41.56 | Maliuta | ~sipnat |
| 06:41.57 | infobot | [~sipnat] Quick guide on configuring Asterisk + SIP behind NAT: http://www.aocomputing.net/?p=3 otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions . Typically, you MUST configure these settings with appropriate values in the [general] section of sip.conf: nat, canreinvite, externhost or externip, and localnet. |
| 06:41.57 | ChannelZ | Barring any other strange routing happening due to the VMs, your packets either aren't making it out of your firewall or into his. |
| 06:46.08 | syntaxx | Maliuta, thats on the both sip account? or just the one that cannot hear? |
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| 06:48.54 | Maliuta | syntaxx: on the asterisk server ... but it may be his nat that's causing the issue (rtp and SIP are done via udp, so there is no connection forwarding unless it's specific) |
| 06:49.05 | Maliuta | syntaxx: IAX might work better for you in that situation |
| 06:50.20 | syntaxx | Maliuta, i am under ldap though.. |
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| 06:55.50 | ChannelZ | errhmm.. what? |
| 06:56.49 | *** join/#asterisk schmidts (~schmidts@Lo20-rt01.lm33.sil.at) |
| 06:56.52 | schmidts | good morning |
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| 07:06.15 | ChannelZ | I have to poop |
| 07:06.23 | ChannelZ | I mean.. good morning.. |
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| 07:24.17 | jacc0 | good morning all! |
| 07:24.29 | jacc0 | how can I show (in CLI) what timing is in use? |
| 07:24.54 | jacc0 | timinig test? |
| 07:25.02 | jacc0 | or will that always use dahdi? |
| 07:29.31 | schmidts | jacc0 if dahdi is available i guess its allways used |
| 07:30.09 | james_zhu | open the logger.conf |
| 07:30.22 | kaldemar | nope, it may use another timer apart from dahdi. |
| 07:30.23 | james_zhu | set the timedate |
| 07:32.57 | james_zhu | ;dateformat=%F %T ; ISO 8601 date format |
| 07:32.58 | james_zhu | ;dateformat=%F %T.%3q ; with milliseconds |
| 07:33.02 | james_zhu | in asterisk-1.8 |
| 07:33.12 | james_zhu | comment out |
| 07:37.27 | jacc0 | ? |
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| 07:39.18 | kaldemar | jacc0: timing test or module show like timing will tell you. |
| 07:40.17 | jacc0 | @kaldemar: you are right |
| 07:40.35 | jacc0 | dahdi shows use count 1 |
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| 07:41.24 | jacc0 | Someone told me timerfd is the default; not dahdi |
| 07:41.52 | jacc0 | and that is the main reason for asterisk default install being unstable imho |
| 07:42.27 | kaldemar | if you have both timerfd and dahdi modules, timerfd will be used. |
| 07:42.53 | jacc0 | I will need to run tests some more days to see if asterisk will keep running stable without taking up al the the systems resources |
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| 08:49.59 | aliasbex | can i configure user specific meta information in the users.conf and retrieve them again in the dialplan again? |
| 08:51.11 | aliasbex | custom variables like i.e. fallbackext=xx to conditinally dial it |
| 08:51.29 | Faustov | what could be wrong if I do channel originate sip/x/number dial... and tone selection is not working, while if he dials the server, dtmf is properly recognizing key presses |
| 08:51.50 | aliasbex | that would be the most comfortable way to extend extension specific meta info.... i think :) |
| 08:53.59 | kaldemar | aliasbex: yes, see function AST_CONFIG |
| 08:54.06 | aliasbex | i have sometimes dtmf problems with the g722 codec |
| 08:54.15 | aliasbex | k. thanks, kaldemar |
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| 08:57.38 | jacc0 | asterisk is still only taking op 122mb when using dahdi :) |
| 08:57.46 | aliasbex | kaldemar, cool. that was exactly what i wanted :D |
| 09:00.22 | ezano | How may I set a variable with the username into the dialplan |
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| 09:01.12 | ezano | setvar=MAVAR=username doesn't works |
| 09:01.47 | aliasbex | Set(${GLOBAL(VAR)}=VALUE)? |
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| 09:02.33 | ezano | okay, but my VALUE must be dynamic :p |
| 09:02.53 | jacc0 | or Set(db(username/valuename)=value) |
| 09:03.06 | aliasbex | i just wanted to mention the global scope... it still can be dynamic :) |
| 09:03.21 | jacc0 | the same goes for DB |
| 09:03.22 | ezano | oh thanks I'll try immediatly |
| 09:08.20 | kaldemar | ezano: with the username? |
| 09:08.30 | kaldemar | ezano: which username? |
| 09:09.05 | ezano | username define into database of a2billing and into the softphone |
| 09:09.08 | ezano | this is the same |
| 09:11.55 | ezano | I want call my voicemail with a softphone for example I hit 5555 to do that, then I have a prompt which ask me my voicemail number which is equal with the username |
| 09:12.12 | ezano | I want drop this step automatically into the dialplan |
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| 09:20.23 | irroot | linux 3.0.1 / dahdi 2.5.0 coming right up |
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| 09:27.40 | aliasbex | i wanted to build a dynamic extension macro for calling local extensions. but the problem is the validation if the extension really exists... how would i do that properly? |
| 09:28.26 | aliasbex | i could also solve it with the ast_config func, but that would always trigger an error which is dirty |
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| 09:37.10 | jacc0 | @irroot: what is new in dahdi 2.5? there is nothing in the changes file |
| 09:37.44 | singler | As I understand it will be up soon |
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| 09:40.25 | jacc0 | okay, found a changes file that hold some changes ; http://downloads.asterisk.org/pub/telephony/dahdi-linux/ChangeLog-2.5.0 |
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| 09:40.45 | irroot | yeah going through it |
| 09:40.59 | irroot | nothing stands out in the tools |
| 09:41.17 | tzafrir | jacc0, hardware EC handling is different. Also many internal changes (though that's not really user-visible) |
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| 09:41.50 | as001 | hello. I want to call sip account from asterisk server 1 which is on asterisk server 2 in style DIal(SIP/999@ip.ad.re.ss) How can I find out on what context on server2 that call will fall ? Can I control it by Dial command on server 1? Thank you |
| 09:42.36 | singler | as001: you cannot control or see remote context |
| 09:42.55 | as001 | how can I know on what context that call will fall on server 2? |
| 09:43.02 | singler | on IAX as I remember there was something about it, but not SIP |
| 09:43.19 | singler | check configuration on remote server (sip.conf) |
| 09:43.32 | singler | or connect to cli and observe |
| 09:43.33 | irroot | tzafrir thx ... see lots of xorcom fixups ... |
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| 09:45.00 | *** join/#asterisk TimeRider (~steve@host-92-27-131-175.static.as13285.net) |
| 09:45.28 | as001 | Yes I found out that on cli. So basicly I must register server 2 on server 1 and then tell on server 1 in what context server2 has been registered ? |
| 09:45.59 | as001 | Problem is i don't want calls from 2 to 1 in that context. I want in different context. |
| 09:46.39 | singler | contexts on server1 and server2 may differ |
| 09:47.11 | singler | when you define peer on server, you can set context=something |
| 09:47.24 | as001 | yes i meant that. |
| 09:47.47 | singler | that "something" can differ on servers |
| 09:48.16 | as001 | yes of course. Thanks i will try that. I thought there is easier way to force it somehow in Dial on server1. |
| 09:48.54 | *** join/#asterisk war9407 (war@c-71-62-61-74.hsd1.va.comcast.net) |
| 09:50.11 | *** join/#asterisk tuxxie (~Ryan@rrcs-70-63-90-226.midsouth.biz.rr.com) |
| 09:50.41 | singler | no, because nonasterisk SIP devices/servers would not understand it |
| 10:16.19 | jacc0 | The memory leak that has been buggin me seems to be solved; I'm now using dahdi as timing source and nolonger using timerfd |
| 10:16.50 | jacc0 | they did better remove "timerfd" from the official release; or at least not set it as default timing source |
| 10:17.18 | jacc0 | it is the cause of at least 2 different deadlocks and 1 memory leak |
| 10:17.57 | jacc0 | n ow updating bug report |
| 10:18.07 | irroot | jacc0 timerfd has a problem with locking threads more than deadlocks the thread blocks on a read when checking timer |
| 10:18.56 | jacc0 | timerfd causes asterisk to no longer respond to incomming sip packages |
| 10:19.29 | jacc0 | is that the result of "a problem with locking threads"? |
| 10:20.58 | *** join/#asterisk MarKsaitis (~MarKsaiti@host81-137-245-117.in-addr.btopenworld.com) |
| 10:22.15 | irroot | jacc0 it holds a lock and blocks when checking timer that lock is held indefinately this looks like a dead lock but its not really |
| 10:22.44 | *** join/#asterisk e7e5 (~rudenko@188.134.2.33) |
| 10:26.00 | jacc0 | okay, clear |
| 10:26.47 | singler | if it is not a deadlock, how it should be called? |
| 10:26.48 | jacc0 | it more like a deadtread then :P |
| 10:27.19 | singler | threadlock maybe? |
| 10:27.33 | *** join/#asterisk MarKsaitis (~MarKsaiti@host81-137-245-117.in-addr.btopenworld.com) |
| 10:27.34 | singler | :) |
| 10:27.45 | jacc0 | hehehe |
| 10:28.00 | *** join/#asterisk Rufus (Rufus@unaffiliated/rufus) |
| 10:28.41 | singler | hm, some major connection disturbance :) |
| 10:28.54 | *** join/#asterisk weinerk (~user@unaffiliated/weinerk) |
| 10:29.31 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 10:30.45 | jacc0 | removing or fixing timerfd would close a lot of open issues |
| 10:31.02 | jacc0 | in the major and blocking cathegory |
| 10:34.19 | *** join/#asterisk aberrios (~aberrios@195.171.4.82) |
| 10:35.29 | *** join/#asterisk Dibbler (~Dibbler@87-194-103-72.bethere.co.uk) |
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| 10:43.08 | *** join/#asterisk hariom (~hariom@117.195.186.152) |
| 10:43.49 | hariom | if incoming audio is recorded in some directory that is not in asterisk sounds directory, how to play it back ? |
| 10:43.55 | hariom | I am using agi |
| 10:44.09 | singler | specify full path |
| 10:45.31 | hariom | singler: Then is it possible to keep the prompts in the directory other than /var/lib/sounds/... ? |
| 10:46.09 | singler | test it and find out ;) |
| 10:48.32 | hariom | singler: it works :) |
| 10:49.01 | hariom | obviously. |
| 10:49.16 | singler | :) |
| 10:51.33 | djuhl30 | Ugh where's the best place to buy a VOIP phone in South Korea? Any ideas? |
| 10:52.02 | hariom | singler: any way you know by which these prompts can be played from a remote system ? |
| 10:52.32 | jacc0 | @hariom; make a samba share? |
| 10:53.08 | hariom | jacc0: hmm.. ok, will need to learn how to set a samba server. |
| 10:53.15 | kaldemar | or use any other preferred method of mounting a remove drive. |
| 10:53.29 | *** join/#asterisk felimwhiteley (~quassel@46.7.101.58) |
| 10:53.34 | *** join/#asterisk skrusty (~skrusty@83.166.171.74) |
| 10:53.42 | hariom | kaldemar: you mean nfs? |
| 10:54.26 | *** join/#asterisk Sertys (~cwalker@hotel-palas.com) |
| 10:54.55 | *** join/#asterisk SoulRaven (SoulRaven@89.41.156.132) |
| 10:55.29 | kaldemar | hariom: i mean whatever you prefer to use. |
| 10:55.39 | kaldemar | samba is just one choice. |
| 10:55.39 | hariom | got it. |
| 10:55.44 | hariom | ok |
| 10:56.01 | kaldemar | for quick testing, i like sshfs. |
| 10:56.49 | hariom | In that case the major point of concern might be sending prompts to asterisk over a network. But I guess Lan environment can provide fat pipe |
| 10:56.59 | hariom | ok |
| 10:57.03 | hariom | looking at it |
| 10:57.54 | kaldemar | why would you want to play prompts over network anyway? |
| 10:59.19 | hariom | kaldemar: one reason is to make sure that asterisk server remains untouched. Another is to get the recorded audio to the application server. |
| 11:01.48 | *** join/#asterisk kleszcz (tick@80.54.23.253) |
| 11:06.13 | *** join/#asterisk cyborg-one (1000@85-238-121-42.broadband.tenet.odessa.ua) |
| 11:10.12 | wonderworld | hariom: you can customize the sounds directory in asterisk.conf |
| 11:10.46 | wonderworld | be sure to read the first line of the file ;) |
| 11:11.53 | *** join/#asterisk awk (~awk@equipoise.hack.co.za) |
| 11:15.59 | *** join/#asterisk davlefou (~david@41.225.9.81) |
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| 12:38.34 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 12:38.34 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 10.0.0-beta1 (2011/07/22), 1.8.5.0 (2011/07/11), dahdi-linux 2.4.1.2 (2011/04/11), dahdi-tools 2.4.1 (2011/03/03), libpri 1.4.12 (2011/07/06) -=- Visit the new official Asterisk wiki: wiki.asterisk.org -=- http://xkcd.com/806/ |
| 12:39.03 | leifmadsen | ~ask |
| 12:39.03 | 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. |
| 12:39.05 | maldous | i have a server address, an outbound proxy, an address and a label. i'm so unsure what to configure in sip.conf |
| 12:39.14 | maldous | too quick. ;) |
| 12:39.27 | maldous | i had this working once before, but it was an unusual sip.conf configuration, that I can't figure out. |
| 12:39.46 | jacc0 | in sip.conf |
| 12:40.04 | *** join/#asterisk radic (~radic@tmo-096-183.customers.d1-online.com) |
| 12:40.23 | jacc0 | for the phones extension add dtmfmode=info/rfc2833/inband |
| 12:40.40 | Faustov | jacc0: with the slashes? |
| 12:40.41 | jacc0 | look in the phones config what it is configured to use |
| 12:40.44 | jacc0 | no |
| 12:40.46 | jacc0 | one of them |
| 12:40.54 | Faustov | I got the rfc |
| 12:41.01 | jacc0 | look in the phone config to see witch one it uses |
| 12:41.10 | Faustov | I don't control the other end's phone |
| 12:41.13 | Faustov | it's a landline btw |
| 12:41.23 | Faustov | but in any case, when he dials in, it's ok |
| 12:41.47 | Faustov | perhaps I need to "overwrite" the dtmf setting in the extensions.conf, in that context? |
| 12:42.22 | kaldemar | Faustov: no. |
| 12:42.25 | jacc0 | hmm, imho originating from a phone should use the same dtmfmode as calling from that phone |
| 12:42.44 | Faustov | exactly my thoughts ;( |
| 12:45.12 | jacc0 | maybe add an answer() above your meetme in the context |
| 12:45.20 | Faustov | I have |
| 12:45.30 | Faustov | I always have answer() and hangup() in my contexts ;) |
| 12:45.49 | Faustov | also, just tried the same channel originate command but to a phone I control - and it works fine |
| 12:47.26 | jacc0 | can you hear audio on the originated phone? what is the set dtmfmode for the originated phone? |
| 12:48.16 | jacc0 | is the phone that work the same type of phone configured the same way? |
| 12:50.45 | jacc0 | is your asterisk server behind nat or firewall ? is the originated phone behind nat or firewall? |
| 12:53.06 | jacc0 | I'm beginning to think your rtp stream is not comming true on your asterisk; nat issue |
| 12:55.39 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 12:55.39 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 12:56.27 | *** join/#asterisk timahvo1 (~rogue@41.212.123.197) |
| 12:57.53 | *** join/#asterisk timeshell (~chatzilla@gw.lusi.on.ca) |
| 12:58.04 | jacc0 | install tshark on your asterisk server |
| 12:58.07 | jacc0 | and run: |
| 12:58.23 | jacc0 | tshark -R "sip || rtp || rtpevent" |
| 12:58.29 | Faustov | jacc0: yes, audio is not a problem, just key-presseses get ignored there. I don't know what the settings are there, as it is a landline |
| 12:59.09 | timeshell | Greetings. We have been experiencing serious side tone issues since upgrading our dahdi drivers to latest release and asterisk to 1.8.4.4. from 1.6.0.26. The side tone only appears on the Polycom SIP phones, not the PSTN side. Doesn't seem to affect SIP to SIP calls. Any thoughts??? |
| 12:59.12 | jacc0 | Hmm, then you might be f@#$%d |
| 12:59.38 | jacc0 | dahdi's latest release would be 2.5 |
| 12:59.46 | timeshell | I have been trying to resolve this for weeks and am really getting frustrated. |
| 12:59.52 | jacc0 | not much people in here have experiance with that |
| 13:00.04 | jacc0 | becaus it has come out today |
| 13:00.11 | jacc0 | try an older version |
| 13:00.37 | timeshell | jacc0 Hey, you were helping me a few weeks ago when the phones were losing their registration weren't you? |
| 13:00.40 | Faustov | jacc0: can I re-set the dtmf mode to "auto" in that particular context? |
| 13:01.00 | jacc0 | nope, don't think so |
| 13:01.24 | Faustov | I could perhaps change it globally |
| 13:01.45 | jacc0 | I guess the provider usses a different config for incomming and outgoing calls |
| 13:02.11 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-tejyihmracmxdazy) |
| 13:02.31 | jacc0 | start tshar -R "sip || rtpevent" |
| 13:02.55 | jacc0 | and see if you are recieving sip-info packets that hold dtmf or maybe rtpevent packets that hold dtmf |
| 13:03.39 | jacc0 | and record the call with mix monitor to hear if there is any inband dtmf |
| 13:03.52 | jacc0 | of non of these are there; you can stop trying |
| 13:04.06 | jacc0 | and call the landline provider |
| 13:04.24 | jacc0 | or the sip provider |
| 13:04.35 | kaldemar | or look into dahdi settings. |
| 13:04.44 | jacc0 | true |
| 13:05.01 | jacc0 | if you use digium card |
| 13:05.03 | *** join/#asterisk Guest8383 (~Geek@unaffiliated/cain) |
| 13:06.07 | kaldemar | or any other card that dahdi uses. |
| 13:06.13 | jacc0 | true |
| 13:06.32 | Faustov | no, it's my sip provider who routes the calls further to landlines |
| 13:06.41 | *** join/#asterisk darkskiez (~darkskiez@2001:470:9278:5:2e0:4cff:fe68:1e29) |
| 13:06.49 | jacc0 | then start tshark on your asterisk server |
| 13:06.57 | Faustov | was a dtmfmode=auto setting supported in 1.6.2? |
| 13:07.03 | jacc0 | <PROTECTED> |
| 13:07.26 | jacc0 | and see if there are any sip-info or rtpevent packages with DTMF |
| 13:07.41 | jacc0 | and record the call with mix monitor and lissten if there is any inband dtmf |
| 13:07.49 | kaldemar | Faustov: then the endpoint being a landline is irrelevant. what matters is your connection to the provider and whatever happens after that. |
| 13:07.59 | jacc0 | of none of them are there: you can stop what you are doing and call your sip provider |
| 13:08.41 | Faustov | jacc0: I see |
| 13:09.27 | Faustov | kaldemar: same provider works ok if another end-point is used |
| 13:09.51 | jacc0 | hmm\ |
| 13:10.23 | kaldemar | Faustov: "auto" for dtmfmode is supported in 1.6.2. |
| 13:10.47 | jacc0 | maybe the guy on the other end could generate dtmf with some other device and hold it next to the phones mic? |
| 13:11.02 | *** join/#asterisk Dovid (Dovid@office.mypbxmanager.net) |
| 13:11.05 | Faustov | lol, I've used this hack a while ago |
| 13:11.07 | jacc0 | using his mobile phone or some software on his laptop? |
| 13:11.27 | Faustov | people almost crucified me for even suggesting it |
| 13:11.56 | kaldemar | and they were right. |
| 13:12.24 | singler | also dtmf can be transfered inband |
| 13:12.44 | Faustov | argh |
| 13:12.47 | singler | I would recomend recording rtp stream and replay it with wireshark |
| 13:12.50 | Faustov | just tried again and it worked |
| 13:12.51 | Dovid | i know to make an array and one value to do it. I do <?PHP $foo = array(); $foo[] = x ; $foo[] = y but how would I add two values, say a name and a number |
| 13:12.54 | Faustov | he entered the corret pin |
| 13:13.06 | Faustov | perhaps it is just a broken landline phone? |
| 13:13.32 | Dovid | oops. |
| 13:13.46 | singler | Dovid: add array to array |
| 13:28.01 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 13:28.07 | *** join/#asterisk Cain (~Geek@unaffiliated/cain) |
| 13:29.38 | timeshell | From Changelog: Since the 2.4 branch was created a new feature was merged into dahdi-linux where it was possible to mix software echocan and hardware echocan on a channel. |
| 13:29.47 | timeshell | Why would this be done? |
| 13:30.19 | timeshell | Would a hw ec not completely eliminate echo and then the sw ec be added to clean it up? |
| 13:32.01 | jaytee | if you run hw ec you don't need or want to run sw ec and I'm pretty sure that dahdi won't enable sw ec if it detects a hw ec module |
| 13:33.29 | timeshell | According to the section from the changelog I just posted above, Dahdi DOES does do this in version 2.4 |
| 13:34.15 | timeshell | And incidentally, I had hw ec on my card when I purchased it and it didn't work! I removed it and went to OSLEC |
| 13:35.29 | timeshell | But since upgrading from dahdi 2.2.0.2 to 2.4.1.2 I have been experiencing serious sidetone issues |
| 13:35.45 | timeshell | I'm at my wits end trying to figure this out. |
| 13:44.25 | singler | timeshell: try 2.5.0, it is fresh out |
| 13:47.50 | *** join/#asterisk catphish (~catphish@gateway.office.atechmedia.net) |
| 13:48.45 | leifmadsen | The_Boy_Wonder: F7ESH! |
| 13:49.49 | *** join/#asterisk logicwrath (~no@c-68-62-24-205.hsd1.mi.comcast.net) |
| 13:49.49 | *** join/#asterisk coppice (~coppice@211.95.105.223) |
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| 13:55.26 | catphish | is it sane to run asterisks with 50,000+ extensions? |
| 13:55.51 | catphish | spanned accross 1000+ included config files |
| 13:56.12 | catphish | and similar numbers of sip peers |
| 13:58.17 | irroot | small problem compiling dahdi_tools 2.5.0 without libnewt you need to now explicitly use --without-newt |
| 13:59.52 | leifmadsen | catphish: if it works, sure :) |
| 14:00.06 | leifmadsen | catphish: although I'd think there should be a better way of building the system :D |
| 14:00.07 | *** join/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 14:00.15 | catphish | good answer, sadly it doesn't |
| 14:00.23 | catphish | about to shard it into multiple processes |
| 14:00.29 | catphish | unless anyone has a smarter idea |
| 14:00.40 | leifmadsen | are you talking like, 50,000+ extensions in dialplan, or 50,000+ end points? |
| 14:00.58 | catphish | end points? |
| 14:01.08 | leifmadsen | so ya, that's a lot of end points :) |
| 14:01.14 | leifmadsen | are they all registering at the same time? |
| 14:01.15 | leifmadsen | to the same box? |
| 14:01.20 | catphish | yep |
| 14:01.25 | leifmadsen | ya, that's probably not going to work |
| 14:01.31 | *** join/#asterisk chazzam (~chazz@173-24-239-247.client.mchsi.com) |
| 14:01.31 | leifmadsen | sounds like you need a SIP proxy |
| 14:01.59 | catphish | actually i was about to run multiple asterisks and just point hosts at different ports |
| 14:02.28 | leifmadsen | that would also work |
| 14:02.35 | leifmadsen | then you're getting into a distributed model |
| 14:02.48 | catphish | that's going to have to be it |
| 14:02.50 | leifmadsen | you'll want to use something like OpenAIS (res_ais) to distribute the device state |
| 14:03.07 | catphish | that doesn't matter really |
| 14:03.16 | catphish | each extension is basically a separate customer |
| 14:03.21 | catphish | they don't need to talk |
| 14:03.23 | leifmadsen | then you may want either a dedicated box for voicemail, or use something like IMAP or ODBC storage so everyone can login to voicemail from whatever box their on, and have the data stored in the same location |
| 14:03.28 | leifmadsen | oh ok |
| 14:03.49 | catphish | i have a box that should be able to handle 2000+ calls without transcoding |
| 14:04.03 | anonymouz666 | that's a lot |
| 14:04.07 | catphish | yep :) |
| 14:04.14 | leifmadsen | if you don't distribute the voicemail, make sure the same customers connect tothe same boxes if you're using voicemail |
| 14:04.27 | leifmadsen | obviously messages and such would be stored on only one box |
| 14:04.28 | catphish | 12-core xeon boxen |
| 14:09.19 | catphish | having trouble reloading the config |
| 14:09.34 | catphish | when using one huge asterisk |
| 14:09.43 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 14:12.43 | *** join/#asterisk d_preston215 (~chatzilla@static-76-161-250-54.t1.cavtel.net) |
| 14:14.45 | catphish | are there any known limits on config size? |
| 14:14.53 | catphish | or optimizations to config reloading |
| 14:15.06 | irroot | looks like latest sangoma drivers dont play nice with linux 3.0 |
| 14:16.28 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 14:16.28 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:17.50 | coppice | that is true of a number of things, which seems to be why fedora has a 2.6.40, which is really 3, so they can sidestep this until all the other packages settle down |
| 14:17.56 | *** join/#asterisk b0ot (~Jinxed---@147.177.56.254) |
| 14:17.59 | b0ot | hello |
| 14:19.40 | irroot | coppice there other problems too things marked deprecated now removed have to be fixed nice idea re 2.6.40 mmm may do that |
| 14:27.05 | b0ot | I'm trying to connect my andriod phone to my asterisk call manager on my private network through wireless |
| 14:27.10 | b0ot | any tutorials on how to do it |
| 14:27.16 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 14:27.43 | *** join/#asterisk n3hxs (~ed@63.68.135.4) |
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| 14:29.22 | irroot | b0ot the fools here with droids got it working try sipdroid ?? |
| 14:31.04 | Kobaz | csipsimple > * |
| 14:31.25 | Kobaz | all the other android soft phones are terrible |
| 14:31.45 | timeshell | Bria |
| 14:31.57 | Kobaz | still crappy |
| 14:32.00 | Kobaz | heh |
| 14:32.05 | timeshell | Not |
| 14:32.08 | timeshell | One of the best. |
| 14:32.16 | Kobaz | how many accounts can you add? |
| 14:32.35 | timeshell | LOL on Droid? Dunno. I use it on iPhone. |
| 14:32.40 | timeshell | I use 2 accounts right now. |
| 14:32.58 | catphish | leifmadsen: i'm going to try config_mysql |
| 14:33.00 | Kobaz | i tried I think every soft phone for android, and they either allow one account or two |
| 14:33.17 | catphish | it seems that running that many extensions isn't the problem, but reloading the config is painful |
| 14:33.27 | timeshell | It appears that I can do more than 2 with Bria on iPhone. |
| 14:33.36 | timeshell | I'd be surprised if they limited it on droid. |
| 14:34.12 | Kobaz | csipsimple is really great. you can get to the very detailed config and change just about anything to do with the sip config |
| 14:34.19 | b0ot | irroot, I'm mostly confused on how to setup the wireless to just extend |
| 14:34.27 | b0ot | my private lan i think |
| 14:34.38 | timeshell | Just configured a fake third... |
| 14:35.11 | Kobaz | timeshell: so maby the bria on iphone isn't so bad |
| 14:35.15 | Kobaz | i've only used the android apps |
| 14:36.03 | Kobaz | anyway, time to putz with the snowblowers before winter comes |
| 14:36.48 | timeshell | winter? snowblowers? Where are you? |
| 14:37.06 | Kobaz | central-eastern new york |
| 14:37.23 | timeshell | Wow, there's like 3 more months at least before snow. |
| 14:37.26 | Kobaz | yeah |
| 14:37.32 | Kobaz | and three months to get these things running |
| 14:37.37 | timeshell | Go fix your car or something. |
| 14:37.39 | timeshell | :p |
| 14:38.09 | Kobaz | it sucks working on equipment in a cold garage when it's snowing and you need to blow but the blowers aren't blowing because you forgot to work on them all summer |
| 14:38.30 | timeshell | lol, I guess. I don't have one! |
| 14:38.44 | n3hxs | Try rebuilding an in frame engine while laying in 3 feet of snow in the Sierras... |
| 14:39.27 | timeshell | I prefer to save my money and put more heave ho into snow |
| 14:39.38 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 14:39.45 | Kobaz | n3hxs: sounds like buckets of fun |
| 14:44.58 | *** join/#asterisk ruben23 (~RLACUMBA@121.97.111.142) |
| 14:45.34 | ruben23 | hi guys |
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| 14:52.12 | *** join/#asterisk adyn (~adyn@pdpc/supporter/active/adyn) |
| 14:52.33 | *** join/#asterisk eXcAliBuR (~eXcAliBuR@206.162.174.6) |
| 14:52.49 | n3hxs | Well, maybe shovels full Kobaz. |
| 14:53.16 | eXcAliBuR | i need help wit my polycom ip 321 phone, after doing a upgrade to lastest firmware, the phone restarts then it can't connect to the boot server anymore. any ideas? |
| 14:54.02 | eXcAliBuR | if i tell it to use the tftp server that i have running on my computer it will work... but i want it to use the one on my asterisk box |
| 14:54.27 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 14:59.08 | *** join/#asterisk Eckrall (~kevin@eckrall.co.uk) |
| 15:00.13 | Kobaz | heh |
| 15:02.57 | merlin8282 | What are all the conference applications available ? I'm just trying (since many days now) to set up nway conference, but don't know how to install/configure dahdi kernel module, so I tried to do nway with ConfBridge, but it doesn't work |
| 15:03.21 | ruben23 | guys any idea im having asterisk server using voip trunk and when calls comes in the customer i calls most of the time are in very low volume the line is good but very low volume i already adjusted softphones volume is full but still the voice of the one i call is low, any idea guys...? |
| 15:04.04 | eXcAliBuR | tell them to speak up? |
| 15:05.05 | catphish | can anyone point me in the direction of mysql table definitions? |
| 15:05.13 | *** join/#asterisk tyman (~tyler@173-12-219-189-Fresno.hfc.comcastbusiness.net) |
| 15:06.15 | _Corey_ | catphish: contrib/realtime/mysql |
| 15:06.29 | catphish | i love you :) |
| 15:07.59 | _Corey_ | um ok |
| 15:08.00 | _Corey_ | lol |
| 15:08.15 | Eckrall | Hi all, does anyone know if 1.6 still has a limit of 64 pickup groups? |
| 15:08.20 | merlin8282 | make love, not war :p |
| 15:08.26 | merlin8282 | (they said) |
| 15:09.40 | coppice | make both - get married |
| 15:11.27 | anonymouz666 | haha |
| 15:11.52 | *** join/#asterisk Jcook_5xData (~Jcook_5xD@173.162.32.1) |
| 15:12.07 | tzanger | coppice: how true |
| 15:13.03 | ruben23 | eXcAliBuR: they already speaking at loud still the volume is low.. |
| 15:13.11 | merlin8282 | mmm, let it say another way: would meetme work if I install dahdi kernel module, even if no hardware is in the server ? Will it not cause any problem ? |
| 15:14.22 | Eckrall | merlin8282: it will work, but we have found that the audio can be really crappy |
| 15:14.30 | catphish | _Corey_: sadly it looks like i need to work out extensions.conf |
| 15:14.36 | catphish | (table) |
| 15:15.03 | merlin8282 | Eckrall: ok, i'll give it a try anyway... |
| 15:15.05 | *** join/#asterisk singler (~singler@78-60-139-121.static.zebra.lt) |
| 15:15.14 | catphish | ah, someone on the wiki has posted one |
| 15:16.09 | _Corey_ | catphish: http://pastebin.com/1xZWDa2P |
| 15:17.17 | merlin8282 | mmm, just installed dahdi-source on the server (squeeze). Now I have "build_tools debian drivers include Makefile" in /usr/src/modules/dahdi/ |
| 15:17.30 | catphish | _Corey_: i found 2 conflicting versions on the wki |
| 15:17.39 | merlin8282 | how do I compile it ? Simply make, or is there a better way to do this ? |
| 15:17.42 | catphish | one like that and one that uses 'application' |
| 15:18.12 | _Corey_ | catphish: That's from a production 1.4 system. I'm not sure if there have been changes, though you can debug easily enough |
| 15:18.28 | catphish | thanks :) |
| 15:19.24 | singler | catphish: you can also check contrib dir in asterisk source, there sometimes are usefull samples (eg mysql sqls) |
| 15:19.55 | singler | merlin8282: how did you install id? |
| 15:20.18 | merlin8282 | singler: aptitude install dahdi-source |
| 15:20.24 | catphish | singler: i did that first thanks |
| 15:21.07 | *** join/#asterisk MarKsaitis (~MarKsaiti@94-30-69-205.xdsl.murphx.net) |
| 15:22.21 | singler | merlin8282: I think if you are installing source from repo, then you can install binary package too. I guess that source is for other programs which may need it. But using vanilla sources you can do make and later make install, so probably and dahdi-source can |
| 15:24.13 | merlin8282 | singler: aaaaah, it's in dahdi-linux -_- |
| 15:24.18 | merlin8282 | thanks! |
| 15:24.35 | singler | np |
| 15:24.49 | p3nguin | You can get source from the repo? Weird. |
| 15:25.09 | *** join/#asterisk hobodave (~hobodave@pdpc/supporter/professional/hobodave) |
| 15:25.14 | singler | I guess it is like linux-headers |
| 15:25.21 | p3nguin | That's not source. |
| 15:25.23 | singler | some programs need this |
| 15:25.34 | *** join/#asterisk retentiveboy (~pdugas@74-95-28-33-Atlanta.hfc.comcastbusiness.net) |
| 15:25.44 | singler | I know that linux-headers is not a source, it is just an example |
| 15:26.13 | merlin8282 | singler: ok, and what should be the name of the kernel driver ? |
| 15:26.25 | singler | eg, if I would be installing sangoma drivers from E1/T1, I would need dahdi source, and I would be in problem if I used dahdi from package |
| 15:26.29 | merlin8282 | I don't find any dahdi or so in /lib/modules/ |
| 15:27.07 | singler | merlin8282: what exactly did you do already? |
| 15:27.47 | merlin8282 | actual state is: asterisk (from squeeze) installed via aptitude, not more. (I just purged the dahdi-source) |
| 15:28.14 | singler | did you download dahdi from asterisk.org? |
| 15:28.20 | merlin8282 | no |
| 15:28.36 | merlin8282 | Just installed via aptitude dahdi-linux |
| 15:28.40 | singler | did you install dahdi from apt? |
| 15:28.54 | merlin8282 | yes |
| 15:29.24 | merlin8282 | I've dahdi and dahdi-linux installed, when searching with dpkg --get-selections "*" | grep dahdi |
| 15:29.29 | *** join/#asterisk Tuju (~tuju@176.75.219.213.sta.estpak.ee) |
| 15:30.54 | singler | try to install dahdi-firmware-nonfree |
| 15:31.09 | *** join/#asterisk irieKEN (~irieKEN@rrcs-76-79-107-76.west.biz.rr.com) |
| 15:31.39 | singler | and did you try to install just dahdi? |
| 15:32.06 | merlin8282 | yes, only dahdi. It's for being able to use MeetMe |
| 15:32.19 | merlin8282 | or what do you mean ? |
| 15:32.23 | singler | I mean package named dahdi |
| 15:32.36 | singler | http://packages.debian.org/search?keywords=dahdi&searchon=names&suite=stable§ion=all |
| 15:34.04 | Bezzie | if you're on debian squeeze and attempting to get the dahdi source etc |
| 15:34.09 | Bezzie | there's some funkiness i noticed with those packages |
| 15:34.34 | Bezzie | for starters, it didnt actually install any modules for me etc when i installed it etc.. had to untar the bzipped source it dropped somewhere |
| 15:34.39 | Bezzie | i dont recall the details unfortunately :( |
| 15:34.46 | Bezzie | but i remember it behaving oddly. |
| 15:35.12 | irieKEN | Anyone know why I might be intermittently getting this message? "Failed to read an alternate host or port in SDP. Expect video problems." Endpoint is an old Nokia SIP hardphone. Happens about 60% of the time, and the calls will drop 15-20 seconds in ("no reply to our critical packet"). |
| 15:35.19 | singler | Bezzie: as I understand he wants just modules |
| 15:35.25 | merlin8282 | Bezzie: the tbz2 is dropped in /usr/src/, and when uncompressing it's in /usr/src/modules/dahdi/ |
| 15:35.34 | merlin8282 | singler: right |
| 15:35.35 | Bezzie | Yep. I had to compile the modules manually and install them |
| 15:35.42 | Bezzie | wasnt complicated |
| 15:35.49 | Bezzie | but the deb package didnt actually install the modules |
| 15:36.07 | merlin8282 | Bezzie: ok, so I do compile... |
| 15:36.20 | Bezzie | I don't believe that's the desired end state, but yes, I had to. |
| 15:36.41 | Bezzie | I will admit i did not spend much time troubleshooting, it was for a proof of concept project I was working with recently involving a sangoma usbfxo |
| 15:36.58 | irieKEN | If it's any help, the Asterisk server is running 1.8.4.1, and IS behind a firewall, but ports 5060,5061, and 10000-20000 UDP are forwarded to it. |
| 15:37.14 | Bezzie | but some research on google showed that a few people had the issue where the modules were not installed by installing the debian package |
| 15:37.24 | Bezzie | and that was the workaround i found/used :P |
| 15:37.39 | singler | irieKEN: try disabling video support (no idea if it will help) |
| 15:38.08 | singler | well I allways build from source (asterisk and dahdi) |
| 15:38.20 | Bezzie | nods. |
| 15:38.28 | irieKEN | singler: Let me give that a spin. |
| 15:38.30 | Bezzie | I originally was using the digium repository. |
| 15:38.39 | Bezzie | but 1.8 in their repository had crash issues. |
| 15:39.02 | Bezzie | so it was either do source, or by stick with stock debian's :P |
| 15:39.15 | singler | :) |
| 15:39.20 | Bezzie | For the expected return on my time into it, I just made it work :P |
| 15:39.53 | catphish | is there any way to avoid defining contexts statically when using a mysql configuration |
| 15:40.32 | singler | catphish: what do you mean? |
| 15:41.04 | irieKEN | singler: No love. The logs don't show the "failed to read alternate host or port" message, but the call still gets cut off due to non-reply to critical packet. |
| 15:42.02 | catphish | singler: when defining extensions as a mysql table, it doesn't work unless you also define each context in extensions.conf |
| 15:42.17 | catphish | with a switch line |
| 15:42.44 | singler | irieKEN: did you check verbose log or debug log? Also you can try disabling timers or use tcpdump to capture packets and try to analyze them |
| 15:43.19 | singler | catphish: yes, and you want that contexts would work without that? |
| 15:43.30 | catphish | correct |
| 15:43.49 | irieKEN | singler: I'm watching /var/log/asterisk/full. I've done a packet capture, and have been able to trace the call all the way through; looks like it's communicating properly in both directions. |
| 15:44.33 | singler | catphish: sorry, but I cannot help you with that.. |
| 15:44.44 | irieKEN | singler: I haven't disabled the timers. How do you recommend doing it? As an asside, my Aastra 6730i and Nokia N810 seem to work fine. It's just this older Nokia that's misbehaving. |
| 15:45.15 | singler | irieKEN: in sip.conf you should find commented config |
| 15:45.50 | catphish | then again it doesn't matter, a reload of thousands of empty contexts is extremely fast |
| 15:46.57 | Jcook_5xData | would it be possible in a dial plan to do some thing like 'if queue{6501} !empty then join else do nothing' the problem I have now is.. the queue is set not join empty but the call move to it if the queue is empty it move back to the one it started in possibly losing it place |
| 15:49.29 | singler | you can, but I do not remember function/app name, try checking out voip-info.org or in asterisk "core show function/application " and tab-complete, you should find something related to queue |
| 15:49.56 | singler | then read help of it |
| 15:51.59 | *** join/#asterisk godmachine-x6 (~godmachin@unaffiliated/godmachine-x6) |
| 15:52.33 | Jcook_5xData | ok I will check it out |
| 15:52.43 | irieKEN | singler: Still have the same behavior after disabling the RTP timeouts:S |
| 15:54.00 | *** join/#asterisk chazzam (~chazz@173-24-239-247.client.mchsi.com) |
| 15:54.37 | singler | I was talking about session timers, but if that will not help, I am out of suggestions.. |
| 15:55.12 | irieKEN | singler: Sorry. Do you happen to recall what the session timer variables are? |
| 15:58.12 | singler | check from line 524: http://svnview.digium.com/svn/asterisk/branches/1.8/configs/sip.conf.sample?revision=325935&view=markup |
| 15:58.27 | singler | session-timers=refuse |
| 16:00.48 | irieKEN | singler: Thanks. Setting session-timers to refuse doesn't appear to have made a difference. I did notice " Retransmission timeout reached on transmission" any idea where I could tweak that? |
| 16:06.39 | singler | scroll up my link, there will be more settings, I guess it is time for you to start googling, I am out of suggestions :) |
| 16:07.30 | irieKEN | singler: Thanks for your help:) I had actually turned to the IRC as a last resort, after spending the last 3 days pushing my Google-Fu to its limits. |
| 16:08.21 | *** join/#asterisk moy (~moy@65.204.112.138) |
| 16:09.53 | singler | np, you can try different versions of asterisk, or tcp connection if it is supported on the phone, did you try to search for formware update for phone? btw does it work if phones call it other directly? |
| 16:13.19 | irieKEN | singler: There's no firmware update; Nokia never officially released the hardware to market, and it also doesn't support TCP (circa 2007). It does, however, work with other SIP carriers, which is why I'm still trying to get it work with my system. |
| 16:14.35 | singler | in phones path launch tcpdump, write dump and compare packets to your own :) |
| 16:14.52 | singler | it may take some time, but maybe less than 3 days :) |
| 16:15.21 | irieKEN | singler: Alright I'll go back to doing that. The odd thing is that about 1 in 3 calls works just fine through the Aasterisk server. |
| 16:16.04 | singler | then compare packets in this case too, it may be easier to find difference |
| 16:16.13 | singler | do you use wireshark? |
| 16:16.39 | singler | I can give you some basic tips :) |
| 16:16.59 | singler | but if you pro at wireshark thren there is no need |
| 16:17.12 | irieKEN | singler: I have used it, and have used its SIP tracer, but I'll use any tips I can get;) |
| 16:17.51 | singler | do you have some dump with sip call? |
| 16:18.38 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 16:20.25 | irieKEN | singler: Not on hand. I'll have to create one. |
| 16:21.04 | singler | no problem, then I will just say what I want, and you'll try later |
| 16:21.12 | irieKEN | singler: I may have to cut this short; my job may require that I actually do work. |
| 16:21.17 | irieKEN | singler: Sure:) |
| 16:22.41 | singler | on telephony menu there is voip calls selection, there voip calls can be viewed and analyzed. selecting desired call you can view packet flow, listen to it and some other things |
| 16:23.07 | singler | clicking on flowcharts arrows will select packets in main view |
| 16:24.02 | singler | also there is a prepare filter button, which can be used to filter main view (you need to press apply in main windows near filter) only that call |
| 16:24.55 | singler | also there is export function where you can save displayed/select/all packets to files, eg you can save them as txt and diff them or do some other things |
| 16:25.03 | singler | EOF :) |
| 16:29.01 | *** join/#asterisk irroot (~irroot@41.51.158.204) |
| 16:29.34 | *** join/#asterisk F|ReSTaRT (~dlyh@unaffiliated/firestart) |
| 16:44.45 | irieKEN | singler: Again, thanks:) |
| 16:47.30 | singler | np :) |
| 16:53.59 | *** join/#asterisk wonderworld (~ww@port-92-201-187-209.dynamic.qsc.de) |
| 16:57.14 | *** join/#asterisk rdahlin_1_ (~rdahlin_1@78-73-19-238-no168.tbcn.telia.com) |
| 17:02.06 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 17:17.23 | *** join/#asterisk gravin (~gravin@204.34.49.60.brf01-home.tm.net.my) |
| 17:23.09 | *** join/#asterisk timahvo1 (~rogue@41.212.123.197) |
| 17:23.19 | MrTelephone | can somoene walk across the stree and slap out adtran? |
| 17:24.12 | chazzam | <PROTECTED> |
| 17:24.16 | chazzam | ;p |
| 17:24.24 | chazzam | what'd adtran do? |
| 17:24.48 | MrTelephone | I bought an awesome multiservice switch from them and it doesn't support caller name on DMS100 |
| 17:25.26 | paulc | MrTelephone: What's the multiservice switch model #? |
| 17:27.09 | MrTelephone | atlas 550 |
| 17:28.12 | malcolmd | i'm giving their south tower an evil look |
| 17:29.07 | MrTelephone | dude, the callername comes in as DISPLAY IE instead of a FACILITY IE. why is it so hard to program it to decode that? |
| 17:32.47 | *** join/#asterisk cerberus_za (~coert@196-210-206-227.dynamic.isadsl.co.za) |
| 17:33.00 | MrTelephone | There programming staff is on floor 13 i guess |
| 17:33.33 | malcolmd | that's a few floors above the roof, so i guess they're in free-fall now |
| 17:33.44 | anonymouz666 | anyone in here already setup heartbeat with asterisk? |
| 17:33.58 | Qwell | anonymouz666: Are you going to Astricon? |
| 17:34.30 | anonymouz666 | Qwell: No Sir. |
| 17:34.35 | Qwell | too bad |
| 17:34.37 | paulc | MrTelephone: Google "Atlas 550 caller name" and the top link is a 2 page sales-y PDF. It says "Will pass Caller ID name and number information from the PRI.." - I'd tell them that then get bolshy and British with them.. but that's just me ;-) |
| 17:34.43 | Qwell | Asterisk redundancy: Handling the inevitable. |
| 17:34.43 | Qwell | Jason Parker |
| 17:34.43 | Qwell | Digium |
| 17:35.04 | *** join/#asterisk brdude (~brdude@12.155.183.30) |
| 17:35.30 | chazzam | Qwell be speaking at the astricon |
| 17:35.32 | anonymouz666 | Qwell: I wish I could see this presentation |
| 17:35.51 | anonymouz666 | but it is so far away from here |
| 17:36.05 | Qwell | excuses |
| 17:36.09 | anonymouz666 | Qwell: I can see the PDFs on the website |
| 17:36.22 | chazzam | won't get to go either |
| 17:37.06 | pabelanger | Qwell's going to be buying the beer that day :D |
| 17:37.12 | MrTelephone | I'll present something |
| 17:37.20 | anonymouz666 | where will be this year? |
| 17:37.25 | chazzam | an adtran dropping from a roof MrTelephone ? |
| 17:37.30 | MrTelephone | hahah |
| 17:37.59 | *** join/#asterisk tamiel (~tamiel@65.204.112.138) |
| 17:38.41 | drmessano | Funny, I have a similar talk on Asterisk I have given |
| 17:39.11 | chazzam | I almost want to ask if it was open sourced |
| 17:39.22 | Qwell | drmessano hates freedom |
| 17:39.44 | drmessano | My talk is centered around keeping a box in the server room with 1.2 on it, as a decoy |
| 17:39.54 | chazzam | blinks |
| 17:39.59 | drmessano | So if someone wants to modify "the system", I show them to that box |
| 17:40.10 | drmessano | Keeping the production box away from prying eyes |
| 17:40.17 | MrTelephone | i want to do a presentation on failover without dropped calls |
| 17:40.36 | drmessano | If you don't want to drop calls, use sturdier rack shelves |
| 17:40.45 | drmessano | or lighter boxes |
| 17:40.48 | drmessano | or bigger screws |
| 17:40.52 | drmessano | It's not that difficult |
| 17:41.04 | anonymouz666 | remote pickup call is possible today? |
| 17:41.08 | Qwell | plug the hole in the bottom of the server |
| 17:41.10 | MrTelephone | now that you mention it i think my asterisk server weighs 50 pounds |
| 17:41.42 | drmessano | You should also stick with Drmessano Law of Open Source and Duct Tape |
| 17:41.48 | chazzam | asterisk scf? |
| 17:42.17 | drmessano | If you're using open source on a box that's too big to stick to a wall with duct tape, you overdid it |
| 17:42.30 | chazzam | heh |
| 17:42.41 | chazzam | well, you can stick a person to a wall with ducttape |
| 17:42.42 | MrTelephone | no kidding |
| 17:42.50 | anonymouz666 | Queue A is ringing phone A on box A, phone B is member of Queue A in box B. can phone B somehow pickup the call is ringing on phone A? |
| 17:42.52 | chazzam | so that's a fairly relative assessment |
| 17:43.13 | chazzam | you have the queue distributed across two systems? |
| 17:43.39 | jaytee | imagine a room where the walls and ceiling are covered with velcro and you have a velcro jumpsuit with velcro gloves and boots |
| 17:43.40 | drmessano | chazzam: Indeed.. But I have a different rule for employees... It mostly involves the connection between their job performance and how well they bounce when dropped from the second floor balcony |
| 17:43.43 | MrTelephone | can we make asterisk a cluster by design? |
| 17:44.21 | chazzam | heh |
| 17:44.28 | chazzam | MrTelephone: asterisk scf? |
| 17:44.32 | MrTelephone | if someone phones box Z can someone on box F pick it up? And if someone trips over the power cord on Box F can everything be automatically shifted to box D? |
| 17:44.33 | *** join/#asterisk webguynow (~webguynow@c-24-1-222-204.hsd1.il.comcast.net) |
| 17:44.55 | chazzam | wasn't that the demo from astricon last year for asterisk scf? |
| 17:44.57 | MrTelephone | What is the point of ISDN redirect? |
| 17:44.57 | drmessano | SCF can do that |
| 17:45.00 | chazzam | the power cord thing? |
| 17:45.08 | jaytee | if someone trips over the power cord then that's an OSHA violation :-) |
| 17:45.14 | chazzam | lol |
| 17:45.26 | drmessano | If they trip over the power cord, the server is too close to the table in the break room. You FAIL |
| 17:45.40 | drmessano | To correct, move the table |
| 17:45.53 | chazzam | you poor sods, you got jilted into having your break room in the server room |
| 17:45.56 | webguynow | I saw cluecon (but couldn't go) but I did attend iptcomm.org IIT conference last week - from an application developer's background, but love to get into VoIP and looking for a new gig. |
| 17:46.03 | MrTelephone | you popped a breaker on south wall so your running 4 x 6ft extension cords from the north wall to power your phone system |
| 17:46.13 | webguynow | chicago area here |
| 17:46.41 | *** join/#asterisk italorossi (~italoross@201.76.154.127.intranet.digi.com.br) |
| 17:46.58 | drmessano | Having the break room table in the server room can actually cut your server monitoring costs greatly |
| 17:47.12 | leifmadsen | MrTelephone: Live call failover is only possible in Asterisk SCF |
| 17:47.38 | chazzam | and it lets you have a cluster by design |
| 17:47.54 | drmessano | I am actually considering a chapter on server placement in MY Asterisk book |
| 17:47.59 | chazzam | heh |
| 17:48.13 | chazzam | quick leifmadsen, duck! |
| 17:49.21 | MrTelephone | I should read more. I seen SCF and didn't even know what it stood for. |
| 17:49.33 | leifmadsen | drmessano: sweet! :) |
| 17:49.42 | leifmadsen | Scalable Communications Framework |
| 17:50.39 | *** part/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 17:50.43 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 17:50.43 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 17:50.48 | leifmadsen | ~asteriskscf |
| 17:50.54 | leifmadsen | wow really? |
| 17:50.57 | MrTelephone | why didn't you decide to add scf as a module to the current asterisk? |
| 17:51.04 | drmessano | Chapter 7 is I think I am gonna be sick!: 101 places for your Asterisk server besides the handicap stall in the Men's room |
| 17:51.04 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 17:51.06 | leifmadsen | MrTelephone: because that would not be possible |
| 17:51.24 | leifmadsen | MrTelephone: Asterisk SCF is totally separate of Asterisk in terms of how it is built and how it works |
| 17:51.38 | leifmadsen | it's not something that could be added as a "module" |
| 17:52.21 | leifmadsen | infobot: asteriskscf is the Asterisk Scalable Communications Framework. More information about Asterisk SCF is available on the wiki at https://wiki.asterisk.org/wiki/display/TOP/Asterisk+SCF+Home |
| 17:52.21 | infobot | leifmadsen: okay |
| 17:52.41 | chazzam | ~asteriskscf |
| 17:52.42 | infobot | it has been said that asteriskscf is the Asterisk Scalable Communications Framework. More information about Asterisk SCF is available on the wiki at https://wiki.asterisk.org/wiki/display/TOP/Asterisk+SCF+Home |
| 17:53.31 | leifmadsen | infobot: no, asteriskscf is <reply> Asterisk SCF is the Asterisk Scalable Communications Framework. More information about Asterisk SCF is available on the wiki at https://wiki.asterisk.org/wiki/display/TOP/Asterisk+SCF+Home |
| 17:53.31 | infobot | leifmadsen: okay |
| 17:53.36 | leifmadsen | ~asteriskscf |
| 17:53.36 | infobot | Asterisk SCF is the Asterisk Scalable Communications Framework. More information about Asterisk SCF is available on the wiki at https://wiki.asterisk.org/wiki/display/TOP/Asterisk+SCF+Home |
| 17:53.38 | drmessano | Asterisk SCF is the answer to "I have 100,000 clients and need failover, since Asterisk can't do it, screw you guys, I am gonna use SER/Kamailio! ... What say you now???" |
| 17:53.39 | leifmadsen | better |
| 17:53.40 | *** join/#asterisk Docfxit (~Console1@netblock-75-79-6-10.dslextreme.com) |
| 17:53.47 | leifmadsen | drmessano: yes |
| 17:54.13 | MrTelephone | sip srv records already help us with load balancing. It's getting the call to the right endpoint |
| 17:54.20 | MrTelephone | you just had to copy SERS registry module |
| 17:55.06 | MrTelephone | rtp still has to pass through asterisk to be able to minipulate the audio streams? |
| 17:55.17 | MrTelephone | or whatever |
| 17:56.04 | MrTelephone | I'm probably out to lunch |
| 17:57.05 | Docfxit | I'm having trouble with one voicemail box. When I call it I get silence instead of the default greeting. Any ideas what might fix it? |
| 18:00.41 | chazzam | do you ever get any audio from it? |
| 18:01.10 | Docfxit | In /var/spool/asterisk/voicemail/default/105 I have moved the files busy, greet, unavail to a different folder. |
| 18:01.35 | Docfxit | chazzam: Yes |
| 18:02.08 | Docfxit | After I move those files do I need to restart Asterisk? |
| 18:04.24 | chazzam | yes |
| 18:07.03 | MrTelephone | Wait until your 'core sip channels' is greater than 45 before you 'core stop now' |
| 18:08.05 | *** join/#asterisk LittleFool (~LittleFoo@over-dozed.com) |
| 18:08.25 | MrTelephone | people are still making money off vhosts? |
| 18:09.07 | MrTelephone | Why would somoene replace an hp printhead when the black ink says low? |
| 18:09.24 | LittleFool | I want to route 5 numbers to different extensions but asterisk keeps using the first route, any suggestions? |
| 18:14.12 | *** join/#asterisk jpcansa (~JP@201.196.95.195) |
| 18:16.45 | Qwell | LittleFool: You've not shown us anything, so no, any suggestions we might have are completely useless. |
| 18:17.59 | MrTelephone | some wife beater just slashed my sister in laws tires for calling child services. people like that should just hang themselves |
| 18:18.06 | LittleFool | Qwell: ow sorry, im using patterns like this right now _X.971 _X.972 untill 7 but only 971 is used no matter which number im calling |
| 18:18.31 | Qwell | Those aren't valid patterns. |
| 18:18.42 | LittleFool | cos of the dot? |
| 18:18.50 | Qwell | everything after . is ignored. |
| 18:19.28 | LittleFool | is there any other way to match "anything" i mean any count of digest? |
| 18:20.18 | LittleFool | cos the number could be 9 or 12 digest long |
| 18:21.58 | _Corey_ | put the dot at the end |
| 18:22.48 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 18:22.53 | LittleFool | but wouldnt _X971. match 0971.... or 3971... |
| 18:23.22 | _Corey_ | yeah... what are you trying to match? |
| 18:23.33 | singler | MrTelephone: no, they should be heavily beatten, broken legs and arms and castrated. And left to rot in jail |
| 18:23.45 | LittleFool | the 971 is at the end but i dont know how many digest are before it |
| 18:23.53 | _Corey_ | oh, I see |
| 18:24.13 | LittleFool | it could be 8,9 or 11 |
| 18:25.08 | singler | LittleFool: define a few extens |
| 18:25.42 | _Corey_ | Are the last 4 digits the only significant ones? |
| 18:25.48 | LittleFool | yes |
| 18:26.13 | _Corey_ | If so, you could substring them off and then just use Goto() to send the call to the 4-digit exten |
| 18:27.06 | MrTelephone | singler, i like that idea too |
| 18:27.11 | LittleFool | hmmm im using freepbx and im pretty nooby :D |
| 18:29.42 | _Corey_ | LittleFool: Well, it can get weird w/FreePBX but you could send your trunk into some custom context, then in there do "exten => _X.,1,Goto(from-trunk,${EXTEN:-4},1)" |
| 18:29.58 | _Corey_ | you may get more mileage in #FreePBX |
| 18:31.25 | LittleFool | okey thx _Corey_ |
| 18:32.57 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 18:34.19 | LittleFool | can i put two DID numbers in one inbound route? |
| 18:34.40 | LittleFool | somehow? |
| 18:34.45 | Qwell | Asterisk has no concept of "inbound routes". Ask in #freepbx. |
| 18:35.12 | LittleFool | right, its dial plan in asterisk |
| 18:35.32 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v012-217.mobile.uci.edu) |
| 18:36.56 | _Corey_ | Either way, you're asking whether something can be done in the FreePBX GUI, not Asterisk... The #Freepbx guys can probably explain |
| 18:36.59 | *** join/#asterisk neurosys (~neurosys@65.204.112.138) |
| 18:38.34 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 18:38.34 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 18:52.37 | *** join/#asterisk clintc (~clintc@n128-227-109-39.xlate.ufl.edu) |
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| 19:05.50 | *** part/#asterisk clintc (~clintc@n128-227-109-39.xlate.ufl.edu) |
| 19:13.10 | *** join/#asterisk vfabi (~fabi@host-static-188-237-250-140.moldtelecom.md) |
| 19:18.15 | *** join/#asterisk cerienjean (~iper@95.138.77.91) |
| 19:29.29 | *** join/#asterisk lanning (~lanning@173-8-187-197-SFBA.hfc.comcastbusiness.net) |
| 19:32.24 | *** join/#asterisk Trinitel (5eaa61e1@gateway/web/freenode/ip.94.170.97.225) |
| 19:34.22 | Trinitel | hi all. Is there a way to have some visual notification (flashing key or something similar) on polycom phones (ip331 or similar) for parked calls? |
| 19:34.49 | pdtpatrick | in dundi.conf .. what are all these mac address mapped to? |
| 19:35.21 | pdtpatrick | entityid ? |
| 19:36.58 | *** join/#asterisk gogasca (~Adium@nat/cisco/x-kiaaeksjnoktutiu) |
| 19:37.31 | *** part/#asterisk LittleFool (~LittleFoo@over-dozed.com) |
| 19:42.04 | *** join/#asterisk Trinitel (5eaa61e1@gateway/web/freenode/ip.94.170.97.225) |
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| 19:47.25 | Trinitel | hi, somebody to help? |
| 19:50.13 | Katty | can i borrow someone to help troubleshoot a no reply to our critical packet (see doc/sip-retransmit.txt)issue that's giving me a headache. |
| 19:50.28 | Katty | i have ports 5060, and rtp 10k-20k open |
| 19:50.58 | Katty | 5060 seems to be perfectly fine, the audio is nonexistant |
| 19:51.08 | Katty | sip.conf shows that nat=yes |
| 19:51.25 | Katty | however sip show peers says Nat is N |
| 19:51.54 | jaytee | that means it is nat'd |
| 19:52.07 | Katty | ok |
| 19:52.13 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v012-217.mobile.uci.edu) |
| 19:52.24 | Katty | does that error mean that asterisk can't reach the ip on the other end to send rtp? |
| 19:52.25 | jaytee | but do you have the same ports open on your firewall to match the RTP? |
| 19:52.34 | Katty | i'll doublecheck rtp.conf |
| 19:52.55 | Katty | yeah |
| 19:52.58 | cerienjean | good test is to set rtp debug on and see to which ip the packets go to - local or nated ? |
| 19:53.06 | Katty | that's a good idea. |
| 19:53.26 | jaytee | the default in rtp.conf is 10000-20000. You would need to make sure the firwall has port forwarding to your * box for ports 10000-20000 to the inside IP address of your * server |
| 19:53.35 | cerienjean | if you have the right ip, then its probably network, otherwise, ast config |
| 19:53.37 | Katty | yeah i've got that one done jaytee |
| 19:53.46 | Katty | how do i turn on rtp debug |
| 19:53.57 | leifmadsen | rtp set debug on |
| 19:54.00 | Katty | ah nm found it |
| 19:54.01 | leifmadsen | rtp set debug off |
| 19:54.03 | jaytee | sip debug and wireshark are your friends |
| 19:54.06 | leifmadsen | ya, you found it right ther e:) |
| 19:54.30 | Katty | how do i enable it for just the extension? |
| 19:54.36 | Katty | rtp set debug on 132? |
| 19:55.16 | jaytee | rtp set debug on|off|ip |
| 19:55.29 | jaytee | so you can't do peer names like with sip debug |
| 19:55.58 | cerienjean | you can set it for one IP |
| 19:56.09 | cerienjean | rtp set debug ip .... |
| 19:56.10 | Katty | my syntax is failing me |
| 19:56.27 | cerienjean | but if you are waiting for the wrong ip, wont do no good |
| 19:56.34 | cerienjean | cant do it for one ext - just ip |
| 19:56.40 | Katty | no it's not liking the syntax. *grump* |
| 19:59.38 | Katty | http://paste2.org/p/1577056 <- syntax spaz. |
| 19:59.58 | Katty | any changes on that in 1.6.2.5? |
| 20:01.08 | beek | waves to Katty |
| 20:01.18 | Katty | here we go, i got some info |
| 20:01.46 | Katty | http://paste2.org/p/1577062 <- updated |
| 20:01.49 | Katty | waves at Bezzie |
| 20:01.50 | Katty | oh |
| 20:01.53 | Katty | waves at beek |
| 20:03.02 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 20:03.09 | Katty | http://paste2.org/p/1577064 <- put more in there, any thoughts? |
| 20:04.55 | *** join/#asterisk blee (~blee@67.219.119.70.cfl.res.rr.com) |
| 20:05.12 | Katty | what's interesting is that i can hear the phone ringing |
| 20:05.19 | Katty | but i guess that's not technically audio |
| 20:05.39 | Katty | as soon as vm tries to play audio, it spazes |
| 20:05.48 | pabelanger | no, ringback is usually produced by the local phone |
| 20:06.03 | Katty | makes sense. |
| 20:06.16 | Katty | i can recieve a call |
| 20:06.22 | pabelanger | pb a SIP debug of your call |
| 20:06.26 | Katty | the phone rings, i answer it...and the polycom appears to get 'stuck' |
| 20:06.27 | Katty | kk |
| 20:08.24 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v012-217.mobile.uci.edu) |
| 20:09.10 | Katty | having that same problem, can't enable debug for a single phone |
| 20:09.14 | Katty | which is making this a disaster |
| 20:09.30 | jaytee | can't enable sip debug for a single peer? |
| 20:09.41 | Katty | no |
| 20:09.44 | Katty | let me try ip |
| 20:10.34 | Katty | won't work on ip either |
| 20:10.40 | Katty | does this dump into a log file? |
| 20:11.15 | jaytee | no, it dumps to the console |
| 20:11.38 | jaytee | but you could launch a remote console and use TEE to fork the console output to a file |
| 20:11.56 | jaytee | asterisk -rvvvvvvvv | tee logfilename |
| 20:12.21 | jaytee | weird that the sip peer won't work. won't work for me either and I know I've done it before though |
| 20:12.33 | chazzam | might be happier if you add a '-n' and maybe a '-T' to the asterisk command though |
| 20:12.42 | chazzam | to disable colors and enable timestamps |
| 20:12.48 | Katty | i think i got most of it |
| 20:12.55 | Katty | http://paste2.org/p/1577077 |
| 20:13.27 | Katty | 103 |
| 20:13.29 | Katty | what on earth is that |
| 20:13.53 | Katty | oh nm, |
| 20:15.32 | Katty | i see the retransmit attempts in there |
| 20:17.02 | *** join/#asterisk jeffspeff (~jeffspeff@173-11-144-149-houston.txt.hfc.comcastbusiness.net) |
| 20:20.12 | *** join/#asterisk jeffgus (~jeffgus@green.zimage.com) |
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| 20:27.12 | *** join/#asterisk _Raptor_ (raptorblue@andariel.informatik.uni-erlangen.de) |
| 20:28.42 | Katty | does reload include sip.conf? |
| 20:29.36 | Katty | jeesus christ on a bicycle |
| 20:29.44 | Katty | localnet issue |
| 20:29.45 | Katty | typo. |
| 20:30.02 | Katty | i thought i was losing my mojo. |
| 20:30.11 | Katty | instead i'm just getting careless. |
| 20:30.26 | Katty | jaytee: thanks for your help sweety. |
| 20:31.07 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 20:33.28 | jaytee | katty, no problem |
| 20:34.36 | jaytee | katty, found out I have to type either ip "address" or peer "peername" when doing either sip debug on a peer or rtp debug on an ip address. just the peername or ip address without the word peer or ip throws that error |
| 20:38.52 | *** join/#asterisk asilva (~andre@2801:88:1000:2::12) |
| 20:39.15 | asilva | anyone here knows a contact of Astricon organization personnel ? |
| 20:40.57 | MrTelephone | sip reload |
| 20:41.18 | leifmadsen | asilva: lking -at- digium -dot- com |
| 20:41.21 | leifmadsen | Lisa King |
| 20:41.29 | asilva | leifmadsen, thank you!! |
| 20:47.30 | MrTelephone | it costs like a million dollars to goto atricon :( |
| 20:49.41 | _Corey_ | The hotel people literally just called me trying to sell me suite upgrades for only $599 a night |
| 20:50.08 | tzanger | _Corey_: that's a deal! |
| 20:50.28 | _Corey_ | yeah, I told her I'd have a hard time putting that one on the expense report though |
| 20:50.44 | _Corey_ | vs. the $150 /night conference rate... :) |
| 20:52.18 | jaytee | bet they even charge for the canapes |
| 20:59.46 | *** join/#asterisk tamiel (~tamiel@65.204.112.138) |
| 21:00.50 | *** join/#asterisk adyn (~adyn@pdpc/supporter/active/adyn) |
| 21:03.15 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 21:03.44 | *** join/#asterisk lcat (~lcat@187.45.252.108) |
| 21:04.45 | leifmadsen | Asterisk 1.8.6.0-rc1 is now available. More information at http://www.asterisk.org/node/51669 |
| 21:05.02 | anonymouz666 | beautiful |
| 21:05.22 | anonymouz666 | let me see if the changelog is scary |
| 21:06.16 | leifmadsen | :) |
| 21:06.19 | *** join/#asterisk hardwire (~spencersr@12.17.188.86) |
| 21:06.37 | hardwire | is meetme an rtp aligning jitter buffer in essence? |
| 21:07.13 | hardwire | cause I have some phones that become massively delayed, and they often have jitter problems on their own (jb not turned on for them either) |
| 21:08.08 | malcolmd | there's a new receive-side jitterbuffer implementation in asterisk 10. useful if called before applications like confbridge or meetme |
| 21:08.19 | hardwire | danke.. I'll poke at it. |
| 21:08.30 | Qwell | malcolmd: We need a function that introduces jitter now. |
| 21:08.54 | hardwire | Qwell: thats what wirefilter is for. |
| 21:08.56 | malcolmd | good luck. i don't think i actually tested it against meetme, just against confbridge. so, if you run into something w/ meetme, please let us know. |
| 21:09.01 | Qwell | malcolmd: I'm still quite fond of the "jpah" "echocan". |
| 21:09.02 | hardwire | hugs wirefilter. |
| 21:09.36 | malcolmd | Qwell: wait, you still use your namesake? |
| 21:09.47 | Qwell | never! |
| 21:09.52 | Qwell | yes, sometimes |
| 21:10.08 | anonymouz666 | closes his eyes at changelog |
| 21:10.12 | anonymouz666 | :) |
| 21:10.49 | anonymouz666 | too many fixes on basic stuff, I don't know if it is good or bad :P |
| 21:11.07 | hardwire | malcolmd: at the moment this is 1.6. I may turn on JB for these peers anyways soon... however the delay is upwards of 5 seconds (no CPU spiked either). |
| 21:11.16 | hardwire | so hmm.. |
| 21:11.21 | malcolmd | wow |
| 21:11.24 | hardwire | indeed. |
| 21:11.33 | hardwire | I need to narrow it down to actual packet delay or not. |
| 21:11.43 | hardwire | Haven't been able to reproduce outside of meetme. |
| 21:12.03 | hardwire | Infact.. I've never reproduced. I'm mostly in it for sport. |
| 21:12.38 | malcolmd | ah, a trophy hunter, well good luck, good sir. we'll see you at the adventurers club after the safari |
| 21:15.12 | *** part/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
| 21:15.22 | anonymouz666 | two days fighting |
| 21:15.26 | anonymouz666 | heartbeat working with asterisk |
| 21:15.35 | MrTelephone | how did you do it? |
| 21:15.43 | malcolmd | with a baseball bat |
| 21:15.50 | MrTelephone | I was going to do the same but my adtran box doesn't support caller name |
| 21:16.00 | MrTelephone | I don't think i'll buy another piece of adtran equipment again |
| 21:16.12 | MrTelephone | junk |
| 21:18.01 | *** part/#asterisk neurosys (~neurosys@65.204.112.138) |
| 21:21.56 | *** join/#asterisk devmikey (~irc@96.46.249.230) |
| 21:29.03 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
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| 21:47.06 | pdtpatrick | Question .. i keep seeing this and need a bit of hand holding .. |
| 21:47.06 | pdtpatrick | [Aug 10 13:39:06] NOTICE[898]: chan_iax2.c:7662 socket_process: Host 10.1.134.2 failed to authenticate as dundi_exten |
| 21:47.51 | *** join/#asterisk corretico (~luis@201.197.141.74) |
| 21:48.00 | pdtpatrick | dundi_exten looks like this |
| 21:48.02 | pdtpatrick | http://pastebin.com/GPFF0L3t |
| 21:49.21 | *** join/#asterisk xpot-mobile (~james@166-70-100-198.ip.xmission.com) |
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| 22:05.29 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-tejyihmracmxdazy) |
| 22:19.01 | *** join/#asterisk cerberus_za (~coert@196-210-206-227.dynamic.isadsl.co.za) |
| 22:28.10 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 22:28.43 | pdtpatrick | where is dundi secret please.. |
| 22:33.45 | *** join/#asterisk jayson_r (~jayson@cpe-071-076-046-081.sc.res.rr.com) |
| 22:36.36 | *** join/#asterisk jayson_r (~jayson@cpe-071-076-046-081.sc.res.rr.com) |
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| 23:00.43 | *** join/#asterisk justdave (~dave@unaffiliated/justdave) |
| 23:13.08 | justdave | are there asterisk packages for CentOS 6 yet? |
| 23:13.31 | justdave | following the yum setup directions on asterisk.org, yum is getting a 404 trying to download a package list |
| 23:13.48 | *** join/#asterisk kn0x (~pinochle@74.63.68.42) |
| 23:13.54 | kn0x | do the sangoma wanpipe modules replace ztdummy/dahdi_dummy? |
| 23:13.59 | kn0x | if I have a quad pri /A104d |
| 23:15.12 | justdave | I know you need the dahdi source present to compile wanpipe, and asterisk only sees dahdi |
| 23:15.30 | justdave | don't know if it replaces them or if it just tricks them into thinking the wanpipe drivers are dahdi hardware |
| 23:19.56 | pabelanger | justdave: I don't believe so, but on somebodies list |
| 23:20.40 | justdave | guess I should wipe this new machine I just built and put CentOS 5 on it for now then |
| 23:21.00 | justdave | or install from source again like I used to :) |
| 23:22.50 | kn0x | wish i could wipe this machine.. |
| 23:23.02 | X-Rob | justdave, there are. |
| 23:23.30 | X-Rob | justdave : https://github.com/FreePBX/HiPBX/tree/master/rpms |
| 23:25.33 | pdtpatrick | Question .. im seeing Host ip failed to authenticate as dundi_exten .. can anyone help ? |
| 23:25.56 | WIMPy | dundi_exten? |
| 23:26.11 | WIMPy | "HiPBX" sounds dangerous. |
| 23:27.37 | pdtpatrick | WIMPy, here's the iax file |
| 23:27.39 | pdtpatrick | http://pastebin.com/13FLe2FQ |
| 23:28.05 | X-Rob | WIMPy, why dangerous? |
| 23:28.30 | WIMPy | X-Rob: Someone might decide it could be confused with "HiPath". |
| 23:29.02 | X-Rob | Lucky I'm in Australia then, isn't it, where we don't have retarded trademark laws like that. |
| 23:29.03 | pdtpatrick | WIMPy, and then on the other server.. it says .. Call rejected by ip: no authority found |
| 23:29.38 | WIMPy | pdtpatrick: What are you doing to get that error? But I'm sure, you don;t want to set qualify and probably not trink, either. |
| 23:30.42 | WIMPy | pdtpatrick: Do you have the secret correctly in your dundi announce? |
| 23:31.53 | pdtpatrick | one of our boxes went down and i rebuilt a new on .. i configured vxml and brought asterisk back up |
| 23:32.03 | pdtpatrick | now the box won't register with the others |
| 23:32.31 | pdtpatrick | and im showing the old server still |
| 23:33.09 | WIMPy | Did you change the ID? |
| 23:33.37 | pdtpatrick | i kept the same id |
| 23:33.40 | pdtpatrick | should i change it ? |
| 23:33.52 | pdtpatrick | this is the old id |
| 23:33.53 | pdtpatrick | 00:16:3e:7b:7f:14 10.1.132.30 (S) Symmetric Unavail OK (1 ms) |
| 23:34.06 | *** join/#asterisk Tech_Travis (~Travis@cpe-76-168-191-127.socal.res.rr.com) |
| 23:34.28 | pdtpatrick | you're saying i should change entityid to something else? |
| 23:34.30 | pdtpatrick | entityid=00:16:3E:7B:7F:14 |
| 23:34.37 | WIMPy | It doesn't matter as long as it's the same on all machines. |
| 23:35.16 | WIMPy | Not the same ID, but the id and the corresponding peer, that is. |
| 23:35.18 | pdtpatrick | oh.. how can i delete the old one .. since its mapping to the wrong ip ? |
| 23:35.52 | pdtpatrick | i ran dundi show peers and it shows the old server ip |
| 23:36.01 | WIMPy | In your dundi.conf. |
| 23:36.34 | pdtpatrick | they are using FQDN instead of ip |
| 23:36.49 | pdtpatrick | show i flush dundi peers or is that a no no ? |
| 23:37.37 | WIMPy | Ok, so the IP for that fqdn has changed and Asterisk hasn't noticed, yet? |
| 23:37.43 | pdtpatrick | yup |
| 23:38.01 | WIMPy | Sounds like a dnsmgr thing. |
| 23:38.03 | pdtpatrick | and so when the calls come in .. it just says.. [Aug 10 16:16:56] NOTICE[898]: chan_iax2.c:7662 socket_process: Host 10.1.134.2 failed to authenticate as dundi_exten |
| 23:41.55 | pdtpatrick | WIMPy, dnsmgr is disabled |
| 23:41.55 | *** join/#asterisk corretico (~luis@201.197.141.74) |
| 23:44.45 | WIMPy | pdtpatrick: Ah, I think it caches forever then. |
| 23:45.17 | WIMPy | At least that was the impression I had, when I tried. |
| 23:45.38 | pdtpatrick | interesting |
| 23:45.56 | pdtpatrick | what's the best way to have it detect new peer? |
| 23:46.11 | WIMPy | reload? |
| 23:46.13 | carrar | <PROTECTED> |
| 23:46.16 | carrar | nice |
| 23:46.26 | carrar | wonders where that doc is |
| 23:46.56 | carrar | <PROTECTED> |
| 23:46.59 | carrar | err |
| 23:46.59 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 23:47.04 | carrar | <PROTECTED> |
| 23:47.08 | carrar | yet the column exists |
| 23:48.17 | leifmadsen | that's why I like using res_adaptive_odbc |
| 23:48.22 | leifmadsen | it does most/all of that for you |
| 23:48.41 | carrar | was trying to avoid using additional layers |
| 23:49.26 | pdtpatrick | WIMPy, dnsmgr is disabled on all systems. |
| 23:50.41 | leifmadsen | ya, but ODBC is much better supported |
| 23:50.44 | carrar | probably have to use obdc, psql real time psql drive seems broken or I am doign something wrong |
| 23:50.46 | leifmadsen | you'll find less issues with it |
| 23:51.09 | carrar | the cdr part works great directly wih pgsql |
| 23:53.35 | *** join/#asterisk Cesare (~Adium@creati59.lnk.telstra.net) |
| 23:55.31 | pdtpatrick | WIMPy, got it pass through.. u had to change the entityid, flush dundi and then reload. |
| 23:55.36 | pdtpatrick | at least that's what happened for me |
| 23:55.41 | pdtpatrick | now i gotta tackle another problem |
| 23:57.15 | pdtpatrick | Question for anyone .. any ideas on troubleshooting this: http://pastebin.com/rJ0ydFzi |
| 23:58.48 | pabelanger | pdtpatrick: where did you get the Vxml application? Asterisk does not ship with that |