| 00:00.16 | *** join/#asterisk WIMPy (~wimpy@e183095026.adsl.alicedsl.de) |
| 00:13.01 | *** join/#asterisk xpot-mobile (~james@166-70-100-198.ip.xmission.com) |
| 00:15.35 | ideas1 | Is freepbx to asterisk what virtualmin is to apache? |
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| 00:32.14 | *** part/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 00:33.27 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 00:56.02 | ChannelZ | more or less |
| 00:59.02 | ideas1 | Thanks ChannelZ, is there any advangtes to running asterisk without a panel like freepbx or elastixs? |
| 00:59.46 | SunTsu | yeah - you get support in this channel ;) |
| 01:03.43 | ChannelZ | Yeah and you're dialplan doesn't do 500 things just to make one phone ring |
| 01:04.06 | ChannelZ | s/you're/your/ |
| 01:04.12 | WIMPy | Or to confues you or anyone trying to help you. |
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| 01:08.26 | ChannelZ | http://ow.ly/1dRUqK |
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| 02:48.55 | *** join/#asterisk lkthomas (lkthomas@n119236143112.netvigator.com) |
| 02:48.57 | lkthomas | hey guys |
| 02:49.07 | lkthomas | does asterisk have dialplan generator ? |
| 02:49.19 | lkthomas | also, does asterisk contain web GUI interface ? |
| 02:50.11 | WIMPy | What do you expect from a "dialplan generator"? |
| 02:50.28 | WIMPy | And, no, but you can get additional GUIs. |
| 02:50.43 | lkthomas | which gui is so call complete? |
| 02:51.13 | WIMPy | does not understand that question. |
| 02:51.43 | lkthomas | the dialplan generator, I am expecting I could do dialplan writing with auto function explain...etc like writing php code |
| 02:52.11 | lkthomas | which web gui is not in broken stage ? |
| 02:52.37 | WIMPy | You can use AGI to use whatever language you want. |
| 02:53.11 | lkthomas | not AGI |
| 02:53.16 | WIMPy | They all have limitations. freepbs seems the most used. |
| 02:53.25 | WIMPy | ~freepbx |
| 02:53.25 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there, or fpbx |
| 02:54.09 | lkthomas | does freepbx only contain the scripts or pack with asterisk ? |
| 02:54.35 | WIMPy | If you want a distro, look at AsteriskNOW. |
| 02:54.54 | WIMPy | You can chose between the Asterisk GUI, FreePBX or no GUI. |
| 02:55.05 | lkthomas | errr, no, I just want to know if freepbx and asterisk is not pack together |
| 02:55.15 | WIMPy | no |
| 02:55.28 | WIMPy | Two parts |
| 02:55.40 | lkthomas | are they integrate well ? |
| 02:56.06 | WIMPy | You need to ask that in #freepbx. |
| 02:56.11 | lkthomas | k |
| 02:56.38 | lkthomas | freepbx looks goo |
| 02:56.40 | lkthomas | good* |
| 02:56.53 | lkthomas | I have a question |
| 02:57.02 | lkthomas | what is DAHDI ? |
| 02:57.45 | WIMPy | The drivers for Digiums telephony hardware. |
| 02:59.17 | lkthomas | so is it a digital line or analog line device ? |
| 02:59.32 | WIMPy | Both |
| 02:59.48 | lkthomas | depends on card model right ? |
| 02:59.52 | WIMPy | yes |
| 03:00.04 | lkthomas | but why FXO and FXS card appear ?! |
| 03:00.12 | lkthomas | isn't DAHDI replace them all together ? |
| 03:00.50 | WIMPy | DAHDI is the software for FXS, FXO, BRI and PRI cards. |
| 03:00.59 | WIMPy | Or one of them. |
| 03:01.41 | lkthomas | I see, so it's a driver not a card name |
| 03:01.49 | WIMPy | correct |
| 03:02.24 | lkthomas | still confusing about PRI and FX_ card |
| 03:02.43 | lkthomas | PRI basically could do whatever FX_ card do right ? |
| 03:03.09 | WIMPy | PRI/BRI is digital, FXO/FXS is analog. |
| 03:03.41 | lkthomas | OH, I remember that mentioned in the book before :P |
| 03:04.06 | lkthomas | it's hard to play with real circuit at home |
| 03:04.18 | WIMPy | Why? |
| 03:04.27 | lkthomas | line card and the circuit cost a lot |
| 03:04.48 | *** join/#asterisk hmmhesays (~hmmhesay@174-126-194-60.cpe.cableone.net) |
| 03:04.54 | WIMPy | Depends on your situation. |
| 03:05.11 | WIMPy | Usually you have a phone line, don't you. |
| 03:05.19 | lkthomas | nope |
| 03:05.20 | hmmhesays | anyone in the atlanta area that can check a cid number for me? |
| 03:05.26 | lkthomas | only mobile phone |
| 03:05.36 | WIMPy | But if it's analog, the card is costly. |
| 03:05.53 | lkthomas | digital line card cost so much |
| 03:06.01 | lkthomas | is it like 1-2K USD ? |
| 03:06.08 | WIMPy | Only if you live in the wrong area. |
| 03:06.47 | WIMPy | NFI. Here it starts at 20 EUR. |
| 03:07.17 | lkthomas | WIMPy: sorry to change topic a bit, is there have complex dialplan "template" which predefine everything for a standard PBX system such as 911 config, voice mail...etc |
| 03:07.48 | WIMPy | There are many examples. |
| 03:08.16 | WIMPy | But you will alwyas have to define your local own devices. |
| 03:08.22 | lkthomas | WIMPy: because writing from scratch would miss a lot of shit |
| 03:08.29 | lkthomas | yes |
| 03:08.44 | WIMPy | Yes, so do the examples :-) |
| 03:11.00 | lkthomas | very funny that most of the things are hard coded in config file, but I try cisco phone system and it allow me to change voice mail password on fly, how is it done with asterisk then ? |
| 03:11.30 | WIMPy | Either via the voicemail menu or by editing the config file. |
| 03:11.46 | lkthomas | huh? what is "voice mail menu" ? |
| 03:12.08 | WIMPy | The thing you call to listen to your messages. |
| 03:12.25 | WIMPy | That will also allow you to configure your voicemail. |
| 03:12.30 | lkthomas | no wait, isn't password also hard coded in voicemail.conf ? |
| 03:12.39 | lkthomas | the menu will change this config file ? |
| 03:12.47 | WIMPy | yes |
| 03:12.57 | lkthomas | OH well, speechless |
| 03:13.03 | WIMPy | Or you do it. |
| 03:13.05 | lkthomas | isn't this will impose security risk ? |
| 03:13.18 | WIMPy | How? |
| 03:13.26 | lkthomas | user is actually accessing the core config file via menu |
| 03:13.42 | WIMPy | Yes. So what? |
| 03:13.51 | WIMPy | You cant dictate changes. |
| 03:14.07 | lkthomas | if bug happen..etc and change other user password, that will be in problem |
| 03:14.25 | WIMPy | Shit happens. |
| 03:14.44 | WIMPy | You can use a database instead, if you want. But that might explode as well, if it has a bug. |
| 03:14.55 | lkthomas | hmm |
| 03:17.16 | lkthomas | again, except keep on google, any site you could suggest to obtain sample config ? |
| 03:17.45 | WIMPy | There must be thousands. |
| 03:20.06 | *** join/#asterisk lucasb (~lucasb@S0106000c42710923.ok.shawcable.net) |
| 03:20.55 | lkthomas | I am going to join a new company which use asterisk, I think they must have a standard template and all functions should be included |
| 03:21.05 | lkthomas | I just need to call those functions |
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| 03:36.36 | lkthomas | WIMPy: you there ? |
| 03:36.42 | WIMPy | yes |
| 03:36.55 | lkthomas | WIMPy: how big of your asterisk are running now ? |
| 03:37.08 | lkthomas | in terms of number of servers |
| 03:37.08 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 03:37.21 | WIMPy | 3 |
| 03:37.59 | lkthomas | did you do function split between servers or just 3 branches of asterisk ? |
| 03:38.21 | WIMPy | 3 locations |
| 03:38.26 | lkthomas | I see |
| 03:38.43 | lkthomas | connecting using IAX2 ? |
| 03:39.00 | WIMPy | yes |
| 03:39.23 | lkthomas | anything you need to tune in betweek ? |
| 03:39.26 | lkthomas | between*] |
| 03:40.03 | WIMPy | ? |
| 03:40.08 | lkthomas | latency tuning |
| 03:40.24 | WIMPy | No |
| 03:40.41 | WIMPy | I'm using 10ms frames. But you don't have to. |
| 03:40.58 | lkthomas | huh? you could adjust the time within frame ? |
| 03:41.10 | WIMPy | yes |
| 03:41.28 | lkthomas | how to determine that number then? |
| 03:41.42 | *** join/#asterisk ketas (ketas@ketas6-sixxs.si.pri.ee) |
| 03:41.48 | WIMPy | By configuring it. |
| 03:41.58 | lkthomas | no, I mean, how do you judge ? |
| 03:42.20 | WIMPy | Bandwidth or latency. |
| 03:42.31 | lkthomas | so you test before apply ? |
| 03:43.02 | WIMPy | Jo. I just used the smallest latency, not caring about the extra bandwidth. |
| 03:43.26 | lkthomas | so smaller latency = higher bandwidth usage, am I correct ? |
| 03:43.35 | WIMPy | correct |
| 03:43.59 | lkthomas | if the link is loosy, should I increase that ms number ? |
| 03:44.07 | WIMPy | smaller frames = less latency but more overhead. |
| 03:44.29 | WIMPy | It it is lossy you shouldn't use it for voip. |
| 03:44.41 | WIMPy | And it depends on the pattern of losses. |
| 03:44.41 | lkthomas | LOL |
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| 03:48.20 | tyman | Has anyone seen a stream of this error (http://pastebin.com/EFPQNsrt) when on a confbridge? |
| 03:49.03 | lkthomas | WIMPy: to troubleshoot askterisk, I could enable detail debug log to analysis, right ? |
| 03:49.47 | WIMPy | You are more likely to watch the debug output at its shell. |
| 03:50.40 | lkthomas | kind of, foreground output |
| 03:51.13 | lkthomas | WIMPy: how do you do HA and what's the plan for DR on asterisk ?! |
| 03:51.18 | WIMPy | No. That will usually be another task connecting to the server. |
| 03:51.20 | lkthomas | never heard anyone doing HA |
| 03:51.32 | lkthomas | what is another task ? |
| 03:51.38 | WIMPy | Look at Asterisk SCF. |
| 03:51.43 | WIMPy | What is DR? |
| 03:51.56 | lkthomas | what is asterisk fail ? |
| 03:52.11 | lkthomas | I see |
| 03:52.13 | lkthomas | SCF |
| 03:52.22 | WIMPy | Many are doing loadbalancing or failover. |
| 03:53.55 | lkthomas | WIMPy: I think it's possible to do it with keepalived (IPVS) |
| 03:54.47 | WIMPy | There are many possible scenarios. |
| 03:54.49 | JuStIcIa_ | or using vrrpd |
| 03:55.16 | JuStIcIa_ | is more easy |
| 03:55.20 | lkthomas | how do you guys implement it then ? |
| 03:55.35 | lkthomas | JuStIcIa_: how do you deal with presistance ? |
| 03:56.37 | JuStIcIa_ | what do you mean ? |
| 03:56.47 | lkthomas | errr |
| 03:56.50 | lkthomas | nevermind |
| 03:56.53 | lkthomas | are you using vrrpd ? |
| 03:56.58 | JuStIcIa_ | one time |
| 03:57.07 | lkthomas | is it work with UDP then ? |
| 03:57.07 | JuStIcIa_ | wa snot good at all |
| 03:57.43 | JuStIcIa_ | tcp |
| 03:57.54 | lkthomas | IAX is running UDP |
| 03:58.07 | JuStIcIa_ | http://tools.ietf.org/html/rfc3768 |
| 03:58.36 | JuStIcIa_ | every vrrp deploy most be create witn this rule |
| 03:58.37 | WIMPy | VOIP is usually UDP only. |
| 03:58.46 | WIMPy | Unless you're using TLS. |
| 03:58.59 | JuStIcIa_ | yeah but vrrp is not, is just for redundancy |
| 03:59.18 | JuStIcIa_ | he set one virtual IP for every equitment |
| 03:59.32 | JuStIcIa_ | that virtual ip is your gateway |
| 04:00.26 | JuStIcIa_ | but is not soo good because if you have some problem with your configuration the daemon dont act |
| 04:00.49 | JuStIcIa_ | he works when you have one machine down |
| 04:01.28 | *** join/#asterisk xpot-mobile (~james@70-91-210-237-BusName-Utah.hfc.comcastbusiness.net) |
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| 04:07.20 | lkthomas | does asterisk be able to utilize multiple core cpu ? |
| 04:09.25 | WIMPy | It uses threads, yes. |
| 04:09.36 | lkthomas | ok |
| 04:09.45 | WiretapWork | yes, asterisking is being able to utilising multiple cpusing |
| 04:10.06 | lkthomas | any rule of thumb to do sizing ? |
| 04:10.20 | WIMPy | no |
| 04:10.38 | lkthomas | how many connection could be handle by core2dual processor ? |
| 04:11.16 | WiretapWork | 50,000 connectings, sir |
| 04:11.41 | lkthomas | so it should be 25K phone calls |
| 04:11.47 | lkthomas | as SIP is p2p protocol |
| 04:12.03 | lkthomas | that's a lot |
| 04:12.18 | WiretapWork | you are asking about connectings, sir. not about phonecallings |
| 04:12.26 | lkthomas | hmm |
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| 04:18.21 | GreatSUN | re |
| 04:23.48 | dym | Lol |
| 04:23.54 | dym | Formal conversations on IRC :D |
| 04:26.48 | *** join/#asterisk gxdssoft (~gxdssoft@190.236.107.216) |
| 04:29.40 | WIMPy | Hmm. early media seems to be broken. Wasn't there some fix in the last verison? Seems to have gone wrong. |
| 04:31.57 | *** join/#asterisk GreatSUN (~greatsun@188.20.12.162) |
| 04:32.10 | lkthomas | how do you guys config call transfer ? |
| 04:32.33 | WIMPy | not |
| 04:32.33 | *** join/#asterisk jkroon (~jkroon@dsl-241-241-185.telkomadsl.co.za) |
| 04:32.40 | lkthomas | ? |
| 04:33.09 | WIMPy | There's nothing to configure. |
| 04:33.45 | WIMPy | Fishy. |
| 04:33.53 | lkthomas | no wait, like I got call from third part and want to transfer to you, how to deal with it ? |
| 04:34.04 | WIMPy | I get early media while dialing, but as soon as it starts ringing it stops. |
| 04:34.23 | WIMPy | Press the transfer button on your phone? |
| 04:34.48 | lkthomas | does voip phone could communicate with asterisk without config ? |
| 04:34.53 | *** join/#asterisk xpot-mobile (~james@70-91-210-237-BusName-Utah.hfc.comcastbusiness.net) |
| 04:35.06 | lkthomas | what if the phone isn't voip phone ? |
| 04:35.09 | WIMPy | For that part: yes. |
| 04:35.20 | WiretapWork | dym: fonejacker, internet service providings |
| 04:35.38 | WIMPy | There is that DTMF feature hack. |
| 04:35.51 | lkthomas | say again ? |
| 04:35.54 | WIMPy | But I'm pissed now and going to sleep. |
| 04:36.18 | WIMPy | What? |
| 04:36.53 | *** join/#asterisk galaxywatcher (~galaxywat@pdpc/supporter/active/galaxywatcher) |
| 04:36.55 | lkthomas | if the phone isn't voip phone |
| 04:36.59 | lkthomas | how does it handle call transfer ? |
| 04:37.16 | WIMPy | There is that DTMF feature hack. |
| 04:37.43 | lkthomas | any docs show how to do that ? |
| 04:37.50 | WIMPy | Or you can do it via dahdi. It seems to work, but not tested in production. |
| 04:38.11 | WIMPy | features.conf |
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| 04:39.17 | lkthomas | ok, brb |
| 04:39.20 | lkthomas | lunch :) |
| 04:39.24 | lkthomas | WIMPy: have a good rest |
| 04:39.45 | WIMPy | Will try |
| 04:40.01 | WIMPy | Need to try that with other channels tomorrow. |
| 04:40.41 | lkthomas | hmm |
| 04:41.51 | JuStIcIa_ | WIMPy: tomorow im will test the Ip phone Nortel 1212 |
| 04:42.15 | JuStIcIa_ | one person told me i just need upgrade the phone with the new firmware |
| 04:42.34 | JuStIcIa_ | i just need the tftp server running and the unistim |
| 04:42.48 | JuStIcIa_ | but well tomorow is another day |
| 04:42.51 | JuStIcIa_ | cya |
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| 05:11.58 | akke | I'm evaluating Kerio Operator, which turns out to be running Asterisk internally. Everything seems to work straight but when I configure multiple SIP peers with the same provider asterisk seems to be doing things wrong? |
| 05:12.25 | akke | For example, when I receive a call on SIP account A asterisks debug messages look like the incomming call is coming from account B |
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| 05:12.42 | akke | doesn't asterisk support registering multiple SIP accounts with the same host? |
| 05:13.35 | akke | as all SIP UDP messages are comming from the same host, maybe asterisks can't always figure out for with account the received message was sent? |
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| 05:20.14 | kaldemar | akke: it depends on how the provider works. see the sample sip.conf for how asterisk matches incoming calls to devices. |
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| 05:21.00 | akke | kaldemar: well, calls are being routed correctly but in the debug messages it looks like the call is coming from the wrong sip account, so I was concerned about possible errors... |
| 05:22.07 | akke | the main reason I'm going to debug mode is the folowing: I have an IVR menu configured but when I call it the first couple of seconds are missing and because of that I don't hear the first couple of words the IVR is talking... |
| 05:22.22 | akke | any idea what could be the cause of that? |
| 05:26.53 | kaldemar | akke: if your peers are defined with a static ip and it happens to be the same for all of them, it all calls will match a single peer unless there are usernames that match first. |
| 05:27.40 | akke | okay |
| 05:28.15 | kaldemar | akke: enable rtp debug and see if there are packets going out of your asterisk as soon as the call starts. if so and the same silence occurs with all calls, there's little you can do about it but ask your ITSP. |
| 05:29.24 | akke | stranges thing is that doesn't seems to happen on all calls. When I call with my mobile (provider X) it doesn't happen. But when my gf calls with her mobile (prover Y) a couple of words are missing at the beginning |
| 05:30.15 | akke | when I reconfigure my IVR, in kerio connect, to only allow g729 pass-trough codec, the IVR works correctly on both our mobiles. But when I configure IVR2 with g729 pass-trough too, non of the IVR's work at all |
| 05:30.21 | akke | so it's kind of a strange problem here... |
| 05:31.49 | kaldemar | does it happen with all calls from provider Y? does it happen with all calls from your gf's mobile, be the destination asterisk or another mobile phone? i've seen that silence behavior on mobile networks and on some mobile phones. |
| 05:31.52 | akke | I also noticed that IVR1 and IVR2 start with the same message but IVR2 is missing 4 words at the beginning while IVR1 is only missing 3 |
| 05:32.24 | akke | kaldemar: it happens with calls from provider Y and provider Z (landline) too |
| 05:32.28 | akke | provider X doesn't seem to be a problem |
| 05:33.04 | akke | it doesn't happen when she calls some other mobile or anything |
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| 05:33.46 | kaldemar | if it would be because of your asterisk box, one could assume it to happen on calls from every provider. |
| 05:34.12 | akke | kaldemar: yeah, that's what I thought too. That's why i'm scratching my head right now :( |
| 05:35.04 | akke | I've been playing arround with my codec settings between my asterisk box and my SIP provider too, without success |
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| 06:12.28 | schmidts | good morning |
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| 06:17.29 | ChannelZ | alohahahaha |
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| 07:51.13 | *** join/#asterisk Polysics (~luca@host210-142-static.228-95-b.business.telecomitalia.it) |
| 07:53.49 | Polysics | hello |
| 07:54.30 | Polysics | i need to hold an user in waiting while i find an operator for him, then connnect them - what's the best way? |
| 07:54.43 | Polysics | i need to implement a ocmplicated queue system i can't do with queues |
| 07:54.57 | Polysics | asterisk 1.8 |
| 07:56.13 | lkthomas | I only know queues |
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| 07:56.38 | lkthomas | what's the problem with queues ? |
| 07:58.14 | Nasga | Polysics: queue + agents pause/unpause should do this ? |
| 07:58.32 | kaldemar | or just an attended transfer... |
| 07:58.43 | Polysics | Nasga, i have a complex (do not ask me why) system for deciding who answers |
| 07:59.02 | lkthomas | can you do grouping ? |
| 07:59.05 | lkthomas | group queues ? |
| 07:59.19 | Polysics | i will cross-post stuff even if i shouldn't |
| 07:59.29 | Polysics | i have a list of people on the system that have one or more languages and specializations, and belong to different companies |
| 07:59.35 | Polysics | a caller selects a language and specialization, then is put in waiting and the system finds him: someone that belongs to a company that has the "trusted" field set, with that lang and spec |
| 07:59.41 | Polysics | if no one has the spec, only the lang |
| 07:59.48 | Polysics | if no one from trusted companies has the lang, again with not trusted companies |
| 07:59.55 | Polysics | if there is a match but he is already on the phone, the caller stays in waiting |
| 08:00.12 | Polysics | additionally, operators that are connected via SIP are preferred over cellphone calls |
| 08:00.42 | Polysics | and all this logic is still subject to change |
| 08:00.47 | Polysics | so ideally, i would put someone in waiting, decide who to dial, then dial him |
| 08:01.40 | Nasga | i have similar stuff in agi |
| 08:01.58 | Polysics | i didn't mention i can do it all in Adhearsion |
| 08:02.12 | Polysics | which is AGI |
| 08:02.29 | Polysics | my problem is at a lower level: which "functions" do i need to build the above? |
| 08:06.44 | Polysics | breaking down the problem, i "simply" need to park a call, dial the destination, bridge the two if destination answers |
| 08:06.53 | Polysics | that's the easiest flow |
| 08:07.06 | lkthomas | on CLI, when I do queue add member, does it means I could assign a caller to a specific agents to answer call ? |
| 08:07.45 | Nasga | ikthomas, you add an agent for the queue, not for a specific caller |
| 08:08.12 | lkthomas | Nasga: isn't all agent already predefined on config file ? what situation need to execute this command ? |
| 08:08.45 | Nasga | you can add sip line to queue directly |
| 08:08.54 | Nasga | that's how i see this method |
| 08:09.10 | Nasga | and you can build an interface to manage queue without reload every times |
| 08:09.26 | lkthomas | you mean like GUI ? |
| 08:09.45 | lkthomas | but when I reload, everything I did on CLI will be wipe away ? |
| 08:10.31 | Nasga | last time i checked on ast 1.2, he keep my cli stuff |
| 08:10.45 | Nasga | event after a restart, but i need to check this |
| 08:10.47 | Polysics | depends on what yo udo |
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| 08:11.12 | Polysics | anyway, can anyone please point me to the basic building blocks for the above? |
| 08:11.34 | Polysics | park a call, dial the destination (chosen by something * doesn't care about), bridge the two if destination answers |
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| 08:12.40 | wdoekes2 | Polysics: you can use a queue with Local members |
| 08:13.23 | Polysics | wdoekes2, and what logic would the local members have? |
| 08:13.40 | wdoekes2 | a local channel is a piece of dialplan |
| 08:14.40 | lkthomas | in a large call center, is there have a monkey to monitor AST all the time and execute CLI according to need ? |
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| 08:19.39 | Polysics | a monkey? |
| 08:19.53 | Polysics | wdoekes2, i forgot to mention i am doing this in adhearsion |
| 08:20.02 | Polysics | logic will be handled there |
| 08:20.29 | Polysics | i basically just can't figure out how to park an incoming call somewere, thenbridge it to a new channel if it answers |
| 08:20.33 | lkthomas | yeah, monkey |
| 08:20.58 | Polysics | i suppose they use AGI/AMI :-D |
| 08:21.09 | Polysics | less poop flying around, less bananas needed too |
| 08:21.18 | lkthomas | hmm |
| 08:21.20 | Polysics | notice i said "less poop", not "no poop" |
| 08:21.21 | Polysics | :-D |
| 08:21.26 | lkthomas | LOL |
| 08:21.35 | lkthomas | someone else please help Polysics, I am newbie on AST |
| 08:22.06 | lkthomas | Polysics: does something fuck up or what? US time should be midnight now |
| 08:23.08 | Polysics | i don't get what your problem is |
| 08:23.29 | Polysics | for automating activities, i usually recommend AGI/AMI over AST |
| 08:23.36 | Polysics | a real programming language is better |
| 08:23.36 | lkthomas | hmm |
| 08:23.43 | Polysics | though 1.8 has Lua support |
| 08:23.50 | lkthomas | I am not a programmer |
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| 08:24.34 | Polysics | lua looks like JS and Ruby made out while drunk |
| 08:24.38 | Polysics | i like it :-D |
| 08:24.41 | lkthomas | LOL |
| 08:24.53 | lkthomas | I am engineer, and I find it very hard to control AST |
| 08:25.04 | Polysics | it will be a little difficult to build logic in any kind of way without coding |
| 08:25.04 | lkthomas | because it's not as simple as scripting |
| 08:25.15 | lkthomas | I know |
| 08:26.50 | Nasga | the main issue is to fallow channels ids witch change at every actions... |
| 08:27.25 | Polysics | again, you can use some kind of DB as a "working memory" |
| 08:27.27 | Polysics | i use Redis |
| 08:29.42 | Polysics | would it be an improper applciation to just use confbridge for my problem? |
| 08:31.55 | kaldemar | Polysics: when you have determined the correct channel or extension, you could use the Originate app to connect the two. |
| 08:32.22 | Polysics | kaldemar, and about the parking part? |
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| 08:32.33 | Polysics | does originate require both channels to be already up? |
| 08:34.09 | lkthomas | I am reading book for AST, it's like reading programming book |
| 08:34.32 | kaldemar | Polysics: no, originate will dial one channel and connect it to an extension or application. |
| 08:35.22 | Polysics | kaldemar, depending on my logic, it might still be ok - but i would like to implement some sort of "FollowMe" on the callees too |
| 08:35.39 | Polysics | isn't there an app that simply bridges two open channels? |
| 08:37.43 | kaldemar | Polysics: unless the current channel is one of the two, no. |
| 08:38.06 | Polysics | well, the "current" channel should be the calling one |
| 08:38.28 | Polysics | the whole thing is initiated when A calls *, gets parked, B is chosen then dialed |
| 08:38.31 | kaldemar | and if it is parked, it won't execute any dialplan commands. |
| 08:38.37 | Polysics | crapola |
| 08:38.49 | Polysics | wouldnt' that be an AMI command though? |
| 08:39.07 | kaldemar | with origination, when B answers, A's parking lot could be dialed and the two would be connected. |
| 08:39.30 | Polysics | parking lots work like that? |
| 08:40.06 | Polysics | then, couldn't i have a ghetto FollowMe where i originate a call between B and *, do my things, then if it is ok just dial the parking lot? |
| 08:40.08 | kaldemar | when you park a call, you get a parking extension. when that extension is dialed, the caller is connected to the parked channel. |
| 08:40.28 | kaldemar | Polysics: that's what the originate application would do for you. |
| 08:40.28 | Polysics | in taht case, it would be * to dial the parking lot |
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| 08:40.54 | Polysics | and i can put any logic i care for in the * leg of the call, BEFORE even dialing the parking lot |
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| 08:41.20 | Polysics | i wonder how i can handle B not picking up and moving to C |
| 08:43.16 | kaldemar | Originate(Local/extenofB@context,exten,parkedcalls,${PARKED_EXTEN_SET_BY_YOU},1) |
| 08:43.30 | kaldemar | the extensions can handle situations where B wouldn't answer etc. |
| 08:46.06 | Polysics | i just need to wrap my head around at how to make it work with AGI/AMI/Adhearsion |
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| 09:35.04 | jacc0 | any news about when 1.8.6-RC is going to be released? |
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| 09:37.53 | wasanzy | comparing asterisk-gui and freepbx, which one has more futures? |
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| 09:54.22 | kaldemar | wasanzy: future or features? |
| 09:54.58 | wasanzy | oh sorry, I mean features |
| 09:55.06 | wasanzy | I want to use one of them |
| 09:55.48 | wasanzy | oh is my friend Kaldemar, how are you? |
| 09:57.01 | kaldemar | freepbx probably has more features. |
| 09:57.50 | wasanzy | oh ok |
| 09:57.53 | wasanzy | good |
| 09:58.30 | kaldemar | people at #freepbx will know more. |
| 09:58.39 | wasanzy | oh ok |
| 09:59.22 | wasanzy | am installing vmware so I can run my test again since am still having trouble with the sound and every one here is busy with their machines |
| 10:03.21 | kaldemar | are all your machines virtual ones? |
| 10:03.44 | wasanzy | none of them is virtual yet |
| 10:05.37 | kaldemar | wasanzy: when you tested earlier, were they virtual machines? |
| 10:06.08 | wasanzy | no they were not |
| 10:06.48 | kaldemar | wasanzy: there was nothing unusual or wrong about your sip debug yesterday, if you saw RTP packets going in each direction with rtp debug enabled, the sound issue is probably not related to asterisk at all. |
| 10:07.47 | wasanzy | ok let me enable rtp debug so we can debug with that and see |
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| 10:48.37 | awk | ARG!@#%(*%@#!*%&@##@%*(!%*(!#@!@#%(* for you guys who are using centos, what is the 'package' addon for sox to convert mp3 to wav... asterisk all of a sudden can't understand what a mp3 file is when I use convert |
| 10:50.46 | johnnyasterisk | recompile sox with lame support |
| 10:51.32 | johnnyasterisk | you may need libmad and lame-devel installed and then recompile sox |
| 10:51.44 | irroot | or pipe the output of mpg123 to sox :P |
| 10:53.37 | awk | converted with mpg123 thanks.. |
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| 11:04.36 | puzzled | hi |
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| 11:18.49 | irroot | hi the buzy has me |
| 11:23.16 | leifmadsen | awk: ya on centos (redhat, fedora, etc...) they are pretty staunch about not supporting MP3 natively -- you'd need sox and mp3 support that was compiled in from something like rpmforge or something -- there are guides you can find on google for how to get it setup. |
| 11:24.28 | jacc0 | MP3 encoding is NON-FREE i believe |
| 11:24.59 | leifmadsen | right |
| 11:25.05 | leifmadsen | which I think is the issue redhat et al have with it :) |
| 11:25.42 | coppice | mp3 decoding is not really free either |
| 11:27.28 | leifmadsen | nothing is free |
| 11:27.57 | irroot | freedom even comes at a price |
| 11:29.14 | coppice | can you get a good discount on it? |
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| 11:33.15 | irroot | hehe coppice you got any spare ?? |
| 11:33.20 | jacc0 | leifmadsen: ny clue when 1.8.6-RC is going to be released? |
| 11:33.30 | leifmadsen | probably next week |
| 11:33.34 | leifmadsen | whenever the sprint ends |
| 11:35.06 | irroot | i have 2 possible blockers im working on and should have ready too go ... 4 1.8.6 |
| 11:35.26 | jacc0 | :D |
| 11:35.33 | jacc0 | keep up the good work ! |
| 11:35.49 | irroot | hehe that is if the busy will release me :P |
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| 11:36.46 | irroot | been abducted |
| 11:51.05 | MrTelephone | What kind of mainboard/processor combos do you poeple usually use for small office installations? |
| 11:51.52 | irroot | MrTelephone how small |
| 11:52.06 | MrTelephone | 12 phones |
| 11:52.08 | irroot | use a HT-425 Atom 1.8 for <10 |
| 11:52.16 | irroot | its tiny |
| 11:52.23 | irroot | integrated system |
| 11:52.34 | MrTelephone | Reliable? |
| 11:52.38 | irroot | but no slots |
| 11:52.49 | irroot | yeah suprisingly |
| 11:52.56 | MrTelephone | Yeah I think im going to buy an adtran unit to do the sip -> fxo conversion |
| 11:53.05 | irroot | i use USB ISDN mostly |
| 11:53.10 | MrTelephone | nice |
| 11:53.28 | MrTelephone | I want to fire this up and not have to worry about it for 3 years |
| 11:53.34 | MrTelephone | solid state drive? |
| 11:54.10 | irroot | using 250Gb 2.5'' |
| 11:54.17 | WIMPy | Hi irroot. Did you see my qestion yesterday regarding you misdn1 git? |
| 11:54.25 | irroot | but SSD should be possible |
| 11:54.37 | irroot | nah sorry abducted by the busy |
| 11:54.46 | irroot | WIMpy shoot |
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| 11:55.36 | WIMPy | I wanted to publish a link where you gould git clone that, in case someone wants to try it, but I didn't find the correct url. |
| 11:55.44 | MrTelephone | i5 mini-itx looks alright |
| 11:55.50 | MrTelephone | more expensive though |
| 11:56.20 | coppice | i5 mini itx is usually noisy |
| 11:56.34 | MrTelephone | fan noise? |
| 11:57.39 | MrTelephone | I didn't even know they still made celerons. |
| 12:00.11 | MrTelephone | Doesn't have to be super tiny but wall mountable. |
| 12:04.15 | irroot | WIMPy i put it on me SVN http://pbx.smartdns.co.za/viewsvn/ |
| 12:04.58 | irroot | WIMPy i put it on me SVN http://pbx.smartdns.co.za/svn |
| 12:05.07 | irroot | pushed it there after chat |
| 12:05.37 | irroot | it requires change to configure.ac in * |
| 12:09.48 | E-bola | look at the fit-pc2 |
| 12:09.57 | E-bola | its super small power efficient and quiet |
| 12:10.04 | E-bola | and has dual ethernet so can be a router as well |
| 12:10.10 | MrTelephone | Chenbro makes a really nice ITX case but doesn't have wall mount brackets :( |
| 12:10.21 | E-bola | It comes with mounting brackets as well :) |
| 12:11.10 | *** join/#asterisk valera (~valera@88.211.48.174) |
| 12:11.17 | MrTelephone | that is pretty kick ass. The website graphics are not the best though |
| 12:11.36 | valera | hello, anyone got success using asterisk 1.8 with vonage softphone lines ? |
| 12:12.06 | MrTelephone | vonage doesn't work with asterisk at all :( |
| 12:12.11 | MrTelephone | as far as I know |
| 12:13.05 | valera | MrTelephone: well it does :) e.g. there's no problem to connect asterisk as a client to their softphone lines, however after upgrade from 1.6 to 1.8 I am experiencing some issues |
| 12:13.42 | valera | particularly with incoming calls |
| 12:14.23 | kaldemar | valera: what kind of issues? |
| 12:15.17 | valera | kaldemar: I cant figure out why incoming calls are stopped working while they used to work fine with 1.6 |
| 12:16.52 | irroot | valera sip trace / verbose output on pb will help |
| 12:16.56 | irroot | ~pb |
| 12:16.56 | 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. |
| 12:17.17 | kaldemar | maybe it's the pedantic setting in sip.conf. its default value was changed from no to yes between 1.6.2 and 1.8.0. |
| 12:17.28 | *** join/#asterisk Faustov (user@gentoo/user/faustov) |
| 12:17.47 | valera | kaldemar: could be hmmm, where it would be listed in changelog ? |
| 12:18.08 | irroot | UPGRADE / CHANGES |
| 12:19.38 | kaldemar | UPGRADE.txt for 1.8.X mentions it. |
| 12:22.53 | WIMPy | irroot: That doesn't seem to allow an anonymous checkout. |
| 12:23.29 | irroot | oh maybe i need to check that out :P |
| 12:23.49 | irroot | or you want the admin pw ?? |
| 12:24.01 | *** join/#asterisk dlu_ (~dlu@ip-62-143-210-16.unitymediagroup.de) |
| 12:24.13 | dlu_ | hi @all |
| 12:24.29 | WIMPy | irroot: No. I wanted to put up a link for others. |
| 12:24.52 | irroot | im kidding ... ill work on it sorry |
| 12:25.33 | MrTelephone | valera: does vonage offer business trunk lines? |
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| 12:32.49 | valera | MrTelephone: not in UK afaik, but prior to 1.8 upgrade - I've been using ~20 softphone lines + 6 over FXO using their boxes without big problems |
| 12:33.20 | valera | MrTelephone: trick was to switch everything to g729 - so its not being recoded and just forwarded |
| 12:33.34 | valera | after that for a few months it worked as a machinegun |
| 12:34.53 | valera | but I was stupid/tempted to upgrade to 1.8 :) |
| 12:34.59 | valera | now its time to pay for that |
| 12:41.06 | MrTelephone | Just be patient it will be fixed I'm sure. Is it something to do with the sip messaging changes? |
| 12:41.17 | MrTelephone | Did you sip debug your connections? |
| 12:41.25 | valera | MrTelephone: yes |
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| 13:00.03 | thecardsmith | what kind of gear do you need from the telco to send/recieve TXT messages to/from an asterisk box? ...currently i'm all PRI, and in the USA, and I'm guessing that dog won't hunt |
| 13:00.13 | leifmadsen | right |
| 13:00.30 | thecardsmith | can you do it with SS7 or SIP, or... *shrugs* I am mentally polluted from working with PRIs :) |
| 13:01.58 | E-bola | with asterisk or just from the box? |
| 13:02.14 | E-bola | Yuo can get a gsm modem |
| 13:02.24 | E-bola | we use it for alarms etc. |
| 13:03.07 | thecardsmith | so if i have existing numbers, i'd have to port them to a mobile provider to receive texts on them, right? ...cause i'd still want voice to go over them, as well |
| 13:03.22 | thecardsmith | i could do it just from the machine, and outside asterisk, just was assuming... maybe asterisk could do it |
| 13:03.38 | thecardsmith | port them to a mobile provider if I use a GSM modem* (correction to above, sorry) |
| 13:04.02 | E-bola | Yes you'd have to port them |
| 13:04.06 | puzzled | thecardsmith: iirc with something like a box from mobigater.com you can send & receive sms |
| 13:04.14 | E-bola | You could in theory get the voice via the modem as well but then it becomes a bit more complex |
| 13:04.17 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 13:08.02 | MrTelephone | How do you get asterisk to match a peer based on the called number. Do you set type=peer or type=friend? |
| 13:08.38 | ssureshot | I was hoping for some insight / explanation .. this was in my old zapata.conf and I'm moving to dahdi,,, asterisk 1.8.4.4,,, http://pastebin.com/74G7ENfP .. what channels are these refering to? I have a t1 so it's only channels 1-23.. I have a paging extension that uses zap/97 and need to recreate this |
| 13:11.33 | kaldemar | ssureshot: looks like 4 analog channels, 2 FXS and 2 FXO. |
| 13:12.29 | kaldemar | there's really nothing to recreate without any further scenarios or configuration. |
| 13:13.56 | *** join/#asterisk seraphie (~erin@75.76.38.159) |
| 13:14.27 | WIMPy | ssureshot: Apart from added features, the dahdi config should be the same as for zap. |
| 13:16.36 | ssureshot | I'm lacking in understanding,, if I add this config to my new dahdi install the channels don't work.. What keywords can I google or what portion of the manual will explain this to help me understand |
| 13:17.35 | WIMPy | If you say you've got a T1, that config won't work with zap, either. |
| 13:19.03 | WIMPy | And if you are looking for channel 97, that's likely to be the 1st channel after 4 T1s. |
| 13:19.36 | kaldemar | signalling is also wrong unless you have a channel bank. |
| 13:22.33 | ssureshot | let me post a few portions of the config,,, this is the internal paging system so It shouldn't really use a t1 portion of the channel I assume. but like I said I don't understand this portion yet.... http://pastebin.com/UqrMiZ97 |
| 13:24.21 | ssureshot | and zaptel.conf http://pastebin.com/f6TaURdZ |
| 13:24.59 | WIMPy | Where are the T1s? |
| 13:25.30 | WIMPy | The config for channels 1-96 is missing. |
| 13:26.17 | *** join/#asterisk screenn (~screenn@178.151.86.196) |
| 13:27.48 | ssureshot | I only have 1 T1,, is it possible to create a channel using the fxoks ? this is the whole zaptel.conf http://pastebin.com/7ctpAUxh |
| 13:28.15 | *** join/#asterisk bchia (~Adium@nat/digium/x-uxwodvplccxrdpij) |
| 13:28.24 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 13:29.11 | WIMPy | You must have a 4 port card with only 1 port configured. |
| 13:29.28 | ssureshot | you are correct sir |
| 13:29.54 | kaldemar | looks like there was a phone connected to what was zap/97. |
| 13:29.56 | WIMPy | So the 4 FXOs are on 97-100. |
| 13:30.20 | kaldemar | 2 FXS's and 2 FXO's. |
| 13:30.31 | WIMPy | Right. |
| 13:30.55 | kaldemar | ssureshot: what is the T1 connected to? |
| 13:31.24 | *** join/#asterisk luminforce (~luminforc@rrcs-71-42-183-54.sw.biz.rr.com) |
| 13:31.24 | ssureshot | new config is an astribank |
| 13:31.43 | ssureshot | old server has a 4 port digium card |
| 13:32.04 | WIMPy | didn't get the part with new hardware. |
| 13:32.22 | ssureshot | lol ok explanation. |
| 13:32.40 | WIMPy | Well, if that is only 1 T1, you fxo/fxs channels will become 25-28. |
| 13:33.39 | ssureshot | Upgrading old asterisk box that that had a 4port digium card, with new server that has an astribank,, gong from version 1.2 to 1.8.4.4.. big jump multiple hurdles I learned a crap load |
| 13:33.40 | tzafrir | ssureshot, just use dahdi_genconf |
| 13:33.40 | kaldemar | AFAIK astribank is connected to an asterisk box with USB, not a PRI or T1 CAS. |
| 13:33.44 | WIMPy | That's annoying, but just the way it is with zap/dahdi. |
| 13:33.58 | kaldemar | tzafrir: or am i wrong? |
| 13:34.17 | tzafrir | Remove from chan_dahdi.conf / dahdi-channels.conf stuff you don't need. But having them in system.conf is harmless |
| 13:34.18 | WIMPy | kaldemar: correct |
| 13:34.44 | tzafrir | kaldemar, correct |
| 13:35.45 | tzafrir | So it's configured like an analog PCI card |
| 13:35.51 | ssureshot | how do I get the paging channel then with the new system config... let me go check if there is something plugged into that 4th port on the current box,, I can't seem to remember |
| 13:36.31 | MrTelephone | I have this problem where an endpoint connects with the same source IP:PORT. When the endpoint is triggered for anonymous/caller id block asterisk does not know which peer to authenticate too. What kind of means do other sip servers have to know which endpoint is what? |
| 13:36.33 | ssureshot | tzafrir: thats what I'm coming to believe Ill be right back |
| 13:36.47 | WIMPy | ssureshot: You need to replace zap/97 with dahdi/25. |
| 13:36.55 | MrTelephone | It would be against the privacy rfc to put anything in the sip message indicating the from user.. |
| 13:36.56 | *** join/#asterisk bchia (~Adium@nat/digium/x-eykidujbvgmmvjiu) |
| 13:36.57 | *** join/#asterisk adyn (~adyn@pdpc/supporter/active/adyn) |
| 13:37.46 | MrTelephone | You can't expect a sip server to hash up every users secret with the current nonce in test. If you had 1000 users that would nail the processing load. |
| 13:38.07 | *** join/#asterisk jwendell (~wendell@unaffiliated/jonh-wendell/x-9044817) |
| 13:38.38 | MrTelephone | leifmadson, have you run into this issue anywhere? |
| 13:39.24 | jwendell | hi, folks. I have a queue, and have calls logged into cdr (mysql). my question is: what's the difference between 'NO ANSWER' and 'BUSY' incoming calls? all of them have duration > 0 |
| 13:39.47 | jwendell | (these values are in the disposition column) |
| 13:39.56 | *** join/#asterisk johnnyasterisk (~johnnyast@89.18.71.45) |
| 13:40.05 | *** join/#asterisk johnnyasterisk (~johnnyast@89.18.71.45) |
| 13:40.06 | kaldemar | jwendell: they are self explanatory. |
| 13:40.22 | jwendell | I wish they were... |
| 13:41.04 | jwendell | I have lots and lots of BUSY (more than ANSWERED and NO ANSWER together) |
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| 13:41.25 | ssureshot | WIMPy: would I still need the fxoks entries in config files ? |
| 13:41.36 | jwendell | when an incoming call is considered 'BUSY'? |
| 13:42.06 | WIMPy | ssureshot: Sure. But with channel 25-28. |
| 13:42.28 | ssureshot | and there is actually a second card in this box that has the paging system hooked up,, I need to do some more research on the hardware ... I wonder if this is a different type of card |
| 13:42.32 | WIMPy | ssureshot: It will be the same, just with cahnnels 25-96 removed. |
| 13:42.40 | MrTelephone | From: "Alice" <sip:6551156d569c4b7d945f310ff10943c5@anonymous.invalid>; When my endpoint is anonymous it blanks out the "Alice" part as well |
| 13:43.15 | WIMPy | ssureshot: Ok, so you don't have that hardware for the new system, yet? |
| 13:44.00 | MrTelephone | So there should be a user= field in the proxy-authorization header. Proxy-Authorization: .... realm="sip.cisco.com" user="fluffy" |
| 13:44.09 | kaldemar | jwendell: when the outgoing (from asterisk's point of view) leg is busy. |
| 13:44.47 | ssureshot | WIMPY: well I thought we were all set ,, I'm looking into this card right now.. if it's just a second 4 port t1 then I don't see why the astribank can't handle everything |
| 13:45.10 | tzafrir | Subdolus, what Astribank, exactly? |
| 13:45.13 | WIMPy | ssureshot: No, it's a 4 port analog card. |
| 13:45.13 | tzafrir | dahdi_hardware -v |
| 13:46.17 | ssureshot | usb:001/002 xpp_usb+ e4e4:1162 Astribank-modular FPGA-firmware |
| 13:46.17 | ssureshot | <PROTECTED> |
| 13:46.17 | ssureshot | XBUS-00/XPD-00: T1 (24) Span 1 DAHDI-SYNC |
| 13:46.23 | *** join/#asterisk bchia (~Adium@nat/digium/x-lexllmrkhpbmymno) |
| 13:47.15 | ssureshot | analog,, got ya,, I have one in my current backup server .. crap I thought this was all complete |
| 13:48.59 | ssureshot | all good information thank you guys,,, let me do some research and I'm sure Ill be back |
| 13:49.13 | WIMPy | MFBS |
| 13:49.47 | WIMPy | I was trying to find out what is wrong with early media since yesterday night. |
| 13:50.01 | WIMPy | Just to find out it's my provider that's broken. |
| 13:52.43 | *** join/#asterisk bchia (~Adium@nat/digium/x-papadpczldrmykmy) |
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| 13:54.22 | ssureshot | yeah,, my experience with all telco is that it usually ins't the hardware,, the lines usually effed up.. once I verify the hardware I just request a loopback .. they jump up then after the crap about possibly being billable |
| 13:55.23 | WIMPy | The "line" is FUBAR anyway. It's NGN SIP shit. |
| 13:56.34 | luminforce | has anyone seen a problem where a mid-call invitte from a remote sip provider (user on hold) causes asterisk to record the cdr as ended (without sending BYE), and then the call continues for several minutes? |
| 14:00.00 | ssureshot | tzafrir: once I connect my paging system to the astribank will it auto create the fxo/fxs ports when it sees the hardware? |
| 14:00.25 | ssureshot | this particualr astribank is an XR-0000 which supports analog also |
| 14:01.08 | tzafrir | ssureshot, from your paste there it has a single E1/T1 port, right? |
| 14:01.15 | tzafrir | And no analog modules |
| 14:02.26 | ssureshot | <PROTECTED> |
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| 14:04.58 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:05.00 | ssureshot | ok nm, I have to order the proper ports good deal |
| 14:07.53 | *** part/#asterisk capt_cassimir (~arch@178.73.217.49) |
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| 14:19.28 | valera | any patches online to build dahdi with recent linux kernels ? |
| 14:19.37 | valera | 3.0.0 rc6 or something |
| 14:20.08 | irroot | valera you welcome to do it :P |
| 14:20.44 | valera | irroot: hahaha, thanks ok, I hope its only about macros...declarations style change |
| 14:21.25 | WIMPy | Are there changes again? |
| 14:21.51 | valera | well, on debian wheezy was not able to build it |
| 14:21.56 | valera | so postponed a bit |
| 14:22.01 | valera | will be back to it later today |
| 14:25.06 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 14:25.34 | WIMPy | irroot: What's the latest linux, you tried with your misdn1 branch BTW? |
| 14:25.49 | irroot | WIMpy using 2.6.38.8 |
| 14:26.46 | jacc0 | are there duch voices for festival? and where can I find them? |
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| 14:27.01 | *** part/#asterisk dlu_ (~dlu@ip-62-143-210-16.unitymediagroup.de) |
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| 14:27.35 | jacc0 | *dutch |
| 14:28.19 | *** part/#asterisk mrfaison (~mrfaison@200.13.124.36) |
| 14:29.42 | *** join/#asterisk spck (~spck@h75-100-71-130.mdsnwi.dedicated.static.tds.net) |
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| 14:34.15 | spck | hi guys, I found a bug, it was "fixed" in trunk, but not fixed correctly. I have the proper solution, but I'm not sure where to go from here. |
| 14:35.00 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 14:35.00 | *** mode/#asterisk [+o malcolmd] by ChanServ |
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| 14:38.32 | WIMPy | spck: Upload your patch to issues.asterisk.org. |
| 14:39.18 | *** join/#asterisk wasanzy (~emmanuel@196.201.43.55) |
| 14:39.59 | raden | Katty, !!! :) :) |
| 14:43.54 | spck | if i do a patch do I need to sign the code release? |
| 14:44.18 | WIMPy | yes |
| 14:44.27 | carrar | in cursive |
| 14:44.42 | spck | my signature is the only time i use cursive |
| 14:45.46 | irroot | do you need a pen for that ?? |
| 14:46.12 | carrar | has to be old school ink |
| 14:46.22 | spck | fountain pen? |
| 14:46.34 | carrar | yeah |
| 14:47.05 | *** join/#asterisk Sean-Der (~Removable@108-90-184-88.lightspeed.toldoh.sbcglobal.net) |
| 14:47.09 | carrar | preferably with a feather coming out one end |
| 14:47.15 | *** part/#asterisk Sean-Der (~Removable@108-90-184-88.lightspeed.toldoh.sbcglobal.net) |
| 14:47.21 | carrar | or just using a feather |
| 14:48.44 | wasanzy | hi guys |
| 14:49.05 | wasanzy | to help me resolve my ongoing problem with calls, I have some questions |
| 14:49.07 | *** join/#asterisk davlefou (~david@41.225.9.81) |
| 14:50.57 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 14:51.30 | wasanzy | when two people uses twinkle to call each other, first of all, does the call passes through the astrisk server and how is the sound or audio transmitted? will the sound be transmitted through the asterisk server as well? |
| 14:52.20 | kaldemar | wasanzy: depends on your configuration and network setup. |
| 14:53.01 | wasanzy | ok,let say am in a NAT setup |
| 14:54.01 | kaldemar | if directmedia/canreinvite is yes the audio stream won't go through asterisk. |
| 14:54.41 | kaldemar | what kind of a NAT setup? everything on the same side of a single NAT? |
| 14:55.08 | wasanzy | so which option is best now? because I have directmedia=no under one account |
| 14:55.24 | *** join/#asterisk ph8 (~ph8@unaffiliated/ph8) |
| 14:55.35 | wasanzy | yes every thing is the same |
| 14:56.21 | wasanzy | we have public IP assign to a router and the router assign lan IPs automatically to the client machines |
| 14:56.50 | wasanzy | and the test am doing, the three machines are in the same network |
| 14:57.28 | kaldemar | then they should work whatever the setting is. |
| 14:57.39 | wasanzy | so doyou think I should have directmedia/canreinvite to yes? |
| 14:58.42 | kaldemar | if you want the audio to not go through asterisk. your choice. |
| 14:59.35 | *** join/#asterisk dms (~Adium@nat/digium/x-maogqgptmuzbmwip) |
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| 15:00.27 | wasanzy | yes, I guess the audios going through the asterisk is why am having the problem, so I will change the settings and see |
| 15:02.42 | wasanzy | can I set those parameters under the [general] so it affect all the accounts at once? |
| 15:11.42 | *** join/#asterisk Jcook_5xData (~Jcook_5xD@173.162.32.1) |
| 15:14.08 | *** join/#asterisk davlefou (~david@41.225.9.81) |
| 15:15.10 | Jcook_5xData | I am running 1.6.2.18 with DAHDI Version: 2.3.0.1. I have has a few reports of user have a phone ring in their ear when talking to a outside line. anyone else have this problem |
| 15:15.29 | Qwell | umm |
| 15:15.45 | Qwell | leifmadsen was talking about something similar to that the other day. Are you using Polycom phones? |
| 15:15.58 | Jcook_5xData | yes 301 & 501 |
| 15:16.07 | Qwell | old |
| 15:16.17 | Qwell | maybe he'll pipe up when he returns |
| 15:17.13 | Jcook_5xData | sip version 3.1.7.0134 with is the oldest they will support |
| 15:19.40 | carrar | try different ploycom firmwares? |
| 15:19.45 | carrar | polycom |
| 15:19.58 | irroot | Qwell you a smart chap ... what happens if a channel is created and linked to the channels list but autoservice/pbx is never started on it |
| 15:21.03 | Jcook_5xData | not yet. but I can give it a try |
| 15:23.08 | Jcook_5xData | correction 3.1.7.0134 with is the newest they will support |
| 15:23.13 | leifmadsen | Jcook_5xData: I have that problem, but it's not from Asterisk |
| 15:23.32 | leifmadsen | Jcook_5xData: the phone appears to be generating the ringing (this system is also SIP only) -- placing the call on Hold and then Unhold causes the ringing to stop |
| 15:23.40 | leifmadsen | if that doesn't happen, then it's a different issue |
| 15:23.52 | Jcook_5xData | nope that what happening |
| 15:23.56 | carrar | oh thats a feature :) |
| 15:25.03 | Jcook_5xData | well the phone are all sip. we have a pri use digium card |
| 15:25.12 | *** join/#asterisk davlefou (~david@41.225.9.81) |
| 15:25.52 | Jcook_5xData | leifmadsen, where you able to fix it? |
| 15:25.59 | leifmadsen | no that's as far as I got |
| 15:26.04 | leifmadsen | I think it's a polycom firmware bug |
| 15:26.26 | Jcook_5xData | hmmmm, what firmware are you running |
| 15:26.32 | leifmadsen | if I had fixed it, I would have just told you the fix instead of the work around |
| 15:26.36 | leifmadsen | 3.3.1 Rev F |
| 15:27.29 | Jcook_5xData | wow if it is a bug that long standing |
| 15:28.14 | leifmadsen | hard to say, I've only run into it at one location |
| 15:30.20 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 15:30.21 | Jcook_5xData | Just crop up here yesterday. before that no one reported a problem like that. |
| 15:30.50 | leifmadsen | shrugs |
| 15:33.30 | Jcook_5xData | hey... I don't know. The only thing I remember changing was adding was "busy-limit = 1" to the user.conf |
| 15:35.13 | _Corey_ | Jcook_5xData: What's your server load like? |
| 15:36.08 | Jcook_5xData | htop states Load avarage 0.88 0.82 0.72 |
| 15:36.31 | Jcook_5xData | using 161/1000mb |
| 15:36.52 | *** part/#asterisk irroot (~irroot@pbx.distrotech.co.za) |
| 15:39.42 | luminforce | can someone help me confirm a bad message on a sip trace... http://pastebin.com/ijXHQWyw |
| 15:40.42 | luminforce | at line 122 our vendor sends a mid-call re-invite (on hold i think), but at line 188 when they ACK our OK (do they need to?) they send a 5060=5060 for the port number, so OpenSIPS never relays the ACK to Asterisk (it's an invalid formatted message). |
| 15:41.35 | luminforce | asterisk times out after 3 seconds, and thinks the call ends, however the vendor bills the call for another few hundred seconds (until a session timer fires and it's not found0. |
| 15:42.13 | maxgo | hello... anyone that has realtime sip users, has ever seen the invite messages going out from one phone to another, with the destination number in the From: header instead of the reall caller id number? |
| 15:48.43 | *** join/#asterisk seraphie (~erin@75.76.38.159) |
| 15:53.32 | mateu | I'd like to call UnpauseQueueMember in a distributed manner, i.e. for a set of queues spread across multiple asterisk instances (cluster). |
| 15:53.48 | mateu | what might be a design approach to doing that from the dialplan of any giving instance? |
| 15:54.47 | *** join/#asterisk davlefou (~david@41.225.9.81) |
| 15:55.12 | *** join/#asterisk m_tadeu (~quassel@89.181.90.166) |
| 15:56.32 | serafie | mateu: FastAGI |
| 15:56.35 | serafie | https://wiki.asterisk.org/wiki/display/AST/AGICommand_EXEC |
| 15:56.53 | mateu | I'm aware of fastcgi |
| 15:59.41 | mateu | pardon, fastagi |
| 16:02.54 | wasanzy | I have some error again when two people tried to call each other : http://pastebin.com/HkXis4nq please what is it about? |
| 16:03.54 | mateu | isn't sure yet how 'EXEC UnpauseQueueMember' would get called remotely from an agi. |
| 16:04.25 | Qwell | mateu: Forget that it's "remote". That isn't relevant. |
| 16:04.58 | mateu | how does the Unpause get the remote aterisk instance? |
| 16:05.03 | mateu | get to* |
| 16:05.28 | Qwell | again, forget that it's remote. How would an unpause get to an asterisk instance that called a local script? |
| 16:05.29 | wasanzy | any help please? |
| 16:05.31 | Qwell | The answer is no different. |
| 16:05.41 | Qwell | It's just stdin/stdout. |
| 16:06.39 | wasanzy | hmmm |
| 16:07.16 | mateu | <PROTECTED> |
| 16:07.26 | *** join/#asterisk esupports (~yp@74-92-58-226-NewEngland.hfc.comcastbusiness.net) |
| 16:07.37 | *** join/#asterisk andres833 (~andres833@190.26.210.155) |
| 16:08.42 | wasanzy | so who should own that file? asterisk or the system user? |
| 16:08.44 | esupports | Can someone help me with a "your call cannot be completed as dialed" problem? |
| 16:09.53 | *** join/#asterisk binilivi (~jorixFA37@115-64-27-246.static.tpgi.com.au) |
| 16:09.55 | Qwell | esupports: freepbx? |
| 16:10.01 | samandiriel | I have a quick question. with READ(), if someone hits pound, what is the value of the READ variable? Is it #, or blank, or something else? |
| 16:10.11 | esupports | yes; ver 1.6.0 |
| 16:10.15 | serafie | mateu: there are FastAGI modules and libraries. python has a good one, StarPy, which the Asterisk testsuite uses. You could check the test suite out of subversion and have many examples to look at. |
| 16:10.49 | mateu | Qwell, serafie thanks for the advice/suggestions |
| 16:10.50 | *** join/#asterisk davlefou (~david@41.225.9.81) |
| 16:11.55 | wasanzy | kaldemar: are you there? |
| 16:12.14 | serafie | samandiriel: I would venture to guess that the variable would be blank. |
| 16:12.51 | *** join/#asterisk tyman (~tyler@173-12-219-189-Fresno.hfc.comcastbusiness.net) |
| 16:13.12 | samandiriel | serafie: that's what I think too, but wanted to see if anyone knew or had tested |
| 16:14.35 | esupports | Qwell. incoming calls to the trixbox are fine. internal transfers are fine. It won't make external calls |
| 16:14.35 | serafie | You could test it by printing the value to logs (Verbose() or NoOp()) |
| 16:14.46 | Qwell | ugh |
| 16:14.48 | Qwell | #trixbox |
| 16:14.54 | Qwell | ~trixbox |
| 16:14.54 | infobot | well, trixbox is SH1TB0X. Basically a CRAPPY, closed-source distro. STAY AWAY! |
| 16:15.02 | *** join/#asterisk becca_r (~IceChat77@72.165.148.230) |
| 16:15.14 | becca_r | morning all |
| 16:15.35 | wasanzy | is it a good message? Remotely bridging SIP/emma-00000000 and SIP/elartey-0000000, this is rtp debug when a call is made |
| 16:16.23 | esupports | Thanks Qwell |
| 16:16.47 | wasanzy | because the call went through but no one could be heard over the call |
| 16:18.06 | *** join/#asterisk seelen (~seelen@190.29.29.212) |
| 16:18.50 | kaldemar | wasanzy: that means rtp is going (or should be going) directly between the peers. |
| 16:19.09 | wasanzy | oh ok |
| 16:19.25 | wasanzy | but why are they not hearing each other talk? |
| 16:19.58 | wasanzy | should I change the type from "friend" to "peer"? |
| 16:20.36 | kaldemar | no idea. dump network traffic on the peer machines and see if the packets are going. |
| 16:20.50 | kaldemar | no, leave it as is. |
| 16:20.57 | wasanzy | ok |
| 16:21.38 | wasanzy | tcpdump? |
| 16:25.11 | samandiriel | Yah, I figured I would ask first to see if anyone just knew before going thru the bother of t esting |
| 16:25.19 | seelen | Hello, I have a big problem with 4 different servers with asterisk 1.8.5 |
| 16:25.37 | seelen | the asterisk SIP crash |
| 16:26.00 | seelen | when I look netstat -anp | grep 5060 |
| 16:26.14 | kaldemar | wasanzy: yes. also check that there is no firewall blocking the traffic on the machines. |
| 16:26.17 | seelen | the number of packages grow but the estensions don't work |
| 16:26.24 | *** join/#asterisk davlefou (~david@41.225.9.81) |
| 16:27.00 | seelen | no any information on the logs, no errors in the system |
| 16:27.00 | kaldemar | seelen: look at asterisk CLI instead. |
| 16:27.16 | seelen | 2 different OS (debian and centos) |
| 16:27.29 | seelen | kaldemar, no relevant info in the CLI |
| 16:27.34 | wasanzy | ok |
| 16:27.48 | seelen | kaldemar, the extensions looks well with sip show peers |
| 16:28.00 | wasanzy | the port to be allow on the two machine is 5060 for the rtp right? |
| 16:28.15 | seelen | kaldemar, but no data processed by the asterisk in SIP protocol |
| 16:28.22 | kaldemar | seelen: nothing with verbosity and sip debug enabled? |
| 16:29.17 | seelen | kaldemar, nothing with verbosity no test with debug because all system are in production machines and has large number of calls |
| 16:29.45 | kaldemar | wasanzy: no, 5060 is for the SIP. in your pastebins the twinkles use 8000. and it's UDP. |
| 16:31.10 | wasanzy | hmmm, am confuse because in the twinkle port 5060 was specified , then I have to reconfigure the twinkle again |
| 16:31.52 | kaldemar | wasanzy: don't screw it up, you're mixing SIP and RTP. |
| 16:32.35 | esupports | Qwell, all asleep at #trixbox. r u able to help me with some basics? |
| 16:32.49 | wasanzy | I didn't actually specified the port for twinkle, but it asigned automatically |
| 16:32.56 | kaldemar | wasanzy: 5060 is the SIP port as it should be. RTP is a different protocol and twinkle already uses port 8000 for it. |
| 16:33.10 | wasanzy | ok |
| 16:33.22 | kaldemar | 5060 is the default port for SIP. |
| 16:33.52 | wasanzy | so I should dump trafics for port 8000 on the two machines? |
| 16:34.39 | seelen | kaldemar, http://pastebin.mozilla.org/1275952 |
| 16:34.49 | seelen | kaldemar, that is the core show locks |
| 16:35.26 | seelen | kaldemar, http://pastebin.mozilla.org/1275953 |
| 16:35.30 | *** join/#asterisk binilivi (~jorixFA37@115-64-27-246.static.tpgi.com.au) |
| 16:35.49 | seelen | kaldemar, and this the backtrace threads |
| 16:36.40 | seelen | kaldemar, the problem is randmo, but occurs every day |
| 16:37.48 | seelen | kaldemar, to solve i need to kill the process becaus the asterisk don't respond any reload, stop command |
| 16:38.05 | Qwell | esupports: nope |
| 16:38.17 | esupports | :( |
| 16:38.39 | *** join/#asterisk angryuser_laptop (~angryuser@90-156-167-83.reverse.alphalink.fr) |
| 16:41.50 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 16:43.24 | seelen | please i need any clue of this, but I know that the info is not sufficient, how can I know if is a asterisk 1.8 bug? |
| 16:46.47 | seelen | this is the core show locks in other server http://pastebin.mozilla.org/1275995 |
| 16:51.43 | *** join/#asterisk Praise- (~Fat@unaffiliated/praise) |
| 16:53.08 | seelen | with debug enable i have this on the log |
| 16:53.09 | seelen | audiohook.c: Failed to get 160 samples from read factory 0x8ae44b0 |
| 16:54.05 | seelen | full log exit http://pastebin.mozilla.org/1276034 |
| 16:55.53 | *** join/#asterisk esupports (~yp@74-92-58-226-NewEngland.hfc.comcastbusiness.net) |
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| 17:00.41 | *** join/#asterisk becca_r (~IceChat77@72.165.148.230) |
| 17:00.41 | *** join/#asterisk 16WAABI7K (~cS@smtp.master-zone.net) |
| 17:01.34 | *** join/#asterisk Bipul (~bipul@unaffiliated/bipul/x-4918593) |
| 17:02.40 | Bipul | hellow my zoiper account says Fals registeration when i am fixing my asterisk IP |
| 17:04.38 | *** join/#asterisk Praise (~Fat@unaffiliated/praise) |
| 17:06.25 | *** join/#asterisk moy (~moy@CPE002719f00364-CM0026f3a1701d.cpe.net.cable.rogers.com) |
| 17:06.25 | *** join/#asterisk becca_r (~IceChat77@72.165.148.230) |
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| 17:10.14 | wasanzy | kaldemar: you are right, and when I did the dump so many packet passed but no sound could be heard |
| 17:10.34 | *** join/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 17:12.08 | wasanzy | you can see the dump for one machine here: http://pastebin.com/W9z1SAnp |
| 17:15.21 | wasanzy | and the second machine's dump is also here: http://pastebin.com/w6JWpFc2 |
| 17:15.52 | wasanzy | every thing looks ok to me, so I don't know what else should be wrong |
| 17:16.14 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 17:22.09 | pabelanger | seelen: try timerfd or dahdi for timing. pthreads is kinda a pig |
| 17:27.20 | seelen | pabelanger, how can I change it ? |
| 17:28.02 | pabelanger | seelen: download install DAHDI, then recompile asterisk |
| 17:28.09 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 17:28.35 | seelen | pabelanger, how can I use timerfd and not dahdi ? |
| 17:29.48 | seelen | pabelanger, the xtrange thing is that I already have a server with DAHDI, with the same problem |
| 17:29.49 | pabelanger | seelen: The information is listed in CHANGES, basically kernel 2.6.25+ and glibc 2.8 |
| 17:30.20 | *** join/#asterisk vinhdizzo (~vinh@dhcp-053179.ics.uci.edu) |
| 17:30.29 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 17:30.29 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 17:31.48 | *** join/#asterisk Deetz (~pauldittr@174.4.128.244) |
| 17:34.40 | *** join/#asterisk drynish (~drynish@modemcable142.218-37-24.mc.videotron.ca) |
| 17:37.57 | drynish | I have an issue with a digium card fcfxo a x100p compatible card. When I run dahdi_test the result are only 99.93% but asterisk team says taht good result are around 99.975% result and we can expect errors when we are in that situation |
| 17:44.34 | drynish | I don't have any clue right now ;( |
| 17:44.43 | *** join/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 17:45.11 | seelen | pabelanger, how can I change the timinig order of res_timing_dahdi.so, res_timing_timerfd.so, res_timing_pthread.so or i just disable pthread |
| 17:45.16 | seelen | pabelanger, ? |
| 17:45.28 | wasanzy | kaldemar: are you there? |
| 17:45.50 | *** join/#asterisk angryuser_laptop (~angryuser@90-156-167-83.reverse.alphalink.fr) |
| 17:47.58 | Deetz | I've got a stuck channel and when I do channel request hangup nothing happens like it has in the past. Anything I can do other than restarting the server? |
| 17:49.48 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 17:49.48 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 17:50.20 | *** join/#asterisk moy (~moy@173.239.155.74) |
| 17:51.28 | *** join/#asterisk singler (~singler@78-60-139-121.static.zebra.lt) |
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| 17:54.30 | *** join/#asterisk beardy_ (~beardy@unaffiliated/beardy) |
| 18:01.16 | *** join/#asterisk jkroon (~jkroon@dsl-241-241-185.telkomadsl.co.za) |
| 18:03.07 | *** join/#asterisk beardy (~beardy@unaffiliated/beardy) |
| 18:04.19 | *** join/#asterisk darksk1ez (~mhb@darkskiez.ipv6.darkskiez.co.uk) |
| 18:13.28 | WIMPy | Deetz: You should at least mention the channeltype. |
| 18:14.43 | Deetz | SIP sorry |
| 18:15.33 | WIMPy | Ok. And now you could tell us where it is stuck. |
| 18:15.34 | Deetz | WIMPy: sip unregister be about the only option I saw |
| 18:15.40 | Deetz | Dialing into voicemail |
| 18:16.14 | WIMPy | Shouldn't do anything. |
| 18:16.38 | WIMPy | Hmm. Voicemail should timeout by itself. |
| 18:16.55 | Deetz | yeah it looks like there was some issue |
| 18:17.01 | Deetz | as there were .lock files in the inbox and old messages |
| 18:17.15 | Deetz | which looks like a problem that was patched ~7 years ago |
| 18:17.21 | WIMPy | If you have thread debugging enabled, put the output of 'core show locks' on a pastebin. |
| 18:19.46 | *** join/#asterisk JonathanRose (~jonathan@nat/digium/x-nrvfquzofqiewjod) |
| 18:21.15 | Bipul | 16505239916<-- can any one find the location and what kind of number is this ? |
| 18:22.03 | Deetz | mm doesn't look like I do as don't even have command core show locks. Thread debugging is option you'd add before compiling? |
| 18:22.14 | WIMPy | Might be in missisipi if the number is complete. |
| 18:22.24 | WIMPy | yes |
| 18:22.34 | Deetz | Bipul: http://www.localcallingguide.com/ |
| 18:22.40 | WIMPy | It's an option in menuselect. |
| 18:23.36 | *** join/#asterisk bjhaid (~abejide@41.203.81.194) |
| 18:23.37 | Deetz | using yee old freepbx so blah blah. But seems useful as well as menuselect. Mmm deadlocks |
| 18:25.06 | bjhaid | I just configured an asterisk box and I keep getting errors while trying to call into the box through an e1 line, when i check the logs, i see Channel unacceptable 6 |
| 18:25.10 | bjhaid | any suggestions please |
| 18:26.42 | WIMPy | Is it a full E1 or some fractional thing? |
| 18:27.06 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 18:27.06 | WIMPy | Does it have directional channels? |
| 18:28.26 | *** join/#asterisk malcolmd_ (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 18:28.27 | *** mode/#asterisk [+o malcolmd_] by ChanServ |
| 18:37.05 | *** join/#asterisk retentiveboy (~pdugas@74-95-28-33-Atlanta.hfc.comcastbusiness.net) |
| 18:40.16 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 18:47.13 | pabelanger | seelen: You cannot change the order, it is defined within asterisk. Your best to noload => res_timing_phreads.so in modules.conf |
| 18:47.51 | seelen | pabelanger, ok tkns |
| 18:48.26 | seelen | pabelanger, any way to see what is using at this time ? |
| 18:54.20 | pabelanger | seelen: *CLI> timing test |
| 18:57.42 | brainiac | does anyone know how I can get a sip invite to go out on 5070? |
| 18:58.38 | pabelanger | brainiac: set port=5070 for your peer in sip.conf |
| 18:59.26 | seelen | pabelanger, tnks! |
| 19:00.08 | brainiac | I still need to receive on port 5060. Will port=5070 cause a problem with that? |
| 19:02.57 | brainiac | thx |
| 19:06.24 | *** join/#asterisk screenn (~screenn@178.151.86.196) |
| 19:15.24 | pabelanger | brainiac: no, they are 2 different setting. |
| 19:15.32 | *** join/#asterisk irroot (~irroot@197.168.155.9) |
| 19:21.55 | brainiac | pabelanger: I got it working. Thank You. |
| 19:24.05 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 19:24.06 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 19:28.04 | *** join/#asterisk angryuser_laptop (~angryuser@90-156-167-83.reverse.alphalink.fr) |
| 19:29.25 | ssureshot | I've installed my backup analog digium card and have the modules loaded ect..... when I run dahdi_genconf it finds the fxs ports but not the fxo ports.... any reason off the top of your head |
| 19:30.22 | malcolmd | when the driver for the card is loaded does it report the fxo modules? (check your dmesg output) |
| 19:30.23 | WIMPy | Analog is evil. |
| 19:31.01 | ChannelZ | is broken? |
| 19:32.41 | ChannelZ | does dahdi_scan show anything interesting |
| 19:35.21 | seelen | I have many logs like this audiohook.c: Read factory 0x8c12d38 was pretty quick last time, waiting for them. |
| 19:35.30 | seelen | nay clue?, how can I solve this |
| 19:38.39 | ssureshot | ChannelZ: malcolmd: I see exactly whats happening now.... Power (molex) isn't plugged into the card (This is ok as I only need the FXO port) but.... dahdi_genconf is labeling the FXO port as an FXS port... dmesg sees the FXS as module 0, 1 and FXO as module 2,3 ... genconf is seeing module 3,4 as the fxs |
| 19:39.18 | ssureshot | wonder if I just need to manually correct it |
| 19:40.09 | *** join/#asterisk wonderworld (~ww@port-92-201-110-175.dynamic.qsc.de) |
| 19:42.23 | dym | Does anyone have a stable Asterisk HA environment running? Would Asterisk, DRDB, Corosync and heartbeat play well together? |
| 19:45.57 | ChannelZ | ssureshot: no... and FXO port uses FXS signalling, and vice-versa |
| 19:46.09 | ChannelZ | an FXO even |
| 19:46.38 | ssureshot | ChannelZ: so either way I need to have the extra power to the card then? |
| 19:46.48 | ChannelZ | no if all you're using is the FXO ports |
| 19:47.23 | *** join/#asterisk becca_r (~IceChat77@72.165.148.230) |
| 19:47.28 | ChannelZ | The FXS needs the extra juice to power phones |
| 19:47.59 | ssureshot | ChannelZ: let me post some out put and see if you can point me in the right direction with the config.. |
| 19:48.15 | ChannelZ | What I'm saying is your config is probably right, it's just you that are confused |
| 19:48.45 | ChannelZ | dahdi_scan should show what the ports actually are |
| 19:50.18 | ChannelZ | However in your DAHDI config, an FXO port will be "fxsks" and an FXS port will be "fxoks" for instance |
| 19:51.11 | seelen | any advance with this bug https://issues.asterisk.org/view.php?id=19234, same simptoms in 4 servers with asterisk 1.8.5 |
| 19:51.37 | ssureshot | ChannelZ: so what looks like an FXO port is actually an FXS port in dahdi ? |
| 19:52.00 | ssureshot | http://pastebin.com/qzsa3vAy |
| 19:52.00 | ChannelZ | No. An FXO port is an FXO port, but it uses FXS signalling. |
| 19:52.15 | ChannelZ | An FXS port is an FXS port, but uses FXO signalling. |
| 19:58.12 | ssureshot | Channelz: IC,, I'm kinda confused I guess,,, in this old zap config,,,, http://pastebin.com/ZwurHVV0 ... is channel 97 FXO or FXS |
| 20:00.26 | ChannelZ | FXS |
| 20:01.39 | *** join/#asterisk ferdna (~yup@cpe-67-10-220-35.elp.res.rr.com) |
| 20:02.49 | ssureshot | ChannelZ: so it's opposite what the configuration states then |
| 20:03.13 | ChannelZ | well OK if that's what you want to say :) |
| 20:03.40 | ChannelZ | It's more so that the configuration is looking at it from a different point of view, but yes essentially it's opposite |
| 20:03.47 | *** join/#asterisk joren (~j@173-12-163-81-oregon.hfc.comcastbusiness.net) |
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| 20:05.52 | ssureshot | ChannelZ: I read this all wrong then,, and was hoping I could use this card in this server but since it's using FXS I can't use it.. I have to purchase the proper module from xorcom for the astribank I am going to use.. |
| 20:06.07 | ssureshot | So I need to understand this better so I don't order the wrong crap |
| 20:06.54 | joren | Hey, does anyone know if it's possible to force asterisk to contact an iax server through a specific interface instead of the default route? I guess that might not make allot of sense, and I'm probably go about it wrong, but my isp split a chunk of bandwidth to a spare IP that I want to dedicate to voip communication. |
| 20:07.06 | ssureshot | <----- goes to RTFM |
| 20:07.21 | irroot | joren "ip route help" |
| 20:08.02 | ssureshot | oh and thanks for the help ChannelZ |
| 20:08.10 | joren | I was imaging it could be done with that, I'll mess around with that for a bit |
| 20:08.54 | WIMPy | ssureshot: Did I get something wrong, or aren't both of your cards 2fxs+2fxo? |
| 20:10.42 | ssureshot | WIMPy: yes and no.... I have an Astribank for the main T1 Interface but I also have need of one analog card that communicates with our PA system |
| 20:11.06 | ssureshot | thats where the Digium card comes in |
| 20:11.16 | WIMPy | ssureshot: Understood. I was under the impression you were trying a card from your backup? |
| 20:11.58 | ssureshot | WIMPy: yes it is the exact same card as the primary... but my server has no molex power adapter so I'm kinda screwed |
| 20:12.33 | WIMPy | Or just missing an adapter? |
| 20:13.41 | ssureshot | It doens't have standard molex at all.. it's not missing it just never had lol.. dell poweredge 1750.. |
| 20:14.22 | ssureshot | I called dell and asked if they had some type of converter but no luck |
| 20:14.38 | WIMPy | I'd be surprised if it doesn't have 12V somewhere. |
| 20:14.54 | *** join/#asterisk jeffik (~chatzilla@TOROON63-1176243424.sdsl.bell.ca) |
| 20:15.15 | ssureshot | yeah I'd have to tap the power supply and would rather not rig a production server if tyou know what I mena |
| 20:16.03 | WIMPy | Sometimes you have to correct some mistakes the manufacturer made :-) |
| 20:16.23 | ssureshot | but I might still look into it .. as I don't feel like spending 1k on these parts I need |
| 20:16.55 | ssureshot | ha .. I hear ya |
| 20:17.08 | WIMPy | How many analog ports do you need? Only one for a tlephone? |
| 20:17.23 | WIMPy | Well, "telephone". |
| 20:20.46 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 20:24.51 | nmjnb | I tried changing the password for the Asterisk manager but it still accepts the old password. I changed pass in /etc/asterisk/manager.conf and /etc/amportal.conf |
| 20:25.11 | nmjnb | according to the documentation on freepbx |
| 20:25.15 | atheos | nmjnb did you do manager reload ? |
| 20:25.29 | nmjnb | no? |
| 20:25.34 | nmjnb | but I rebooted |
| 20:25.51 | atheos | that should have done the trick |
| 20:26.07 | nmjnb | well, I'm still able to log in with freepbx/fpbx |
| 20:28.31 | *** join/#asterisk lusty (~Adium@124-171-3-105.dyn.iinet.net.au) |
| 20:28.34 | mickecarlsson | nmjnb that user/password is for FreePBX not manager |
| 20:28.35 | ssureshot | WIMPy: yes I only need one line |
| 20:29.04 | *** join/#asterisk WiretapWork (~wiretap@unaffiliated/wiretap) |
| 20:29.17 | mickecarlsson | nmjnb: that backdoor was removed in FreePBX 2.9 |
| 20:29.19 | nmjnb | mickecarlsson: ok, so what is the manager, and how do I change freepbx password? |
| 20:29.42 | mickecarlsson | Please swith to #freepbx and as the question there |
| 20:29.54 | mickecarlsson | s/as/ask |
| 20:29.59 | WIMPy | ssureshot: A BRI card and a POTS adapter should be a lot cheaper than a direct solution. |
| 20:30.06 | dym | Does anyone have a stable Asterisk HA environment running? Would Asterisk, DRDB, Corosync and heartbeat play well together? |
| 20:30.30 | nmjnb | mickecarlsson: ok, thanks |
| 20:33.07 | *** join/#asterisk Dibbler (~Dibbler@87-194-103-72.bethere.co.uk) |
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| 20:35.48 | *** part/#asterisk luminforce (~luminforc@rrcs-71-42-183-54.sw.biz.rr.com) |
| 20:46.47 | ferdna | i need a directory... when ever someone calls me asterisk sends name stored in a phone directory... is this possible? |
| 20:47.33 | WIMPy | Sure. Call an AGI or do a database query or whatever. |
| 20:48.35 | ssureshot | ferdna: you can use the built in directory that reads the vm file I believe that app is Directory |
| 20:48.59 | WIMPy | For local extensions. |
| 20:49.31 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 20:49.45 | ssureshot | true,,I assumed thats what was meant.. |
| 20:49.45 | ferdna | WIMPy, yeah i dont need local extensions... |
| 20:50.15 | ssureshot | ha! I'm too new to help |
| 20:50.18 | ferdna | ssureshot, http://www.voip-info.org/wiki/view/Asterisk+cmd+Directory |
| 20:50.23 | WIMPy | Local extension should have their caller ID set. |
| 20:51.18 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 20:53.28 | *** join/#asterisk nightrid3r (~nightrid3@91.176.0.177) |
| 20:56.11 | ferdna | WIMPy, where do i set them? in extensions.conf? |
| 20:56.30 | WIMPy | Set what? |
| 20:56.43 | ferdna | [14:49] <WIMPy> Local extension should have their caller ID set. |
| 20:57.01 | ChannelZ | sip.conf |
| 20:57.04 | WIMPy | In the peers definitions. |
| 20:57.17 | ferdna | callerid="caller id info" |
| 20:57.18 | WIMPy | sip.conf, iax.conf, etc. |
| 20:57.20 | ferdna | right there? |
| 20:57.28 | WIMPy | yes |
| 20:58.01 | ferdna | WIMPy, i have this format in there: |
| 20:58.14 | ferdna | callerid="Ext XXXXX" <XXXX> |
| 20:58.18 | ferdna | is that correct? |
| 20:58.28 | WIMPy | yes. |
| 20:59.11 | ferdna | ok thanks =) |
| 21:00.00 | dym | Is there some sort of Hybrid phone, that can use a SIP and some ISDN line and negotiate between them? (as in maybe relay calls, etc?) |
| 21:00.13 | dym | The phone directly has to support both |
| 21:00.37 | WIMPy | I think the was a Samsung thing. |
| 21:01.01 | dym | also call redirection? as in i get a call via the sip line and redirect it to another isdn phone? |
| 21:01.23 | WIMPy | i don;t think a phone will do that. |
| 21:01.40 | WIMPy | But some plastic router could. |
| 21:02.30 | WIMPy | Why not let Asterisk do that? |
| 21:02.34 | dym | Okay, following scenario: |
| 21:02.57 | dym | I have a phone infrastructure based on isdn |
| 21:03.01 | dym | number 123-0 - 123-5 |
| 21:03.10 | dym | now i port out 1230 to an online sip trunk |
| 21:03.27 | dym | and the phone that used to be 123-0 needs to be reachable from the outside |
| 21:03.43 | dym | i thought of maybe having a sip account on it and connecting it to the online asterisk |
| 21:03.54 | dym | but then again it wont be able to relay calls to the other internal phones |
| 21:04.01 | WIMPy | Ah, no Asterisk at that location? |
| 21:04.07 | dym | nah, online online |
| 21:04.16 | dym | sadly |
| 21:04.22 | dym | any idea? |
| 21:04.47 | dym | calls from the 123-0, which is now "outside" have to be redirected to the former 123-0 which is now inside. |
| 21:04.49 | WIMPy | I don't see how you could connect a local ISDN PBX and a remote SIP PBX, unless the ISDN PBX supports SIP itself. |
| 21:05.13 | dym | so only way would be assigning former 123-0 123-5 or sth |
| 21:05.25 | WIMPy | Or you have a spare port that could be connected to a gateway. |
| 21:05.25 | dym | so the outside 123-0 can be relayed to 123-5 |
| 21:05.39 | dym | on the isdn pbx? |
| 21:05.47 | WIMPy | Where's the point in that? |
| 21:05.52 | WIMPy | yes |
| 21:06.10 | dym | is there gateways for that? |
| 21:06.15 | dym | in which form? |
| 21:06.18 | WIMPy | Sure |
| 21:06.27 | dym | got an example? |
| 21:06.28 | WIMPy | Plastic routers. |
| 21:06.33 | WIMPy | AKA IADs. |
| 21:06.52 | dym | Do you have a link? |
| 21:07.04 | WIMPy | AVM, SMC, Sphairon, Siemens, Draytek, D-Link, ... you name it :-) |
| 21:08.01 | WIMPy | But make sure it can do that without being used as router. |
| 21:08.33 | WIMPy | Some models won't even try to do SIP when the uplink isn't up. |
| 21:08.57 | WIMPy | It's often called ATA mode. |
| 21:11.46 | ferdna | WIMPy, one more thing... type=friend... what should i have in there? peer or friend? |
| 21:12.16 | WIMPy | For what? |
| 21:12.27 | ferdna | for the extensions |
| 21:13.06 | WIMPy | Unless you authenticate them via their IP, you need friend. |
| 21:13.29 | ferdna | oh ok =)... thanks |
| 21:15.10 | WiretapWork | I just gave the new guy 'the keys' to the PBX |
| 21:15.22 | WiretapWork | wonder how long before I get the phonecall 'I broke it, halp' |
| 21:15.36 | dym | :D |
| 21:15.41 | *** join/#asterisk lusty (~lusty@lusty.unix.com.au) |
| 21:15.57 | WIMPy | WiretapWork: Did you already announce your price? |
| 21:16.23 | *** part/#asterisk lusty_ (~Adium@124-171-3-105.dyn.iinet.net.au) |
| 21:16.51 | dym | WIMPy: Could you link me to a product that suites your description? Cant really find one ad hoc |
| 21:17.05 | beek | WiretapWork: ... or your favorite brands of brew? |
| 21:17.10 | WiretapWork | lol |
| 21:17.21 | WIMPy | dym: Just something that converts SIP to S0? |
| 21:17.36 | WiretapWork | WIMPy: not yet, but I will be charging standard sysadmin rates of NZ$125/hr |
| 21:17.48 | dym | Well yeah. I need to be able to negotiate calls inbetween with an external sip account |
| 21:17.51 | WiretapWork | just set up a weekly backup sched |
| 21:17.51 | dym | @ WIMPy |
| 21:18.11 | WIMPy | goes to xe |
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| 21:21.22 | WiretapWork | I didn't give him the root login |
| 21:21.27 | WiretapWork | only the login to the asterisk console user :P |
| 21:22.50 | WIMPy | dym: I can tell you things that are too big (expensive) for that purpose. Where can you buy? |
| 21:23.20 | dym | mhh |
| 21:23.37 | dym | shouldnt be too expensive and compatible with a simple ISDN port |
| 21:23.45 | dym | buy - well... online? |
| 21:24.33 | WIMPy | dym: But where? I gouess the chance to find such a thing in an US shop are rather slim. |
| 21:24.53 | dym | .de |
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| 21:25.52 | WIMPy | Ok. Then search for some Fritz!Box. The 7270 will do, but a smaller model would be more appropriate. |
| 21:26.00 | dym | geez |
| 21:26.10 | dym | thats an entire router/dialup/etc appliance |
| 21:26.19 | WIMPy | You can also watch the local electronic store for bargains. |
| 21:26.37 | dym | But also thats not quite what im looking for i guess. |
| 21:26.45 | WIMPy | Reichelt had Samsung things for 12EUR some month ago. |
| 21:26.47 | dym | Say I had an ISDN pbx locally |
| 21:27.13 | dym | and i needed to be able to connect the pbx to an external sip account |
| 21:27.19 | WIMPy | The FBs will convert from SIP to ISDN and vice versa. |
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| 21:27.47 | dym | mhh |
| 21:27.50 | WIMPy | That's what happens if you order an ISDN "line" from KDG or 1&1. |
| 21:28.18 | dym | im aware of that, but i'd be able to negotiate calles inbetween both technologies |
| 21:28.22 | dym | calls* |
| 21:28.47 | dym | can i pm? maybe its more clear in german |
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