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| 02:08.48 | dijib | ive got a bitshaping issue, 5060 inbound is choppy. i have scheduling setup and this is the report of such. http://pastebin.com/2rPX3BQq |
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| 02:34.35 | p3nguin | How would you determine if 5060 is "choppy" or not? |
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| 03:24.58 | raden | lol |
| 03:25.11 | p3nguin | *shrug* |
| 03:33.27 | philipp64|laptop | dijib: it's not 5060 that matters, it's your SDP/RTP media stream. |
| 03:34.17 | philipp64|laptop | you could use tcpdump for a capture of the media stream and then look at the packet timestamps, maybe right a script that calculates the average inter-packet spacing, as well as standard deviation, etc. |
| 03:34.21 | philipp64|laptop | that's what I would do. |
| 03:34.42 | philipp64|laptop | but your handsets should have enough elastic buffer to accommodate most jitter. |
| 03:35.08 | philipp64|laptop | if not, try decreasing your sampling interval from 30ms to 20ms. |
| 03:35.32 | philipp64|laptop | if you still have audio problems, that's likely packet loss or delay. not jitter. |
| 03:36.40 | p3nguin | He's not even shaping his RTP. |
| 03:36.49 | p3nguin | At least not that I can determine. |
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| 04:50.25 | ThinkGNU- | hello asterisk channel |
| 04:51.17 | ThinkGNU- | I'm learning asterisk right now and I have a couple of questions. Maybe someone could answer them and put my mind at ease |
| 04:54.06 | ThinkGNU- | does anyone mind answering a couple of my questions? |
| 05:00.23 | [TK]D-Fender | ~ask |
| 05:00.23 | 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. |
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| 05:05.17 | ThinkGNU- | sorry, didn't see the status icon in my notification panel that there were updates to the channel |
| 05:06.07 | ThinkGNU- | I am going to have 23 phones and maybe a max of 10-12 concurrent calls. Will a 3Ghz AMD64 be enough processing power using ulaw as the codec? |
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| 05:09.18 | [TK]D-Fender | ThinkGNU-: more than enough |
| 05:10.09 | ThinkGNU- | for some reason I'm having doubts. don't know why |
| 05:10.40 | ThinkGNU- | the phone system is such a big thing at our business that if it screws up, I could have some problems |
| 05:11.40 | ThinkGNU- | what's a good way to stress test the system without actually having a bunch of users make calls? |
| 05:11.46 | [TK]D-Fender | Fear leads to anger ..... anger leads to hate .... anger leads to pain ... pain leads to the Cheescake Factory ... Cheesecake leads to cardiac arrest.... |
| 05:12.13 | [TK]D-Fender | ThinkGNU-: Trust me your needs are SUB -petty |
| 05:12.27 | [TK]D-Fender | ThinkGNU-: My watch could handle that load.... and it's ANALOG |
| 05:12.36 | ThinkGNU- | lol |
| 05:12.49 | ThinkGNU- | well, that's good to know |
| 05:13.06 | ThinkGNU- | and, you can use your watch to find north...since it's analog |
| 05:14.02 | ThinkGNU- | and cheesecake...that sounds good |
| 05:14.25 | ThinkGNU- | SIP is something I'm still coming to grips with too |
| 05:15.33 | ThinkGNU- | I have had a few issues with no audio and such so I'm still learning the nuances of SIP |
| 05:15.55 | [TK]D-Fender | ~sipnat |
| 05:15.55 | 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, directmedia, externhost or externaddr, and localnet. |
| 05:16.00 | [TK]D-Fender | ~nat |
| 05:16.01 | infobot | methinks nat is Network Address Translation Usable in Asterisk sip.conf file with externip, localnet, and localmask setup properly. See docs. |
| 05:16.18 | [TK]D-Fender | in 1.6+ "canreinvite" has been replaced by "directmedia" |
| 05:16.26 | [TK]D-Fender | those are the rules where NAT is involved |
| 05:16.33 | [TK]D-Fender | and the #1 common issue |
| 05:16.34 | ThinkGNU- | oh, I didn't know about directmedia |
| 05:16.53 | ThinkGNU- | i've set externip and localnet |
| 05:17.01 | [TK]D-Fender | nat=yes <- as well |
| 05:17.03 | ThinkGNU- | I'll set localmask and directmedia |
| 05:17.07 | ThinkGNU- | I have net=yes |
| 05:17.24 | [TK]D-Fender | under [general], however ITSP entries should almost always be "nat=no" for those peers |
| 05:17.54 | ThinkGNU- | that's internet telephone service providers right? |
| 05:18.00 | [TK]D-Fender | Correct |
| 05:18.10 | ThinkGNU- | what tricks are they doing to avoid that? |
| 05:20.53 | [TK]D-Fender | They have public IP's, and may have separate media & signalling servers, etc |
| 05:21.01 | ThinkGNU- | ah |
| 05:21.03 | [TK]D-Fender | so you have to trust them to handle their own business. |
| 05:21.23 | [TK]D-Fender | So they are "nat=no". YOU are another matter, as are other peers you may have. |
| 05:21.59 | ThinkGNU- | I've been messing with a few of the free VoIP providers such as ekiga.net |
| 05:22.21 | ThinkGNU- | I have it working to some degree. I've got sound going one direction |
| 05:22.31 | ThinkGNU- | I'll tweak those settings that I missed though |
| 05:23.40 | [TK]D-Fender | Soundsing like you're going to do just fine.... |
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| 05:24.50 | ThinkGNU- | well, I'll be popping in here frequently I think. Are you in often [TK]D-Fender? |
| 05:25.00 | [TK]D-Fender | yup |
| 05:25.40 | ThinkGNU- | alright |
| 05:25.55 | ThinkGNU- | I'll catch you later then |
| 05:25.59 | ThinkGNU- | thanks for the help! |
| 05:26.00 | [TK]D-Fender | Sounds good... |
| 05:26.05 | [TK]D-Fender | You're welcome |
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| 05:38.04 | din3sh | tgif |
| 05:38.06 | din3sh | :D |
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| 05:58.03 | raden | p3nguin, u around |
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| 06:36.37 | schmidts | good morning |
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| 07:01.06 | din3sh | mrning |
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| 08:12.37 | arekm | hi, is it possible to limit number of concurrent connections in a context? assume I have context to-gsm and to-outside - both use the same isdn connection. How can I limit number of concurrent connections in to-gsm to a 4 for example? |
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| 08:18.57 | kaldemar | arekm: use GROUP functions. "core show functions like GROUP" |
| 08:20.59 | din3sh | kaldemar hello |
| 08:21.13 | din3sh | am having sip retransmit errors |
| 08:21.15 | din3sh | http://paste2.org/p/1997268 |
| 08:21.27 | The_Ball | Anybody know of a open source or free-as-in-beer web voip client? |
| 08:21.28 | din3sh | any idea why? |
| 08:26.30 | kaldemar | din3sh: you're not getting a response back. usual cause for those is invalid NAT settings. asterisk is sending to 192.168.2.18:5060, is that address correct? |
| 08:26.57 | din3sh | yes it is the phone which initiated the call, but cancellled |
| 08:27.16 | din3sh | the call |
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| 08:27.26 | kaldemar | din3sh: is the phone behind a NAT? |
| 08:27.29 | din3sh | i have no NAT settings |
| 08:27.40 | din3sh | all on same subnet/network |
| 08:28.03 | kaldemar | doesn't look like it. :) |
| 08:28.18 | kaldemar | you have 192.168.2.18 and 172.17.0.10 in there. |
| 08:29.50 | arekm | kaldemar: seems working, thanks ! |
| 08:32.47 | din3sh | kaldemar even phones on 172... have same problem |
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| 08:40.29 | kaldemar | din3sh: show your sip.conf and a CLI output of a call with sip debug enabled. |
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| 09:04.23 | krotos | hi all |
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| 09:33.50 | din3sh | sorry kaldemar, caught up with another issue meanwhile |
| 09:34.12 | din3sh | queue members are not ringing, asterisk saying :WARNING[8175] app_queue.c: Realtime field uniqueid is empty for member |
| 09:38.28 | kaldemar | surely that's not all asterisk says. |
| 09:44.59 | din3sh | hehe |
| 09:45.36 | din3sh | resolved |
| 09:50.45 | din3sh | kaldemar: DEBUG[8354] res_rtp_asterisk.c: Setting the marker bit due to a source update |
| 09:50.51 | din3sh | what would this mean? |
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| 10:00.51 | FlashDeluxe | hi! is there a current print-to-fax-tool for linux clients (e.g kubuntu/ubuntu etc)? In former times there was kdeprintfax, but it doenst exist anymore.. :( |
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| 10:26.44 | catphish | is there any reason why allowing incoming calls from 0.0.0.0/0 is a bad idea? |
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| 10:30.12 | ijpalmer | Hi I have a problem where Asterisk doesn't seem to send out a chan_end event in the CEL data. I have noticed this only in channel barge at the moment. I'm using * 1.8.5 |
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| 10:37.13 | kaldemar | catphish: if you have proper device names, secrets, guest calls are not allowed and your dialplan is not full of holes, not really. |
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| 10:47.49 | din3sh | you have to define th events you want to log in cel.conf |
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| 10:49.21 | ijpalmer | din3sh: Thanks for replying. I have named the events and normally I get the chan_end event. Problem is on one occasion I didn't get this event which we rely for an app we have plugged in. This has so far only occurred when the channel barge had ended |
| 10:50.37 | din3sh | apps=all? |
| 10:54.03 | ijpalmer | I have apps=dial,park |
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| 10:56.15 | din3sh | dahdibarge / |
| 10:56.17 | din3sh | ? |
| 10:56.27 | din3sh | thats an application i guess? |
| 10:56.33 | din3sh | try adding it to apps= |
| 10:57.00 | ijpalmer | Thing is in all instances of channel barge it seems to work, it was just this one occasion |
| 10:57.27 | din3sh | ok |
| 10:57.54 | din3sh | am trying to install iaxmodem, how do we add respawn in inittab centos 6.2? |
| 10:58.13 | ijpalmer | what I'm worried about is if there is a bug in Asterisk where it sometimes doesn't send out this event |
| 10:59.37 | din3sh | there's no bug in asterisk |
| 10:59.37 | din3sh | :p |
| 11:00.39 | ijpalmer | fair enough lol. I'll keep monitoring the situation for more instances, I'll also add the channel barge app into apps= of cel.conf. Cheers for your help |
| 11:01.10 | din3sh | ;) |
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| 12:46.33 | dug429 | If a UA receives a 401 with a stale=true response, is the expected result of that for the UA to register again with a new call-id? Or is it common for the call-id to be the same and the UA to expect a new nonce? |
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| 12:56.04 | [koss] | what asterisk distribution has the easiest auto provisioning,, specifically for polycom |
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| 13:01.28 | [TK]D-Fender | [koss], Anything running FreePBX has a well-maintained 3rd party provisioning module. |
| 13:11.26 | din3sh | yep |
| 13:11.49 | din3sh | elastix running on freepbx engine has pretty neat endpoint module |
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| 13:18.13 | [TK]D-Fender | Elastix runs a forked embedded trashed-up version of FreePBX that is not supported by the community. |
| 13:20.26 | [koss] | there used to be a distro voiceroute/druid that had awesome auto polycom provisioning |
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| 13:20.28 | catphish | kaldemar: what do you mean by "guest calls not allowed"? that was what i was proposing allowing |
| 13:23.16 | bowzak | where should i start when trying to troubleshoot pstn calling? drivers seem to be installed fine.. no errors when i make the call.. but nothing rings. |
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| 13:25.21 | sofh | Hello all |
| 13:26.11 | sofh | i've some questions regarding asterisk , basic most is : is it possible that asterisk fwd each connecting user to some parent sip proxy ? |
| 13:27.54 | [TK]D-Fender | sofh, Based on..? |
| 13:28.20 | sofh | i should clear more |
| 13:28.29 | [TK]D-Fender | [be] |
| 13:29.01 | sofh | what i want , sip user tries to register to my asterisk box , but i dont want asterisk to handle its registaration neither routing , it should just pass it to some other sip server to register |
| 13:29.16 | sofh | same way media should pass from user to asterisk and then to that proxy server where its registered |
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| 13:29.32 | [TK]D-Fender | bowzak, start by describing what you're actually using, showing configs, and showing status dumps that show that everything appears the way it should for calls to have a chance of coming in... |
| 13:30.13 | [TK]D-Fender | sofh, * does not do this. You seem to be wanting to use it as a proxy. It is not a proxy. |
| 13:30.53 | bulkorok_afk | sofh: take a look at kamailio with mediaproxy module |
| 13:31.18 | sofh | actually i want to have benefits of IAX trunking ,but want to have my routing,billing from other switch |
| 13:31.26 | sofh | seems not possible what i want |
| 13:31.33 | bulkorok | why IAX? |
| 13:31.40 | sofh | to save bandwidth |
| 13:31.51 | sofh | its saves about 60% of bandwidth as per to my experience |
| 13:32.24 | [TK]D-Fender | sofh, Correct, it isn't possible |
| 13:32.25 | catphish | why would iax save bandwidth? |
| 13:32.47 | [TK]D-Fender | catphish, IAX2 trunk mode. |
| 13:32.59 | sofh | catphish, well you should asterisk developers ,why they made IAX trunking to share overhead packets |
| 13:33.09 | sofh | but it realy works awesome |
| 13:33.20 | [TK]D-Fender | you just completely out a word there.... |
| 13:33.38 | catphish | i guess sip has a fair overhead for call setup |
| 13:33.39 | [TK]D-Fender | And it isn't "overhead packets" |
| 13:33.44 | catphish | but it doesn't compare to the data itself |
| 13:33.51 | [TK]D-Fender | And this has nothing to do with "SIP" or "call setup" |
| 13:34.07 | sofh | when its IAX how SIP can be involved |
| 13:34.13 | sofh | its IAX offcourse |
| 13:35.10 | catphish | i'm confused, what were we comparing iax to? and for what purpose is it 60% lighter? |
| 13:35.53 | bowzak | ... using freepbx on Rhino box with Rhino PCI 8FXO card. The local line is in Iraq, but I see that the drivers are using FXS Kewlstart |
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| 13:36.18 | catphish | or are you just talking about iax trunk vs multiple sockets? |
| 13:36.26 | bowzak | when i dial out.. i get "called DAHDI" and DAHDI answered |
| 13:37.13 | [TK]D-Fender | sofh, IAX2 is not a routed protocol. Sounds like you should simply PORT FORWARD to the other server |
| 13:37.40 | [TK]D-Fender | bowzak, PASTEBIN the bits I have asked about |
| 13:37.41 | [TK]D-Fender | ~pb |
| 13:37.41 | infobot | [~pb] 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.us , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ ; or install pastebinit with yum or aptitude. |
| 13:37.42 | [TK]D-Fender | ^^^ |
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| 13:54.01 | jaytee | I want to upgrade a server from asterisk 1.6.2.x (compiled) with DAHDI Complete 2.4.1 to Asterisk 1.8.11.1 RPM with DAHDI 2.6.1 RPM If I back up my configs in /etc/asterisk and /etc/dahdi and run the RPMs will that work ok or should I delete the files in /usr/lib/asterisk/modules first? |
| 13:54.38 | [TK]D-Fender | trash modules.... |
| 13:54.45 | jaytee | nods |
| 13:57.09 | jaytee | including the dahdi modules in /lib/modules? |
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| 14:06.02 | [TK]D-Fender | jaytee, DAHDI will overwrite and "strays" don't tend to be an issue in my experience |
| 14:06.18 | [TK]D-Fender | jaytee, * modules however... get an extra in there and autoload will FUBAR you |
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| 14:10.49 | jaytee | [TK]D-Fender, thanks. I'd read about problems upgrading with old modules still in /usr/lib/asterisk/modules causing problems but wasn't sure about DAHDI |
| 14:13.44 | jaytee | I'm hoping that the new Cisco switches with QoS and this upgrade will fix my problem. If I make a call from one of the Polycom 331 phones to another Polycom on the inside network or dialing out to my cell I get no ringing and when the other phone answers I get no audio. If I hangup and dial again the second call gets ringing indication and two way audio works fine. |
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| 15:09.56 | ryduh | hello. trying to figure out if a debug msg means what I think it means |
| 15:10.03 | ryduh | "Peer doesn't provide T.38 UDPTL" |
| 15:10.38 | ryduh | my sip provider doesn't provide t.38 support if that shows up right? |
| 15:11.41 | ryduh | i think it is, since i also see this |
| 15:11.41 | ryduh | Got T.38 Re-invite without audio. Keeping RTP active during T.38 session. Callid 72f61c1003bbd6633f16c6a72fb46885@sip.flowroute.com |
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| 15:34.58 | *** join/#asterisk paolosupino (~paolo-sup@net-2-38-88-100.cust.dsl.vodafone.it) |
| 15:35.07 | paolosupino | hi |
| 15:35.42 | paolosupino | Question: Is it possible to suspend e certain internal extension voicemail? |
| 15:37.50 | *** part/#asterisk clintc (~clintc@n128-227-12-58.xlate.ufl.edu) |
| 15:38.21 | [TK]D-Fender | paolosupino, It's your dialplan... if does whatever you tell it to do. |
| 15:38.27 | [TK]D-Fender | it* |
| 15:40.05 | *** join/#asterisk doug (doug@breakout.horph.com) |
| 15:40.10 | doug | hullo |
| 15:40.39 | doug | i'm hoping to build a nice menu |
| 15:41.13 | doug | when you call, it says "thanks for calling my debt resolution line. if you are not calling on behalf of a collection agency, you have probably called this number in error." |
| 15:41.28 | doug | "please choose from the following menu" |
| 15:41.48 | doug | "please press 1 if you want to submit a payment plan for consideration" |
| 15:42.13 | doug | "please press 2 if you want to read from a script that you hope will inspire me to pay the entire balance immediate" |
| 15:42.24 | doug | so i can hand-hack this into extensions.conf |
| 15:42.35 | [TK]D-Fender | Correct |
| 15:42.36 | doug | but i figured someone probably has come up with a nice, easy system for doing this by now... |
| 15:42.40 | doug | any recommendations? |
| 15:42.52 | [TK]D-Fender | And no... this is your menu, with your options, and your own recordings |
| 15:42.55 | aberrios | doug if you're good with dialplan it'll take an hour tops |
| 15:43.06 | [TK]D-Fender | s/hour/10minutes |
| 15:43.13 | aberrios | if you're awesome with dialplan 10 mins |
| 15:43.15 | doug | yeah, my conf is already a giant mess 'o spaghetti |
| 15:43.15 | aberrios | =D |
| 15:43.38 | doug | what i'm really hoping for is something that's already built in all the nice little convenience features |
| 15:43.39 | [TK]D-Fender | No, if you're OK in dialplan and are a moderately stable person to do the recording and don't need 20 takes each |
| 15:43.55 | [TK]D-Fender | doug, Nope. |
| 15:43.55 | doug | like * for going back, # to repeat, etc. |
| 15:43.59 | [TK]D-Fender | doug, It's all you. |
| 15:44.04 | carrar | If you're not stable, please video record yourself doing it |
| 15:44.16 | carrar | might be funny |
| 15:44.29 | doug | would have been funny when i was starting out |
| 15:44.31 | [TK]D-Fender | carrar, When two epileptics make a deal ... do they shake on it? |
| 15:44.34 | aberrios | putting system recordings on at christmas when everyone is half cut makes for good recordings =D |
| 15:44.40 | carrar | heh |
| 15:44.59 | doug | when i was like "okay! this is a popular telephony system, and i'm sure intelligent people have been involved in designing the configuration language..." |
| 15:45.24 | doug | although i suppose that's what gave rise to ael |
| 15:45.52 | doug | although i'm not completely conviced that ael is an improvement |
| 15:46.36 | doug | can i mix&match extensions.ael and extensions.conf? |
| 15:47.48 | [TK]D-Fender | doug, "* to repeat" is ONE line of dialplan. |
| 15:48.15 | [TK]D-Fender | doug, And you are blowing this way out of proportion. |
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| 15:49.26 | doug | one line of dialplan for each menu, sure. |
| 15:49.55 | doug | but one line i'd rather not worry about if i don't have to. |
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| 15:51.10 | angryuser | Hello when i dump asterisk traffic with tcpdump i have 100% of voip traffic out of seq, asterisk 10.2.1 any ideas ? |
| 15:51.12 | DarthExpeditor | I'm looking for a hardware recommendation for a 50 SIP handset system using both SIP and Analog lines for termination with MOH, VoiceMail, DIDs, IVR, and Call Queues. |
| 15:51.17 | angryuser | Bug in wireshark ? |
| 15:51.26 | [TK]D-Fender | doug, If you're too lazy for writing a few lines of dialplan, then perhaps PBX administrtion isn't for you.... |
| 15:52.28 | shinao1 | hi guys please i have a problem with a dundi+iax2 trunk network.. im trying to do a dundi lookup and it complains of LAGGED connections and wont search. Is there a way to make dundi/iax2 ignore the latencies, and what is the maximum latency they'll suffer before breaking up? |
| 15:54.50 | doug | hm, maybe you just don't have the experience with writing useful systems and therefore lack the perspective of someone who has.. |
| 15:54.50 | aberrios | thats just lazy :) jfdi |
| 15:54.54 | doug | every line is a liability. |
| 15:55.26 | *** join/#asterisk nighty- (~nighty@static-68-179-124-161.ptr.terago.net) |
| 15:55.33 | [TK]D-Fender | You're right. Divest yourself of all of the lines! |
| 15:55.48 | doug | as many as possible. |
| 15:56.00 | doug | which sometimes is all of them... |
| 15:56.05 | doug | although not always. |
| 15:56.42 | doug | in any case, it's never "divest," more "delegate." |
| 15:56.55 | doug | and while that carries risk of its own, it's often (in my experience) a better route. |
| 15:57.30 | doug | especially when building something for which there are years of history and example in multitude |
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| 16:03.26 | *** join/#asterisk jeffspeff (~jeffspeff@12.49.160.131) |
| 16:04.21 | p3nguin | catphish: If you are going to allow guest calls, be sure to set up good dial plan to manage them, and not let them access your toll extensions. |
| 16:04.37 | p3nguin | catphish: And IAX2 saves bandwidth because it does trunking. SIP does not have trunking. |
| 16:05.31 | [TK]D-Fender | http://www.quickmeme.com/meme/3p039w/ |
| 16:06.01 | p3nguin | raden: I'm here now. |
| 16:07.49 | doug | cool service. i've often felt the loss of the tools i had as a playground bully. |
| 16:11.56 | *** join/#asterisk emora (~emora@213.236.11.83) |
| 16:13.26 | paolosupino | [TK]D-Fender: And how do I disable voicemail for a specific internal extension? |
| 16:13.51 | [TK]D-Fender | paolosupino, it's your dialplan.. you don't disable voicemail.. you simply don't call it |
| 16:14.04 | *** join/#asterisk Russ (foobar@ip68-106-254-4.ph.ph.cox.net) |
| 16:15.45 | paolosupino | [TK]D-Fender: that wasn't my intention. My intention was that I don't want callers to be able to leave messages for a certain internal extension… but I also want to do it on as temporary thing, not a permanent one. |
| 16:16.05 | p3nguin | Then don't call VoiceMail() in the dial plan. |
| 16:16.16 | p3nguin | Extensions only have what you write. |
| 16:16.21 | [TK]D-Fender | paolosupino, Then add some logic for shoosing whether to do action A vs B. |
| 16:16.22 | p3nguin | Phones are not extensions. |
| 16:16.31 | [TK]D-Fender | paolosupino, "core show application gotoif" <- |
| 16:17.39 | [TK]D-Fender | choosing* |
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| 16:29.02 | catphish | p3nguin: that's fine, my dialplans are extremely carefully managed |
| 16:29.15 | catphish | p3nguin: and what is trunking? |
| 16:29.49 | catphish | i don't fully understand why sending all signals down one socket should be that much more efficient |
| 16:30.07 | catphish | except for avoiding the unnecessary chatiness of sip |
| 16:30.17 | [TK]D-Fender | Has nothing to do with SIP |
| 16:30.31 | catphish | [TK]D-Fender: please read the conversation |
| 16:30.37 | [TK]D-Fender | I did |
| 16:30.41 | p3nguin | Are you familiar with dot1q? |
| 16:30.42 | [TK]D-Fender | And commented earlier |
| 16:30.43 | paolosupino | [TK]D-Fender: OK so I have to build it into the dial plan logic… Can you help me with building the logic? |
| 16:31.04 | catphish | [TK]D-Fender: did you? |
| 16:31.16 | [TK]D-Fender | paolosupino, I already gave you the key command... you need to come up with the means of deciding what the answer is to "A or B?" |
| 16:31.25 | [TK]D-Fender | catphish, I did. |
| 16:31.40 | catphish | p3nguin: if you're talking to me then yes |
| 16:32.08 | [TK]D-Fender | catphish, Bandwidth savigs are from RTP, not SIP |
| 16:32.14 | [TK]D-Fender | SIP sets up calls, it is not the VOICE |
| 16:32.18 | catphish | [TK]D-Fender: i don't see you say anything on the subject since p3nguin said "IAX2 saves bandwidth because it does trunking. SIP does not have trunking." |
| 16:32.37 | [TK]D-Fender | RTP is encoded in UDP packets with 20kbit of WASTE EACH |
| 16:32.46 | [TK]D-Fender | RT is not SIP <- |
| 16:32.49 | [TK]D-Fender | Know your protocols |
| 16:33.00 | catphish | i know them well its ok |
| 16:33.33 | [TK]D-Fender | So with ULAW, 20 calls over RTP have 64kbps of data, and 20kbps of UDP header. EACH |
| 16:33.44 | catphish | err |
| 16:34.15 | catphish | ok i get that, does IAX allow multiple streams to share a packet? |
| 16:34.16 | [TK]D-Fender | IAX Trunking combines the voice data of MULTIPLE calls over a SINGLE packet. so say 5 x 64kbps worth of calls with only 20 kbps of waste instead of 100 < |
| 16:34.46 | catphish | when talking about packet overhead, try to use bytes, not bps, but i understand now :) |
| 16:34.52 | talntid | wow. I didn't knw that |
| 16:34.53 | [TK]D-Fender | Now scale this with a lighter codec like GSM610 : 13kbps data VS 20kbps overhead for UDP |
| 16:35.14 | catphish | yeah, that makes a whole bunch of sense for large volumes of low bandwidth codec calls |
| 16:35.20 | catphish | very cool |
| 16:35.53 | [TK]D-Fender | catphish, "bytes" alone doesn't cover it... voice is a measure of time. it's "bytes out of how many over what period of time" |
| 16:36.33 | [TK]D-Fender | packetizations rate is an important factor |
| 16:36.48 | [TK]D-Fender | This is based on 20ms packets. |
| 16:37.07 | *** part/#asterisk drudge` (~drudge@unaffiliated/drudge/x-837452) |
| 16:37.12 | [TK]D-Fender | catphish, Your earlier error I corrected was in the use of the term "overhead packets". |
| 16:37.20 | coppice | 10s packets does more for you than bundling streams |
| 16:37.22 | catphish | of course, if you fill 1500 byte frames with one call you're not going to have much of an issue |
| 16:37.29 | catphish | apart from the quality :) |
| 16:37.32 | [TK]D-Fender | catphish, There is no entire "packet" that is "overhead". it is the HEADER of EACH packet. |
| 16:37.50 | p3nguin | packet overhead vs overhead packet? |
| 16:37.55 | [TK]D-Fender | coppice, Yeah it does things like "awkward silence" in PL scenarios ;) |
| 16:38.12 | coppice | "over" |
| 16:38.26 | [TK]D-Fender | p3nguin, talks does funny Yoda, hmmmMMM!??!?!? |
| 16:38.28 | [TK]D-Fender | ;) |
| 16:38.43 | p3nguin | Very silly you are. |
| 16:38.48 | catphish | [TK]D-Fender: i mean per-packet overhead (thats how i usually express transmisison overhead, rather than as a bps figure) |
| 16:39.02 | catphish | ie x byte packet headers |
| 16:39.20 | [TK]D-Fender | catphish, well you seem to have gotten it now.... |
| 16:39.20 | catphish | but as i said, it makes sense now |
| 16:39.36 | catphish | we've been barking up the same tree for some time now :) |
| 16:40.02 | catphish | though you seemed very offended that i was comparing iax with sip |
| 16:40.09 | [TK]D-Fender | catphish, Actually that'd be "flogging the deceased equine" ;) |
| 16:40.09 | catphish | instread of sip+rtp :) |
| 16:40.12 | catphish | lol |
| 16:40.18 | catphish | stoopid horse |
| 16:40.40 | coppice | digium don't seem to be supporting IAX on their new kit, so I assume they consider it dead |
| 16:40.50 | [TK]D-Fender | coppice, Which new kit? |
| 16:41.01 | p3nguin | We haven't used IAX in how many years now? |
| 16:41.04 | coppice | the phones and gateway boxes |
| 16:41.23 | catphish | oo its home time now |
| 16:41.26 | catphish | have a nice weekends |
| 16:41.32 | [TK]D-Fender | coppice, On phones I understand.. it's rather pointless there.. on a larger gateway ... well now THAT would be dumb :) |
| 16:42.06 | [TK]D-Fender | coppice, If their new T1/E1/J1 boxes didn't offer it... then I'd say there was a serious failing... |
| 16:42.23 | coppice | there have been several drafts for bundling in RTP, but they seem to fade away each time |
| 16:42.26 | p3nguin | |
| 16:42.41 | coppice | the blurb for the digium gateway boxes only mentions SIP |
| 16:42.51 | [TK]D-Fender | coppice, That irks me... |
| 16:42.52 | Kobaz | i think it supports iax |
| 16:43.02 | Kobaz | i remember seeing all kinds of stuff listed on the data sheet |
| 16:43.08 | [TK]D-Fender | Checking now... |
| 16:45.22 | [TK]D-Fender | \Yup... no mention of IAx2 |
| 16:45.34 | Kobaz | mmm |
| 16:45.35 | ruben23 | hi guys please help me with this error any idea with my asterisk -----> http://pastebin.com/6YcrCyZy |
| 16:46.00 | Kobaz | corrupt file? |
| 16:46.47 | [TK]D-Fender | IAX2 as never really designed for phones and not even profitable BW wise unless you're on 3/n-way. So understandable the phones don't have it. Their ateways however can push upwards of 60 calls.... this is VERY significant |
| 16:48.15 | Kobaz | the packet savings is pretty minimal, especially since 99% of the time i would think they would be on a lan |
| 16:49.34 | Kobaz | it's great to save what you can on the WAN, but for LAN stuff would it really matter? |
| 16:49.46 | [TK]D-Fender | Kobaz, at G.729 it would save 2/3rds of the BW |
| 16:50.07 | [TK]D-Fender | Kobaz, That is not "small" |
| 16:52.40 | Kobaz | iax trunking doesn't give you any savings on the media payload, it combines media under fewer packets |
| 16:52.50 | Kobaz | you're still sending the same payload |
| 16:53.43 | Kobaz | if you want g729, then use g729, you don't have to use iax to get it |
| 16:54.37 | coppice | Kobaz: saving media payload is one of the two key selling points for iax |
| 16:55.15 | *** join/#asterisk DagMoller (~aguirre@unaffiliated/dagmoller) |
| 16:55.20 | coppice | its sad that one of the drafts for doing that with RTP has not reach release status |
| 16:55.29 | DagMoller | there is a way to remove a lot of useless "NoOp" messages produced by AEL extensions? |
| 16:56.14 | [TK]D-Fender | Kobaz, I used the payload to exemplify the overhead of UDP headers using RTP vs IAX2 |
| 16:56.35 | [TK]D-Fender | Kobaz, that's 20kbps EACh for 9kbps of PAYLOAD. |
| 16:56.48 | [TK]D-Fender | 9.6* |
| 16:57.07 | [TK]D-Fender | so 2/3 of your total BW is wasted on header |
| 16:57.39 | Kobaz | coppice: not it isn't... the main selling point is reducing header payload |
| 16:57.52 | p3nguin | dagmoller: Rewrite the dial plan with less NoOp()s. |
| 16:58.02 | [TK]D-Fender | IAX2 = 9.6 * 23 (lets say full PRI-T1) + 20kbps = 240.8kbps |
| 16:58.09 | coppice | Kobaz: could you explain the difference? |
| 16:58.13 | ruben23 | Kobaz: aprticular like the recordings..? |
| 16:58.26 | DagMoller | p3nguin, extension.ael when parsed, auto generate a lot of NoOp |
| 16:58.40 | p3nguin | dagmoller: Rewrite the dial plan to have less. |
| 16:58.47 | Kobaz | coppice: headers tell you information about the packet... the media gives you a chunk of audio |
| 16:58.47 | [TK]D-Fender | RTP = (9.6 +20) * 23 = 680.8kbps |
| 16:58.53 | DagMoller | p3nguin, my dialplan have no NoOp messagens |
| 16:59.17 | [TK]D-Fender | DagMoller, Yes, AEL parses out to bloated standard dialplan logic. |
| 16:59.18 | p3nguin | dagmoller: Then your question and statement do not correlate. |
| 16:59.23 | Kobaz | [TK]D-Fender: yeah, like i said before, savings on the WAN is great, but for lan it won't make a big impact, unless your switches are overloaded |
| 16:59.24 | [TK]D-Fender | DagMoller, Expect waste & filler |
| 16:59.37 | *** join/#asterisk anonymouz666 (~anonymouz@189.25.142.176) |
| 16:59.37 | p3nguin | does not use AEL. |
| 16:59.42 | [TK]D-Fender | Kobaz, Yeah, on a LAN, who cares .... but that isn't everybody... and this IS a big number |
| 17:00.03 | Kobaz | yeah i was talking about the end-result on a lan is minimal unless you have a crappy network |
| 17:00.08 | DagMoller | p3nguin, -- Executing [99RECEPCAO@CF-Local:25] NoOp("DAHDI/31-1", "Finish if_if_CF-Local_795_796") in new stack |
| 17:00.10 | coppice | Kobaz: its all part of the media packets |
| 17:00.22 | Kobaz | coppice: media packets.. not media payload |
| 17:00.36 | Kobaz | coppice: media payload is media payload, it needs to get from A to B anyway |
| 17:00.43 | Kobaz | coppice: you can't get any savings there |
| 17:00.54 | DagMoller | p3nguin, this is auto generated by ael parser |
| 17:01.10 | Kobaz | if you want a smaller media payload, use a different codec, using iax2 trunking wont help you there |
| 17:01.34 | [TK]D-Fender | <DagMoller> there is a way to remove a lot of useless "NoOp" messages produced by AEL extensions? <- go change the AEL parser. |
| 17:01.34 | coppice | Kobaz: you seem to be working hard to score idiot points |
| 17:01.45 | DagMoller | :] |
| 17:01.47 | DagMoller | :/ |
| 17:02.05 | Kobaz | coppice: i'm trying to clarify your understanding of what iax trunking actually does |
| 17:02.17 | [TK]D-Fender | facepalms... |
| 17:02.34 | Kobaz | and you are fighting me on it... so. I'll just let you continue in ignorance |
| 17:02.44 | coppice | [TK]D-Fender: a little knowledge is a dangerous thing :-) |
| 17:03.03 | Kobaz | apparently so |
| 17:03.16 | [TK]D-Fender | coppice, so is "none" and "lots". Depends whose HANDS ;) |
| 17:04.16 | Kobaz | what fender is saying and what I'm saying go hand in hand... fender pasted out the bandwidth savings by reducing header size (not reducing media side) |
| 17:04.23 | Kobaz | *size |
| 17:04.51 | *** part/#asterisk ipiera (~Paul@ipiera.plus.com) |
| 17:04.54 | [TK]D-Fender | Kobaz, And you missed my point that showing what I did was to show the SIGNIFANCE of the the savings as a measure of % waste. |
| 17:05.03 | [TK]D-Fender | Kobaz, Which is why I chose the payload I did |
| 17:05.23 | [TK]D-Fender | Kobaz, Because that ration of 9.6/20 is a lot more obvious than 64/20 |
| 17:05.33 | [TK]D-Fender | ratio* |
| 17:05.50 | p3nguin | 9.6 is g729? |
| 17:05.54 | [TK]D-Fender | You decided to go fucknuts on my choice of codec :p |
| 17:05.55 | [TK]D-Fender | p3nguin, Yes |
| 17:06.38 | Kobaz | hehe |
| 17:07.18 | Kobaz | [TK]D-Fender: okay... that was a good choice for showing signifigance |
| 17:07.28 | [TK]D-Fender | Qwell, Go pummel your G100/G200 devs into adding IAX2 support! |
| 17:07.44 | Qwell | G what now? |
| 17:07.51 | coppice | [Tk]D-Fender: did you know the guy that came up with RTP and SIP is now the CTO of the FCC? sad, huh? |
| 17:08.37 | [TK]D-Fender | Qwell, http://www.digium.com/en/products/gateway/g200/#documentation |
| 17:08.45 | [TK]D-Fender | Qwell, Your new gateways |
| 17:09.00 | Qwell | You know, I'd never actually seen those model numbers before... |
| 17:09.44 | coppice | I guess they're called 100 and 200 because they don't have 100 or 200 channels :-\ |
| 17:09.48 | [TK]D-Fender | coppice, I know SIP is too loose a protocol.. not sure on the judgements around RTp itself though... but yeah it doesn't scream "standards" to me... even if everyone claims to "support" them |
| 17:10.01 | Kobaz | hah, sip.... standards |
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| 17:14.01 | coppice | [TK]D-Fender: its not so much loose as misguided. "H.323 is complex. We can do something simpler.". Now H.323 is looking real simple :-) |
| 17:14.23 | coppice | it is pretty vague, though |
| 17:14.45 | coppice | SDP is particularly nasty |
| 17:15.29 | [TK]D-Fender | coppice, What little I'd read on it was that it was a fairly close direct translation of existing TDM protocols which meant that it should be easy to take in a traditional circuit and get most of the sme functionality out of it... |
| 17:15.56 | [TK]D-Fender | Q.931 I think was one of the related #'s |
| 17:16.41 | coppice | H.323 is pretty much a respin of ISDN. It has a 100 years of thinking about call making built into it |
| 17:16.55 | Naikrovek | modern |
| 17:18.11 | Naikrovek | i remember having an ISDN line in my house, and an account at the local ISDN ISP. holy lord that sucked |
| 17:18.41 | doug | isdn, kick ass. |
| 17:19.06 | doug | hm, h.323 might be like ISDN in that they are both used for video |
| 17:19.09 | Naikrovek | it worked great but it was slow at the time. hard for me to change my perspective to how slow things were then |
| 17:19.18 | doug | which is like saying ice cream is like broccoli because they are both for eating |
| 17:19.20 | Kobaz | a whole 128kbit of data, it was great |
| 17:19.24 | Naikrovek | yeah |
| 17:19.34 | Kobaz | lightning fast |
| 17:19.45 | coppice | H.323 is like ISDN, because much of the specs is common to both |
| 17:19.48 | Naikrovek | 14.4kb/s download. i remember being flabberghasted by such a high speed |
| 17:19.55 | [TK]D-Fender | I miss the days of being able to whistle up a 300 baud carrier ;) |
| 17:20.01 | Naikrovek | i don't |
| 17:20.46 | doug | i think h.323 takes some hints from ISDN, but ISDN is way more than just a way to transmit video |
| 17:21.11 | Naikrovek | i think that's what he's saying |
| 17:21.11 | doug | i guess it's like comparing ice cream and a corral full of steer |
| 17:21.38 | Naikrovek | isdn is still in very high use today |
| 17:21.50 | Naikrovek | radio and voice over industries rely on it heavily |
| 17:21.51 | coppice | doug: why the fascination with video? very little H.323 or ISDN carries video |
| 17:21.52 | [TK]D-Fender | yeah .. in PRI form ;) |
| 17:23.14 | doug | pri on the desktop |
| 17:23.52 | doug | huh, guess i haven't used h.323 outside of a video context |
| 17:24.41 | coppice | the H.323 message and the ISDN messages are essentially the same. The call model is exactly the same |
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| 17:30.45 | jeffspeff | does confbridge support kicking a specific user from an active conference? |
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| 17:31.08 | Qwell | jeffspeff: In Asterisk 10, confbridge is feature-compatible with meetme. |
| 17:31.22 | Qwell | (for features that anybody cared about) |
| 17:31.28 | jeffspeff | lol |
| 17:31.38 | jeffspeff | i'll alter my google search then. thanks. |
| 17:31.45 | Qwell | ~asteriskwiki |
| 17:31.45 | infobot | somebody said asteriskwiki was http://wiki.asterisk.org |
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| 17:34.21 | jeffspeff | Qwell, i'm seeing the CLI command for it, is there any way to make it more end-user friendly in an admin menu? |
| 17:34.45 | Qwell | dunno |
| 17:43.12 | shinao1 | hi guys please i have a problem with a dundi+iax2 trunk network.. im trying to do a dundi lookup and it complains of LAGGED connections and wont search. Is there a way to make dundi/iax2 ignore the latencies, and what is the maximum latency they'll suffer before breaking up? |
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| 19:02.01 | philipp64|laptop | is there a way to configure a dialplan (in sip.conf and extensions.conf) for an OPTIONS message I'm getting? my peer Taqua is sending me a "sip:ping@my.ip.add.ress" as the To: field... |
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| 19:08.07 | [TK]D-Fender | philipp64|laptop, The only thing I ever heard for this is if there is a match it doesn't respond 404... but it still doesn't "do" anything. |
| 19:08.17 | [TK]D-Fender | philipp64|laptop, What are you trying to accomplish? |
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| 19:19.09 | philipp64|laptop | [TK]D-Fender: trying to get my Asterisk box to stop sending 404's to the other guy (the Taqua) and send 200 OK instead. |
| 19:19.34 | [TK]D-Fender | make an estension to match their inbound request |
| 19:19.52 | [TK]D-Fender | It won't actually "execute", it'll supposedly just allow it to respond OK |
| 19:19.57 | philipp64|laptop | I've set my default context to INVALID so that my box can't be used to pirate my service from random SIP attacks out there. |
| 19:20.19 | philipp64|laptop | so I'm getting a message about "no extension 's' found in context INVALID" |
| 19:21.33 | anonymouz666 | s,1,Noop - would solve your problem |
| 19:22.00 | philipp64|laptop | it might also make me open to people placing random outbound calls. |
| 19:24.54 | philipp64|laptop | too bad there isn't a "optionscontext" like there's a regcontext and a subscribecontext. |
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| 19:36.12 | leifmadsen | philipp64|laptop: whatever context the options hit should be considered publicly accessible anyways, which means you need to secure it like any other public facing interface. Also I don't see how s,1,NoOp() makes open to anyone placing calls |
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| 19:42.39 | [TK]D-Fender | <philipp64|laptop> it might also make me open to people placing random outbound calls. <- only if you put very unintelligent dialplan in there. |
| 19:45.25 | philipp64|laptop | Ok, so my default context is INVALID and I have: |
| 19:45.28 | philipp64|laptop | [INVALID] |
| 19:45.30 | philipp64|laptop | exten => s,1,Noop(INVALID) |
| 19:46.00 | philipp64|laptop | there's hopefully no way to place an outgoing call. |
| 19:46.21 | philipp64|laptop | I suppose I could put a Hangup() instead... |
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| 19:52.25 | [TK]D-Fender | philipp64|laptop, "hopefully"? This is your context. Did you put anything else in there? |
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| 19:58.41 | philipp64|laptop | nope, just that. |
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| 19:58.57 | philipp64|laptop | I'm just use to hackers being able to find arcane exploits regularly. |
| 19:59.23 | anonymouz666 | named was the paradise for script kiddies |
| 19:59.36 | anonymouz666 | every night, a new bash# |
| 20:00.59 | anonymouz666 | "0-day exploit" |
| 20:03.17 | anonymouz666 | Ahh I just remembered the e-zine called phrack |
| 20:03.31 | anonymouz666 | don't know if still exist |
| 20:03.45 | philipp64|laptop | php and HPOV still give me night sweats. |
| 20:03.56 | anonymouz666 | HPOV? |
| 20:04.05 | philipp64|laptop | HP OpenView. |
| 20:04.49 | anonymouz666 | isn't this software that costs tons of money? |
| 20:04.54 | philipp64|laptop | yes. |
| 20:06.45 | anonymouz666 | replace it with bigbrother or nagios :-P |
| 20:09.27 | teratoma | they bought openview for 1.5 billion, it must be awesome |
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| 20:33.48 | coppice | product names beginning with open are so last century |
| 20:40.13 | anonymouz666 | openfax for asterisk |
| 20:41.13 | coppice | I registered opencall.org in the last century. after several years HP's lawyers asked for it |
| 20:41.35 | anonymouz666 | did you sell ? |
| 20:42.12 | coppice | yes, because it was now this century and opencall seemed ancient :-) |
| 20:43.04 | philipp64|laptop | until it comes into retro style, then it will be "reopen"... |
| 20:44.36 | coppice | their opencall product had such a low profile when I registered opencall.org that googling opencall only trawled talent agency links, and opencall.com was a talent agency site |
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| 21:32.52 | krotos | hi guy! Someon of you use iperf for testing connection / measure jitter etc? |
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