| 00:02.44 | leifmadsen | ok, so use what it recommends :) |
| 00:10.07 | bintut | leifmadsen: yes, i just did and it's now loaded.. now, it's time to test it.. ;) |
| 00:17.38 | *** join/#asterisk BuenGenio (~Gene@059148208218.ctinets.com) |
| 00:26.52 | *** join/#asterisk DocAwesome (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 00:26.52 | *** mode/#asterisk [+o DocAwesome] by ChanServ |
| 00:28.38 | bintut | gtg now.. thanks to all.. ;_) |
| 00:28.42 | bintut | ;) |
| 00:28.44 | bintut | waves |
| 00:35.35 | joker2u | . |
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| 01:03.13 | rixon | why is Digium no longer selling their skype channel driver? |
| 01:03.38 | rixon | tomorrow is the last day it's available for activation. |
| 01:03.47 | rixon | I wonder if I should hurry up and get it... |
| 01:04.29 | SunTsu | http://slashdot.org/story/11/05/24/2010222/Microsoft-Kills-Skype-For-Asterisk |
| 01:08.00 | rixon | thanks suntsu. :-( |
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| 01:39.02 | WIMPy | @@*$!@^*! |
| 01:39.19 | WIMPy | WTF does a loopback switch not work in realtime? |
| 01:41.57 | WIMPy | Ok. It looks like it does, if you don't use a pattern. |
| 01:42.18 | carrar | mmm loopback |
| 01:43.19 | WIMPy | Looks like the solution to favourite issue. |
| 01:43.52 | carrar | solution to everything |
| 01:44.05 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 01:44.16 | WIMPy | Damn. That's it. |
| 01:44.47 | WIMPy | Why didn't anyone ever mention it when I ranted about not being able to edit the dialed number in realtime? |
| 01:49.22 | DocAwesome | goddamn slashdot -- I doubt MS had anything to do with it |
| 01:49.37 | DocAwesome | Skype has been dragging their feet with Asterisk integration ever since they signed the deal |
| 01:49.44 | DocAwesome | (at least that's what it has looked like) |
| 01:49.53 | DocAwesome | anyways, bedtime |
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| 02:02.23 | WIMPy | Damn. That's so F***ing hot. |
| 02:03.06 | WIMPy | You are all losers. You didn't tell me! |
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| 02:37.32 | x86 | wow, 1.8.5.0 really changed things up from the 1.8.1 that I was running :) |
| 02:37.40 | x86 | no more menuselect... kinda nifty |
| 02:41.55 | x86 | oh menuselect is there, just had to configure first |
| 02:43.20 | ChannelZ | mmmyeah |
| 02:43.55 | WIMPy | You always had to configure first. Depending on the result you get different menuselects. |
| 02:46.26 | ChannelZ | "Same as it ever was.... same as it ever was..." |
| 02:49.37 | x86 | yeah it's been a while :) |
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| 03:22.58 | aaroneous | looking for audiocodes MP-11X firmware.. their website isn't terribly useful.. anyone around who could help a brotha out? |
| 03:25.08 | aaroneous | (also looking for related files required during update) |
| 03:30.39 | *** join/#asterisk batphone (~boo@unaffiliated/batphone) |
| 03:31.00 | batphone | will someone please help me configure my tdm400p? |
| 03:31.29 | batphone | what is this dahdi crap, and what happened to ZAPTEL? |
| 03:31.48 | batphone | tough crowd. |
| 03:32.07 | WIMPy | zaptel was renamed to dahdi. |
| 03:32.16 | batphone | seriously. which module color is fxs... |
| 03:32.47 | batphone | and thats.. it? |
| 03:33.04 | batphone | what modules do i load for this tdm card? |
| 03:33.15 | WIMPy | Apart from some updates since then, yes. |
| 03:33.25 | batphone | wctdm is there but ztcfg/dahdi_cfg -vvv does not see it with that module loaded |
| 03:33.28 | batphone | there is power to the card. |
| 03:34.18 | batphone | this is for my son.. |
| 03:34.55 | batphone | i have a couple of polycoms and some wireless routers running linux that are getting hooked up to this system in my house. |
| 03:35.12 | batphone | IF YOU ARE A BILL COLLECTOR, HANG UP. |
| 03:35.17 | batphone | hehe ;D |
| 03:35.24 | WIMPy | see dahdi_sacn, dahdi_hardware and dahdi_genconf. |
| 03:35.48 | batphone | is that going to tell me what module to load? |
| 03:36.03 | WIMPy | s/sacn/scan/ |
| 03:36.07 | batphone | ah, i see i didnt configure the RTC |
| 03:36.22 | batphone | its loading the dahdi dummy driver for it. |
| 03:36.40 | batphone | thus is ubuntu |
| 03:37.14 | batphone | all dahdi_scan shows is the dummy driver |
| 03:37.37 | batphone | dahdi hardware complains about wctdm being needed |
| 03:37.46 | batphone | instead of this netjet drive |
| 03:38.04 | ChannelZ | netjet? |
| 03:38.35 | batphone | # dahdi_hardware |
| 03:38.36 | batphone | driver should be 'wctdm' but is actually 'netjet' |
| 03:39.00 | ChannelZ | I don't know where that came from |
| 03:39.14 | ChannelZ | Is your card dead? What does lspci show you? |
| 03:39.19 | batphone | hah, so, rmmod netjet, ok |
| 03:39.24 | batphone | then modprobe wctdm |
| 03:39.28 | batphone | and that error goes away |
| 03:39.33 | batphone | no i see it |
| 03:39.43 | ChannelZ | so you just didn't load the drivers |
| 03:39.43 | *** part/#asterisk Tas01019 (~IceChat77@c-76-106-227-203.hsd1.fl.comcast.net) |
| 03:39.45 | batphone | 04:02.0 Communication controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface |
| 03:39.48 | ChannelZ | /etc/init.d/dahdi start usually |
| 03:40.02 | ChannelZ | Wait, I thought you said you had a TDM400? |
| 03:40.17 | batphone | hmm |
| 03:40.18 | batphone | i did too |
| 03:40.24 | ChannelZ | Is that a clone from China or something |
| 03:40.27 | batphone | ah, this is an old card |
| 03:40.29 | batphone | they look alike |
| 03:40.36 | batphone | so anyways. should still work. |
| 03:40.36 | WIMPy | Isn't that Tioger thing a bog standard HFC-S card? |
| 03:41.44 | batphone | dude i dont care about high fructose corn syrup ok? |
| 03:41.53 | batphone | so quit trying to confuse me. |
| 03:42.55 | batphone | is this thing on? |
| 03:43.08 | ChannelZ | Is it an old X100? http://www.voip-info.org/wiki/view/Asterisk@Home+Handbook+Wiki+Chapter+5 |
| 03:43.10 | WIMPy | Ok, looks like they also did FXO/FXS cards. |
| 03:43.18 | batphone | i can look at my other cards |
| 03:43.21 | batphone | but i will have to reboot |
| 03:43.32 | ChannelZ | oh.. well actually someone mentions that for the 400 too |
| 03:43.50 | batphone | i would have this on the old pbx but it has one of those old intel boards with the clip-on fan that breaks |
| 03:43.53 | WIMPy | What? |
| 03:44.10 | batphone | WIMPy: nothing. |
| 03:44.33 | batphone | i had a hell of a time getting the card in there |
| 03:44.41 | WIMPy | No, ChannelZ: What is mentioned? |
| 03:44.44 | batphone | i would take a pic |
| 03:45.18 | ChannelZ | "The TDM cards are recognized as: (don't ask me why) Network controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface" |
| 03:45.19 | batphone | would it just be easier to put in a modem |
| 03:45.56 | WIMPy | Maybe, but it won;t work :-) |
| 03:46.01 | batphone | WIMPy: i thought i had put a TDM400P in there. i have a bunch of cards laying around and thought they were all the same. |
| 03:46.17 | ChannelZ | Well it apparently might be a 400 |
| 03:46.28 | ChannelZ | wctdm would be the driver for it in any case |
| 03:46.41 | ChannelZ | so load it and see if dahdi_scan sees it |
| 03:46.43 | batphone | the one in my hand definetly says Quad TDM PCI\nWildcard TDM400P |
| 03:46.52 | batphone | i cant se the bottom of this card in the pc |
| 03:47.08 | batphone | . |
| 03:47.10 | batphone | hah! |
| 03:47.29 | WIMPy | unload/blacklist the netjet driver and try the wctdm. |
| 03:47.44 | batphone | i had to bend my graphics card out of the way. its a TDM400P registering with the PCI as a Tiger 300 |
| 03:47.59 | batphone | WIMPy: i did |
| 03:48.17 | WIMPy | And what happened? |
| 03:48.18 | batphone | i think i need to configure the span now |
| 03:48.46 | batphone | WIMPy: now i can see the card with dahdi_hardware comand |
| 03:49.54 | batphone | ChannelZ: dahdi_scan does not see it |
| 03:50.27 | batphone | i miss seeing the green lights come on |
| 03:50.44 | batphone | * * * * |
| 03:51.13 | batphone | you used to just be able to stop zaptel, reload the modules in order, and ztcfg -vvv |
| 03:51.31 | batphone | i had a great gentoo init script at some point. this sucks. i feel so behind the times. |
| 03:53.32 | batphone | to be precise, i feel more like a beggar. i think i will shut down Xinerama so I can use the hardware accleration on my card to plan on just one screen. |
| 03:53.43 | batphone | you know, because Xinerama doesn't work with Twinview... |
| 03:54.20 | batphone | then i will play some enemy territory on a botty server |
| 03:54.31 | batphone | and be late to work in the morning, again |
| 03:55.13 | batphone | i'll leave work early, again |
| 03:55.34 | batphone | and come ask you more questions. |
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| 04:04.08 | echinos | does 1.8 not use s and n for start and next in extensions.conf? |
| 04:04.39 | WIMPy | What's that supposed to be? |
| 04:05.05 | WIMPy | Or are you mixing up extensions and priorities? |
| 04:05.42 | echinos | I thought you could use those on extensions |
| 04:06.16 | echinos | it was OK in the previous version of asterisk, but in 1.8 it can't find where to go in the contexts |
| 04:06.20 | WIMPy | Explain |
| 04:06.33 | echinos | IE exten => 4,s,<do stuff> |
| 04:06.48 | echinos | exten => 4,n,<do next thing> |
| 04:06.53 | echinos | etc. |
| 04:07.06 | WIMPy | That has never been valid, AFAIK. |
| 04:07.17 | WIMPy | You always start with 1. Then you can use n. |
| 04:07.28 | echinos | well, I just found a quirk then :/ |
| 04:07.41 | echinos | that would explain it :) |
| 04:08.03 | WIMPy | There is an s etension, i.e. s,1,DoSomething. |
| 04:09.47 | echinos | as in, what you get when you come in on that context and is the default extension? |
| 04:09.57 | echinos | Much better, all fixed. |
| 04:10.10 | echinos | tyvm |
| 04:10.23 | WIMPy | It's not a default it is the no extension given extension. |
| 04:10.31 | echinos | ah, ok. |
| 04:10.41 | echinos | Next up, get my SPA connected :) |
| 04:10.48 | echinos | thanks again, off to bed for me. |
| 04:11.02 | WIMPy | Good plan. |
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| 05:13.12 | ChannelZ | He must dream in pig latin |
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| 06:41.57 | schmidts | good morning |
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| 06:51.08 | syntaxx | im trying asterisk + ldap integration however im getting ERROR[4383] res_config_ldap.c: No variable metric in entry '?'(category: general) for file 'sip.conf' any idea? |
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| 06:53.16 | ChannelZ | misconfiguration |
| 06:53.26 | ChannelZ | thanks, I'll be here all week! |
| 06:54.38 | ChannelZ | probably a type-o, missing quote, or something |
| 06:56.21 | syntaxx | ChannelZ, is that for me? |
| 06:56.31 | ChannelZ | yes |
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| 06:57.17 | syntaxx | ChannelZ, im not sure if its a misconfiguration cause any word i typed there im getting the same thing except for the general |
| 07:00.55 | ChannelZ | shrugs - I don't know crap about LDAP sorry |
| 07:01.16 | ChannelZ | but the '?' suggests some sort of parsing error to me |
| 07:01.34 | syntaxx | hm ok thanks |
| 07:07.45 | ChannelZ | you could pastebin your res_ldap.conf for people to look over |
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| 07:42.44 | batphone | need help with a tdm400p and config |
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| 07:51.09 | ChannelZ | still |
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| 07:58.27 | batphone | still |
| 07:58.33 | batphone | hey, i got a phone registerd ;P |
| 07:58.56 | batphone | which means i wrote a sip.cfg, a phone.cfg, extensions.conf, asterisk.conf, sip.conf |
| 07:59.08 | batphone | i just dont remember the trick to these zaptel/dahdi files |
| 07:59.28 | batphone | i dont have any reason to think the card is bad |
| 08:06.13 | batphone | maybe its a permissions issue somewhere |
| 08:06.30 | batphone | for whatever reason, dahdi_hardare detects the card, but dahdi_genconf does not write the config |
| 08:08.42 | batphone | ah |
| 08:08.46 | batphone | had the signalling backwards |
| 08:08.53 | batphone | lets see if asterisk approves ;P |
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| 08:09.27 | batphone | :D |
| 08:09.33 | batphone | time for a dial plan!!! |
| 08:09.39 | batphone | thanks for all your help ;) |
| 08:09.50 | batphone | but not really |
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| 09:11.04 | *** join/#asterisk jacc0 (~jacc0@178.231.16.197) |
| 09:11.07 | jacc0 | hi all |
| 09:11.29 | jacc0 | How do I add a header like : Outgoing calls required adding <param name="extension-in-contact" value="xxxx"/> to the SIP profile |
| 09:11.33 | jacc0 | ? |
| 09:24.38 | jacc0 | sipaddheader(param name="extension-in-contact" value="${EXTEN}") maybe? |
| 09:28.57 | *** join/#asterisk Rico29 (~rico@sar-b123.olm.fr) |
| 09:29.03 | Rico29 | hello all |
| 09:29.04 | jacc0 | anyone? |
| 09:29.15 | jacc0 | I have problems with outgoing calls to a mitel |
| 09:29.22 | jacc0 | hi rico29 |
| 09:29.34 | Rico29 | i went here a couple days ago, for a sip stack crash problem |
| 09:30.33 | jacc0 | the mitel returns a 404 |
| 09:31.46 | Rico29 | schmidts and wdoekes2 recommend me to do a core show locks, and to get a gdb trace |
| 09:31.51 | Rico29 | I've done that |
| 09:31.56 | Rico29 | what can I do now ? |
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| 09:37.50 | jacc0 | sorry; lost my internet connection |
| 09:37.58 | jacc0 | did anyone answer? |
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| 09:48.11 | jacc0 | sorry; lost my internet connection |
| 09:48.15 | jacc0 | did anyone answer? |
| 10:00.32 | *** join/#asterisk jacc0 (~jacc0@94.157.18.47) |
| 10:00.39 | jacc0 | sorry; lost my internet connection again |
| 10:00.42 | jacc0 | did anyone answer? |
| 10:01.29 | Faustov | no |
| 10:03.05 | Rico29 | can anyone help me ? |
| 10:03.19 | Rico29 | i've posted on asterisk issue tracker :https://issues.asterisk.org/jira/browse/ASTERISK-18173 |
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| 10:11.37 | wasanzy | hello all |
| 10:12.40 | wasanzy | when programming an IVR system, which of the files is best to define your dialplan? is it extension.conf or the extension.ael? |
| 10:13.16 | schmidts | Rico29 i will take a look at your case |
| 10:13.25 | Rico29 | thanks a lot schmidts |
| 10:15.05 | schmidts | Rico29 i have to search but thats a know bug |
| 10:15.19 | Rico29 | this is a good thing |
| 10:15.35 | Rico29 | i'm waiting for your reply, or any indications... |
| 10:16.54 | wasanzy | please any one to advice the best practice of my question? |
| 10:17.42 | schmidts | Rico29 i am not 100% sure but i think this https://reviewboard.asterisk.org/r/1313/ is the solution for your problem, do you know how to apply a patch (the downloaded diff) to asterisk? |
| 10:18.26 | Rico29 | i've already done that, but can you remember me the correct syntax ? |
| 10:19.24 | schmidts | normally it should work with patch -p0 but it depends on your asterisk version and which this patch was made against |
| 10:21.53 | schmidts | Rico29 the easiest way would be to check out the latest 1.8 version directly and just use this, cause this patch is allready in the source tree |
| 10:26.11 | wasanzy | hmmm |
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| 10:32.07 | Rico29 | schmidts, ok, so the 1.8.5.0 should fix my problem ? |
| 10:32.28 | Rico29 | or may I take the latest svn 1.8 branch ? |
| 10:33.01 | schmidts | the latetest svn 1.8 branch i think this patch will come in 1.8.6 ;) |
| 10:33.17 | Rico29 | ok |
| 10:33.35 | Rico29 | and no need to apply the patch in this svn release, file is already patched ? |
| 10:36.27 | schmidts | if you get something above revision 329299 yes it will ;) |
| 10:37.15 | Rico29 | rev 329526 |
| 10:37.26 | schmidts | looks fine ;) |
| 10:37.30 | Rico29 | ok, i'll try to recompile this evening |
| 10:37.32 | Rico29 | thanks a lot |
| 10:37.39 | schmidts | your welcome |
| 10:37.45 | Rico29 | :) |
| 10:42.51 | Rico29 | schmidts > in fact, i just recompiled 1 min ago |
| 10:43.06 | Rico29 | so now, wait'n'see... |
| 10:43.44 | schmidts | :D |
| 10:44.05 | Rico29 | hope it will do the trick |
| 10:47.20 | Rico29 | why dont'I have the "core show version" command in my CLI ? |
| 10:47.38 | schmidts | dont know, it should be there |
| 10:47.43 | Rico29 | mmh |
| 10:48.12 | Rico29 | not important... |
| 10:48.20 | Rico29 | [root@SRV-IPOP1 tmp]# asterisk -r |
| 10:48.20 | Rico29 | Asterisk SVN-branch-1.8-r329471, Copyright (C) 1999 - 2011 Digium, Inc. and others. |
| 10:50.03 | *** join/#asterisk m_tadeu (~quassel@89.181.60.178) |
| 10:50.04 | *** join/#asterisk aberrios (~aberrios@195.171.4.82) |
| 10:52.24 | m_tadeu | hi, I'm having a small trouble here. I'm using SIP for calls, and every once in a while, RTP packages flow without sound(both ways). Has this happened to anyone here? Or anyone has an idea on what is going on? |
| 10:59.52 | *** join/#asterisk eject_ck (~eject_ck@62.205.134.210) |
| 10:59.56 | eject_ck | Hi all |
| 11:00.00 | eject_ck | What does it mean ? |
| 11:00.01 | eject_ck | [2011-07-26 13:57:26] WARNING[16490]: res_rtp_asterisk.c:424 create_new_socket: Unable to allocate RTP socket: Address family not supported by protocol |
| 11:00.02 | eject_ck | [2011-07-26 13:57:26] WARNING[16490]: chan_h323.c:978 __oh323_rtp_create: Unable to create RTP session: Address family not supported by protocol |
| 11:00.02 | eject_ck | <PROTECTED> |
| 11:03.32 | *** join/#asterisk MDesade (~chatzilla@ip24-251-93-137.ph.ph.cox.net) |
| 11:05.42 | *** join/#asterisk den512 (~dradon@carbon.gonicus.de) |
| 11:10.28 | kaii | eject_ck: please enable sip debug and put a full log of the call als well as all relevant parts of your sip.conf on pastebin. also, we would need to know your asterisk version |
| 11:10.30 | schmidts | eject_ck this means that the client which tries to talk to your asterisk doesnt offers a codec which asterisk supports, or which is allowed for this client |
| 11:10.41 | kaii | eject_ck: you said you use SIP, but your log says "h323" |
| 11:11.01 | wasanzy | when programming an IVR system, which of the files is best to define your dialplan? is it extension.conf or the extension.ael? |
| 11:11.06 | kaii | oops mixed up two peoples question ;) sry |
| 11:11.22 | kaii | m_tadeu: please enable sip debug and put a full log of the call als well as all relevant parts of your sip.conf on pastebin. also, we would need to know your asterisk version |
| 11:11.40 | kaii | m_tadeu: most possibly this is because of NAT between the two endpoints |
| 11:12.07 | schmidts | wasanzy it only depends on what you can handle easier, there are only few limitations between both |
| 11:12.29 | schmidts | wasanzy IMHO you should try to do "programming logic" outside asterisk in an AGI |
| 11:13.45 | wasanzy | but in ur opinion, which file is best to use considering their limitations? |
| 11:14.01 | m_tadeu | kaii: well I was testing calls between the same 2 phones. about 10%of the calls have no sound, always withe the same conditions, same peers, etc. The asterisk version is 1.8.5 |
| 11:14.22 | schmidts | wasanzy i use the conf file but most of my servers are still 1.2 :D |
| 11:14.42 | wasanzy | ok |
| 11:14.48 | wasanzy | thank you |
| 11:15.57 | wasanzy | now am about to setup the IVR in ss7 network, do you have any documentation you can recommend? |
| 11:18.03 | *** join/#asterisk puzzled (~patrick@puzzled.xs4all.nl) |
| 11:18.06 | kaii | m_tadeu: does the same happen when you put "directmedia=no" in each sip peers configuration? |
| 11:20.33 | m_tadeu | kaii: let me check what directmedia does...sounds something like canreinvite, but should do something different |
| 11:24.31 | schmidts | m_tadeu not its exactly the same ;) |
| 11:28.29 | m_tadeu | schmidts: I was reading about it now. but still...the end is the same, right? or asterisk is in the path or not. would apreciate a better distinction between directmedia and canreinvite |
| 11:31.12 | *** join/#asterisk tzafrir_laptop (~tzafrir@2001:41b8:203:dc11:21c:23ff:fe20:991f) |
| 11:32.04 | Gugge | m_tadeu: canreinvite was renamed to directmedia |
| 11:32.44 | Gugge | if you have canreinvite in your config it may not be used for anything :) |
| 11:33.29 | m_tadeu | how can I check if it's being used or not? that may be my problem |
| 11:34.10 | Gugge | check the source for canreinvite |
| 11:34.47 | m_tadeu | I'm using packages...is there way through the console? |
| 11:35.27 | Gugge | im pretty sure 1.8 dont have canreinvite |
| 11:35.42 | Gugge | but im not gonna check :) |
| 11:38.25 | m_tadeu | Gugge: sure, that's my job here :) was wondering if there is a way to check that on the console...since I'm using canreinvite, the mechanism should be flaged somewhere as active or not |
| 11:39.50 | Gugge | sip show peer <peer> |
| 11:39.57 | Gugge | and check the "directmedia" output |
| 11:41.23 | m_tadeu | Gugge: in deed...It's set to "no" |
| 11:49.03 | wasanzy | please what hardwares do I need to be able to setup an IVR system, excluding networkcard and the PC? |
| 11:50.52 | *** join/#asterisk timahvo1 (~rogue@41.215.1.35) |
| 11:52.21 | schmidts | wasanzy it depends on what do you want to do |
| 11:53.49 | wasanzy | ok, just a moment, but for the mean time, am going to use ss7 protocol |
| 11:54.49 | schmidts | wasanzy i dont know much about ss7 but AFAIK you will need some PSTN connection like an ISDN interface card |
| 11:55.38 | wasanzy | ok |
| 12:04.11 | *** join/#asterisk billmania (~bill@38.98.130.98) |
| 12:06.00 | Rico29 | I have a lot of warnings including : chan_sip.c:3349 __sip_xmit |
| 12:06.51 | Rico29 | can anybody tell me what it means ? |
| 12:08.15 | kaii | Rico29: that information is very vague. i guess asterisk is warning you that there is some issue when transmitting sip messages to some destination. |
| 12:13.42 | *** join/#asterisk nicola_pav (~chatzilla@mail2.tikalnetworks.com) |
| 12:14.08 | *** join/#asterisk nightrid3r (~kvirc@91.176.173.127) |
| 12:18.16 | nicola_pav | hello. anyone familiar with FFA, fax for asterisk? I am getting the following error: Channel 'SIP/audiocodes-00000028' FAX session '40' is complete, result: 'FAILED' (FAX_NO_FAX), error: 'T1_TIMEOUT', pages: 0, resolution: 'unknown', transfer rate: '2400', remoteSID: '' |
| 12:18.23 | nicola_pav | any hints please? |
| 12:18.27 | *** part/#asterisk apten (~apten@carbon.gonicus.de) |
| 12:28.45 | *** join/#asterisk SuPrSluG (~SuPrSluG@8.22.96.106) |
| 12:29.10 | m_tadeu | I have something that I fond weird here....I send an invite and get 6 unauthrized responses from asterisk....is this normal? |
| 12:32.21 | m_tadeu | another strange thing is, on those silent calls, rtp is only running from the client to asterisk and no rtp packages are going to the client |
| 12:32.52 | Rico29 | schmidts, I recompiled asterisk with svn branch, and app_queue was not available in asterisk functions list (as well as the core show version function) |
| 12:33.03 | Rico29 | and app_queue was selected in menuselect |
| 12:35.20 | leifmadsen | Rico29: there is no func_queue.... |
| 12:35.38 | Rico29 | app_queue |
| 12:35.41 | Rico29 | sorry |
| 12:35.45 | leifmadsen | yes, it's in the Applications list |
| 12:35.50 | leifmadsen | not the Functions list |
| 12:35.54 | Rico29 | yes |
| 12:35.58 | leifmadsen | yes |
| 12:36.10 | Rico29 | and it was selected in my menuselect > applications |
| 12:36.21 | Rico29 | but not available in asterisk |
| 12:36.27 | leifmadsen | did you load app_queue.so? |
| 12:36.32 | leifmadsen | module show |
| 12:36.37 | leifmadsen | module show like queue |
| 12:37.23 | Rico29 | I went back on my old buggy version (sip stack crashing) |
| 12:37.39 | Rico29 | and will test again this eveening (production environment) |
| 12:39.33 | Rico29 | i'm trying to reproduce it on a VM |
| 12:40.56 | *** join/#asterisk azv4 (~kyleblake@host-66-202-7-218.pit.choiceone.net) |
| 12:43.26 | m_tadeu | checking the communication flow in wireshark, I saw that h261 is being used, even dough video is not enabled...is this ok? |
| 12:46.31 | SuPrSluG | if the rtp is one way from the client and the client sending h261 and asterisk has video disabled this would be normal. asterisk has been told to ignore video |
| 12:47.37 | *** join/#asterisk irroot (~irroot@41.124.132.252) |
| 12:48.25 | Rico29 | worked fine on my VM |
| 12:48.27 | irroot | hello |
| 12:49.12 | irroot | in malelane outside kruger park on way to mozambique for the day to avoid cold weather :P |
| 12:51.30 | *** join/#asterisk Deeewayne (~dwayne@68-64-220-110.static.forethought.net) |
| 12:51.30 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 12:56.05 | *** join/#asterisk Kamineko (~ender@moya.rainside.sk) |
| 12:59.30 | *** join/#asterisk tonsofpcs (~tonsofpcs@cpe-69-205-240-64.stny.res.rr.com) |
| 13:07.45 | *** join/#asterisk Jcook_5xData (~Jcook_5xD@173.162.32.1) |
| 13:08.48 | Jcook_5xData | Is there away to pause the cli? I was just wondering because it hard the trace a problem with it scrolling |
| 13:11.39 | Jcook_5xData | leifmadsen, Did you fix your ring in the ear problem? |
| 13:11.48 | eject_ck | kaii: I've tried to make h323 call to my ISP, no SIP there |
| 13:11.50 | leifmadsen | Jcook_5xData: no, I just am using the work around |
| 13:12.01 | eject_ck | Asterisk crashes during call :) |
| 13:12.14 | leifmadsen | Jcook_5xData: don't watch it like that -- output it to a text file using tee or something |
| 13:12.57 | *** join/#asterisk tzafrir_laptop (~tzafrir@2001:41b8:203:dc11:21d:e0ff:fe72:129b) |
| 13:13.19 | anonymouz666 | leifmadsen: I bought your book |
| 13:13.25 | Jcook_5xData | leifmadsen, ok I was seeing there a way. away do you see my post to you a few day ago about how I fix it. |
| 13:13.38 | leifmadsen | Jcook_5xData: no I did not |
| 13:14.13 | leifmadsen | anonymouz666: yay! another shiny nickle! :) |
| 13:14.19 | Jcook_5xData | here is what I did it seem to fix it. under my sip user I had 'busyallow=1' |
| 13:14.31 | Jcook_5xData | then I just remove it |
| 13:15.15 | Jcook_5xData | not sure if that your cause but it fix it for me |
| 13:16.26 | leifmadsen | Jcook_5xData: oh ya I saw that -- busyallow does not exist though |
| 13:16.32 | leifmadsen | that's not an option in sip.conf |
| 13:17.17 | leifmadsen | and there is no mention of it in chan_sip.c at all |
| 13:17.43 | *** join/#asterisk hobodave_ (~hobodave@pdpc/supporter/professional/hobodave) |
| 13:19.01 | Jcook_5xData | weird.. I don't know why it clear up after removing it. |
| 13:23.10 | wasanzy | who have experience building an IVR system using ss7 protocol? I am doing lot of reading reading right now with out making any thing out of it. |
| 13:23.24 | *** join/#asterisk rgsteele (~rgsteele@apo.aweber.com) |
| 13:24.12 | wasanzy | am mainly interested in knowing the components I will need, in order to build this system, so please if any one have an idea or docs, I will appreciate it |
| 13:28.02 | *** join/#asterisk moy (~moy@173.239.155.74) |
| 13:35.11 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 13:35.42 | nightrid3r | wasanzy : i'm also interested in figuring out this ss7 stuff. just waiting till i can scarpe up the $$$ to buy a card |
| 13:38.22 | *** join/#asterisk dms (~dms@nat/digium/x-kqannvmckssbbdqt) |
| 13:40.49 | *** join/#asterisk timahvo1 (~rogue@41.215.1.35) |
| 13:42.29 | m_tadeu | ok, here it goes...sometimes there is no sound on SIP calls(about 20% of the calls). The peers are the same, all behind nat, asterisk behind another nat. here's the pastebin with sip debug http://pastebin.com/6KNqarnz |
| 13:42.53 | m_tadeu | would apreciate all the help :)I'm getting nuts |
| 13:43.12 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 13:44.02 | leifmadsen | m_tadeu: I've seen that when your rtp.conf doesn't match the open ports on your firewall |
| 13:47.38 | m_tadeu | leifmadsen: there was a mismatch, alright...I'll test it for a while.....thanx for the eyes :) |
| 13:49.15 | leifmadsen | np :) |
| 13:52.05 | jaytee | I tried installing Fail2Ban on a CentOS 5.6 using Asterisk: The Definitive Guide. Everything looks ok, fail2ban service running, iptables shows the fail2ban-ASTERISK entry in the INPUT chain but when I test banning registration attempts for an account with the wrong password it never bans the IP or logs anything in the fail2ban.log file Not sure what to check next. |
| 13:54.33 | wasanzy | nightrid3r: both of us need the help, those who hv done it b4 in the room |
| 13:55.15 | wasanzy | please any one with any idea? am highly waiting to appreciate any one with an idea |
| 13:56.46 | kaii | wasanzy: idea: ss7 has nothing to with IVR. if you want to use the ss7 protocol, you need an interface card and pstn connection capable of ss7. |
| 13:58.06 | kaii | wasanzy: how to do an IVR, which is totally independent of the ss7 protocol, is documented very well in several places over the internet. example: http://ofps.oreilly.com/titles/9780596517342/asterisk-IVR.html |
| 14:01.07 | *** join/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 14:01.19 | *** join/#asterisk MarKsaitis (~MarKsaiti@78-61-148-80.static.zebra.lt) |
| 14:01.40 | wasanzy | kaii: So I don't need any special hardware or protocol if I want to do IVR? |
| 14:01.52 | kaii | no. |
| 14:02.02 | kaii | just read that link i sent you |
| 14:02.44 | wasanzy | I went through and it didn't mention any additional hardware but only application |
| 14:03.15 | kaii | wasanzy: that is because IVR is just a feature of asterisk. you dont need any hardware for that. |
| 14:03.28 | wasanzy | ok |
| 14:03.34 | kaii | wasanzy: BUT you need hardware to use that ivr, because without a phone you cannot talk to asterisk. |
| 14:04.03 | wasanzy | so what is the relationship between asterisk and ss7? |
| 14:04.04 | kaii | (well .. you can. but it generally doesnt help you to talk to asterisk using a sound card in the server) |
| 14:04.25 | kaii | wasanzy: ss7 is a protocol for large scale phone network connections |
| 14:04.44 | kaii | wasanzy: just as sip, h323, isdn-pri, and several others |
| 14:05.12 | kaii | wasanzy: its a method of talking to asterisk. ivr is a way how asterisk behaves when you call it. |
| 14:05.35 | *** join/#asterisk mintos (~mvaliyav@115.242.102.209) |
| 14:06.00 | wasanzy | so IVR has nothing to do with ss7 protocol right? |
| 14:06.15 | kaii | right |
| 14:07.05 | wasanzy | then doing the IVR, how do u connect ur phone to the asterisk? buy some special card? |
| 14:07.43 | *** join/#asterisk mintos (~mvaliyav@115.242.102.209) |
| 14:08.22 | m_tadeu | leifmadsen: now it only seem to happen when the caller enters a queue. sometimes, there is no sound....had that experience too? |
| 14:08.37 | leifmadsen | m_tadeu: yes, you need to answer the channel prior to entering the queue |
| 14:09.07 | kaii | wasanzy: honestly, read a book. |
| 14:09.51 | m_tadeu | leifmadsen: I do....wait 2 secs, answer, some blablabla, then queue. |
| 14:10.01 | leifmadsen | then I have no idea |
| 14:10.13 | wasanzy | hmmm, the link u sent didn't mention how to communicate with ur phone yet, do u hv another link? |
| 14:10.27 | *** join/#asterisk aiksa[LV] (~aiksaLV]@212.70.182.16) |
| 14:10.33 | kaii | leifmadsen: pure interest .. why do you need to answer before joining a queue? |
| 14:10.43 | leifmadsen | kaii: because it works? |
| 14:10.56 | leifmadsen | my guess is because for some reason the Queue() application doesn't always answer the line correctly |
| 14:10.58 | leifmadsen | for whatever reason |
| 14:11.07 | kaii | wasanzy: click on the title of that page |
| 14:11.30 | wasanzy | ok |
| 14:11.36 | wasanzy | thank for ur help |
| 14:11.44 | kaii | leifmadsen: hm interesting .. never had this problem |
| 14:11.53 | leifmadsen | I don't get it often, but I've seen it before |
| 14:12.07 | leifmadsen | there are many ITSPs and things always seem to work differently |
| 14:12.11 | jaytee | I saw bigfoot once! |
| 14:12.21 | leifmadsen | anyways, doing an Answer() before Queue() doesn't hurt anything |
| 14:12.21 | kaii | jaytee: thumbs up |
| 14:12.34 | kaii | leifmadsen: well, i costs the caller money |
| 14:12.52 | leifmadsen | not in north america |
| 14:12.59 | leifmadsen | well not usually anyways |
| 14:13.05 | kaii | fine for americans |
| 14:13.10 | leifmadsen | that's all I deal with |
| 14:13.15 | anonymouz666 | why not in north america? |
| 14:13.17 | leifmadsen | anyways, this isn't really something I'm trying to debate |
| 14:13.22 | anonymouz666 | here you call it is answered, you pay |
| 14:14.39 | kaii | leifmadsen: i was not going to start a debate, sorry. |
| 14:14.52 | leifmadsen | kaii: that's ok |
| 14:14.55 | kaii | we're just "silly" europeans ;-) |
| 14:15.29 | leifmadsen | I'm just saying, I've had a problem with queue() not playing audio, and answering it before joining the queue resolved that problem. In that situation I didn't care about answering, so I moved on and didn't investigate why it worked and why it didn't work without it |
| 14:15.55 | anonymouz666 | I had this problem too |
| 14:16.21 | anonymouz666 | it was a problem with 183/SDP early media in SIP |
| 14:23.14 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 14:26.21 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 14:27.11 | p3nguin | jaytee: Did you make sure to set up the logging part right? If your logs aren't in the right format, or if you aren't logging to the file fail2ban is reading, it won't ever work. |
| 14:29.41 | *** join/#asterisk merlin8282 (~merlin828@AStrasbourg-554-1-233-240.w86-204.abo.wanadoo.fr) |
| 14:29.54 | merlin8282 | Hiho ! |
| 14:30.21 | merlin8282 | I just installed asterisk 1.6 on a debian Squeeze and wondered what this [stdexten] is, in extensions.conf... |
| 14:30.28 | *** join/#asterisk vinhdizzo (~vinh@pool-173-60-112-55.lsanca.fios.verizon.net) |
| 14:32.03 | p3nguin | merlin8282: That's what we call a SAMPLE FILE. |
| 14:32.39 | merlin8282 | p3nguin: sure, but I mean, there's no reference to this context anywhere else. |
| 14:32.51 | p3nguin | If you are only trying to understand the SAMPLE file rather than use it, pastebin it and I'll take a look. |
| 14:33.43 | p3nguin | Or maybe someone else with that example on his system will look at theirs and help. |
| 14:36.28 | jaytee | p3nguin, I followed the section in The Book. The logpath is /var/log/asterisk/messages in the iptables-asterisk entry in jail.conf |
| 14:37.28 | *** part/#asterisk sekil (~sekil@78.24.104.73) |
| 14:37.44 | merlin8282 | p3nguin: http://pastebin.com/pjbZeCTL here it is, complete original template from squeeze. |
| 14:37.50 | jaytee | when I run fail2ban-regex against the log it matches the on some 20 log entries but when I test trying to ban live by having a softphone register with a bad password it logs but never bans. |
| 14:38.26 | p3nguin | jaytee: Can you try to make it register in rapid-fire succession? |
| 14:40.16 | jaytee | p3nguin, it makes about 2 registration attempts every second |
| 14:40.33 | p3nguin | merlin8282: There are at least three other contexts which include the stdexten context. |
| 14:48.36 | *** join/#asterisk ddickenson (~ddickenso@67-198-0-5.static.grandenetworks.net) |
| 14:50.42 | p3nguin | jaytee: Would you show me one of the lines that is logged to /var/log/messages.log when you have a failure? |
| 14:51.29 | p3nguin | or whatever your actual logpath is. |
| 14:53.23 | *** join/#asterisk hetii (~hetii@87.99.51.172) |
| 14:53.36 | hetii | Hello :> |
| 14:54.05 | *** join/#asterisk BuenGenio (~Gene@cm61-10-82-188.hkcable.com.hk) |
| 14:54.16 | p3nguin | /var/log/asterisk/messages, as you said earlier. |
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| 14:59.40 | *** mode/#asterisk [+o malcolmd] by ChanServ |
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| 15:03.39 | WIMPy | I've got a problem with channel variables I set via setvar in sip.conf. They don't seem to exist when the dialplan is entered, but first when there is a match ion the dialplan. Is that a bug or |
| 15:03.46 | WIMPy | intended behaviour? |
| 15:04.24 | McBoing | I got my old Asterisk server peered with a new install, trying to call back from new to old (old is the production server). Simple test using "exten => 248,n,Dial(SIP/servera/248)" from the new server, is this format not good for Asterisk 1.8.5, its what I use for the 1.2.12 |
| 15:05.33 | p3nguin | wimpy: You lost me at the comma. Does the variable exist in the dial plan or not? |
| 15:05.51 | McBoing | I initially had exten => _XXX,n,Dial(SIP/servera/${EXTEN}) but it didnt catch anything |
| 15:05.53 | m_tadeu | is there a wayt to check the rtp port a current channel is using? both on asterisk and caller? |
| 15:06.37 | p3nguin | mcboing: Do you have a sip peer in sip.conf called "servera"? |
| 15:06.46 | WIMPy | p3nguin: Thet's exactely the issue: no and yes. It exists when the first dialplan application is executed, but it doesn't exist when used in an lswitch. |
| 15:06.50 | McBoing | p3nguin: Yessir, hi BTW |
| 15:06.54 | *** part/#asterisk merlin8282 (~merlin828@AStrasbourg-554-1-233-240.w86-204.abo.wanadoo.fr) |
| 15:07.39 | WIMPy | McBoing: Doesn't look bad to me. What exactely happens? |
| 15:08.04 | McBoing | p3nguin: http://pastebin.com/D8BdQwRz |
| 15:08.20 | p3nguin | wimpy: I use setvar in my sip.conf peers, and the variable always exists when the call hits the dial plan. If it is a problem with inheritance, you could try checking the value and resetting it with the underscores to make it inherit through one or all created channels. |
| 15:10.05 | WIMPy | p3nguin: Err, no. At that point there is no issue any more. The issue is that it doesn't exist while trying to find a match. Just AFTER it has found a match. |
| 15:10.41 | p3nguin | mcboing: You have no priority 1. |
| 15:10.54 | WIMPy | McBoing: Where is you priority 1 for that extension? |
| 15:10.54 | p3nguin | mcboing: Every extension must start with priority 1. |
| 15:11.22 | *** join/#asterisk nightrid3r (~kvirc@91.176.173.127) |
| 15:12.51 | McBoing | p3nguin: Jul 26 11:11:56 NOTICE[3652]: pbx.c:1741 pbx_extension_helper: Cannot find extension context 'incoming' <--- from servera, so its getting there, but why does it expect the context incoming? |
| 15:13.43 | McBoing | is there a parameter I can use to tell it to use another context? |
| 15:13.44 | p3nguin | mcboing: The peer entry in sip.conf has been set to context=incoming. |
| 15:14.08 | McBoing | oh I see |
| 15:18.29 | McBoing | p3nguin: I changed my context to the appropriate one that extension 248 lives in and ran "sip reload" and "reload" and still get error "Jul 26 11:17:19 NOTICE[3652]: pbx.c:1741 pbx_extension_helper: Cannot find extension context 'incoming'" |
| 15:19.16 | *** part/#asterisk dms (~dms@nat/digium/x-kqannvmckssbbdqt) |
| 15:19.24 | p3nguin | mcboing: Are you seeing that on the new system or the old one? |
| 15:19.32 | McBoing | old one |
| 15:20.02 | p3nguin | Does "dialplan show incoming" show you the context you made? |
| 15:20.38 | chazzam | I think he said he changed the definition in sip.conf so it pointed at a different context |
| 15:20.47 | chazzam | but it is still complaining about the previous one |
| 15:20.58 | p3nguin | Well then I need to be updated with the current information. |
| 15:21.04 | McBoing | p3nguin: Yes, on the new server running "dialplan show incoming" shows the incoming context |
| 15:21.26 | jaytee | p3nguin, my fail2ban is now working :-) thanks for the help |
| 15:21.43 | McBoing | p3nguin: I mentioned the change above, but I guess you missed it or I suck at explaining :) |
| 15:21.44 | p3nguin | jaytee: Do you know what turned out to be the problem with it? |
| 15:22.08 | chazzam | McBoing: might be he is wanting a new pastebin |
| 15:22.11 | jaytee | p3nguin, it was a ID 10 T error :-) |
| 15:23.03 | *** join/#asterisk hetii (~hetii@87.99.51.178) |
| 15:23.12 | p3nguin | mcboing: I thought you were saying you wanted to use the context called 'incoming' and you set the values accordingly in sip.conf. chazzam interpreted it differently. |
| 15:23.16 | hetii | re |
| 15:23.19 | hetii | :) |
| 15:23.27 | hetii | Is there some special # about sip protocol ? |
| 15:23.41 | p3nguin | a special hash mark? |
| 15:23.47 | hetii | channel :) |
| 15:23.48 | p3nguin | a special pount sign? |
| 15:23.50 | p3nguin | oh |
| 15:23.57 | Qwell | hetii: not really |
| 15:25.18 | hetii | ok then i need ask here. could someone tell me how should look proper sip communication when i have such scheme: clientsip [127.0.0.1:6060](start make a call)--> sip proxy[127.0.0.1:5050] -> destination_server[213.xxx.xxx.xxx] |
| 15:25.20 | McBoing | p3nguin: here is my setup http://pastebin.com/wAWf1vQ5 |
| 15:26.18 | sunfone | Morning all |
| 15:26.20 | p3nguin | mcboing: lines 11 and 39 are out of whack. |
| 15:26.21 | McBoing | p3nguin: In order for my new server to call an existing extension on the old server I need to change my context I guess on the new server peer setup |
| 15:26.22 | hetii | and is it exist something like transparent sip proxy ? |
| 15:26.58 | McBoing | p3nguin: new asterisk deprecated "username" so I used "defaultuser" as it suggested |
| 15:27.49 | sunfone | An annoyance I have been dealing with a long time - when someone uses our local telco to temporarily forward their (POTS) line, a short blip of a ring occurs every time a call is forwarded, which is unfortunately enough to trigger the Sangoma card to answer the POTS line |
| 15:27.50 | p3nguin | mcboing: line 41 is where the problem seems to be. |
| 15:27.57 | McBoing | p3nguin: every article I read explained it this way, to use the username on the opposite server, no? |
| 15:28.15 | p3nguin | mcboing: It's type=peer. Don't use username or defaultuser. |
| 15:28.16 | sunfone | That then causes the reception phone to ring in our normal setup, and of course no one is there |
| 15:28.42 | McBoing | p3nguin: 41 used to say "incoming" but my old server doesnt have such a context, so I used the one that has the extensions I want to call |
| 15:28.42 | sunfone | Any ideas on how to avoid answering a line that has only this very short ring? |
| 15:29.56 | sunfone | I'm assuming that this happens within the dahdi module, but perhaps it is lower level... in the card? |
| 15:30.21 | McBoing | p3nguin: I used the examples from the Asterisk 3rd edition book exactly, so blame Lief lol |
| 15:30.34 | McBoing | err Leif I meant |
| 15:33.21 | m_tadeu | which one takes a smaller bndwidth? gsm, alaw or ulaw? |
| 15:33.40 | *** join/#asterisk hetii (~hetii@87.99.51.178) |
| 15:33.58 | ChannelZ | gsm |
| 15:34.37 | hetii | sunfone i suppose * will not answer your call till someone pick up a phone ? :> |
| 15:34.51 | *** join/#asterisk nny1 (~SM@cpe-174-107-223-014.sc.res.rr.com) |
| 15:35.20 | McBoing | oh I figured it out, I had the contexts backwards |
| 15:36.17 | hetii | sunfone: you can also play some silent before you answer channel if its routed somewhere else then some user |
| 15:37.56 | WIMPy | I think I will auto-generate localisation contexts. |
| 15:38.34 | hetii | sunfone: or write some AGI script that will do what you want or even more :> |
| 15:41.05 | jaytee | so now instead of calling it Asterisk 1.10 they're gonna call it Asterisk 10 and the one after that will be Asterisk 11. I think I'm just gonna call it "Phil" |
| 15:41.21 | leifmadsen | jaytee: nice |
| 15:42.12 | jaytee | I'm curious to see how the video conferencing will play out in 10. |
| 15:43.00 | leifmadsen | it works really well from what I've played with |
| 15:43.59 | _Corey_ | Anyone used Diastar for video? |
| 15:44.39 | _Corey_ | it seems to have some impressive features... multi-user screen, etc |
| 15:45.00 | m_tadeu | ChannelZ: thanx |
| 15:50.09 | jaytee | the last time I messed with video was with * 1.4. Now it seems like h264 is finally supported as a codec at least in 1.8 |
| 15:50.27 | jaytee | gonna have to play with it some soon |
| 15:51.19 | jaytee | did Cisco buy Digg? seems like every 4th link on Digg is for the Cisco Cius |
| 16:04.04 | Qwell | jaytee: You mean the 'DiggAd by Cisco" links? |
| 16:05.08 | Qwell | also, h264 has been supported forever |
| 16:05.32 | Qwell | r7855 | markster | 2006-01-07 11:54:22 -0600 (Sat, 07 Jan 2006) | 2 lines |
| 16:08.27 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:6575:8f01:4813:eb88) |
| 16:09.10 | wasanzy | kaii: thank you for the feeds, all this while, I was confusing my self with IVR and ss7, I crossed checked and realized what my boss want from me is, to have asterisk configure to use the ss7 protocol, and an Outbound dialer for interfacing with asterisk to make calls from an application |
| 16:09.57 | Qwell | You confused IVR with SS7? |
| 16:10.16 | wasanzy | any one who have an idea about this or even some nice book or docs should please help me |
| 16:10.25 | Qwell | ~buybook |
| 16:10.25 | infobot | You can buy "Asterisk: The Definitive Guide" at http://oreilly.com/catalog/9780596517342 so go buy it SERIOUSLY |
| 16:11.21 | wasanzy | yes, Qwell, I was figuring out how to build IVR using ss7 protocol |
| 16:11.50 | wasanzy | but now I know they are not in relationship |
| 16:12.13 | wasanzy | that book you mentioned, I went through but mentioned nothing about ss7 |
| 16:12.43 | wasanzy | yes, there is IVR in there but not ss7, which I actually want to work on right now |
| 16:13.08 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:6575:8f01:4813:eb88) |
| 16:14.05 | nightrid3r | wasanzy : ss7 is a protocol library |
| 16:14.35 | nightrid3r | its on the download page of asterisk.org |
| 16:14.45 | chazzam | http://svn.asterisk.org/svn/asterisk/branches/1.8/configs/chan_dahdi.conf.sample search for ss7? |
| 16:14.54 | WIMPy | It seems somewhat unlikely that you want to use SS7 if you don;t already know it. |
| 16:15.07 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:6575:8f01:4813:eb88) |
| 16:15.19 | wasanzy | so I guess this will be my solution right?: http://ofps.oreilly.com/titles/9780596517342/asterisk-OutsideConn.html |
| 16:16.06 | wasanzy | yes I want to use ss7 |
| 16:16.43 | Qwell | Do you know what ss7 is? |
| 16:17.10 | wasanzy | I learned it is a protocol |
| 16:17.18 | Qwell | describe what that means. |
| 16:17.59 | wasanzy | hmmm, is unfortunate, I don't know the details of ss7 |
| 16:18.13 | Qwell | Describe what a protocol is. |
| 16:18.14 | wasanzy | can u help? |
| 16:18.51 | wasanzy | a protocol, is what I can say a channel through data is transmitted |
| 16:19.17 | wasanzy | through which data is transmitted |
| 16:19.48 | wasanzy | am I right? am not a networking guy either though |
| 16:19.51 | nightrid3r | wasanzy : http://en.wikipedia.org/wiki/Signaling_System_7 |
| 16:20.15 | Qwell | if you aren't a networking or telecom person, why are you setting up Asterisk? |
| 16:20.35 | Qwell | I mean, I get doing things as a hobby. But you're being paid to do so, professionally. |
| 16:20.39 | carrar | cause it's opensource! |
| 16:21.05 | wasanzy | becos it I want to learn from it |
| 16:21.09 | Qwell | There are just some things that I will never understand. |
| 16:21.20 | carrar | US Gov? |
| 16:21.27 | Qwell | You aren't trying to learn though. You're asking us for every little thing. |
| 16:22.33 | wasanzy | I learn through asking, I don't know that is why I asked, and when I get the answer, next time, I have done it before so, it will become part of me |
| 16:22.48 | Qwell | That isn't learning. That's remembering answers. |
| 16:22.50 | carrar | deep |
| 16:23.15 | carrar | wasanzy, try to change your approach |
| 16:23.16 | wasanzy | ok |
| 16:23.28 | carrar | buy the book |
| 16:23.30 | carrar | read it |
| 16:23.33 | Qwell | 4 times |
| 16:23.34 | carrar | google |
| 16:23.44 | carrar | then come ask! |
| 16:23.50 | *** join/#asterisk engrxyz (~puitpyitr@212.23.51.7) |
| 16:23.56 | wasanzy | good advice |
| 16:23.59 | wasanzy | thank you |
| 16:25.16 | carrar | If I want to build a bridge, maybe the first thing is I pay someone to do it, or become a structural engineer |
| 16:25.48 | wasanzy | hmmmm |
| 16:25.49 | Qwell | safety, pfft |
| 16:26.03 | Qwell | Just build the damn thing now. If it falls, it's somebody elses problem. |
| 16:26.18 | carrar | *cough*china*cough* |
| 16:27.34 | *** join/#asterisk tzafrir_laptop (~tzafrir@2001:41b8:203:dc11:21d:e0ff:fe72:129b) |
| 16:29.40 | p3nguin | mcboing: Now it works? |
| 16:35.32 | jaytee | lol |
| 16:35.41 | *** join/#asterisk coppice (~chatzilla@116.92.17.49) |
| 16:38.37 | Kobaz | why do people have to have such long phone calls |
| 16:38.42 | Kobaz | don't they know i want to reload app_dial |
| 16:39.03 | anonymouz666 | restart now then |
| 16:39.14 | anonymouz666 | it's a great command |
| 16:39.14 | p3nguin | Do you expect it to drop a call if you reload it? |
| 16:39.20 | anonymouz666 | hehe |
| 16:40.09 | Kobaz | p3nguin: there are calls currently using Dial |
| 16:40.13 | Kobaz | i think it would fail to unload |
| 16:40.21 | Kobaz | i mean like, not just reloading, unloading and loading |
| 16:40.25 | p3nguin | Does that mean it won't reload, or will it drop the calls using it? |
| 16:40.34 | Kobaz | it should have a use count |
| 16:40.47 | hetii | did sip proxy need listen on port 5060 ? in most client where i see proxy settings don`t see place for port |
| 16:40.48 | Kobaz | app_dial.so Dialing Application 2 |
| 16:40.50 | Kobaz | yeah |
| 16:41.08 | p3nguin | hetii: 5060 is the port SIP uses. |
| 16:41.54 | hetii | i know but what in the case when i want to use sip proxy on the same server where * is |
| 16:42.07 | leifmadsen | you don't |
| 16:42.15 | hetii | why ? |
| 16:42.16 | p3nguin | That would be like having to tell your web browser to use port 80 when you've entered http://something. |
| 16:42.18 | leifmadsen | or they have to listen on different ports |
| 16:42.49 | p3nguin | Why do you think you need to use a proxy on it? |
| 16:43.08 | hetii | yes but you can do http://something:other port |
| 16:43.35 | Kobaz | <PROTECTED> |
| 16:43.38 | Kobaz | yay |
| 16:43.50 | p3nguin | Yes you can, but the standard port is used when using http unless you define a nonstandard port. |
| 16:43.53 | hetii | i need set a proxy to change sip ACK message from t38modem application because it seams that have a bug |
| 16:44.29 | hetii | and don`t use on his header information from SIP 200 ok response. |
| 16:44.30 | p3nguin | So the standard port being 5060 will be used when using SIP unless you define a nonstandard port. |
| 16:45.19 | p3nguin | If you really need a proxy on the same system, I'd probably use 5060 on the proxy and something like 5080 on asterisk. |
| 16:45.28 | hetii | did some of you have some linux client or tools for sip that can be usefull to define some sip packages and send them ... ? |
| 16:45.51 | p3nguin | sipp does that, I think. |
| 16:45.53 | Qwell | hetii: sipp |
| 16:45.59 | hetii | i cannot because its a production server and all local phones use it :> |
| 16:46.31 | p3nguin | Run them through the proxy too. |
| 16:46.54 | hetii | for local network it is not required |
| 16:47.25 | p3nguin | Use port fowarding at the perimeter. |
| 16:47.38 | hetii | in fack if this will work as i expect it will works like that: t38modem --> sipproxy (to fix headers) --> * |
| 16:47.49 | hetii | all in localhost :) |
| 16:48.18 | p3nguin | Good luck! I'm going to the pool. |
| 16:48.40 | hetii | oright :) let know if i discover something nice :) |
| 16:49.36 | p3nguin | I think after I stop at the pool for a minute, then I'll go to the beach. |
| 16:50.01 | p3nguin | The ocean looks lovely today, so I may have to swim there as well. |
| 16:50.05 | Kobaz | are there any other options other than Dial 'I' to prevent connected line updates |
| 16:50.36 | anonymouz666 | p3nguin: where are you from ? |
| 16:51.17 | *** join/#asterisk MarKsaitis (~MarKsaiti@78-61-148-80.static.zebra.lt) |
| 16:51.32 | p3nguin | I'm in Florida right now. |
| 16:53.39 | azv4 | Does anyone in this channel know of a really good support channel for older phone systems (such as Panasonic Digital hybrid system)? |
| 16:54.00 | hetii | p3nguin: lucky man :) |
| 16:54.28 | anonymouz666 | p3nguin: seems a good thing to do :D |
| 16:56.25 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 17:00.27 | *** join/#asterisk rmunozsb1 (~IceChat77@sb1.serviceobjects.com) |
| 17:00.47 | Kobaz | soooooo |
| 17:00.55 | rmunozsb1 | looking for asterisk PRI expert |
| 17:01.06 | Kobaz | i have a good one for you guys |
| 17:01.30 | WIMPy | ~ask |
| 17:01.30 | 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. |
| 17:01.35 | WIMPy | rmunozsb1: ^^ |
| 17:01.51 | Kobaz | asterisk 1.8.5... Set(CALLERID(name)) Dial(SIP/foo&SIP/bar...etc...lots.more.phones) |
| 17:02.06 | Kobaz | SIP/foo sees original callerid... SIP/bar and thereafter sees the changed callerid |
| 17:03.19 | _Corey_ | Kobaz: What happens if you change the sequence of SIP targets? |
| 17:03.28 | Kobaz | oddly enough, nothing |
| 17:03.39 | Kobaz | SIP/foo still gets the wrong callerid |
| 17:04.34 | _Corey_ | Anything different about the 'foo' peer, trustrpid, settings etc. ? |
| 17:04.41 | rmunozsb1 | Im using clear channel 64k looking to get accurate ISDN fault codes returned |
| 17:04.50 | *** join/#asterisk war9407 (war@c-71-62-61-74.hsd1.va.comcast.net) |
| 17:04.51 | Kobaz | same as all the other ones |
| 17:05.01 | McBoing | p3nguin: yeah now that I have the context's the other way around its fine, even works with the _XXX match |
| 17:05.07 | _Corey_ | that's screwy |
| 17:05.13 | Kobaz | i'm thinking of rebooting the phone, but it seems like an asterisk issue |
| 17:05.19 | Kobaz | moved from 1.6.0 to 1.8.5 |
| 17:05.26 | _Corey_ | yeah, don't see how it would be the phone |
| 17:05.32 | *** join/#asterisk adyn (~adyn@pdpc/supporter/active/adyn) |
| 17:05.39 | rmunozsb1 | @WIMPy |
| 17:05.41 | _Corey_ | Unless all the INVITES are really the same |
| 17:06.06 | WIMPy | rmunozsb1: As in HANGUPCAUSE? |
| 17:06.10 | rmunozsb1 | yes |
| 17:06.19 | McBoing | I am almost convinced I can just put my sip/extensions/voicemail/ conf files from Asterisk 1.2.12 right to the 1.8.5 server, its nice that the config is almost the same |
| 17:06.30 | WIMPy | rmunozsb1: What else? |
| 17:06.46 | Kobaz | actually |
| 17:06.47 | Kobaz | hmm |
| 17:06.49 | Kobaz | what the hell |
| 17:07.05 | rmunozsb1 | well my current config isnt too accurate mostly returning 0 |
| 17:07.32 | *** join/#asterisk alcmagalhaes (~andrelc@venus.planetarium.com.br) |
| 17:07.40 | *** join/#asterisk BenC[UK] (~bcummins@cpc2-lock3-2-0-cust77.6-1.cable.virginmedia.com) |
| 17:08.44 | WIMPy | rmunozsb1: If HANGUPCAUSE is 0, it was you whu cleared the call. |
| 17:08.48 | BenC[UK] | Hi Guys, can someone push me in the right direction - I want to setup a "find me" type thing for the guys here, I want to to be call the asterisk management interface, and get it to call several numbers at the same time, but as soon as one person answers, hang up the rest of the calls |
| 17:09.28 | WIMPy | BenC[UK]: Sounds like a simple Dial(). |
| 17:10.06 | Kobaz | _Corey_: actually the peers aren't exactly identical.. heh,, hmm |
| 17:10.14 | Kobaz | they should have been |
| 17:10.28 | rmunozsb1 | What does that mean WIMPy. I cleared the call as ?? Im new to this project |
| 17:10.54 | WIMPy | rmunozsb1: It was you who hung up. |
| 17:11.20 | WIMPy | rmunozsb1: Maybe you should describe what you're trying to do. |
| 17:12.09 | Kobaz | _Corey_: it was sendrpid |
| 17:12.16 | Kobaz | _Corey_: should have been off, but was set to yes |
| 17:12.32 | _Corey_ | makes sense |
| 17:12.41 | rmunozsb1 | im trying to test numbers without actually ringing them |
| 17:13.21 | WIMPy | rmunozsb1: By setting some digital BC? |
| 17:13.33 | Kobaz | hmm |
| 17:13.46 | rmunozsb1 | I have to step into a meeting. Ill get back to you |
| 17:14.53 | *** join/#asterisk royce3 (4aca4162@gateway/web/freenode/ip.74.202.65.98) |
| 17:15.15 | *** join/#asterisk fenlander (~fenlander@82.152.81.57) |
| 17:15.41 | royce3 | is there a way to do "hot standby" for an acd queue in vanilla asterisk? |
| 17:16.35 | *** join/#asterisk luckyaba (~Lucky@ip72-194-218-169.sb.sd.cox.net) |
| 17:20.10 | *** join/#asterisk moy (~moy@173.239.155.74) |
| 17:20.21 | *** join/#asterisk bent_screwdriver (~bent_scre@74.255.249.66) |
| 17:22.29 | bent_screwdriver | Are the ports requires for SIP firewall entries still 10000-20000 and the default 5060? Seems like i've seen the rtp ports on a different range somewhere but wasn't sure if that was old or new data.... |
| 17:24.18 | WIMPy | Look at your rtp.conf. |
| 17:25.08 | bent_screwdriver | beautiful. never realized thats where it was. it is defaulted to 10000-20000. thanks!!!! |
| 17:30.09 | Qwell | royce3: Rephrase your question. |
| 17:30.43 | *** join/#asterisk elfelvin (~elfelvin@87-194-69-88.bethere.co.uk) |
| 17:34.01 | *** join/#asterisk bn-7bc (~bjarne-im@pdpc/supporter/active/bn-7bc) |
| 17:34.02 | royce3 | I would like for my agents' phones to not ring, but instead when a call is delivered to them, to just received a courtesy tone in their ear, and it should be answered automatically as long as they are in a ready state ( I can do the auto-answer via AMI or AGI I think ) |
| 17:37.33 | rmunozsb1 | hey WIMPy im back |
| 17:37.57 | WIMPy | is full |
| 17:38.18 | WIMPy | But just from lunch. |
| 17:38.59 | rmunozsb1 | your last statement 'setting some digital BC' can you elab |
| 17:39.33 | WIMPy | You are trying not to call them by using a non-audio BC? |
| 17:39.50 | rmunozsb1 | using clear channel |
| 17:40.23 | WIMPy | Where are you? |
| 17:40.34 | rmunozsb1 | US - California |
| 17:41.06 | WIMPy | Ok, then I guess you need to be prepared for that approach to fail. |
| 17:41.30 | rmunozsb1 | how would you approach |
| 17:41.33 | WIMPy | Last time we found out that some telcos don't give sensible information. |
| 17:41.43 | rmunozsb1 | im getting that |
| 17:41.47 | WIMPy | Exactely that way. |
| 17:42.04 | WIMPy | But I know that it works here. |
| 17:42.11 | WIMPy | To the obvious limits that is. |
| 17:43.13 | rmunozsb1 | I read somewhere about getting raw hex message |
| 17:43.44 | WIMPy | That is exactely what ends up in HANGUPCAUSE. |
| 17:44.04 | WIMPy | Just that they are unfortunatly displayed in decimal. |
| 17:44.27 | rmunozsb1 | your pretty knowledgeable abut this |
| 17:44.40 | McBoing | WIMPy is tha man |
| 17:45.06 | WIMPy | I'm just not from North America :-) |
| 17:45.15 | McBoing | aaaahahahaha, you dick |
| 17:45.48 | McBoing | sorry I assume that was sarcasm |
| 17:46.03 | rmunozsb1 | So is it North American Telco that dont give sensible info or is this across the board |
| 17:46.13 | WIMPy | Partially :-) |
| 17:46.52 | McBoing | could I make my own musiconhold? |
| 17:47.04 | WIMPy | If there's one thing I learned in this cahnnel, it's that north american telcos do a lot of strange things. |
| 17:47.54 | rmunozsb1 | Is it unreasonable to try to map telco response |
| 17:48.25 | billmania | It seems there are two definitions for the Dial() option "F" in https://wiki.asterisk.org/wiki/display/AST/Application_Dial. Is the second definition what happens in the absence of a specified context, exten and priority? |
| 17:48.40 | WIMPy | No. But as I stated someone else had to find out that the responses from his telco weren't usable. |
| 17:49.18 | WIMPy | billmania: For definite information do 'core show application dial' at the *CLI. |
| 17:49.39 | billmania | WIMPy: Will do. Thanks. |
| 17:50.16 | *** join/#asterisk moy (~moy@173.239.155.74) |
| 17:51.13 | WIMPy | rmunozsb1: But if you don't get any cause at all, there must be something wrong with what you're trying. |
| 17:51.39 | rmunozsb1 | what do you mean no cause as in no code |
| 17:52.04 | WIMPy | It can not be 0. |
| 17:52.40 | WIMPy | If it is 0, you didn't receive any. Either because you cleared the call yourself or it didn't even get out. |
| 17:53.00 | billmania | I added a comment to that page, clarifying the "F" option. |
| 17:53.03 | rmunozsb1 | I had to shorten the dial script to eliminate the possibility of ringing as it was actually was in some cases |
| 17:53.34 | *** join/#asterisk naveed1 (~naveed1@115.167.4.167) |
| 17:53.38 | rmunozsb1 | So what your saying is I ned to go back into my config |
| 17:53.40 | WIMPy | rmunozsb1: So you're effectively not waiting for a response? |
| 17:53.52 | rmunozsb1 | Well its a tradeoff here |
| 17:53.53 | naveed1 | Hi! all |
| 17:54.21 | rmunozsb1 | I would like to eliminate the 0 but how? |
| 17:54.28 | WIMPy | If you seet the BC to some data service it should never ring a phone. That's totally broken. |
| 17:54.31 | billmania | When I execute Dial("SIP/5101-00000002", "DAHDI/G0/1312#######,30,F") and I hangup before the 30 second timeout and before the 1312####### answers, instead of going to the next priority in the macro, I get "(macro-dialout-trunk, s, 20) exited non-zero on 'SIP/5101-00000002'". |
| 17:54.52 | WIMPy | (but that's at the called partys end) |
| 17:55.29 | rmunozsb1 | what is BC stand for (Sorry im new to the mnemonics) |
| 17:55.44 | billmania | How do I get Dial() to allow the macro to proceed to the next priority? |
| 17:55.54 | WIMPy | Bearer Capability or Transfercapability in Asterisk terms. |
| 17:56.04 | rmunozsb1 | Gotcha :) |
| 17:56.19 | naveed1 | sorry! it is not place for my query but i need some guidance. |
| 17:56.28 | WIMPy | billmania: On what cahnnel? If you hangup, there is no dialplan execution. |
| 17:57.04 | naveed1 | i want to check FOIP on localhost(IAXmodem+asterisk+Hylafax server) server |
| 17:57.13 | WIMPy | rmunozsb1: But a ringing phone is exactely that weirdness I was talking about. |
| 17:57.23 | rmunozsb1 | We set the BC to video |
| 17:57.24 | naveed1 | mean send a fax from My PC and then Received it on my PC. |
| 17:57.25 | billmania | When the channel which originated the call hangs-up before the timeout and before the answer. |
| 17:57.45 | Gugge | billmania: you dont, use the h extension |
| 17:57.49 | naveed1 | any one can guide me whether is it possible |
| 17:57.51 | naveed1 | ? |
| 17:58.02 | billmania | Ah, I think I understand. When I hangup the originating channel, the whole trip through the dialplan is over? |
| 17:58.15 | Gugge | yep |
| 17:58.18 | *** join/#asterisk MarKsaitis (~MarKsaiti@78-61-148-80.static.zebra.lt) |
| 17:58.23 | WIMPy | rmunozsb1: Video contains audio. Better use "(un)restricted digital information". |
| 17:58.40 | rmunozsb1 | CHANNEL(transfercapability)=DIGITAL |
| 17:58.48 | billmania | Gugge: Where would I place a jump to the "h" extension in this scenario? |
| 17:58.59 | Gugge | billmania: you dont |
| 17:59.03 | naveed1 | help plz:( |
| 17:59.18 | Gugge | billmania: the h extension is called when you hangup |
| 17:59.57 | WIMPy | naveed1: It has been done before. Be patient and wait for someone using iaxmodem/hylafax. |
| 18:00.03 | billmania | Gugge: WIMPy: OK, I think I get it. That gives me some clues to resume study of the dialplan and the verbose log. Thanks. |
| 18:00.06 | *** join/#asterisk jmls (~Julian@host217-36-208-155.in-addr.btopenworld.com) |
| 18:00.11 | jmls | evening all |
| 18:00.31 | jmls | does anyone know of a screen popup app that interfaces with outlook and google apps ? |
| 18:00.50 | jmls | for contacts etc |
| 18:01.45 | naveed1 | WIMPy: sir! stuff is available on internet but i am confused in some tech terms..can you tell me the purpose of IAXchannel and how Context works in Asterisk |
| 18:02.18 | WIMPy | ~book |
| 18:02.19 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342. Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://ofps.oreilly.com/titles/9780596517342/, or ~buybook |
| 18:02.23 | WIMPy | naveed1: Try that ^^ |
| 18:02.44 | naveed1 | Wimpy: i will try sir! thanks |
| 18:03.25 | naveed1 | I will read it but can you tell me in few words ..mean what is the purpose? just |
| 18:03.44 | naveed1 | Any way! i will read the book |
| 18:04.13 | WIMPy | IAX is a VOIP protocoll and contexts have several purposes, grouping your dialplan, e.g. for security. |
| 18:06.00 | rmunozsb1 | @WIMPy so the two options are SPEECH and DIGITAL for BC |
| 18:07.16 | WIMPy | rmunozsb1: There are more. See under 'core show function CHANNEL'. |
| 18:08.18 | WIMPy | And note that SPEECH and Digital w/t/a is strictly speech and must not be used for fax. |
| 18:09.52 | *** join/#asterisk retentiveboy (~pdugas@74-95-28-33-Atlanta.hfc.comcastbusiness.net) |
| 18:13.47 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 18:15.51 | rmunozsb1 | Thanks for your help WIMPy |
| 18:15.56 | rmunozsb1 | Im out |
| 18:20.09 | WIMPy | That should probably flash up in big letters as a common mistake when trying to send a fax over a digital line on every fax related howto. |
| 18:23.18 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 18:23.18 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 18:24.32 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v000-096.mobile.uci.edu) |
| 18:32.35 | *** join/#asterisk ccesario_ (~ccesario@201-42-141-53.dsl.telesp.net.br) |
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| 18:41.14 | pdtpatrick | Question .. i've gone off the PRI and now using a sip provider. I am trying to implement fax as the old one has stopped working. Is it possible to fax using SIP? im looking at the voip-info examples and they are suggesting that I need some modules installed. However I am not using PSTN.. is this possible and if so can u please point to an example or wiki .. thanks in advance. |
| 18:41.54 | WIMPy | T.38 |
| 18:42.04 | WIMPy | And some luck. |
| 18:42.19 | WIMPy | Your ITSP needs to support it. |
| 18:42.54 | *** join/#asterisk hfb (~hfb@pool-96-247-108-172.lsanca.dsl-w.verizon.net) |
| 18:43.36 | *** join/#asterisk BuenGenio (~Gene@cm61-10-82-188.hkcable.com.hk) |
| 18:46.19 | aiksa[LV] | a lot of luck may I say |
| 18:46.37 | aiksa[LV] | and an ITSP which understands |
| 18:47.10 | *** join/#asterisk Nasga (~Nasga@AAmiens-157-1-63-179.w86-196.abo.wanadoo.fr) |
| 18:49.42 | pdtpatrick | hehe .. wow is it that hard to get the fax going ? |
| 18:49.50 | McBoing | I wanna toy around with musiconhold.conf, any good places to get some free moh packs? |
| 18:51.47 | *** join/#asterisk Nasga_ (~Nasga@AAmiens-157-1-63-179.w86-196.abo.wanadoo.fr) |
| 18:52.00 | Qwell | ~faxforasterisk |
| 18:52.00 | infobot | faxforasterisk is probably Digium's commercial Fax For Asterisk module is available at http://www.digium.com/en/products/software/faxforasterisk.php |
| 18:52.02 | Qwell | pdtpatrick: trivial |
| 18:52.21 | *** join/#asterisk gogasca (~Adium@nat/cisco/x-bauvjzabbaxamllm) |
| 18:54.21 | malcolmd | McBoing: try opsound.org |
| 18:54.59 | McBoing | cool thanks |
| 18:55.40 | McBoing | I was going to do something goofy like make the moh South Park theme song |
| 18:56.47 | aiksa[LV] | Qwell: trivial if every factor outside is ok |
| 18:59.10 | gogasca | any of u guys have tried new 1.10? I have couple of Tandberg E20s and want to see if new 1.10 support extended fmtp parameters ( I remember in my 1.8 box asterisk stripped fmtp and video call is not negotiated as HD, just basic low quality) |
| 18:59.17 | aiksa[LV] | if for instance ITSP does lcr and SOMETIMES is sending your call further through sip interconnrct which doesnt have t.38 capability good luck debugging that |
| 19:00.09 | leifmadsen | gogasca: you mean Asterisk 10 |
| 19:00.45 | gogasca | yep |
| 19:01.39 | gogasca | missed Kevin Fleming email :) |
| 19:01.46 | *** join/#asterisk fish-bulb (~qcstewart@nat/digium/x-bghqwhtizfnletfd) |
| 19:02.14 | pabelanger | gogasca: media support for Asterisk 10 was re-written, so expect different results from Asterisk 1.8 |
| 19:03.44 | gogasca | thanks will give it a try tonite |
| 19:04.05 | *** join/#asterisk tzafrir_laptop (~tzafrir@2001:41b8:203:dc11:21d:e0ff:fe72:129b) |
| 19:09.50 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 19:10.20 | McBoing | reading an article that mentions I could use internet radio stream for moh with Asterisk, prolly not ideal but I want to test it anyway, just add this under [classes] for musiconhold.conf? default => mp3:/var/lib/asterisk/stream,<STREAM URL> ? |
| 19:13.29 | psilikon | apropos to nothing currently going on but Asterisk is so incredibly well documented. I sometimes I take that for granted until I work on a project that is not blessed with such documentation. |
| 19:13.46 | Hive | McBoing, not sure about the stream part but you should probably put directory=mp3:/var/lib/asterisk/stream,<STREAM URL> under the [default] heading |
| 19:13.54 | Hive | google can probably help you out here |
| 19:14.20 | McBoing | yeah maybe this article is not complete then |
| 19:14.32 | McBoing | so I need the classes AND the directory parameter set right? |
| 19:15.52 | Qwell | Why don't you look at the sample config in Asterisk? |
| 19:16.16 | McBoing | Qwell: I just saw that now >< |
| 19:16.59 | Hive | Like I said, i'm not sure about the parameters for the stream aspect, the [className] part is what you will have to Set(CHANNEL(musicclass)=className) to |
| 19:17.20 | Hive | (google will help you better than I ;-) |
| 19:17.30 | McBoing | thanks |
| 19:18.07 | McBoing | :q! |
| 19:18.11 | McBoing | ah shoot lol |
| 19:18.20 | Hive | Qwell, I dont know if I would describe faxing over voip as trivial :P |
| 19:18.26 | Hive | at least not in my experience haha |
| 19:18.51 | Hive | Happy to help :) |
| 19:19.56 | Qwell | Hive: I didn't say fax over voip. |
| 19:20.05 | Qwell | fax over voip is extremely silly |
| 19:23.58 | Hive | Im a pretty silly person. |
| 19:24.03 | Hive | :D |
| 19:27.35 | *** join/#asterisk dhartman (~dhartman@wilug/newlug/ricko73) |
| 19:32.57 | *** join/#asterisk Rico29 (~Cedric@ANantes-554-1-228-174.w2-1.abo.wanadoo.fr) |
| 19:33.03 | Rico29 | hi again |
| 19:33.24 | Rico29 | got a big problem, need help please |
| 19:33.41 | Rico29 | I'm trying to compile asterisk 1.8 svn branch |
| 19:33.45 | Rico29 | it compiles fine |
| 19:34.10 | Rico29 | but when I do a "core show application Queue", it tolds me than the queue app does not exists (or is not registered) |
| 19:34.21 | Rico29 | app_queue.so is loaded |
| 19:34.48 | Rico29 | and when i do a "module show like", I can't finish the sentence, and asterisk crashes (freeze) |
| 19:35.06 | Rico29 | I have to do an init.d/asterisk restart in another terminal |
| 19:35.17 | Kobaz | are you using svn for a specific reason? |
| 19:35.21 | Kobaz | and not the release? |
| 19:35.35 | Kobaz | and did you delete your old modules before you make install on the new 1.8 ? |
| 19:36.51 | *** join/#asterisk MarKsaitis (~MarKsaiti@78-61-148-80.static.zebra.lt) |
| 19:36.59 | Rico29 | yes, I'm using svn for a bug which is only resolved in svn branch |
| 19:37.19 | Rico29 | I was in 1.8.3.2, and no I didn't delete the old modules |
| 19:37.24 | Rico29 | should I delete them ? |
| 19:38.07 | Kobaz | yes |
| 19:38.19 | Rico29 | from asterisk JIRA : This issue has already been fixed with ASTERISK-17760 a few days ago. It is currently on the SVN v1.8 branch. |
| 19:38.20 | Rico29 | ok |
| 19:38.22 | Kobaz | and build everything from scratch to make sure |
| 19:38.23 | Rico29 | let's go |
| 19:38.27 | Rico29 | ok |
| 19:38.53 | Kobaz | make distclean; ./configure; make install |
| 19:39.20 | *** join/#asterisk galaxywatcher (~galaxywat@pdpc/supporter/active/galaxywatcher) |
| 19:39.24 | Kobaz | actually you'll have to redo make menuconfig too, before make install |
| 19:39.57 | Kobaz | mixing in old modules is *bad* |
| 19:40.10 | *** join/#asterisk fobus912 (~fobus912@41.251.39.254) |
| 19:40.17 | Rico29 | ok, i've removed all old modules, and it seems to work really better ! |
| 19:40.44 | Kobaz | 'work really better' would be 'work much better' |
| 19:41.19 | Rico29 | yes, sorry... I'm french, and it's well known that rench people are not really good in english speaking... :) |
| 19:41.29 | Kobaz | yeah |
| 19:41.50 | Hive | <PROTECTED> |
| 19:42.13 | Kobaz | c'est vraiment beau |
| 19:42.16 | Kobaz | or something |
| 19:42.25 | gogasca | :) |
| 19:42.34 | Kobaz | the direct translation doesn't work exactly into proper english gramma |
| 19:42.36 | Kobaz | r |
| 19:42.54 | Kobaz | i'm having the same problem going the other way |
| 19:43.00 | Kobaz | en->fr |
| 19:44.31 | serafie | c'est tres bonne? |
| 19:44.45 | serafie | or c'est tres bon |
| 19:44.59 | leifmadsen | I enjoy bon bon's |
| 19:45.29 | Kobaz | oui, bon pas beau puet-etre |
| 19:46.09 | Rico29 | :) |
| 19:46.19 | Rico29 | c'est très bon is perfectly french |
| 19:46.37 | Kobaz | perfect no ly |
| 19:46.44 | serafie | "puet-etre" is not. :P |
| 19:46.53 | serafie | "peut-etre" |
| 19:47.04 | Kobaz | oui oui |
| 19:48.28 | Kobaz | petit probleme du l'orthographe |
| 19:49.16 | Kobaz | parle est plus simplement |
| 19:50.02 | *** join/#asterisk CaptainPants (~CaptainPa@nat/digium/x-eczkzyvoltqskwxw) |
| 19:51.00 | *** join/#asterisk WHYS (~drumm@137.28.94.209) |
| 19:51.02 | pabelanger | pomplamoose? |
| 19:51.21 | Rico29 | pamplemousse |
| 19:51.27 | pabelanger | :) |
| 19:51.34 | Kobaz | oui pomplamoose |
| 19:51.36 | chazzam | sings instead |
| 19:51.40 | Kobaz | wait.. what's a pomplamoose |
| 19:51.47 | pabelanger | grapefruit |
| 19:51.58 | chazzam | I think pomplamoose is a band |
| 19:52.02 | Kobaz | je prefere l'orange |
| 19:52.40 | sunfone | Its terrible the things they do to mooses these days |
| 19:52.53 | WHYS | Anyone here used sip trunking from qwest, sprint or norlight? |
| 19:52.55 | Kobaz | i was once bitten by a moose |
| 19:53.08 | sunfone | I thought it bit your sister! |
| 19:53.12 | Kobaz | her too |
| 19:53.30 | sunfone | the moose has been sacked |
| 19:53.34 | Kobaz | poor moose |
| 19:53.45 | Kobaz | but didn't the guy who sacked the moose get sacked? |
| 19:53.53 | sunfone | :) |
| 19:54.07 | Kobaz | The system is going down for reboot NOW! (pts/0) (Tue Apr 12 17:24:34 2011): |
| 19:54.19 | Kobaz | mmm, another asterisk system almost ready to do |
| 19:54.21 | Kobaz | go |
| 19:54.31 | sunfone | Do they even have mooses in England? |
| 19:54.38 | Kobaz | maybe imports? |
| 19:54.40 | sunfone | What is the plural of Moose? |
| 19:55.07 | beardy | Mices! |
| 19:55.19 | psilikon | moose |
| 19:56.37 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 19:57.38 | Rico29 | thanks all for your help |
| 20:00.07 | WHYS | Who offers sip trunking - off the open internet. Anyone use large Tellcos? |
| 20:00.47 | Kobaz | meece |
| 20:01.14 | Kobaz | ~itsp |
| 20:01.14 | infobot | [~itsp] An ITSP is an Internet Telephony Service Provider (or VoIP telephone company). They allow you to either SEND calls to the PSTN (this is called termination), RECEIVE calls from the PSTN (called origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs. |
| 20:01.19 | Kobaz | ~itsplist-us |
| 20:01.19 | infobot | Here are some popular ITSPs (USA) starting with the more respected ones: http://www.teliax.com , http://www.voicepulse.com/connect/ , http://www.broadvoice.com , http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net , http://voip.ms and http://flowroute.com |
| 20:02.03 | Rico29 | ~itsplist-fr |
| 20:02.08 | Rico29 | too bad... |
| 20:02.18 | pabelanger | ~itsplist-ca |
| 20:02.19 | infobot | somebody said itsplist-ca was Here are some popular Canadian ITSPs: http://www.les.net , http://www.babytel.ca , http://www.voip.ms, http://unlimitel.ca |
| 20:03.09 | WHYS | Thanks. I use vitelity now, but for my employer they want something off the wild wild internet. |
| 20:06.33 | *** join/#asterisk jplank (~G_Bove@208-104-67-26.dyn.fttp.comporium.net) |
| 20:09.10 | Kobaz | teh interwebs |
| 20:09.52 | Hive | careful, i hear there be spiders in those webs |
| 20:19.31 | McBoing | so does MusiconHold(default) attempt to play class from musiconhold.conf "default", I keep seeing the error "Music class default requested but no musiconhold loaded." |
| 20:25.27 | nny1 | where does asterisk store recordings pre mix monitor? |
| 20:30.28 | *** join/#asterisk retentiveboy (~pdugas@74-95-28-33-Atlanta.hfc.comcastbusiness.net) |
| 20:32.22 | SuPrSluG | in /var/spool/asterisk/monitor usually |
| 20:35.37 | *** join/#asterisk Aliosity (~aliosity@94-192-160-216.zone6.bethere.co.uk) |
| 20:37.48 | *** join/#asterisk lanning (~lanning@173-8-187-197-SFBA.hfc.comcastbusiness.net) |
| 20:40.05 | Hive | McBoing, from what I understand, the default music on hold class is set by default |
| 20:40.52 | Hive | I found that in the musiconhold.conf file you must make a [className] for the music oh hold you want (which has attributes under it like where to find the sound file) |
| 20:41.08 | Hive | then you have to Set(CHANNEL(musicclass)=className) at least that's how I got mine working :) |
| 20:41.45 | *** join/#asterisk deeperror (~textual@76.226.169.86) |
| 20:42.47 | deeperror | using sip.conf i've created a context=trashcan to dump all unauthenticated inbound calls to. What is the best way to handle these within this context? _X.,hangup() ? |
| 20:42.57 | Hive | exten => 1900yournum,1,Set(CHANNEL(musicclass)=className) |
| 20:43.30 | Hive | you could do that deeperror |
| 20:43.52 | Hive | or do something fun with them :) |
| 20:43.55 | WIMPy | deeperror: That will only catch calls to destinations starting with a digits. |
| 20:44.01 | hetii | hmm |
| 20:44.21 | deeperror | WIMPy ok, then what is the best thing to put in there? nothing? |
| 20:44.28 | deeperror | let them fail with ext not found? |
| 20:44.29 | WIMPy | But why don;t you just disable guests if you don't want them? |
| 20:44.39 | deeperror | option? |
| 20:44.53 | WIMPy | allowguests |
| 20:45.08 | deeperror | hahaha |
| 20:45.20 | deeperror | so set that to no and it will just reject them all? |
| 20:45.33 | deeperror | before it even gets to the context? |
| 20:45.34 | hetii | I build some sip transparent proxy, i`m able to register my phone by it but when i try make a call i got 404 not found |
| 20:46.16 | hetii | i suppose because * see ip of proxy instead of real one. |
| 20:46.42 | Qwell | doesn't sound very transparent then |
| 20:46.58 | deeperror | WIMPy, thanks for the tips! |
| 20:46.58 | *** join/#asterisk capt_cassimir (~arch@178.73.219.103) |
| 20:47.04 | hetii | yes indeed :> |
| 20:47.22 | deeperror | WIMPy, was just playing around with sipvicious and it brought up a lot of concern :)) i'll sleep better now |
| 20:47.33 | hetii | Contact field in SIP i suppose should be set then to real ip of my softphone :> |
| 20:47.44 | hetii | or maybe some more or other one as well ? |
| 20:48.18 | capt_cassimir | Do you guys have a favorite DNS provider for SRV records? |
| 20:48.42 | Qwell | I learned 2 things just now. |
| 20:48.46 | Qwell | 1) There are DNS providers. |
| 20:48.52 | Qwell | 2) Some of them don't support DNS. |
| 20:49.23 | capt_cassimir | @qwell: I have a bunch of domains with a hosting provider, who thinks i am only hosting small websites. Therefor, they don't let me set anything beyond A, CNAME, and MX |
| 20:49.34 | Qwell | Yeah, they fail. |
| 20:49.58 | capt_cassimir | So you think the only way for me to get an SRV record is to spin up a BIND server? |
| 20:50.53 | *** join/#asterisk nightrid3r (~kvirc@91.176.173.127) |
| 20:53.38 | citywok | capt_cassimir: what is this a plesk or webmin instance or something? |
| 20:55.26 | capt_cassimir | citywok: Yeah, it might as well be. I'd rather not say who it is |
| 20:55.38 | capt_cassimir | but i'm flexible and not broke, so i'm in the market for a proper DNS provide |
| 20:55.40 | Qwell | $20 on godaddy. |
| 20:55.57 | capt_cassimir | Go daddy will let me do fancy DNS administration? looking.... |
| 20:56.01 | Qwell | no |
| 20:56.05 | Qwell | nevermind. ignore me. |
| 20:56.12 | russellb | my domains on godaddy have SRV records ... |
| 20:56.17 | russellb | using their DNS hosting |
| 20:56.35 | capt_cassimir | russellb: that's exactly what i want, pulling it up |
| 20:57.46 | jaytee | 3 more days russell! |
| 20:57.56 | Qwell | SPY! |
| 20:58.04 | russellb | true statement |
| 20:58.16 | jaytee | are you gonna wear the red fedora? |
| 20:58.27 | Qwell | serafie paid him not to |
| 20:58.37 | russellb | assuming I get one yes |
| 20:58.40 | Qwell | at least, that was the arrangement |
| 20:58.51 | russellb | Qwell: she isn't going to pay up :-p |
| 20:58.52 | jaytee | I bought one from their online store. |
| 20:58.55 | serafie | um, yeah, i might have to default on that arragement. |
| 20:59.03 | russellb | see!? |
| 20:59.03 | serafie | $1200 is steep. |
| 20:59.09 | russellb | you said $2500 |
| 20:59.16 | Qwell | buys the arrangement at bank auction |
| 20:59.20 | serafie | lol |
| 20:59.20 | Qwell | $2.50! |
| 20:59.34 | russellb | seriously though, anyone want to buy my mini fridge? ^_^ |
| 20:59.36 | nny1 | does asterisk put it's files pre mix-monitor in /var/spool/asterisk/monitor? |
| 20:59.42 | serafie | russellb: how much do you want for it? |
| 20:59.47 | jaytee | I already have a mini-fridge.....sorry |
| 20:59.52 | russellb | serafie: $60? |
| 21:00.30 | capt_cassimir | Somebody getting their RHCE? |
| 21:01.02 | Qwell | RHCE is (way) overrated. |
| 21:01.07 | jaytee | we doan need no steenking certificates! |
| 21:01.23 | russellb | jaytee: didn't I meet you while you were getting dCap? :- |
| 21:01.24 | Qwell | I would take (and pass) the test if it was offered for free. |
| 21:01.25 | russellb | :-p |
| 21:01.37 | capt_cassimir | lol |
| 21:01.58 | nightrid3r | RHEC, can you insert a cd == PASSED :) |
| 21:01.59 | jaytee | russellb, yep when I took the Adv. * class Jared taught back in Nov of 2008 |
| 21:02.14 | capt_cassimir | nightrid3r: You obviously are not familiar with the exam process |
| 21:02.20 | russellb | never got his dCap :-( |
| 21:02.41 | serafie | you can have mine if you promise to stay. :P |
| 21:02.45 | jaytee | me neither, I passed the written and ran out of time and only got 60 on the lab |
| 21:02.59 | russellb | serafie: I don't think it works that way |
| 21:03.07 | serafie | lol damn |
| 21:03.09 | jaytee | if I'd skipped the optional part of installing a T1 card I would have had enough time |
| 21:03.15 | russellb | serafie: but thanks anyway <3 |
| 21:03.35 | russellb | serafie: mini fridge! let's make a deal! |
| 21:04.01 | *** join/#asterisk seraphie (~erin@75.76.38.159) |
| 21:04.05 | jaytee | you've done alot for all of us Asterisk users over the years, russell. So thanks and Godspeed in your new endeavour. |
| 21:04.20 | serafie | Yeah.... M says he bought one for $20 once, so that's kind of his price point. I don't think you wanna let yours go for that little. |
| 21:04.22 | russellb | thanks a lot, jaytee. It has been an incredible experience. :-) |
| 21:04.24 | jaytee | Just don't be a stranger to IRC |
| 21:04.29 | russellb | i won't |
| 21:04.56 | russellb | serafie: $20, huh .. |
| 21:05.18 | russellb | serafie: but this one has special features, including a shelf, a door, and a tiny freezer that could hold like 2 popsicles! |
| 21:05.20 | capt_cassimir | Pfft, i've fished them out of dumpsters for free |
| 21:05.23 | capt_cassimir | you get what you pay for. :) |
| 21:05.25 | serafie | hehe |
| 21:05.26 | jaytee | gotta run, bbl |
| 21:05.58 | russellb | serafie: $40? |
| 21:06.17 | serafie | :P OK, I will give you $40. |
| 21:06.22 | russellb | \o/ |
| 21:06.35 | serafie | ...on Monday |
| 21:06.41 | russellb | >_< |
| 21:06.44 | serafie | jk. |
| 21:06.52 | russellb | I'll still be in town Monday actually |
| 21:07.18 | Qwell | What day do you live? |
| 21:07.29 | Qwell | leave.. |
| 21:07.33 | capt_cassimir | @russellb: Looks like the Premium DNS service at Go Daddy *might* be what i want. Is that what you use? |
| 21:07.52 | russellb | capt_cassimir: i don't think I use premium anything |
| 21:08.09 | russellb | Qwell: packing the truck and cleaning the house over the weekend, driving away Monday afternoon |
| 21:08.13 | Qwell | Party at russellbs on Tuesday! |
| 21:08.42 | russellb | Qwell: sure, if you buy it |
| 21:11.45 | *** join/#asterisk Bezzie (~tzink@static-108-27-36-18.nycmny.fios.verizon.net) |
| 21:16.29 | capt_cassimir | @russellb: You probably got it with the purchase of a domain; meanwhile mine is clever and i'd like to keep it. :) |
| 21:21.37 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 21:26.08 | hetii | hmm |
| 21:31.52 | *** join/#asterisk MikeSix (Mike6@14.1.36.138) |
| 21:33.28 | *** join/#asterisk royce3 (4aca4162@gateway/web/freenode/ip.74.202.65.98) |
| 21:36.52 | *** join/#asterisk singler (~singler@88-119-250-190.static.zebra.lt) |
| 21:43.29 | *** join/#asterisk Praise (~Fat@unaffiliated/praise) |
| 21:45.02 | MikeSix | hi, I've got (what i feel is) a reasonably complex transfer problem between 2 asterisk systems |
| 21:48.21 | Qwell | sounds not complex so far |
| 21:48.36 | *** join/#asterisk m_tadeu (~quassel@89.181.60.178) |
| 21:50.19 | *** join/#asterisk luckyaba (~Lucky@ip72-194-218-169.sb.sd.cox.net) |
| 21:54.10 | WIMPy | It could be. |
| 21:55.30 | hetii | my proxy almost works:) sip can register i can call but ... * send RTP data to wrong destination :< |
| 22:08.28 | *** join/#asterisk nny (~SM@cpe-174-107-223-014.sc.res.rr.com) |
| 22:08.51 | nny | 192.168.1.100 (None) 8c96bca1-1e3862 0x0 (nothing) No Rx: SUBSCRIBE <guest> <-- what does the peer "guest" signify here? (from sip show channels) |
| 22:09.10 | nny | if i understand it that's the remote side of the channel? |
| 22:10.33 | gogasca | a not configured or defined IP which is still sending SIP⦠|
| 22:12.43 | *** join/#asterisk tzafrir_laptop (~tzafrir@static-8-143-241-92-p2mp.broadband.blic.net) |
| 22:15.34 | MikeSix | Qwell: well, the problem is, i have 3 trunks between the 2 systems, a set of queues at system 1, and incoming calls go through system 1, and need to be transfered back through the same system - ive ended up iwth the same call taking 3 channels when transferred |
| 22:15.46 | *** join/#asterisk celord (~celord@201.198.102.2) |
| 22:21.42 | *** join/#asterisk DrDigi (~mmurphy@173-162-2-137-Stockton.hfc.comcastbusiness.net) |
| 22:30.58 | *** join/#asterisk Bipul (~bipul@unaffiliated/bipul/x-4918593) |
| 22:36.12 | *** join/#asterisk mindCrime (~chatzilla@static-50-52-147-222.drhm.nc.frontiernet.net) |
| 22:46.16 | Bipul | if i upgrade my Ubuntu version then is there any Effect on my Asterisk server that has being installed earlear |
| 22:47.31 | *** join/#asterisk jeremy_g (jeremy_g@c-a6a0e755.022-186-73746f3.cust.bredbandsbolaget.se) |
| 22:49.40 | *** join/#asterisk singler (~singler@78-60-139-121.static.zebra.lt) |
| 22:52.29 | hardwire | ok.. what.. the crap. |
| 22:52.54 | hardwire | I have two identical machines (yah right) |
| 22:52.59 | chazzam | upgrading ubuntu has a high probability of breaking ALL THE THINGS |
| 22:53.14 | hardwire | both using the bria soft phone. one is chopped/gapped and delayed substantially. |
| 22:53.17 | *** join/#asterisk WIMPy (~wimpy@e183095026.adsl.alicedsl.de) |
| 22:54.06 | hardwire | it takes the wonky machine 13 seconds to send 400 frames. it takes the good machine 8 seconds to send 400 frames (ulaw) |
| 22:54.41 | hardwire | I've already placed the wonky machine with another one. as well as placed it on a working machines peripherals/net connection |
| 22:54.54 | hardwire | I'm so confused.. it seems like a packetizer issue due to a sound buffer problem.. |
| 22:55.49 | hardwire | in wireshark if I decode the audio using rtp timestamps everything is perfect however... but if I decode it using the packet times.. it's gapped.. packets appear to be sent much slower even though the rtp timestamps are correct. |
| 22:55.54 | hardwire | soooo confuuuused. |
| 22:58.09 | jeremy_g | hardwire:did you start asterisk with a special switch to decode according to packet timestamps? |
| 22:58.15 | jeremy_g | asterisk -? |
| 22:58.29 | *** join/#asterisk Wiretap_Work (~wiretap@unaffiliated/wiretap) |
| 22:58.54 | *** join/#asterisk qjcg (~weechat@216-58-27-134.cpe.distributel.net) |
| 23:00.41 | hardwire | jeremy_g: I'm more confused on why the identical machines would differ. |
| 23:01.15 | hardwire | and swapping out the broken one with another would act identically. |
| 23:01.18 | hardwire | I'm.. grr. |
| 23:01.24 | hardwire | 65% slower rtp! |
| 23:01.44 | hardwire | I don't think decoding based on rtp timestamps would work well... it's sending slower. |
| 23:01.59 | hardwire | it would run out of playable audio really fast. |
| 23:03.46 | hardwire | jeremy_g: I really think you're pullig my leg btw. |
| 23:07.09 | MikeSix | Qwell: I'm not so good with ascii art - should I try and explain the full scenario? |
| 23:11.40 | *** part/#asterisk hetii (~hetii@87.99.51.178) |
| 23:15.23 | jeremy_g | hardwire:you didnt answer my question. how do you choose what to use as reference? |
| 23:17.05 | *** join/#asterisk Cesare (~Adium@creati59.lnk.telstra.net) |
| 23:18.02 | hardwire | jeremy_g: the answer is no. no jitter buffer, nothing. |
| 23:18.44 | hardwire | it appears to be related to the client, asterisk simply responds as soon as it recieves a packet from the client. |
| 23:19.14 | hardwire | in other words.. the distance between client packets is 65% slower than it needs to be.. the response packet from asterisk is 100% what it needs to be |
| 23:19.54 | hardwire | I feel like the client is just sending them slowly... I'll load wireshark on them and test the 'local' environment. |
| 23:19.57 | hardwire | sigh |
| 23:20.00 | WIMPy | So it isn't Asterisk related? |
| 23:20.06 | hardwire | WIMPy: don't be like that WIMPy ! |
| 23:20.36 | WIMPy | is always like that |
| 23:20.41 | hardwire | I REFUSE TO PUT ALL MY YEARS IN THIS CHANNEL HELPING PEOPLE OUT THERE TO THE WOLVES. |
| 23:20.44 | hardwire | heh |
| 23:20.50 | hardwire | I've been here.. like.. 5 years? |
| 23:21.15 | WIMPy | Hard times. |
| 23:21.32 | hardwire | hehe |
| 23:21.46 | hardwire | well if y'all aint seen this sort of weird before.. that sucks for me. |
| 23:22.48 | WIMPy | Oh I just found out how to edit dialed numbers on the fly yesterday, despite of me ranting about that being impossible for some months. |
| 23:25.53 | hardwire | WIMPy: via the dialplan or via sip injection? |
| 23:26.32 | WIMPy | I'm not doing technology dependant stuff. I'm not purely SIP. |
| 23:26.45 | hardwire | what was impossible about it? |
| 23:26.48 | WIMPy | In fact I usually only use one SIP phone. |
| 23:27.00 | WIMPy | Doing it for incomplete extensions. |
| 23:27.19 | hardwire | gotcha |
| 23:27.26 | hardwire | I do some fun catchall stuff |
| 23:27.38 | WIMPy | But now I'm using loopback lswitches and it's so much better. |
| 23:28.08 | WIMPy | Bummer tho, that you can't use channel variables there. |
| 23:28.15 | WIMPy | That would have been ace. |
| 23:28.31 | hardwire | goto/gosub wasn't cool enough? |
| 23:28.39 | *** join/#asterisk m_tadeu (~quassel@89-181-60-178.net.novis.pt) |
| 23:28.44 | WIMPy | No, it just doesn't work. |
| 23:28.58 | WIMPy | They will terminate dialing. |
| 23:33.09 | *** join/#asterisk joeflyde (~Gregory@63.239.202.68.cfl.res.rr.com) |
| 23:36.31 | *** join/#asterisk godmachine-x6 (~godmachin@h239.250.19.98.dynamic.ip.windstream.net) |
| 23:40.53 | *** join/#asterisk WiretapWork (~wiretap@unaffiliated/wiretap) |
| 23:43.51 | nny | what's the proper way to pass variables to System? |
| 23:44.17 | nny | System(command ${variable})? |
| 23:44.23 | beek | yes |
| 23:44.28 | nny | thanks |
| 23:44.32 | beek | Your system script will need to pick it up. |
| 23:45.12 | nny | yeah my script uses command (number) |
| 23:45.17 | nny | so just need to pass that digit |
| 23:46.09 | *** part/#asterisk joeflyde (~Gregory@63.239.202.68.cfl.res.rr.com) |
| 23:49.40 | hardwire | WIMPy: hmm.. I'm interested in testing it out if you wanna shoot me info. |
| 23:49.57 | hardwire | I use the invalid handler to do just about everything relating to invalid. |
| 23:50.28 | hardwire | nny: make sure your command exits quickly. |
| 23:51.04 | hardwire | I've had problems with processes that last longer than a split second causing problems. |
| 23:51.21 | hardwire | like.. asterisk will just zone out for like.. a few seconds.. doing nothing. |
| 23:51.42 | hardwire | I usually have it run scripts that run other scripts in the background because of that. :( |
| 23:52.19 | hardwire | well.. tested locally.. the client is skewing the rtp. |
| 23:52.24 | hardwire | dies |
| 23:53.54 | WIMPy | If you hit the i extension, the user cannot continue to dial, either. |
| 23:54.18 | WIMPy | At least not in the local dialplan. |
| 23:57.48 | *** join/#asterisk riddlebox (~james@75-132-224-120.dhcp.stls.mo.charter.com) |
| 23:59.56 | nny | hardwire: can use TrySystem too i supose |