| 00:00.54 | TiCPU | iax2 show peers show my device as (D) with unspecified IP |
| 00:01.52 | TiCPU | for my other peers I see at least that Asterisk Tx to them to reply, but not my new device |
| 00:05.23 | p3nguin | Does that mean SiAX is broken? |
| 00:06.20 | p3nguin | The D just means that you have used host=dynamic and it must register. A host of null means it hasn't registered. |
| 00:06.36 | *** join/#asterisk Naikrovek (~mbluth@unaffiliated/naikrovek) |
| 00:09.09 | TiCPU | it could ir it could not be, I'm just asking if there is a way to get a more verbose log, SiAX forums says the application is well known to work with Asterisk |
| 00:15.36 | p3nguin | Do you have the ability to sniff on the router? |
| 00:20.09 | TiCPU | yes |
| 00:21.22 | TiCPU | on the internal interface, I see packets from my external IP to the internal Asterisk IP, the device uses a dynamic port to communicate with the Asterisk server at 4569 |
| 00:29.16 | p3nguin | But you determined previously that asterisk did not respond appropriately? |
| 00:55.28 | *** join/#asterisk xpot-mobile (~james@70-91-210-237-BusName-Utah.hfc.comcastbusiness.net) |
| 00:55.37 | *** join/#asterisk jstapleton (~jstapleto@c-24-125-190-50.hsd1.va.comcast.net) |
| 01:04.45 | *** join/#asterisk mjordan (~mjordan@user-69-1-6-49.knology.net) |
| 01:18.56 | *** join/#asterisk rdegges (XtMpnFX7nF@69.164.197.143) |
| 01:19.37 | rdegges | Hey guys, quick question. The ``ImportVar`` application lets you import another channel's variable value--is there an equivalent application that allows you to set another channel's variable somehow via dialplan? |
| 01:19.47 | rdegges | Or, is there some special syntax I can use with Set() to accomplish that? |
| 01:19.59 | rdegges | (you can assume that the remote channel's name is 'test' or whatever) |
| 01:21.01 | rdegges | I realize that I could define a subroutine, eg: [test_blah] exten => s,1,Set(var=value), then do a ChannelRedirect(test,test_blah,s,1) |
| 01:21.12 | rdegges | but that seems really unnecessarily verbose for what I want to accomplish |
| 01:21.27 | rdegges | anyhow--suggestions welcome! thanks <3 |
| 01:36.12 | [TK]D-Fender | rdegges: System(asterisk -rx "dialplan set chanvar ${thechannel} variablename value") |
| 01:36.52 | rdegges | isn't there a more pragmatic way to do it? :o |
| 01:37.04 | rdegges | i could have sworn there was a way to set a channel variable in the dialplan somehow |
| 01:37.40 | p3nguin | That looks like it's in the dialplan. |
| 01:37.40 | [TK]D-Fender | rdegges: There is an AMI command... not sure how much easier it could be. |
| 01:38.02 | [TK]D-Fender | See if they added a CLI execute command in 1.8. I haven't looked at it yet |
| 01:38.03 | rdegges | Well, my goal is to not break out via system(), or use the AMI(), or anything--to just do it in pure dialplan |
| 01:38.16 | rdegges | Just seems a bit silly =/ |
| 01:39.53 | [TK]D-Fender | And my goal is to one day retire to a private tropic island with a Victoria's Secret model who thinks I'm the best thing since sliced bread |
| 01:39.59 | [TK]D-Fender | oooohhh... who am I kidding. |
| 01:40.02 | [TK]D-Fender | I want that NOW :p |
| 01:42.35 | p3nguin | I don't see the problem with doing it dial plan using app_system. |
| 01:42.42 | p3nguin | If it works, it works. |
| 01:42.44 | [TK]D-Fender | Ok, ther IS one. |
| 01:42.56 | [TK]D-Fender | There.. one tiny layer less |
| 01:43.14 | rdegges | Well, it's just ugly =/ |
| 01:43.31 | rdegges | Don't you guys care about the aesthetics? :x |
| 01:43.44 | rdegges | Also, that means it adds additional overhead for the fork() or whatever |
| 01:44.33 | p3nguin | If you're running on a 25 MHz system, I might believe that it actually matters how heavy it is to use System() to run the console dialplan command. |
| 01:45.33 | p3nguin | 25 MHz with 8 kB RAM |
| 01:47.36 | [TK]D-Fender | 8 KB ram? WTF |
| 01:48.03 | [TK]D-Fender | My old XT was 4.77mhz and had 640kB :p |
| 01:56.44 | *** join/#asterisk adolfomaltez (~taro@190.62.236.168) |
| 01:58.10 | p3nguin | A badass machine you had. |
| 02:01.10 | Naikrovek | [TK]D-Fender: in "A Game of Thrones" there's a line where Tyrion is asked how he wants to die. (as in choosing his method of impending death) and he has a great response to it. |
| 02:01.25 | [TK]D-Fender | Naikrovek: I've watched the series |
| 02:01.33 | [TK]D-Fender | Naikrovek: Peter Dinklage = WIN |
| 02:01.39 | Naikrovek | that moment is in the episode. Agreed. |
| 02:01.46 | Naikrovek | in AN episode. |
| 02:02.12 | [TK]D-Fender | A similar line I'm sure. |
| 02:02.26 | [TK]D-Fender | Mine was lifted from the "smart-ass's McDonalds Application" |
| 02:02.39 | Naikrovek | "How do you want to die, halfman?" "... at 80! Drunk, with a mouth on my c**k." |
| 02:02.42 | [TK]D-Fender | that circulated about 15 years ago |
| 02:02.55 | Naikrovek | yeah i saw that app. |
| 02:03.32 | Naikrovek | i'm reading the books now. the novel and the series are very similar. minor bits are out of order, but virtually unchanged. |
| 02:03.38 | Naikrovek | on the 4th book now. |
| 02:03.45 | Naikrovek | just gets better and better and better |
| 02:04.48 | Naikrovek | the 9th or 10th episode of next season is going to be insane. |
| 02:05.04 | Naikrovek | and that's all I'm going to say about that. not wanting to spoil anything. |
| 02:07.33 | SeRi | guys what is the best speech recognition module? or how about the best ivr? |
| 02:07.34 | *** join/#asterisk godmachine-x6 (~godmachin@unaffiliated/godmachine-x6) |
| 02:16.51 | dlynes | SeRi, the best ivr is ryo |
| 02:17.05 | SeRi | dlynes, Thanks |
| 02:17.20 | dlynes | SeRi, i.e. roll-your-own |
| 02:18.03 | SeRi | ;) |
| 02:18.19 | *** join/#asterisk godmachine-x6 (~godmachin@unaffiliated/godmachine-x6) |
| 02:18.58 | dlynes | SeRi, but i'm guessing you meant voice recognition, not autoattendant, right? |
| 02:19.35 | SeRi | yes |
| 02:20.16 | dlynes | SeRi, just thought i'd throw that out there...in pbx-speak, an autoattendant is often called an IVR as well |
| 02:20.47 | SeRi | oooo thanks for the info |
| 02:20.58 | dlynes | SeRi, depending on the particular phone system manufacturer, that is...they have all kinds of different names for it |
| 02:22.02 | dlynes | SeRi, but, I think there's three(?) that work with asterisk? Digium has one called Lumenvox |
| 02:22.44 | SeRi | ahhhh lumenvox. I seen that one. looks like ill explore it |
| 02:29.58 | [TK]D-Fender | dlynes: Works with, not made by |
| 02:31.03 | *** join/#asterisk netmax (~netmax@is.linux-administrator.com) |
| 02:31.29 | TiCPU | p3nguin, it responds by rejecting the reg |
| 02:33.20 | *** join/#asterisk skr (skr@189-30-6-100.smace701.dsl.brasiltelecom.net.br) |
| 02:46.00 | skr | hi! need some thoughts on my setup (spa3102+asterisk). Everything seems to be ok on my setup. If I unplug network to asterisk server, outgoing calls from spa/line1 are routed to pstn normally. So, i wish to do the other way around: if there is an incoming call from pstn, what do I have to do to make spa call line1 *when asterisk server is unreachable* ? (if it isn't the right place to ask |
| 02:46.01 | skr | this, please point me to. and sorry my poor english!) |
| 02:46.05 | *** join/#asterisk mjordan (~mjordan@user-69-1-6-49.knology.net) |
| 02:48.20 | p3nguin | skr: You want calls from the PSTN to go to the phone instead of to Asterisk? |
| 02:51.03 | skr | only if asterisk isn't reachable |
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| 03:03.13 | *** part/#asterisk mjordan (~mjordan@user-69-1-6-49.knology.net) |
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| 03:13.42 | skr | is it in any way related to 'register expires' (3600) or 'ans call without reg' (no) ? |
| 03:14.43 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 03:15.23 | p3nguin | No. |
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| 03:50.15 | *** join/#asterisk Pegasus_RPG (~chatzilla@p5DD427CF.dip.t-dialin.net) |
| 03:51.29 | Pegasus_RPG | Hello. I'm interested in adding the iLBC codec to my asterisk 1.6.2.9 system but the contrib/scripts/get_ilbc_source.sh script is outdated and all instructions I find on the web refer to it. |
| 03:51.49 | Pegasus_RPG | (It also looks like the makefile is outdated) |
| 03:52.28 | Pegasus_RPG | Does anyone have any such instructions for iLBC's current location on Google Code? |
| 03:54.56 | Pegasus_RPG | <rolleyes> nevermind...the get_ilbc_source.sh just looks for the wrong file name |
| 03:55.04 | Naikrovek | why ilbc? |
| 03:55.07 | Naikrovek | out of curiosity |
| 03:55.33 | Pegasus_RPG | Because it's the lowest-bandwidth free-ish codec my HandyTone 386 supports |
| 03:55.41 | Naikrovek | ah |
| 03:55.51 | Naikrovek | god i hate grandstream |
| 03:55.53 | *** join/#asterisk brdude (~brdude@c-24-5-194-184.hsd1.ca.comcast.net) |
| 03:56.00 | Naikrovek | i wish you the best of luck with that little guy |
| 03:56.02 | Pegasus_RPG | And it can't seem to get G.726 right...sounds like crap |
| 03:56.05 | Naikrovek | mine all hoarked |
| 03:56.12 | Pegasus_RPG | haha, it's been a solid performer using ulaw |
| 03:57.46 | Naikrovek | mine died within 3 weeks of purchase, all of them. |
| 03:58.02 | Pegasus_RPG | wow. I guess I've been lucky then |
| 03:58.07 | Naikrovek | they're not recommended in this channel because others have had the same experiences |
| 03:58.08 | Naikrovek | anyway |
| 03:58.12 | Naikrovek | glad yours works |
| 03:58.18 | Naikrovek | you figured out the ilbc problem |
| 03:58.33 | Pegasus_RPG | I think so. Building now |
| 03:58.45 | Naikrovek | why are you worried about bandwidth? do they not have fast ethernet to the asterisk box? |
| 03:59.07 | Naikrovek | the asterisk box can translate the streams to whatever your sip provider supports on the fly |
| 03:59.56 | Naikrovek | it's not always an ideal solution, that translation |
| 04:01.35 | Naikrovek | i'm just curious, i don't mean to pry or challenge your decisions, if you're wondering |
| 04:01.48 | Pegasus_RPG | My DSL is on the edge of DSL service so has only 128kbps upstream |
| 04:01.56 | Naikrovek | ahh remote phone |
| 04:02.11 | Pegasus_RPG | and I had the * box on the LAN but the provider only supports ulaw |
| 04:02.14 | Naikrovek | i have dsl as well. |
| 04:02.19 | Naikrovek | i hate it. |
| 04:02.48 | Pegasus_RPG | I liked it fine where I used to live. solid reliable 1.5Mbps down, 768k up |
| 04:03.01 | Pegasus_RPG | but 768k down and 128k up is a challenge for VoIP |
| 04:06.06 | *** join/#asterisk irroot (~irroot@41.49.195.131) |
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| 06:29.55 | din3sh | Hi all, I'm having a random problem whereby calls from ZAP channels appear "No Answer" in CDR logs but with a billsec value greater than 0 |
| 06:30.07 | din3sh | any idea why/ |
| 06:30.08 | din3sh | ? |
| 06:36.28 | *** join/#asterisk timahvo1 (~rogue@197.176.75.105) |
| 06:41.31 | Pegasus_RPG | Hi again. 'Module 'codec_ilbc.so' was not compiled with the same compile-time options as this version of Asterisk.' How can I find out what those options were? |
| 06:41.42 | Pegasus_RPG | (I'm using a binary package of *) |
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| 06:50.38 | *** join/#asterisk lftsy (~lftsy@install.deckpoint.ch) |
| 06:50.48 | wdoekes2 | Pegasus_RPG: $ asterisk -nrx 'core show settings' | grep Build |
| 06:51.19 | *** join/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net) |
| 06:56.45 | *** join/#asterisk timahvo1 (~rogue@41.81.102.2) |
| 06:59.59 | Pegasus_RPG | wdoekes2: ok, so I only get Build Options: LOADABLE_MODULES but if I pick only that in menuselect, * still complains about the module I build |
| 07:00.21 | *** join/#asterisk heroxbd (~user@ernie02-dmz.awa.tohoku.ac.jp) |
| 07:02.14 | *** join/#asterisk waschtl (~waschtl@HSI-KBW-078-043-090-014.hsi4.kabel-badenwuerttemberg.de) |
| 07:02.38 | din3sh | I'm having a random problem whereby calls from ZAP channels appear "No Answer" in CDR logs but with a billsec value greater than 0 |
| 07:04.41 | wdoekes2 | Pegasus_RPG: well.. in reality it uses AST_BUILDOPT_SUM, a hash of BUILD_DEPS (not the C(PP)FLAGS we see above): $ grep -e "^MENUSELECT_BUILD_DEPS" menuselect.makeopts | sed 's/MENUSELECT_BUILD_DEPS\=//g' | md5sum |
| 07:05.07 | *** join/#asterisk irroot (~irroot@pbx.distrotech.co.za) |
| 07:05.47 | heroxbd | Hi friends, my company has a local switched telephony system. How can I build a gateway between sip and switched telephony system with a voice modem on my PC? |
| 07:05.59 | heroxbd | It is possible? |
| 07:06.26 | Pegasus_RPG | heroxbd: if you want only one SIP channel available on the switched system |
| 07:07.35 | Pegasus_RPG | heroxbd: and I believe only certain voice modem chipsets will work |
| 07:13.09 | heroxbd | Pegasus_RPG: only one SIP channel means only one sip number, and the rest is the whole switched telephony system? |
| 07:14.04 | Pegasus_RPG | heroxbd: I mean only one conversation on the SIP side at a time. (One conversation per modem, that is) |
| 07:14.24 | Pegasus_RPG | How many SIP DIDs you have doesn't matter |
| 07:15.01 | kaldemar | heroxbd: you better forget about the voice modem and use somethign else. for example an ATA. |
| 07:16.00 | heroxbd | Pegasus_RPG: I see. That's okay. I am heading for an alarming system that monitors our server and can call administrator when necessary. |
| 07:16.42 | Pegasus_RPG | oh that's true what kaldemar said. An ATA would be a far easier way to do it, especially for just one channel |
| 07:17.34 | heroxbd | Pegasus_RPG: kaldemar: wow, I didn't know about ATA before, let me see.. |
| 07:19.19 | heroxbd | wow. That's what i need! thanks, kaldemar |
| 07:20.50 | heroxbd | btw, in my company the switched telephone line have 4 wires, is it a different standard than 2-wire ones? |
| 07:21.15 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 07:21.38 | Pegasus_RPG | heroxbd: That could either be two lines per connector (2 pairs) or the second pair is dead |
| 07:21.41 | kaldemar | heroxbd: most likely, unless it is just two analog lines. |
| 07:21.59 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 07:22.04 | *** join/#asterisk Nasga (~Nasga@186.212.10.93.rev.sfr.net) |
| 07:22.12 | kaldemar | or it may be a PRI or a BRI. |
| 07:22.37 | Pegasus_RPG | oh yeah those are 4 wires too |
| 07:22.47 | kaldemar | heroxbd: first step is to find out the technology that your old system supports, then you know what kind of a device you need. |
| 07:23.23 | kaldemar | heroxbd: so start with that. |
| 07:23.27 | *** join/#asterisk jeffgus (~jeffgus@2001:470:f2eb:1::4) |
| 07:24.35 | heroxbd | kaldemar: How to find out the technology of the old system? by looking at the central telephone switch? |
| 07:24.51 | *** join/#asterisk hetii (~hetii@194.181.154.25) |
| 07:25.30 | kaldemar | heroxbd: ask someone who knows the system, look at the manual or the system itself, tell the model of the switch and maybe someone else has the possibility to know for sure. |
| 07:25.40 | heroxbd | the telephones are "dec dterm 85". |
| 07:25.53 | heroxbd | kaldemar: I see |
| 07:26.14 | Pegasus_RPG | heroxbd: you will need to look at the central system itself. The cards may have numbers telling you what they are |
| 07:26.19 | kaldemar | the telephone model is not helpful. the trunk connection does not have to use the same technology. |
| 07:26.38 | heroxbd | I see. I will have a look at the central unit. |
| 07:26.40 | *** join/#asterisk oej (~olle@ns.webway.se) |
| 07:28.24 | heroxbd | just a quick answer, the central switch is NEC APEX3600i |
| 07:29.12 | heroxbd | ahh. maybe it is a IP-based telephone system. |
| 07:30.29 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:add9:c4ae:e34b:ab91) |
| 07:31.12 | kaldemar | heroxbd: brief googling suggests that it might be. find out what protocol it supports. |
| 07:33.10 | heroxbd | kaldemar: yeah, thanks a lot :) |
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| 07:59.41 | *** join/#asterisk SteveWilliams (~SteveWill@59.162.182.218) |
| 08:02.04 | SteveWilliams | Good Morning! All |
| 08:03.56 | SteveWilliams | Please help me. How do I connect people who are called up by our office outbound IVR to our office number? I think it needs Meetme. Here is my dialplan http://pastebin.com/RuYrbn4R. Sorry for being a noob. Please help. |
| 08:04.58 | *** join/#asterisk schmidts (~schmidts@Lo20-rt01.lm33.sil.at) |
| 08:05.00 | schmidts | good morning |
| 08:06.03 | *** join/#asterisk irroot (~irroot@pbx.distrotech.co.za) |
| 08:06.18 | SteveWilliams | ' |
| 08:06.31 | tuxx- | morning |
| 08:07.34 | SteveWilliams | ' |
| 08:09.14 | kaldemar | SteveWilliams: a simple Dial will do that, no need for MeetMe. |
| 08:10.13 | *** join/#asterisk irroot (~irroot@pbx.distrotech.co.za) |
| 08:10.22 | SteveWilliams | @kaldemar please, could you help me with the steps. please... |
| 08:10.34 | *** join/#asterisk Faustov (~fst@gentoo/user/faustov) |
| 08:10.47 | SteveWilliams | @ where do i enter the dial command in my dialplan |
| 08:10.55 | *** join/#asterisk waschtl (~waschtl@HSI-KBW-078-043-090-014.hsi4.kabel-badenwuerttemberg.de) |
| 08:11.23 | irroot | where you want to call out ... once you have done all the processing you want to do |
| 08:12.14 | SteveWilliams | @ irroot , okay, so i can just Dial() out the number from the macro which does the processing?? |
| 08:13.35 | irroot | yeah |
| 08:13.55 | irroot | or when the macro returns might be better from a debug programing pov |
| 08:14.24 | irroot | so check the access rights/call waiting / .... as you need then dial( |
| 08:14.59 | SteveWilliams | @ irroot kaldemar , thanks for your answers. i am going to change my dialplan and see if that works for me (if i do not commit another noob mistake again). Thank you!! |
| 08:15.51 | irroot | we love n00bs gives us somthing to abuse and have fun with like a sadistic cat and mouse :P |
| 08:16.52 | *** join/#asterisk jacc0 (~jacc0@D522448D.static.ziggozakelijk.nl) |
| 08:16.54 | jacc0 | hoi hoi |
| 08:16.59 | jacc0 | good moring |
| 08:17.03 | jacc0 | :) |
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| 08:26.43 | *** part/#asterisk Pegasus_RPG (~chatzilla@p5DD427CF.dip.t-dialin.net) |
| 08:32.34 | WIMPy | Moin |
| 08:36.17 | phiscribe | when are u gonna pay me for that hamburger |
| 09:01.34 | jacc0 | me? |
| 09:04.13 | *** join/#asterisk irroot (~irroot@pbx.distrotech.co.za) |
| 09:04.53 | WIMPy | Hi irroot. Busy again? |
| 09:05.25 | din3sh | Hi all, I'm having a random problem whereby calls from ZAP channels appear "No Answer" in CDR logs but with a billsec value greater than 0 help!!!!!!!!!!!!!!!!!!!!!!!!!!!!!! |
| 09:05.27 | irroot | i am a bit not too hectic back to misdn-v1 |
| 09:06.25 | WIMPy | With the Digium card? |
| 09:06.34 | irroot | i have updated my tree to be more out the box mISDN [v1] will build and install the headers in include/mISDN for mISDNuser and asterisk also -lisdnnet is now -lisdnnet-1 |
| 09:07.11 | irroot | yeah the Digium card works fine with v1 but not v2 cant see differences in the code that is significant |
| 09:07.34 | WIMPy | That's bad. |
| 09:08.47 | jacc0 | I'm trying to clean up the messages log file; there are some warnings in there that are not needed |
| 09:08.58 | WIMPy | Have you considered dahdi? I think it has become quite usable over the last year. |
| 09:09.01 | irroot | so my v1 can coexist with v2 all that i need to do is comment out a modprobe.conf include and put the v2 modules inplace and del v1 |
| 09:09.42 | irroot | yeah i need to work on the installer for that as its in me Gui so need to accomodate it to autodetect and allow using it as option in config |
| 09:09.46 | irroot | not a big job |
| 09:10.00 | WIMPy | Hehe. On my test box I've taken it as far as choosing via lilo :-) |
| 09:10.16 | jacc0 | like this warning after I do a softhangup(): |
| 09:10.19 | jacc0 | WARNING[2898]: app_softhangup.c:122 softhangup_exec: Soft hanging Local/1153-1@shiftCallfiles-9687;2 up. |
| 09:10.34 | jacc0 | I'll make a patch |
| 09:12.23 | *** join/#asterisk SteveWilliams (~SteveWill@59.162.182.218) |
| 09:12.59 | *** join/#asterisk puzzled (~patrick@puzzled.xs4all.nl) |
| 09:14.17 | jacc0 | any clue when 1.8.8.0 will be comming out? |
| 09:14.31 | SteveWilliams | guys, i have changed the dialplan. here it is http://pastebin.com/rQ6SqjU2. but the guy who is receiving the call is not being able to talk to our office number when he presses 1 although 1 is being recognised by Read() |
| 09:14.48 | SteveWilliams | please help. |
| 09:15.30 | *** join/#asterisk aberrios (~aberrios@195.171.4.82) |
| 09:15.56 | WIMPy | unable to talk is a very vague description. |
| 09:16.41 | kaldemar | SteveWilliams: pastebin a CLI output of a clal |
| 09:16.43 | kaldemar | call |
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| 09:18.05 | SteveWilliams | @ kaldemar okay, one moment please |
| 09:18.36 | irroot | WIMPy laringitus :P |
| 09:19.11 | kaldemar | SteveWilliams: you might as well add dtmf to the console line in logger.conf and do a logger reload to be sure that the 1 comes in to asterisk. |
| 09:19.37 | SteveWilliams | @ kaldemar okay |
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| 09:30.38 | SteveWilliams | @ irroot @ kaldemar I am sorry, my bad. I forgot to reload the dialplan previously(I am such a f00l). It is working correctly. Thanks for all your help! Thank you so much. |
| 09:31.01 | irroot | cool nice now you can sit in the stupid corner :P |
| 09:31.46 | WIMPy | If there's space left. |
| 09:32.39 | SteveWilliams | @ irroot @ WIMPy from what it seems from here, there is still a little space left, for my miserly BOSS! |
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| 09:35.35 | irroot | hehe i have life membership :P |
| 09:36.18 | WIMPy | The golden A-card? Valid for life and not transferrable? |
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| 10:56.15 | catphish | can anyone tell me what is likely to cause silent calls where a sip client is behind a nat but the server is not? |
| 10:57.48 | catphish | my understanding was that in that case the firewall / nat flow was very simple (client sends the first udp packet to the server, then the server can reply on that flow that the nat keeps open) |
| 10:59.17 | Chainsaw | catphish: Well, the call setup over SIP has succeeded. |
| 10:59.30 | Chainsaw | catphish: However... the audio stream is sent as an RTP stream, which is established separately. |
| 10:59.56 | Chainsaw | catphish: So there are two flows, one of which succeeds and the other fails. Generally I find that using a STUN server is the easiest fix, if the client supports it. |
| 11:00.30 | catphish | correct, the sip initiation is fine, but the rtp is i guess either not being allowed out, or the resplies aren't being allowed back, or the replies are going to the wrong port |
| 11:00.45 | catphish | i don't understand what would cause that to happen |
| 11:00.58 | catphish | i did look at stun, but i didnt understand how it solves the problem |
| 11:02.58 | catphish | i guess there are different types of nat, i don't fully understand the difference |
| 11:03.02 | catphish | so i'll just try it :) |
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| 11:06.52 | Chainsaw | catphish: The main problem is that RTP is being established separately from SIP, which is why it often fails. |
| 11:07.20 | Chainsaw | catphish: You will still need to specify nat=yes for the relevant peer. |
| 11:07.31 | catphish | yeah i use nat=yes |
| 11:07.48 | catphish | and i'm puzzled as to why that doesn't resolve the problem fully |
| 11:08.07 | catphish | as i understand it nat=yes allows the rtp endpoint to change dynamically as data is received from a phone |
| 11:08.12 | Chainsaw | catphish: You might find this helpful: http://en.wikipedia.org/wiki/STUN#Protocol_overview |
| 11:10.19 | Lipsum | Anyone familiar with auto provisioning of Polycom phones? Only half of the things stated in the Admin Guide SIP 3.1 seem to work when using SIP 3.1.7? For example, the clock settings work like a charm, but language settings are totally ignored? |
| 11:11.12 | Chainsaw | Lipsum: I do automatic provisioning, but my language settings are relatively simple. (i.e. english) |
| 11:13.03 | catphish | Chainsaw: do you know specifically what problems can be solved by stun but not by nat=yes? |
| 11:13.31 | jacc0 | @catphish: discovering the internet ip |
| 11:14.03 | catphish | jacc0: i was under the impression that nat=yes achieved that |
| 11:14.09 | catphish | by waiting for the first rtp packet |
| 11:14.35 | Chainsaw | catphish: Your first RTP packet won't make it. |
| 11:14.49 | Chainsaw | catphish: SIP is simple, the client on NAT initiates it, so all you have to do is reply. |
| 11:14.52 | catphish | why not? the server isn't behind a nat |
| 11:15.03 | Chainsaw | catphish: And now the server would like to open a session to you. It knows what your external IP is. |
| 11:15.24 | catphish | but can't the client initiate the rtp too? |
| 11:15.26 | Chainsaw | catphish: And your $20 Netgear with its fantastic stateful packet inspection firewall will drop it on the floor. |
| 11:15.36 | Chainsaw | catphish: Why? Because it's not a session that a local client initiated! |
| 11:15.54 | Chainsaw | catphish: RTP is going "the wrong way", shall we say. The difficult way. |
| 11:16.07 | catphish | won't the client initiate the rtp session too though as soon as they speak? |
| 11:16.15 | Chainsaw | catphish: No, call setup happens first. Audio happens later. |
| 11:16.27 | *** join/#asterisk puzzled (~patrick@puzzled.xs4all.nl) |
| 11:16.29 | catphish | does call setup happen over rtp? |
| 11:16.32 | Chainsaw | catphish: In fact, you might be hearing all sorts of call setup tones *before* an audio stream is established, because that's out of band. |
| 11:16.52 | Chainsaw | catphish: No, call setup happens over SIP. That part is easy as pie, and it works fine for you. |
| 11:17.07 | catphish | yep that's what i thought, and that works great |
| 11:17.22 | catphish | so once that's done don't both sides immediately start sending audio? |
| 11:17.23 | Chainsaw | catphish: Let's oversimplify this greatly. |
| 11:17.31 | Chainsaw | catphish: NAT breaks anything but outbound sessions. |
| 11:17.37 | catphish | correct |
| 11:17.49 | Chainsaw | catphish: STUN allows both sides to only make an outbound session, and strings those two streams together. |
| 11:18.04 | Chainsaw | catphish: That is the magic. |
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| 11:19.08 | catphish | hmm, that would certainly be necessary if both ends were behind a nat |
| 11:19.34 | Chainsaw | catphish: Even if one end is a NAT, because it is going to throw the inbound RTP away. |
| 11:19.40 | catphish | correct |
| 11:19.45 | Chainsaw | catphish: And that is why you need STUN. |
| 11:19.59 | catphish | but the client behind that nat will eventually send outbound rtp won't it? |
| 11:20.03 | Chainsaw | catphish: Last I checked stun.ekiga.net is broken, but stun.xten.com should do well. |
| 11:20.16 | Chainsaw | catphish: No. The audio stream originates at the server. |
| 11:20.47 | Chainsaw | catphish: You might well successfully send RTP out. Which would be great if you only want to say something without caring what the other end replies with? |
| 11:21.09 | Chainsaw | (But note that robo-calls annoy a great many people, including me, and that I will cease assisting you immediately) |
| 11:21.42 | catphish | but once the client sends an rtp packet to the server, the server will update its record of that client's location and send data back through the punched hole in theory |
| 11:21.47 | catphish | of course i realise that it doesn't work |
| 11:21.52 | catphish | i just don't see why not :( |
| 11:22.19 | Chainsaw | catphish: Because a successful call is a SIP session for call setup, an A->B RTP stream and a B->A RTP stream. |
| 11:22.29 | Chainsaw | catphish: Pick any two, because the third is never going to happen. |
| 11:22.32 | catphish | oh is the stream not symmetric? |
| 11:22.49 | catphish | i assumed the 2 way audio went over the same pair of ports |
| 11:22.54 | catphish | if not that explains everything |
| 11:22.58 | Chainsaw | catphish: You live in a simpler, gentler world. |
| 11:23.04 | catphish | :) |
| 11:23.20 | Chainsaw | catphish: Out here in the parallel universe, NATs... suck. |
| 11:24.40 | catphish | yeah i hate nat |
| 11:24.51 | catphish | but i do at least know what it does |
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| 11:25.24 | catphish | sadly i didnt realise that 2-way audio used 2 separate udp connections |
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| 11:25.54 | Chainsaw | catphish: Now that you know, you'll understand why "1-way audio" equals "NAT". |
| 11:26.13 | Chainsaw | catphish: Or firewalling, or other horrid things. |
| 11:26.25 | catphish | yeah, nat is evil |
| 11:26.45 | catphish | potentially more so because so many NATs have workarounds for SIP and hide the problem |
| 11:28.12 | Chainsaw | catphish: A man in the middle attack on a machine you don't control. |
| 11:29.14 | Lipsum | Chainsaw, do you have an example of the override sip.cfg which works? |
| 11:30.04 | Chainsaw | Lipsum: There is a lot of config in there which is specific to the company I work for. |
| 11:30.29 | Chainsaw | Lipsum: I can look at individual options if you ask me about them, but I'm not comfortable sharing the full file. |
| 11:31.15 | Lipsum | Chainsaw, aight. A few questions. Everything should be inside <sip> tags, right? |
| 11:31.56 | Chainsaw | Lipsum: There is a main enclosing <sip>, yes. |
| 11:32.25 | Chainsaw | Lipsum: Containing <voIpProt>, <dialplan>, etc. |
| 11:32.58 | Lipsum | Yeah, seems right compared to the admin guide. |
| 11:33.47 | Chainsaw | Lipsum: My phones push logs back into the provisioning server using webdav, so I can see how things have been interpreted on the handset. |
| 11:34.10 | Lipsum | And we have <lcl><lcl.ml lcl.ml.lang="Norwegian_Norway" /><ml lcl.ml.lang="Norwegian_Norway" /></lcl> AND <TCP_IP><SNTP tcpIpApp.sntp.daylightSavings.enable="1" tcpIpApp.sntp.address="127.0.0.1" tcpIpApp.sntp.gmtOffset="+3600" /></TCP_IP> inside as well. |
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| 11:34.34 | Lipsum | Both according to the admin guide... However, the language doesn't work at all. The time thing does. |
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| 11:35.41 | jmls | hey |
| 11:35.42 | Lipsum | We also push back log files... But there's nothing in them about any language. |
| 11:35.45 | Chainsaw | Lipsum: It is part of the default load for an IP670 at least. I have Norsk (no-no) as option 13 in Language. |
| 11:35.56 | jmls | can you set more than one group in a SPYGROUP ? |
| 11:36.08 | jmls | set(SPYGROUP=foo,bar) |
| 11:36.19 | jmls | and then chanspy on either foo or bar ? |
| 11:36.20 | Lipsum | Chainsaw, we have "Norwegian_Norway" on that label in the sip.cfg file for IP601. |
| 11:36.42 | Chainsaw | Lipsum: language lcl.ml.lang="English_United_Kingdom" seems to work here. |
| 11:37.28 | Chainsaw | Lipsum: I don't think you should be closing that language tag like that though. |
| 11:37.46 | Chainsaw | Lipsum: In my case the language tag has menu, clock, list & tags subtags nested one level deeper. |
| 11:38.31 | Lipsum | In the sip.cfg or the override sip.cfg? |
| 11:39.20 | Chainsaw | Lipsum: I don't override sip.cfg, I supply my own. |
| 11:39.21 | Lipsum | uhm, we have: <lcl><ml lcl.ml.lang="Norwegian_Norway" /></lcl> |
| 11:39.37 | Lipsum | Not the first thing I pasted. |
| 11:40.07 | Chainsaw | Lipsum: http://pastebin.com/E7sfj64z |
| 11:40.27 | Chainsaw | Lipsum: That is wrangled by linewrapping etc, but you get the idea hopefully. |
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| 11:44.21 | Lipsum | yeah, but thats the sip.cfg file... the doc states that you should not touch it (understandable based on the fact that if you upgrade sip version, youll probably get a new sip.cfg file which then needs to be changed again), and instead make a "company" based sip.cfg which will override the default sip.cfg. |
| 11:44.35 | catphish | right, brought up a stun server |
| 11:44.39 | catphish | we'll see if that helps |
| 11:44.49 | catphish | thanks for the pointers Chainsaw |
| 11:45.02 | Chainsaw | catphish: STUN servers work best when they are not on your own network. |
| 11:45.10 | Chainsaw | catphish: As such, I would recommend testing with stun.xten.com |
| 11:45.52 | catphish | is it a problem to run it on the same ip as the asterisk server? |
| 11:45.57 | Chainsaw | catphish: Yes. |
| 11:46.16 | catphish | well that's annoying |
| 11:46.20 | Chainsaw | catphish: You want to see the external IP of both clients, and the only way to do that is to not be on the same subnet as the Asterisk server or the remote client. |
| 11:46.55 | catphish | does that apply even if i know that the asterisk host is not natted? |
| 11:47.39 | Chainsaw | catphish: Do you want a sure-fire solution, or do you have time to spend? |
| 11:49.15 | *** join/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
| 11:49.29 | catphish | i have some time, though what i don't have is many nats to test with |
| 11:50.11 | Chainsaw | catphish: Then I would suggest that you use stun.xten.com and confirm that the problem disappears immediately. |
| 11:50.24 | zamba | can someone please explain the difference and relation between register and a stanca ([])? |
| 11:50.30 | zamba | i still haven't figured it out |
| 11:50.45 | Chainsaw | catphish: You will then have time afterwards to play around with stund, find out it is unmaintained and won't even build in most cases, that it needs to be outside your own network to work properly, etc. |
| 11:50.51 | zamba | the reason i'm asking is that i have three accounts from the same voip provider, which three different phone numbers |
| 11:50.52 | catphish | fortunately i am very confident about my own network config, just not my customers' |
| 11:50.55 | zamba | so i have three register statements |
| 11:51.17 | catphish | Chainsaw: i downloaded 'stunserver' (someone's project on github) |
| 11:51.24 | catphish | which appears to work |
| 11:51.30 | zamba | and now the syntax is like this: register => number@host:password:number@host:port/number |
| 11:51.37 | zamba | which seems a bit redundant..? |
| 11:51.42 | catphish | however sadly i don't have my own broken nat to test with |
| 11:51.54 | Chainsaw | catphish: No 3G dongles available to you? |
| 11:52.23 | Chainsaw | catphish: Pub/restaurant with WiFi? Starbucks? |
| 11:52.37 | catphish | no, i can run some packet dumps on the stun server though and see if it works as it should with my linux nat |
| 11:52.52 | catphish | it it used for calls? or does the phone just poll it occasionally? |
| 11:52.54 | Chainsaw | catphish: Testing is only valid from outside your own network. |
| 11:53.31 | Chainsaw | catphish: Pick up your laptop, take it somewhere where you have no control over the network, try from there. |
| 11:53.51 | catphish | yes that makes sense, just realised my office isn't actually natted at all from my asterisk servers |
| 11:53.58 | catphish | i'll go next door :) |
| 11:54.06 | Chainsaw | catphish: That should do it, yes. |
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| 11:54.40 | Chainsaw | catphish: Use stun.xten.com as the "known good", and then your own STUN server as an experiment that may fail. |
| 11:54.53 | Chainsaw | catphish: And without any STUN as an example of 1-way audio. |
| 11:55.16 | catphish | Chainsaw: is the stun server used when initiating calls? |
| 11:55.34 | catphish | or just occasionally by an idle client? |
| 11:55.36 | Chainsaw | catphish: I would only expect it to be used once the RTP streams are establishing. |
| 11:55.43 | catphish | ok |
| 11:55.59 | Chainsaw | catphish: Not during the call setup phase. But I haven't read all the RFCs, that is an educated guess, not a statement of fact. |
| 11:56.25 | catphish | thanks |
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| 12:24.22 | eduzimrs | hi, anyone knows what is this > WARNING[21694] func strings.c: No argument specified! its appearing recently at * cli |
| 12:26.01 | [TK]D-Fender | eduzimrs, Perhaps you should showeverything around it... |
| 12:26.25 | [TK]D-Fender | eduzimrs, Because that looks like you made a dialplan error involving strings... |
| 12:29.35 | eduzimrs | hum... tks ill look at |
| 12:30.00 | eduzimrs | [TK]D-Fender probably shoud be |
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| 13:37.56 | scubes13 | anyone using web-meetme 3 on a 64-bit system?I ma having problems compiling cbmysql |
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| 13:46.55 | eppigy | good morning |
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| 13:51.08 | m4xx | when i try to restart my asterisk instance my box freezes |
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| 13:52.45 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 13:53.05 | m4xx | and i have to call the dc and ask them to reboot the box |
| 13:54.01 | [TK]D-Fender | what is an "asterisk instance"? |
| 13:54.43 | jacc0 | @[TK]D-Fender: maybe they mean a running process |
| 13:55.07 | [TK]D-Fender | doesn't run on "maybes" |
| 13:55.41 | m4xx | jacc0: is correct =\ |
| 13:56.24 | eppigy | don't say maybe baby |
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| 13:57.52 | m4xx | not sure what the problem is. nothing else is running on the box. |
| 13:59.01 | jacc0 | I couldn't be sure because I'm a non native english speaker and didn't know the context\ |
| 14:05.37 | Katty | pokes eppigy |
| 14:05.51 | beek | waves to Katty |
| 14:06.01 | Katty | hai beekster! |
| 14:06.02 | Katty | hugs beek |
| 14:06.08 | Katty | how're you dear |
| 14:06.11 | eppigy | pokes Katty back |
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| 14:06.14 | eppigy | :> |
| 14:06.14 | Katty | :> |
| 14:06.15 | beek | Very good... yourself? |
| 14:06.16 | Katty | glomps eppigy |
| 14:06.23 | eppigy | 8[] |
| 14:06.28 | Katty | beek: healin. will be right as rain soon |
| 14:07.01 | Katty | eppigy: we have lots o catching up to do |
| 14:07.41 | beek | Katty: Good to hear! |
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| 14:12.56 | *** join/#asterisk lovetide (~LiuYan@222.125.132.191) |
| 14:13.00 | lovetide | join postgresql |
| 14:13.56 | beek | lovetide: try adding a slash and octothorpe |
| 14:14.23 | lovetide | oops, sorry & thanks |
| 14:14.25 | Naikrovek | slashhash |
| 14:14.47 | *** join/#asterisk willzzz (~Will@gateway.meteor-web.com) |
| 14:14.52 | Katty | i'll hash your slash in a minute. |
| 14:15.05 | willzzz | guys i'm using g729 for outbound trunk but my internal LAN IP phones are u-law (g711) |
| 14:15.06 | willzzz | <PROTECTED> |
| 14:16.23 | eppigy | Katty: yesh lets do it over coffee |
| 14:26.07 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 14:32.32 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 14:34.12 | *** join/#asterisk DennisG (~DennisG@ip5454b5b3.adsl-surfen.hetnet.nl) |
| 14:39.52 | [TK]D-Fender | willzzz, And have you activated your G.729 licenses and verified they are ready at * CLI? |
| 14:40.29 | [TK]D-Fender | willzzz, If so, show us, along with the complete failed call at * CLI with SIP debug enabled. Pleas pastebin all of this |
| 14:40.32 | [TK]D-Fender | ~pb |
| 14:40.32 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com , http://pastebin.us , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ ; or install pastebinit with yum or aptitude. |
| 14:40.38 | willzzz | TK, "exited non-zero on" |
| 14:40.54 | [TK]D-Fender | willzzz, "entire-call". |
| 14:41.35 | carrar | Katty drinks coffee? |
| 14:42.18 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 14:42.40 | Katty | no, no she doesn't |
| 14:42.53 | eppigy | aww i thought you were a caffeine junky like me |
| 14:42.58 | Katty | i am! |
| 14:43.03 | eppigy | oh :D |
| 14:43.21 | eppigy | well we will catch up over caffeinated beverages |
| 14:45.34 | willzzz | D, http://pastebin.com/RLErgfsM |
| 15:09.58 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 15:09.58 | *** topic/#asterisk is #asterisk Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 10.0.0-beta2 (2011/09/27), 1.8.7.1 (2011/10/17), dahdi-linux 2.5.0.1 (2011/09/08), dahdi-tools 2.5.0.1 (2011/09/08), libpri 1.4.12 (2011/07/06) -=- Visit the official Asterisk wiki: wiki.asterisk.org -=- http://xkcd.com/806/ |
| 15:10.08 | *** join/#asterisk kpettit (~kpettit@99-116-144-138.lightspeed.hstntx.sbcglobal.net) |
| 15:10.59 | kpettit | Is there a list anywhere of SIP phones/models with features that work with asterisk? I've only got experience with Polycom and wanted to try some other brands. |
| 15:11.20 | Chainsaw | kpettit: Between you and me, you already have the best... |
| 15:11.24 | Naikrovek | kpettit: I don't know, but if you find one, please let me know. |
| 15:11.39 | Naikrovek | but yes you already have the best. the only thing Polycom needs tos upport that it doesn't is the SIP REASON header |
| 15:11.57 | kpettit | Chainsaw, I like Polycom. But they are pretty pricey. So I wanted to find a lower cost alternative where most of the features work. |
| 15:12.09 | kpettit | ah. |
| 15:12.35 | Naikrovek | pricy? kk. |
| 15:12.39 | navaismo | kpettit Try with Aastra, Grandstream, Snom Yealink |
| 15:12.39 | irroot | im more a snom fan they have cleaned up all the issues ive had and they good value |
| 15:12.42 | Naikrovek | we bought them because they were cheap |
| 15:12.43 | WIMPy | What features? |
| 15:13.05 | irroot | yealink are ok once they get working |
| 15:13.21 | WIMPy | Oh, I know a lot of issues with the Snoms, but others seem to be worse. |
| 15:13.21 | navaismo | I preffer Aastra |
| 15:13.27 | kpettit | There is one phone I was anxious to try becuase it has a openVPN client built right in. But I have no idea how many of the normal phone features work with asterisk . Yealink was the brand on that. |
| 15:13.54 | WIMPy | Snom 370 also has OpenVPN support. |
| 15:14.02 | navaismo | SIP PHONES works on asterisk |
| 15:14.10 | kpettit | navaismo, All of those work well with asterisk? Mainly I'm concerned about DND, Forwarding, Hold, Call Parking, etc using the native phone buttons |
| 15:14.16 | WIMPy | The 8xx probably as well, I haven;t really looked at them. |
| 15:14.30 | irroot | kpettit our one "partner" uses yealink its actually not 1/2 bad HD fro entry models and openvpn provisioning is easy enough |
| 15:14.32 | navaismo | yes kpettit all of them work on asterisk |
| 15:14.33 | DennisG | I have a lot of experience with Aastra and the price / quality value is very good! |
| 15:14.47 | WIMPy | kpettit: Should work with almost any phone. |
| 15:14.50 | navaismo | yes Go Aastra, |
| 15:15.19 | kpettit | Cool. Great. I'll try some out. I have a little bit of budget so I can try 2-3 different kind of phones. |
| 15:15.30 | DennisG | Yealink is cheap but it's just like a toy if you compare it with Aastra, Snom or Polycom haha :) |
| 15:15.39 | kpettit | Maybge I'll try Aastra and Yealink |
| 15:15.41 | DennisG | and the Yealink has a poor translation for some languages.. |
| 15:15.50 | WIMPy | I like the task switching concept of Avaya. |
| 15:15.59 | kpettit | DennisG, I saw that on some of the forums for it. |
| 15:16.17 | kpettit | WIMPy, haven't heard of that. |
| 15:16.18 | irroot | the new yealink fw has iorned out issues |
| 15:16.48 | navaismo | Yealink its the cheaper alternative |
| 15:17.01 | DennisG | yeah it's just what you need.. |
| 15:17.22 | kpettit | Thanks for the suggestions, I'll give a few of them a try. |
| 15:17.28 | DennisG | n.p. |
| 15:18.00 | kpettit | FYI Mitel really sucks. I had a few of those around a changed them to SIP, but beyond basic service most of the buttons don't work with asterisk. |
| 15:18.02 | DennisG | btw.. cisco is also cheap! |
| 15:18.44 | kpettit | I was wondering about that. Is Cisco phones changed to SIP very decent? Do all the phone features work with asterisk? |
| 15:18.49 | *** join/#asterisk _Corey_ (~chatzilla@mail1.infradapt.com) |
| 15:20.53 | DennisG | i've worked with some Cisco phones and with some Cisco ATA's and the ATA's are very simple.. the phones are just what you expected from cisco. just a lot of features. |
| 15:21.05 | DennisG | which phone features do you need for Asterisk? |
| 15:21.15 | DennisG | Like BLF support or something like that? |
| 15:21.18 | WIMPy | kpettit: Avaya = Lucent+Tenovis+Nortel |
| 15:21.32 | Chainsaw | kpettit: Cisco phones can be extremely temporamental. |
| 15:21.37 | kpettit | I want to make sure the buttons on the phone for DND, Forwarding, Paging, Address Book, etc work. |
| 15:21.39 | Chainsaw | kpettit: I had 7960s and wouldn't recommend them to anyone. |
| 15:22.11 | DennisG | witch cisco you can assign special values for some hard buttons.. like forwarding = ** |
| 15:22.37 | DennisG | about DND i don't know.. i always use Aastra if the customer need the DND functionality |
| 15:22.46 | kpettit | ahh. That's good to know. Some people are set on Cisco so it's good to know what I can do with them. |
| 15:22.47 | DennisG | because Aastra have a lot of buttons and options for this |
| 15:22.47 | WIMPy | kpettit: That's all local funciotns, not really related to Asterisk. |
| 15:22.53 | DennisG | with* |
| 15:22.59 | *** join/#asterisk brdude (~brdude@c-24-5-194-184.hsd1.ca.comcast.net) |
| 15:23.32 | kpettit | Got ya |
| 15:23.41 | DennisG | but if you need just some simple SIP functions (answer, hangup, transfer) then almost every SIP device is enough for the project.. |
| 15:25.17 | kpettit | I'm mainly trying to pick 2 or 3 model's (Cheap, OK and Very Nice) to offer to customers. there are so many out there just trying to widdle the list down to ones that work better |
| 15:25.51 | p3nguin | Good old "model is." |
| 15:26.08 | Qwell | ugh |
| 15:26.10 | Qwell | please don't buy Cisco |
| 15:26.27 | Qwell | I will laugh at you for paying $100 extra per phone, for a license to use the phone. |
| 15:26.44 | p3nguin | Black market wins again! |
| 15:27.01 | kpettit | Qwell, Ouch, that sucks. You have to do that even if you are doing SIP? |
| 15:30.06 | *** join/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net) |
| 15:30.19 | p3nguin | You're supposed to pay for licensing fees to use the phones. |
| 15:31.27 | *** join/#asterisk jeffspeff (~jeffspeff@173-11-144-149-houston.txt.hfc.comcastbusiness.net) |
| 15:31.54 | Qwell | kpettit: It's a license to use the software on the phone. |
| 15:32.19 | kpettit | Qwell, that's just wonderfull. Guess those are off my list |
| 15:36.00 | jeffspeff | i'm having trouble with my users getting a busy signal when calling our DID. We have been back and forth over more than a week withour SIP provider without making any progress. If somebody could take a look at these logs and help point us in the right direction it would be greatly appreciated. The 3rd call is what failed. sip debug log --> http://pastebin.com/cX4seUyM sip cli output --> http://pastebin.com/sQiM |
| 15:36.00 | jeffspeff | h4wT |
| 15:36.02 | navaismo | we never pay anything for using cisco phones, this only apply on USA? |
| 15:37.33 | [TK]D-Fender | navaismo, No, theft is multi-national complaint. |
| 15:37.43 | [TK]D-Fender | compliant* |
| 15:37.48 | *** join/#asterisk vinhdizzo (~vinh@vqn-portege.ics.uci.edu) |
| 15:38.59 | Qwell | kpettit: http://ntbox.com/cisco-openletter.html |
| 15:39.02 | WIMPy | And many widely used licencing models are illegal in big parts of the world. |
| 15:39.13 | Qwell | kpettit: feel free to skip the first part, and go straight to their answer |
| 15:39.21 | WIMPy | So location usually does matter. |
| 15:39.59 | *** join/#asterisk vinhdizzo (~vinh@vqn-portege.ics.uci.edu) |
| 15:40.11 | Qwell | kpettit: they spent a *long* time getting this sanitized by their lawyers before sending it to me. |
| 15:40.11 | kpettit | Qwell, Ah cool. Reading that now |
| 15:40.23 | Qwell | I want to say it was like 10 weeks |
| 15:41.51 | kpettit | wow. I think having to deal with licensing on a phone really sucks. |
| 15:42.34 | Qwell | it wouldn't be so bad if the licenses weren't freaking $100 |
| 15:42.50 | Qwell | (also if their phones didn't suck) |
| 15:43.37 | kpettit | no kidding. They prices on the phones looked good until you consider adding that $100 to the price. |
| 15:43.44 | Qwell | well, no, not really. heh |
| 15:43.49 | Qwell | They're expensive even without the license :p |
| 15:44.00 | Qwell | (a bit more expensive than competition) |
| 15:44.14 | Qwell | not a hell of a lot, but still |
| 15:44.25 | kpettit | yeah. |
| 15:44.55 | Qwell | You're paying for the logo, really. |
| 15:45.28 | jeffspeff | could somebody tell me what is meant by this? [Oct 20 10:13:52] DEBUG[1334] chan_sip.c: Processing session-level SDP o=FreeSWITCH 1319106380 1319106381 IN IP4 63.211.239.14... UNSUPPORTED. |
| 15:46.02 | jeffspeff | i'm seeing several unsupported sdp messages with calls from my sip provider |
| 15:47.15 | puzzled | jeffspeff: dunno. maybe it's sending video stuff in the sdp which you may have turned off? use wireshark to make a trace and then analyze it for more details |
| 15:48.04 | WIMPy | Hmm. I just wanted to check what shops said about Cisco licenses and had to find out that almost none sell them. Probably because of that. |
| 15:48.08 | *** join/#asterisk blizzow (~jburns@67.50.165.58) |
| 15:49.25 | p3nguin | Because of what? |
| 15:49.35 | WIMPy | The licence issue. |
| 15:49.50 | p3nguin | They don't sell the license because of the license issue? |
| 15:50.04 | WIMPy | Do you need different licenses depending on what you want to connect the phone to? |
| 15:50.12 | WIMPy | No the phone. |
| 15:50.21 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 15:50.31 | Qwell | WIMPy: The other related problem is that nobody sells it because nobody knows about it. |
| 15:50.38 | WIMPy | YouIf you need the licence, you can't sell the phone without. |
| 15:50.51 | Qwell | They can, and do. |
| 15:51.31 | p3nguin | The easiest place to get the phones without a license is ebay. |
| 15:51.38 | WIMPy | In the US. |
| 15:52.27 | Qwell | If I put a pirated app on a computer and sell that on eBay, it does not mean that anybody can use that app just because they paid me for the computer. |
| 15:52.33 | WIMPy | How do you find out if it has a licence? |
| 15:52.41 | Qwell | You don't. |
| 15:52.53 | Qwell | They are non-transferrable. |
| 15:53.16 | WIMPy | See that's an US thing again. |
| 15:53.22 | p3nguin | Don't confuse "can" with "allowed to legally." If you sell the pirated app, they most likely *can* use it. |
| 15:54.09 | WIMPy | So that might become an interesting situation of you buy Cisco phones abrouad. |
| 15:55.01 | puzzled | WIMPy: if you buy a Cisco phone called "spare" then it comes without a license. afaik the other ones offered online come with a license |
| 15:55.26 | p3nguin | That's from a registered reseller. |
| 15:55.37 | puzzled | but why Cisco when there is Polycom who does not make you buy a support contract so you can download firmware that fixes *bugs* |
| 15:55.38 | p3nguin | The term "spare" isn't something bootleggers ever use. |
| 15:55.50 | puzzled | Cisco does |
| 15:55.59 | WIMPy | Hmm. The whole device as spare. Might work. |
| 15:58.10 | Qwell | This is Cisco. Their legal budget is bigger than some Fortune 500 companies are big. |
| 15:58.33 | p3nguin | It is my understanding that it is cheaper to buy a spare from a reseller than it is to buy one that isn't considered spare from the same reseller.. |
| 15:59.01 | puzzled | correct. the spare does not carry the additional cost of the SIP or H.323 license that comes with the other SKU's |
| 15:59.25 | Qwell | s/SIP or H.323/firmware use/ |
| 15:59.49 | puzzled | Qwell: iirc the SCCP firmware is free so not sure about that |
| 15:59.53 | Qwell | incorrect |
| 15:59.57 | puzzled | :) |
| 16:00.02 | WIMPy | Qwell: That's probably true for M$ as well, but still they continousely find out the hard way, that the law isn;t the same everywhere. |
| 16:00.26 | WIMPy | puzzled: That's what I thought so far as well. |
| 16:00.40 | Qwell | According to Cisco, *any* firmware requires a license. |
| 16:01.11 | Qwell | see the link above |
| 16:01.37 | puzzled | WIMPy: there's a link on this page that points to a site with a ton of Cisco firmware. For educational purposes only off course :) http://www.voip-info.org/wiki/view/Cisco+Phones |
| 16:02.19 | WIMPy | puzzled: I do only have a nostalgiv 30vip, so there's nothing for me anyway. |
| 16:03.00 | Qwell | I like my 30vip |
| 16:03.26 | WIMPy | Do you have it running? |
| 16:03.31 | *** join/#asterisk jrose_atDigium (~jon@nat/digium/x-krkdufrazyyxmaxh) |
| 16:03.58 | Qwell | WIMPy: I haven't booted it in like 3 years probably. It worked fine with chan_skinny last I tried though. |
| 16:04.23 | WIMPy | I only got it to work partially and that was with 1.4 I think. |
| 16:04.26 | *** join/#asterisk Ionic (ionic@ionic.de) |
| 16:04.43 | WIMPy | No go since then. |
| 16:28.59 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 16:28.59 | *** topic/#asterisk is #asterisk Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 10.0.0-beta2 (2011/09/27), 1.8.7.1 (2011/10/17), dahdi-linux 2.5.0.1 (2011/09/08), dahdi-tools 2.5.0.1 (2011/09/08), libpri 1.4.12 (2011/07/06) -=- Visit the official Asterisk wiki: wiki.asterisk.org -=- http://xkcd.com/806/ |
| 16:31.44 | WIMPy | yebyen: What were you looking for then? |
| 16:32.00 | *** join/#asterisk Ionic (ionic@ionic.de) |
| 16:32.08 | *** join/#asterisk brdude (~brdude@12.155.183.30) |
| 16:33.30 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 16:36.31 | dym | works from here. |
| 16:36.50 | puzzled | me too |
| 16:37.16 | dym | I've been ddosing it for hours - i should know. |
| 16:37.31 | dym | (: |
| 16:37.45 | *** join/#asterisk irroot (~irroot@41.55.47.152) |
| 16:37.53 | puzzled | hi irroot |
| 16:38.59 | *** join/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net) |
| 16:42.27 | *** join/#asterisk _Corey_ (~chatzilla@mail1.infradapt.com) |
| 16:43.08 | *** join/#asterisk shido6 (~shido6@nat/yahoo/x-izcwfmrzhbiolpzk) |
| 16:46.12 | jsarrel | ?pb |
| 16:46.49 | *** join/#asterisk MarKsaitis (~MarKsaiti@cpc4-rdng22-2-0-cust932.15-3.cable.virginmedia.com) |
| 16:48.20 | p3nguin | ~pb |
| 16:48.20 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com , http://pastebin.us , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ ; or install pastebinit with yum or aptitude. |
| 16:51.14 | pigpen | it has been too long since I have addressed this, but I have sip phones, asterisk 1.8.5, and a sip audiocodes fox unit. During a call, dtmf digits are not passed (like for "press 1 for accounting") |
| 16:51.29 | pigpen | and I cannot remember what the solution is, since it has been so long. |
| 16:52.45 | pigpen | is it the dtmfmode=rfc2833 in the sip.conf? |
| 16:52.52 | pigpen | :q! |
| 16:52.55 | pigpen | dam |
| 16:53.12 | Lipsum | Anyone familiar with Polycom autoprovisioning? I can not make the phone request the language file I've selected? |
| 16:53.34 | Naikrovek | Lipsum: which language is it getting |
| 16:53.55 | Lipsum | It's using the default english (which is by default on the phone) |
| 16:54.01 | pigpen | if you have had it grab a different language once, you may have to format the phone to grab another, but unsure, never did anything with the language selection. |
| 16:54.10 | *** join/#asterisk greenwolf (~chatzilla@cpe-74-77-195-238.buffalo.res.rr.com) |
| 16:54.10 | *** join/#asterisk coppice (~chatzilla@183178203098.ctinets.com) |
| 16:54.24 | Lipsum | If I change the language file, it doesn't ask for anything and just keeps using English. |
| 16:54.40 | Lipsum | pigpen, I've tried formating it as well. |
| 16:55.33 | pigpen | change the language file at 1) the phone 2) web interface of the phone 3) cfg file 4) other |
| 16:58.19 | pigpen | if you are a bad test taker, the best bet would be to select C or #3. |
| 16:58.21 | pigpen | ;-) |
| 16:58.42 | pigpen | and no, I didn't plan that. Just lucky. |
| 16:59.35 | *** join/#asterisk VultureZ (~Vulture@gateway/tor-sasl/vulturez) |
| 17:00.41 | *** join/#asterisk _Corey_ (~chatzilla@mail1.infradapt.com) |
| 17:00.42 | Lipsum | When I curl for the sip.cfg file which I see the phone asks for (through tailing the httpd access log), I see <language lcl.ml.lang="Norwegian_Norway"> |
| 17:01.01 | Lipsum | So the phone gets the language... But still, it's never being asked for. |
| 17:01.26 | [TK]D-Fender | Lipsum, And all the language folders are in your provisioning folder? |
| 17:01.54 | Lipsum | Well, yeah, but the phone never asks for the language, so it really stops before it can find the langauge file. |
| 17:02.12 | pigpen | are you watching the ftp/http/tftp log? |
| 17:02.16 | Lipsum | yeah. |
| 17:02.48 | [TK]D-Fender | verify permissions and the the folder structure is proper |
| 17:03.04 | Lipsum | That's not the problem. |
| 17:03.15 | yebyen | WIMPy: now it's loaded... *shrug* |
| 17:03.24 | Lipsum | The phone never sends GET language-file... |
| 17:03.43 | Lipsum | It never asks for the language file... |
| 17:03.46 | pigpen | [TK]D-Fender, if I cannot pass dtmf during a sip call, like press 1 type thing, on 1.8.5, what is it likely again? I just can't get my memory to kick in on this little coffee. |
| 17:04.17 | yebyen | WIMPy: i was looking for the part on why you want those priority numbers so you can make them overlapping... |
| 17:04.19 | [TK]D-Fender | pigpen, DTMFMODE mismatch. in your peer try eaaach |
| 17:04.21 | Naikrovek | DTMFMODE |
| 17:04.25 | Naikrovek | what he said |
| 17:04.29 | yebyen | WIMPy: thought that was what you were saying, as an example of something you can't do with AEL |
| 17:04.41 | Naikrovek | inband will likely always work, but it's the least desirable option, and won't work if you're using g729 |
| 17:05.26 | pigpen | ah..k I have DTMFMODE=RFC2833 |
| 17:05.47 | pigpen | yeah, using ulaw |
| 17:06.51 | Lipsum | No ideas? :> |
| 17:06.55 | WIMPy | yebyen: Oh, that one. I already removed that again. |
| 17:07.14 | Naikrovek | Lipsum: where are you setting the language |
| 17:07.17 | [TK]D-Fender | Lipsum, reset your choice and try choosing from the phone itself. |
| 17:07.25 | jeffspeff | pigpen, i'm using SIP + ulaw + rfc2833 and our dtmf's work fine... possibly another issue? |
| 17:07.28 | Naikrovek | make sure the conf file you're setting that up in is being read. |
| 17:07.34 | *** join/#asterisk _Corey_ (~chatzilla@mail1.infradapt.com) |
| 17:07.37 | yebyen | WIMPy: ah, so i was too late :) |
| 17:07.51 | Naikrovek | jeffspeff: depends on what his provider expects. if provider doesn't look for RFC2833, then RFC2833 won't work. |
| 17:08.12 | jeffspeff | Naikrovek, true |
| 17:08.17 | Naikrovek | SIP INFO, RFC2833 usually work, though. Mine only works for inband. :( |
| 17:08.43 | WIMPy | yebyen: Put it back again to http://wimpy.yeti.dk/pastebin |
| 17:08.47 | yebyen | WIMPy: well i am going out for lunch again, so you can post it and remove it before I have a chance to read... |
| 17:08.53 | pigpen | jeffspeff, yeah, I have been using it for years with no issue, but, in the 1.8.x's you never know. |
| 17:09.34 | jeffspeff | pigpen, Asterisk 1.8.5.0 built by root @ sip.starengr.local on a x86_64 running Linux on 2011-08-05 20:39:27 UTC |
| 17:09.37 | yebyen | WIMPy: ok :) i have the text |
| 17:09.49 | yebyen | i'll try to figure out what you're doing with it when ig et back |
| 17:10.07 | WIMPy | OKI |
| 17:10.23 | pigpen | Also a note, I have always used digium cards (i.e.: Zap/Dahdi). These are the first installations using Audiocodes SIP FXO units. |
| 17:10.31 | pigpen | jeffspeff, using Audiocodes? |
| 17:10.57 | jeffspeff | pigpen, no, SIP only |
| 17:11.56 | pigpen | yeah, we are all sip too, except when we pass out the fxo on the audio codes |
| 17:12.29 | *** join/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net) |
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| 17:16.39 | pigpen | I will know in a bit if it works. |
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| 17:18.39 | *** part/#asterisk irroot (~irroot@41.55.47.152) |
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| 17:39.33 | scubes13 | have a client that is having an issue where they are having 911 hangups from their location⦠have looked in Reporting and dont see any calls to 911⦠is there any way I can temporarily disable calls to 911 via the phone system so that we know for sure it is not coming from the pbx? |
| 17:39.34 | *** join/#asterisk Fritz09 (~Adium@pop1-1170.catv.wtnet.de) |
| 17:45.16 | Naikrovek | scubes13: your provider probably offers a 911 test number (mine is 711) that you can use to test all you want. |
| 17:45.44 | Naikrovek | but yes, you can route those calls directly to Hangup() I think. Thought, I doubt you want that, really. |
| 17:46.11 | scubes13 | just trying to long enough to make certain the random calls to 911 are not coming from us |
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| 17:46.53 | *** join/#asterisk JonathanRose (~jon@nat/digium/x-hvunklarqwunlkgc) |
| 17:47.45 | Naikrovek | does that client have to dial 9 to get out? i used to have a lot of people try to dial internationally by dialing 911xx when it was supposed to be 9011xx.. |
| 17:48.01 | Naikrovek | they wound up with 911 operator |
| 17:48.21 | Naikrovek | dialing 9 on voip system .. bad. |
| 17:48.24 | Naikrovek | anyway |
| 17:48.37 | Naikrovek | yes, you can route 911 to wherever you want in the dialplan |
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| 17:48.58 | *** mode/#asterisk [+o mnicholson] by ChanServ |
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| 17:56.52 | scubes13 | sorry - no, the client doesnt have to dial 9 to get out |
| 17:57.13 | scubes13 | they are on a POTS line connected to Sangoma card so not using voip at the moment |
| 17:58.16 | scubes13 | these 911 calls are occurring after hours when no one is on site |
| 18:00.11 | p3nguin | Put an explicit extension to 911 in the dialplan. Mine does Playback(no-911). |
| 18:00.22 | p3nguin | "This system does not have 911 services." |
| 18:01.20 | phpboy | has the agent login functionality been done away with? |
| 18:01.24 | phpboy | asterisk 1.8.* |
| 18:01.46 | scubes13 | p3nguin - how would I do that re: explicit extension in dialplan? |
| 18:02.00 | scubes13 | is currently using FreePBX to do configs |
| 18:02.45 | [TK]D-Fender | make a route for 911. Put it first |
| 18:04.10 | scubes13 | ok⦠now how do I tell that route where to go? |
| 18:04.28 | *** join/#asterisk _Corey_ (~chatzilla@mail1.infradapt.com) |
| 18:04.35 | *** join/#asterisk xpot-mobile (~james@dhcp69.emcb.utah.edu) |
| 18:04.59 | scubes13 | [TK]D-Fender right now I just see trunks to select from... |
| 18:05.24 | [TK]D-Fender | scubes13, Actually.. just make sure NO route can hit it. |
| 18:05.55 | *** join/#asterisk Dovid (~Dovid@213.8.121.90) |
| 18:05.59 | scubes13 | ie, just make sure 911 is not in dial plan of any route (only have one outbound route at the moment) |
| 18:07.19 | p3nguin | scubes13: No clue. This isn't a FreePBX channel, and I can't support FreePBX. To do it in asterisk, you open up extensions.conf and add the extension in the appropriate context. |
| 18:07.32 | scubes13 | fair enough |
| 18:09.48 | Dovid | hates freepbx |
| 18:11.14 | *** join/#asterisk _Corey_ (~chatzilla@mail1.infradapt.com) |
| 18:17.36 | Linuturk | scubes13: /join #freepbx |
| 18:18.03 | scubes13 | Linuturk - I did. Thanks. Testing now. |
| 18:18.15 | Linuturk | :-D |
| 18:19.53 | phpboy | :\ |
| 18:20.08 | phpboy | Managing Agents in 1.8.* is quite a mission it seems |
| 18:20.42 | *** part/#asterisk e7e5 (~rudenko@188.134.2.33) |
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| 18:23.36 | nhaynes | anyone have any idea why when i do call forward all it will work for a local number but if you have to put in the area code it doesn't work |
| 18:24.03 | nhaynes | so say i want to forward to my cell phone which isn't a local number for the area it won't work but someone with a local cell phone it does work |
| 18:24.12 | nhaynes | my phone lines come in via pri |
| 18:25.40 | *** join/#asterisk afink (~afink@204.26.87.226) |
| 18:28.35 | [TK]D-Fender | nhaynes, Because that's how your telco works. |
| 18:30.56 | *** join/#asterisk _Corey_ (~chatzilla@mail1.infradapt.com) |
| 18:31.08 | nhaynes | <[TK]D-Fender> is that typical with pri providers? |
| 18:31.30 | nhaynes | like should they be able to fix that? |
| 18:32.30 | [TK]D-Fender | Your description is also confusing to follow. |
| 18:33.14 | nhaynes | If i do a call forward all and then set the number and i put in a local number and call that extension it will forward |
| 18:33.24 | nhaynes | but if it's a number that requires an area code it will not |
| 18:33.32 | willzzz | SIP/2.0 404 Not Found |
| 18:33.37 | willzzz | aastra firmware bug |
| 18:33.39 | willzzz | i hate you aastra |
| 18:33.41 | nhaynes | and it just goes to the voicemail of the extension after however many rings |
| 18:35.11 | willzzz | does anyone here know about the aastra phone gruu problem? |
| 18:35.31 | [TK]D-Fender | willzzz, Where do we see this "bug"? |
| 18:35.53 | willzzz | http://pastebin.com/84FtS5TK |
| 18:35.58 | willzzz | http://forum.sipfoundry.org/index.php?t=msg&goto=43614 |
| 18:36.06 | willzzz | and let me find the link from aastra company devs |
| 18:36.09 | [TK]D-Fender | hnWould help to watch the full debug of the call the fails vs one that works... |
| 18:36.45 | willzzz | TK, the problem is that no matter what the phone disconnects after 4 seconds |
| 18:36.55 | willzzz | read the sipfoundry url |
| 18:37.41 | willzzz | i need to set the sip instance id to 0 |
| 18:38.17 | [TK]D-Fender | willzzz, that forum post said ti was fine with Asterisk and back with SipX |
| 18:39.41 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 18:39.51 | willzzz | http://track.sipfoundry.org/browse/XX-8726?page=com.atlassian.streams.streams-jira-plugin%3Aactivity-stream-issue-tab |
| 18:42.31 | [TK]D-Fender | willzzz, please show your full sip config for general and your peer |
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| 18:46.05 | willzzz | http://pastebin.com/6RtRkNXQ |
| 18:48.25 | [TK]D-Fender | willzzz, Move your register's to the end of [general] |
| 18:48.53 | [TK]D-Fender | [701] is a mess |
| 18:49.40 | willzzz | ? |
| 18:52.05 | [TK]D-Fender | willzzz, http://pastebin.com/Y4ARnx8M |
| 18:52.30 | [TK]D-Fender | willzzz, remove all the commented out junk from there as well and move all registeres to the end of [general] |
| 18:53.11 | *** join/#asterisk _Corey_ (~chatzilla@mail1.infradapt.com) |
| 18:53.15 | [TK]D-Fender | apply all changes and retest. on fail show the complete attempt (your first was not) and your new configs |
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| 19:01.38 | willzzz | http://pastebin.com/nLf5SiRp |
| 19:02.34 | willzzz | my phone sends a CSeq: 102 ACK and SIP/2.0 400 Bad REquest |
| 19:03.58 | willzzz | btw the phone supports g711u 8k and 16k as SEPERATE options |
| 19:04.03 | willzzz | what is the default asterisk 8k or 16k? |
| 19:06.05 | Qwell | 8k |
| 19:06.32 | [TK]D-Fender | willAgain you're not showing the complete call |
| 19:07.04 | willzzz | http://pastebin.com/HEh4YY29 |
| 19:07.41 | Qwell | The aastra is saying the ACK is a bad request? |
| 19:08.09 | [TK]D-Fender | Ok, still not a complete call from beginning to end. I'm done with this. |
| 19:08.12 | willzzz | yes |
| 19:08.23 | willzzz | the PHONER DISCONNECTS AFTER 4 SECONDS |
| 19:08.25 | willzzz | AUTOMATICALLY! |
| 19:09.13 | willzzz | it DOESNT GET TO THAT PART |
| 19:09.17 | willzzz | i'm calling the phone vendor |
| 19:09.20 | willzzz | their SIP implementation si shit |
| 19:09.44 | willzzz | and a different phone using the same login works (siemens)_ |
| 19:12.03 | *** join/#asterisk shido6 (~shido6@nat/yahoo/x-gbefczsjxmepeqmt) |
| 19:12.13 | [TK]D-Fender | My 57i CT works just fine... |
| 19:12.23 | [TK]D-Fender | oh well. |
| 19:12.27 | [TK]D-Fender | heads off for a while |
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| 20:01.45 | citywok | how's thursday treating everybody? |
| 20:02.36 | psilikon | great |
| 20:02.55 | psilikon | dtmf in a conference with Sangoma not so much... |
| 20:03.05 | citywok | lol |
| 20:03.25 | citywok | we have problems with our polycom on a pap2 thinking it hears DTMF tones randomly. it's really annoying. |
| 20:04.17 | psilikon | a conference phone? |
| 20:05.44 | citywok | it's a conference system actually |
| 20:05.52 | citywok | and it always annoys meetme |
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| 20:08.19 | psilikon | citywok, Sounds like dtmf talkoff. |
| 20:09.29 | citywok | hmm. i wonder if it is an issue with the pap2, or the polycom itself. |
| 20:10.02 | citywok | we know it gets triggered by the video conference system, and sometimes a cellphone if the volume is cranked up and somebody hits a digit, but otherwise it still happens at random |
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| 20:15.51 | *** join/#asterisk michael-i (~anonymous@204.11.230.58.static.etheric.net) |
| 20:19.58 | michael-i | I'm a little confused with the new CALLERID function in Asterisk 10. Do I SET things with it like: SET(CALLERID(num,+1XXXYYYZZZZ))? I thought it was always CALLERID(num)=+1XXXYYYZZZZ |
| 20:24.12 | *** part/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
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| 20:30.04 | michael-i | Let me put this a different way. Does the new CALLERID() function in 10 behave differently than in previous versions or does it simply offer more fields to set/get? |
| 20:30.34 | jaytee | for a good breakdown in syntax type "core show function CALLERID" at the CLI |
| 20:32.39 | michael-i | I'm reading that already but it shows a comma separated CALLERID(field,value). That doesn't seem to mesh with the old CALLERID(field)=value when SETing. |
| 20:32.51 | pabelanger | It should be: Set(CALLERID(name)=Asterisk PBX) |
| 20:32.53 | michael-i | just confused what changed |
| 20:33.38 | *** join/#asterisk navaismo (~navaismo@fixed-203-96-202.iusacell.net) |
| 20:33.55 | michael-i | k, thanks pabelanger ⦠the wiki just confused me with the seemingly new syntax |
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| 20:34.50 | *** mode/#asterisk [+o mnicholson] by ChanServ |
| 20:34.57 | michael-i | Specifically, what the "optional CID" does. |
| 20:35.17 | pabelanger | Ya, looking into it now |
| 20:35.18 | p3nguin | We just had this discussion the other day. There is absolutely no documentation on that option CID field. |
| 20:35.26 | p3nguin | It's in 1.8, too. |
| 20:35.37 | p3nguin | For now, just disregard the [,CID] part. |
| 20:35.48 | p3nguin | You probably don't need it. |
| 20:36.42 | *** join/#asterisk shido6 (~shido6@nat/yahoo/x-ltgsnwmxornnpwfx) |
| 20:36.54 | michael-i | Probably not. I just read it as "this is where the new value goes for the field specified" |
| 20:36.56 | *** join/#asterisk mpe (~mpe@31.25.23.177) |
| 20:37.06 | p3nguin | michael-i: http://pastebin.com/8r6kiuvW |
| 20:38.04 | michael-i | p3nguin: hehehâ¦ok, yeah, pretty harmless |
| 20:38.27 | p3nguin | I haven't seen any reason to use that optional field, yet. |
| 20:38.49 | pabelanger | michael-i: p3nguin: the optional CID is only used when trying to get / read CALLERID(..). Basically you are telling CALLERID to use the CID field rather then using the value from the channel |
| 20:39.05 | michael-i | So I guess the next question I have is, is +1XXXYYYZZZZ a valid CALLERID(num) value? |
| 20:39.18 | p3nguin | Yes it is. |
| 20:39.20 | michael-i | pabelanger: makes sense, thanks |
| 20:39.49 | p3nguin | It's valid, but don't expect any telco in USA to accept it. |
| 20:40.07 | michael-i | p3nguin: alrightyâ¦time to complain to my provider then for not setting it (yes, I have privileges on that) |
| 20:40.12 | p3nguin | In my experience, valid NANP caller id is 10 digits. NXXNXXXXXX |
| 20:40.46 | p3nguin | If you're calling to a US number, format it accordingly. |
| 20:41.01 | michael-i | p3nguin: ah..that could be it too. I only know that they are strict to the other format on incoming routing |
| 20:41.12 | p3nguin | If you aren't calling in the US, I'm not the right person to talk to abou CID values. |
| 20:41.20 | michael-i | all us only for now |
| 20:41.32 | p3nguin | Then it's 3145551212, not +13145551212. |
| 20:41.59 | michael-i | stupid question then, what's the +1xxxyyyzzzz format called? So I don't keep having to give examples. |
| 20:42.32 | citywok | michael-i: http://en.wikipedia.org/wiki/E.164 |
| 20:43.07 | michael-i | thanks, I know I should know that by now. |
| 20:43.29 | citywok | and now you do! :) |
| 20:43.31 | p3nguin | Maybe global? |
| 20:43.39 | michael-i | :) |
| 20:44.17 | michael-i | Since I am responsible for setting the Caller ID correctly, I'm a bit paranoid before a trial/error session |
| 20:45.14 | p3nguin | You can also check http://en.wikipedia.org/wiki/Telephone_numbering_plan |
| 20:45.51 | *** join/#asterisk _Corey_ (~chatzilla@mail1.infradapt.com) |
| 20:46.06 | p3nguin | When I dial NANP, my callerID number is set to 10 digits. When I dial anything else, I set it to +1 and 10 digits. |
| 20:47.40 | citywok | apparently you have to be careful with callerid name. our T1 provider Qwest rejects any call that has the callerid(name) value set... fail. |
| 20:48.00 | michael-i | I'll put that in for now as a decent start. Since I also control the name field, I'll just stick to anything but Pentagon, Whitehouse, CIA, etc.. |
| 20:48.15 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 20:48.16 | michael-i | I'll probably leave it blank for now. |
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| 20:48.43 | *** mode/#asterisk [+o mnicholson] by ChanServ |
| 20:48.52 | pabelanger | Updated the documentation for CALLERID, hopefully it little more clearer. |
| 20:49.37 | michael-i | pabelanger: thanks, it makes sense after explanation but was confusing |
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| 20:51.20 | michael-i | apropos nothing: was there any further discussion of adding perl style regex support into the dial plan? |
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| 20:51.44 | pabelanger | michael-i: Ya, I didn't know myself. I had to look at the code to figure it out, so I figure I should update the documentation for the next person |
| 20:52.25 | michael-i | pabelanger: never a good thing |
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| 21:21.24 | talntid | wait_for_answer: Unable to write frame |
| 21:21.40 | talntid | I get that message quite a bit... but things seem to run smooth.... ideas? |
| 21:21.51 | [TK]D-Fender | details. |
| 21:22.47 | talntid | Asterisk 1.6.2.5-0ubuntu1 built by buildd @ yellow on a x86_64 running Linux on 2010-04-16 05:34:19 UTC |
| 21:23.04 | talntid | Dials out to another asterisk box (pbxin) that dials out via Sangoma card |
| 21:23.43 | talntid | Any other details that I need to provide, just let me know and I'll provide |
| 21:24.16 | p3nguin | pabelanger: You misspelled parameter. (paramater) |
| 21:25.32 | pabelanger | heh, I noticed it too and though I fixed it. Guess not |
| 21:25.56 | p3nguin | I only looked at 341664. You may have changed it later. |
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| 21:33.38 | michael-i | I've never setup an install like this, so forgive the ignorance. I register with a provider with a single username but have multiple DIDs which are routed to it. Incoming calls are always landing in my dial plan w/the username as extension. |
| 21:33.57 | michael-i | My register string does not have a /username at the endâ¦so I thought it would route to the DIDs specified. |
| 21:33.57 | p3nguin | Fix your register statement. |
| 21:34.15 | p3nguin | register => user:pass@host |
| 21:34.30 | michael-i | that's how it is right now |
| 21:34.38 | p3nguin | hmm |
| 21:34.45 | p3nguin | Are you using trustrpid=yes in the peer? |
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| 21:35.07 | michael-i | no, it's not set |
| 21:35.30 | p3nguin | Try it. |
| 21:35.47 | p3nguin | Who is your ITSP? |
| 21:35.58 | michael-i | I'll set it |
| 21:39.57 | *** join/#asterisk _Corey_ (~chatzilla@mail1.infradapt.com) |
| 21:39.59 | michael-i | no dice |
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| 21:42.18 | *** join/#asterisk pdtpatrick1 (~pdtpdt@12.249.4.226) |
| 21:42.34 | pdtpatrick1 | Question .. what's the best solution/app to use if one wants to record outbound calls? |
| 21:42.47 | talntid | Monitor() |
| 21:43.12 | pdtpatrick1 | Yeah was going to try that but thought there was something else.. okay - sounds good then. Much appreciated |
| 21:50.00 | pdtpatrick1 | I noticed there's also ChanSpy .. which u cannot use along with Monitor() .. after setting this up .. how do you then spy on the call? If anyone has used this before -- thanks in advance |
| 21:50.17 | p3nguin | pdtpatrick1: MixMonitor() |
| 21:50.28 | talntid | beat me to it. |
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| 21:56.59 | citywok | I think i ran in to a bug in asterisk regarding dialing with T1 & setting callerid(name), can somebody take al ook at the response i got from the carrier? it shows the output from me, their response, and their output from their side of the call. |
| 21:57.00 | citywok | http://pastebin.com/UpHLDcXm |
| 21:57.35 | citywok | according to qwest they sent a message saying we didn't like the name field being set but we're oging to make the call anyway, and then astersk replied back hanging up. |
| 21:59.51 | talntid | looks to me like Asterisk is hanging up, at the request of Qwest |
| 22:00.06 | [TK]D-Fender | citywok: Now get your own PRI debug |
| 22:01.14 | citywok | interesting, okay, will do |
| 22:01.46 | *** join/#asterisk _Corey_ (~chatzilla@mail1.infradapt.com) |
| 22:04.36 | WIMPy | citywok: The message seems pretty clear. And you're not the first with that issue. Just do as they said. |
| 22:05.19 | citywok | WIMPy: i did for a quick fix. we typically send calleridname when we can and set it to something other than the phone#, depending which carrier the call ends p going out most will take it |
| 22:05.42 | citywok | obviously it only works when we are sending callerid# as an 800# instead of a DID |
| 22:06.04 | WIMPy | Is it allowed at all in NI-2 or whatever? Not according to Q.931. |
| 22:07.34 | citywok | not sure about if it's allowed. here's the pri debug http://pastebin.com/CCWWQKct |
| 22:07.39 | citywok | if it isn't allowed shouldn't asterisk know to not send it? |
| 22:08.13 | WIMPy | That's why I asked about the protocol. It doesn't for DSS1. |
| 22:08.14 | citywok | the 253# was the callerid display, 8002662278 is 1800comcast |
| 22:08.40 | *** join/#asterisk shido6 (~shido6@nat/yahoo/x-slhdhkhdgkrpkfqy) |
| 22:09.30 | p3nguin | I doubt asterisk will know what your carrier allows. |
| 22:09.32 | citywok | i do see the carrier say call proceeding |
| 22:09.43 | citywok | but then asterisk kills the call |
| 22:09.47 | WIMPy | That's what they wrote. |
| 22:10.20 | citywok | yea, they may have termed it anyway from the protocol error the message before |
| 22:10.24 | WIMPy | But I see Asterisk sending STATUS, that doesn't make sense. |
| 22:10.58 | WIMPy | And then a hangup which we don't see. |
| 22:11.24 | WIMPy | And shouldn't happen, either. |
| 22:11.48 | citywok | yea that was all pridebug gave me for the call |
| 22:13.02 | WIMPy | So it doesn't even terminate the call? |
| 22:13.22 | citywok | no |
| 22:13.28 | citywok | interesting, in the console output: Span 4: Channel 0/23 got hangup, cause 99 |
| 22:13.36 | citywok | cause 99 is what they send to me when they don't like the display digits |
| 22:13.50 | citywok | Ext: 1 Cause: Info. element nonexist or not implemented (99), class = Protocol Error (e.g. unknown message) |
| 22:14.19 | WIMPy | Yes, it is know to treat any cause as an error, even if in a Status message. |
| 22:14.31 | WIMPy | known |
| 22:15.04 | citywok | alrighty then. is that an expected behavior or something that was unintended? |
| 22:15.07 | WIMPy | But I never know if that's a libpri or a chan_dahdi issue. |
| 22:15.26 | WIMPy | I'd certainly call it a bug. |
| 22:15.53 | WIMPy | It thinks the call has gone, but actually it still exists. |
| 22:16.05 | WIMPy | However there the two traces differ. |
| 22:16.34 | citywok | the traces aren't from the exact same call so it's possible |
| 22:16.43 | WIMPy | Your telco says Asterisk terminates the call, but Asterisk didn't say that. It just said it thinks it's terminated. |
| 22:16.55 | WIMPy | From tyour trace I'd expect the call to time out later. |
| 22:17.18 | *** join/#asterisk jstapleton (~jstapleto@c-24-125-190-50.hsd1.va.comcast.net) |
| 22:17.26 | WIMPy | Sure, but there's different behaviour. Or did you upgrade anything in between? |
| 22:18.06 | citywok | no upgrades or anything, although the trace for the call he sent was out of a different dialplan/context but it should be pretty much dientical |
| 22:18.24 | WIMPy | Well, it probably doesn't time out but will get terminated once it's connected. |
| 22:18.40 | citywok | for my test call for debug i made a quick 2 line dialplan to simply set the name and then dial the trunk |
| 22:19.04 | citywok | i'm in production right now so i can't change the normal dialplan to test it |
| 22:19.23 | citywok | the call terminates on my phone right as i hit dial on my phone |
| 22:19.57 | WIMPy | Sure, but the call on the PRI might last longer. |
| 22:20.03 | citywok | gotcha |
| 22:20.14 | WIMPy | And the dialplan probably shouldn't make a difference. |
| 22:20.47 | citywok | yea that's what i'd figure which is why i did it that way |
| 22:20.59 | citywok | so it's odd the traces don't line up |
| 22:21.33 | WIMPy | Yes |
| 22:21.48 | citywok | but i'm not good enough to even notice that, that's all you haha |
| 22:21.49 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-nweleimzpebuxutg) |
| 22:21.58 | WIMPy | But none of them are good. |
| 22:21.58 | citywok | this is the first time i've ever even had to turn pridebug on |
| 22:22.19 | WIMPy | Oh and BTW: Is it NI-2 you're using? |
| 22:22.33 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 22:22.40 | citywok | pretty sure, let me double check though |
| 22:23.00 | WIMPy | I think that was where I saw that before. |
| 22:23.54 | citywok | switchtype = national, which i'm pretty sure is ni2 |
| 22:24.19 | *** join/#asterisk binbash_ (~peter@server.digitog.nl) |
| 22:25.48 | *** join/#asterisk shido6 (~shido6@nat/yahoo/x-bmftniudsdugqccn) |
| 22:26.00 | _Corey_ | I really love AT&T... seriously, who puts an O and a 0 in the same ticket number? |
| 22:28.17 | WIMPy | BTW: On DSS1 I also get lots of Status messages from the telco complainging about what Asterisk sends. But that happens after connect where it doesn't seem to do any harm. |
| 22:28.29 | navaismo | \o/ the set_as_single_video_src in confbridge asterisk10-beta rules!!! |
| 22:29.21 | WIMPy | will try to avoid using dahdi in the future if possible. |
| 22:37.43 | WIMPy | Happy weekend BTW :-) |
| 22:41.41 | *** join/#asterisk pushpop (~pushpop@pool-173-77-211-176.nycmny.fios.verizon.net) |
| 22:46.03 | *** part/#asterisk pietro (~pietro@88-149-226-181.dynamic.ngi.it) |
| 22:49.36 | p3nguin | It seems like the reason many google voice calls are missed by asterisk is related to the sending of DTMF 1. Is there any way to improve the quality of the digit being transmitted so that it is more often accepted by gv? |
| 22:51.15 | *** join/#asterisk tamiel (~tamiel@ip-183.net-81-220-86.toulouse.rev.numericable.fr) |
| 22:59.45 | *** join/#asterisk ruben23 (~APRILJOEY@121.54.58.140) |
| 23:00.02 | ruben23 | hi guys, I got this error msg: [Oct 20 18:58:54] NOTICE[1654]: chan_sip.c:16378 handle_request_register: Registration from '"150"<sip:150@68.68.31.112>' failed for '121.54.58.140' - No matching peer found |
| 23:00.37 | ruben23 | what could be possible solution for this? |
| 23:00.53 | *** join/#asterisk Praise (~Fat@unaffiliated/praise) |
| 23:02.10 | navaismo | if its in your network and you desire the register create the peer, if its an attack use blockhost or fail2ban |
| 23:02.44 | *** join/#asterisk hfb (~hfb@pool-96-247-56-101.lsanca.dsl-w.verizon.net) |
| 23:03.31 | navaismo | ruben23 ^^ |
| 23:04.22 | p3nguin | Do you want 68.68.31.112 to register using a SIP peer name of 150? |
| 23:04.40 | p3nguin | Err, wait... |
| 23:04.55 | p3nguin | Do you want 121.54.58.140 to register using a SIP peer name of 150? |
| 23:05.19 | p3nguin | or did I have that right the first time? |
| 23:06.56 | p3nguin | I think the second question was the right question. |
| 23:07.13 | navaismo | yep |
| 23:08.41 | p3nguin | So now we wait. |
| 23:20.06 | navaismo | and wait... |
| 23:25.30 | *** join/#asterisk Syrex (~syrex@dsl-146-17-16.telkomadsl.co.za) |
| 23:36.28 | ruben23 | I you want 68.68.31.112 to register using a SIP peer name of 150 |
| 23:37.55 | p3nguin | Isn't that Asterisk's IP address? |
| 23:38.20 | *** join/#asterisk nighty- (~nighty@69-165-220-105.dsl.teksavvy.com) |
| 23:43.13 | p3nguin | And wait... |
| 23:52.54 | SeRi | p3nguin, you in? |
| 23:55.27 | michael-i | It seems the Asterisk build system has gotten much smarter since my last go-round (and more complex!). I'm adding my own module to the apps directory but it needs to link in some extra libs when building the .so. Where can I define my special case for that to happen? |
| 23:55.43 | michael-i | (sorryâ¦Asterisk 10) |
| 23:57.05 | SeRi | p3nguin, when you are avail hit me up. I have everything packed and ready to go. |