| 00:00.10 | ChannelZ | It will probably reject the call for no auth though |
| 00:00.36 | spck | ya getting SIP/2.0 401 Unauthorized |
| 00:00.36 | spck | on the production server now |
| 00:01.01 | ChannelZ | Make a peer on your test box called [production] or something.. |
| 00:01.45 | ChannelZ | configure it similarly to the [asterisk-test] one you made on the production box but set the host=sip.unioncab.com |
| 00:02.09 | ChannelZ | then Dial(SIP/production/810) instead |
| 00:02.52 | dijib | p3nguin, howcome this is dialing to '1' and not SIP/100 |
| 00:02.53 | dijib | ? |
| 00:02.54 | dijib | http://pastebin.com/JNzp7DAi |
| 00:03.29 | ChannelZ | there's no way it can be *dlaing* 1 |
| 00:03.41 | ChannelZ | Your Dial statement is hard-coded to SIP/100 |
| 00:04.03 | spck | what is the value of ${OUTNUM}? |
| 00:05.03 | spck | ok i give up, this isn't that important |
| 00:05.10 | ChannelZ | This is another annoyance of mine, using 'extension' numbers as SIP device names |
| 00:06.25 | leifmadsen | +1 |
| 00:06.26 | dijib | yikes |
| 00:06.33 | leifmadsen | devices, extensions, and people should be entirely abstracted |
| 00:07.02 | leifmadsen | extension numbers are applied to people, and people are applied to a device |
| 00:07.25 | leifmadsen | (which means you should name your devices something unique to the device, such as an ID tag, or a MAC address) |
| 00:08.45 | ChannelZ | It's just a laziness to be able to do things like Dial(${EXTEN}) |
| 00:09.23 | ChannelZ | but causes more trouble than it's worth for beginners who mix up an extension with a device because they called them both "100" |
| 00:09.28 | leifmadsen | aye |
| 00:09.53 | leifmadsen | it's hard enough to understand the logic in your head about an extension number and a device, than to call them the same thing |
| 00:10.02 | ChannelZ | Yeah. |
| 00:15.31 | p3nguin | Most people still can't grasp that extensions are not phones. |
| 00:15.59 | ChannelZ | Bound by the chains of analog PBXes |
| 00:17.00 | p3nguin | dijib: http://pastebin.com/JNzp7DAi says: if OUTNUM is 1, Dial SIP/100, else GOTO extension ${OUTNUM} priority 1, in the current context. |
| 00:17.33 | p3nguin | If OUTNUM has no value, Goto() will choke and die. |
| 00:18.09 | p3nguin | If it contains a value, there had better be an extension of the value in that context. |
| 00:19.10 | ChannelZ | isn't it actually saying if "OUTNUM" is 1 ? |
| 00:21.16 | spck | no real need for the GotoIf if you are doing it that way |
| 00:21.16 | p3nguin | It's a string comparison, so shouldn't it be fine? |
| 00:21.29 | spck | just set it up with Goto and have multiple labels |
| 00:21.31 | p3nguin | How else would you make it conditional? |
| 00:21.40 | p3nguin | You wouldn't. |
| 00:21.49 | p3nguin | So you use the GotoIf(). |
| 00:22.57 | p3nguin | I haven't seen an actual failure yet, so I don't know that there is actually any problem to fix. |
| 00:23.29 | spck | like: http://pastebin.com/28u10xYY |
| 00:23.55 | spck | then you don't have to chain a bunch of if's together |
| 00:24.11 | spck | more like a switch statement then |
| 00:24.14 | ChannelZ | (Yes it's a string comparison, but is 1 is an int not a string) |
| 00:25.21 | ChannelZ | s/but is 1/but 1 is/ |
| 00:25.27 | ChannelZ | barf |
| 00:25.29 | ChannelZ | anyway |
| 00:26.11 | spck | heh, i don't think voip-info gets updated because registering isn't working |
| 00:26.17 | ChannelZ | $["${OUTNUM}" =1] should never test true |
| 00:26.56 | ChannelZ | spck: eh? |
| 00:27.30 | spck | i just tried registering on voip-info.com, doesn't look like it's sending out the emails |
| 00:27.41 | spck | makes it kind of hard to edit the wiki |
| 00:27.47 | ChannelZ | oh. You mean registering on the website |
| 00:27.50 | leifmadsen | spck: the double quote are literally checked |
| 00:27.58 | p3nguin | spck: Using Goto() like that sure would work, but how is it any different from using a GotoIf() the way I did it? |
| 00:28.06 | leifmadsen | so if ${OUTNUM} returned 1, then you're doing a comparison of "1" = 1 |
| 00:28.12 | leifmadsen | which will always return false |
| 00:28.13 | p3nguin | uh oh |
| 00:28.16 | spck | p3nguin: it's scalable |
| 00:28.17 | p3nguin | That was my mistake. |
| 00:28.36 | p3nguin | I didn't quote the 1. |
| 00:28.37 | spck | originally he was asking for an if/else if solution |
| 00:28.43 | dijib | i think im going to have to fix those numbered contexts, and 100,200 is a sandbox. |
| 00:28.56 | p3nguin | I provided the solution, but I didn't quote the 1 in the comparison. |
| 00:29.04 | p3nguin | And since I never saw any failure to debug, I had nothing to fix. |
| 00:29.42 | dijib | contoso |
| 00:29.47 | spck | it's like a switch statement instead of a hard coded if tree |
| 00:32.22 | p3nguin | I don't see any advantage. |
| 00:32.29 | p3nguin | nor disadvantage. |
| 00:33.02 | spck | you can keep adding num-XXXX lines instead of having to write multiple if statements |
| 00:35.25 | spck | switch vs if: http://pastebin.com/gc8BvcPx |
| 00:36.07 | spck | that's just psuedo javascript, but in an asterisk dialplan using GotoIf would be a lot harder to add stuff |
| 00:37.15 | spck | since there is no else if in normal dialplan |
| 00:37.20 | spck | no ael on the other hand... |
| 00:37.26 | spck | s/no/now |
| 00:37.29 | p3nguin | The request was; if OUTNUM = 1, do something, else do something different. I don't see what your problem is with GotoIf(). |
| 00:38.09 | spck | <dijib> ok, have an gotoif or if-else or if-then-else |
| 00:38.09 | spck | <PROTECTED> |
| 00:38.31 | spck | i guess i read that as else if |
| 00:39.12 | spck | sounded like he wanted to chain multiple if's together |
| 00:39.16 | *** join/#asterisk nix8n82-phone (~AndChat@59.sub-174-253-181.myvzw.com) |
| 00:39.20 | p3nguin | If I didn't think GotoIf was capable of doing it, I wouldn't have tried to use it. |
| 00:39.48 | p3nguin | You could even use more than one GotoIf if necessary. |
| 00:39.57 | usrbinfoobar | hmm wierd intel page seems messed up |
| 00:40.07 | usrbinfoobar | keeps redirecting me in circles when trying to download the IPP |
| 00:40.13 | spck | ya but then the context gets all messy with labels |
| 00:40.36 | p3nguin | Surprisingly, that's what labels are for. |
| 00:40.36 | spck | a lot easier and clearer just using the Goto with named extensions |
| 00:41.09 | spck | but why write 3 lines when one will suffice? |
| 00:41.40 | p3nguin | I do it both ways, and I don't see one as being better or worse than the other. |
| 00:41.59 | spck | the way i decide is if i'm going to have to add something to it in the future |
| 00:44.24 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 00:46.10 | spck | anyone got any idea why my phones don't ring when dialing out to another line sporadically? |
| 00:47.47 | dijib | cpu load? |
| 00:47.48 | ChannelZ | squirrels |
| 00:48.11 | spck | started happening after upgrading to 1.8 from 1.6 |
| 00:48.50 | *** join/#asterisk mistergibson (~mistergib@71-36-122-200.ptld.qwest.net) |
| 00:50.13 | mistergibson | shameless plea for newb help : I'm hoping to setup a simple H.323 asterisk server for voip on Ubuntu/Debian. Anyone have a favorite how-to url they want to toss at the Newb? Thanks in advance. :) |
| 00:51.02 | spck | you might try one of the turn key projects like freepbx or pbx in a flash |
| 00:51.19 | mistergibson | spck: thanks, sniffing ... |
| 00:53.55 | *** join/#asterisk Defraz (~Defraz@184-155-137-27.cpe.cableone.net) |
| 00:54.57 | spck | otherwise there's always apt-get install asterisk |
| 00:55.03 | mistergibson | spck: hrm ... not looking to do a custom distro, but rather cobble a config procedure for my existing platform |
| 00:55.17 | mistergibson | spck: yeah, but it is *after* that step is where it gets into the weeds a bit |
| 00:55.59 | mistergibson | spck: Nerd Vittles looks promising |
| 00:56.12 | spck | try "the asterisk book" |
| 00:56.19 | spck | i don't know how to make the bot link it |
| 00:59.05 | dijib | hey p3nguin should i make the switch to iax2? |
| 00:59.28 | dijib | and codec,s what should i use? ulaw good enough? |
| 01:00.37 | mistergibson | spck: this looks good if others come asking : http://www.freepbx.org/support/documentation/administration-guide |
| 01:01.21 | ChannelZ | dijib: ulaw is as good as it gets (*) if you've got the bandwidth |
| 01:01.34 | ChannelZ | * this is partly a lie |
| 01:01.55 | dijib | (*) in asterisk world? |
| 01:02.14 | ChannelZ | * as in disclaimer |
| 01:02.15 | dijib | whats better on bandwidth? |
| 01:02.19 | dijib | ahh. |
| 01:02.33 | dijib | 729 was it? |
| 01:02.46 | ChannelZ | Yeah if you and your ITSP support it |
| 01:03.31 | ChannelZ | The public telephone system at large is 8-bit 8kHz which is ulaw/alaw so sound quality wise everything else is of little use. |
| 01:03.55 | dijib | ahh now i see it. |
| 01:04.00 | catphish_ | will asterisk attempt to negotiate the same codec on both sides of a bridged call where possible? |
| 01:04.13 | dijib | sounds goood at 41000hz |
| 01:04.20 | ChannelZ | Calling SIP-to-SIP there are much higher quality wideband codecs you can use, thus the disclaimer to my lying statement that ulaw was as good as it gets |
| 01:04.20 | dijib | i noticed |
| 01:04.46 | catphish_ | ie if my local phones have g711 disabled, will asterisk try to use a lower bandwidth codec right through to my upstream provider? |
| 01:04.47 | ChannelZ | catphish_: yes |
| 01:04.49 | dijib | wider than ulaw? |
| 01:05.00 | ChannelZ | catphish_: both sides tell the other what codecs they can support |
| 01:05.04 | catphish_ | i know |
| 01:05.12 | catphish_ | what i mean is if the call has 2 legs |
| 01:05.19 | *** join/#asterisk chuckf (~chuckf@ubuntu/member/chuckf) |
| 01:05.21 | catphish_ | will asterisk try to negotiate the same for both legs |
| 01:05.24 | ChannelZ | It only matters what happens between you and them |
| 01:05.38 | catphish_ | to avoid transcoding |
| 01:05.39 | ChannelZ | You don't know/have control over what happens after that. |
| 01:05.50 | ChannelZ | Ideally it would try to use the same one yes. |
| 01:05.58 | catphish_ | well asterisk has control over both legs of the call |
| 01:06.10 | ChannelZ | Oh with that path yes |
| 01:06.13 | catphish_ | so i was hoping it would negotiate them together rather than independently |
| 01:06.17 | catphish_ | thats good |
| 01:06.35 | catphish_ | if course i guess it's best to be explicit with what you want to use |
| 01:06.43 | catphish_ | and know what's supported |
| 01:09.13 | ChannelZ | Well either both sides support the same codec or they don't, so it kind of becomes an "it is what it is" sort of thing |
| 01:10.34 | ChannelZ | You can put the codecs in order of preference and Asterisk will try to pick the best it can |
| 01:10.41 | *** join/#asterisk johnnyasterisk (~johnnyast@89.18.71.45) |
| 01:10.55 | catphish_ | well i was just looking to avoid the situation where a local phone only supports a low bandwidth codes, but my upstream provider supports everything, in that case i wouldn't want to make asterisk tanscode, rather make my provider do it by negotiating the same low bandwidth codes on both sides of the bridged call |
| 01:11.13 | catphish_ | codes negotiation is simple when it's only one leg |
| 01:11.31 | catphish_ | but when it comes to bridging i want to prevent avoidable transcoding |
| 01:11.52 | *** join/#asterisk johnnyasterisk (~johnnyast@89.18.71.45) |
| 01:12.14 | ChannelZ | Depends if * is in the actual media stream or not |
| 01:12.22 | catphish_ | it is |
| 01:13.58 | ChannelZ | AFAIK it will not go out of its way to transcode anything unless forced to by configured codec restrictions |
| 01:14.12 | catphish_ | that's good |
| 01:14.15 | catphish_ | easily tested anyway |
| 01:16.10 | usrbinfoobar | anyone have any idea about g729b (no vad) support? |
| 01:16.58 | *** join/#asterisk james_zhu (~Administr@183.16.212.123) |
| 01:18.06 | *** join/#asterisk [netman] (netman@138.236.76.188.dynamic.jazztel.es) |
| 01:19.29 | *** join/#asterisk [netman] (netman@138.236.76.188.dynamic.jazztel.es) |
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| 01:21.49 | dijib | hey p3nguin it just dawned on me an action i took in the voip.ms console yesterday, regarding the 's' extension. I changed from softphone -> asterisk last night, thats why it wasnt routing. |
| 01:22.05 | dijib | and subsequitly switched back to my number |
| 01:26.23 | *** part/#asterisk russellb (~russellb@asterisk/contributor-and-cool-guy/russellb) |
| 01:38.02 | *** join/#asterisk adolfomaltez (~taro@190.62.200.105) |
| 02:08.22 | *** join/#asterisk sunfone (~jeff@c-67-184-36-158.hsd1.il.comcast.net) |
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| 02:33.42 | *** join/#asterisk dlynes (~dlynes@bas6-hamilton14-1176003212.dsl.bell.ca) |
| 02:34.13 | dlynes | Just curious if anyone knows why the numbering change from asterisk 2.0 to 10.0? |
| 02:34.30 | dlynes | There's no mention of it in the changelog |
| 02:35.08 | robl^ | dlynes: marketing thinks it "sounds better" ;-) |
| 02:35.13 | dlynes | robl^, ah |
| 02:35.37 | dlynes | robl^, Do you happen to know if 10.0 is iax2-compatible with the 1.6.2 series? |
| 02:35.47 | robl^ | no technical reason⦠just a "name/number" change.. |
| 02:35.54 | dlynes | robl^, or sip-compatible for that matter? |
| 02:36.30 | dlynes | robl^, Just asking, because certain 2.4 versions weren't iax2 compatible with the 1.6.1/1.6.2 series |
| 02:36.35 | *** join/#asterisk godmachine-x6 (~godmachin@unaffiliated/godmachine-x6) |
| 02:36.55 | dlynes | robl^, and I was wanting to test it on my home machine |
| 02:37.05 | robl^ | dlynes: they are implementing a protocol.. should be compatible.. if not, I'd assume a bug.. just 10 should have more features/tweaks |
| 02:38.06 | robl^ | keep in mind, 10.0 is a beta. there is likely bugs don't use for production |
| 02:40.37 | dlynes | robl^, yeah...wasn't planning on using it in any production systems |
| 02:40.49 | dlynes | robl^, unless you count my home machine as being production :) |
| 02:41.21 | dlynes | robl^, only person I'm likely to piss off with my home system is my wife, but she's used to it :0 |
| 02:42.09 | dlynes | robl^, I've been running 1.8 on it for a while, and I consider that to be beta level as well |
| 02:43.18 | robl^ | well, 1.8 is at least LTS, so it will be around for a long while. 10 is kinda a playground of testing large architectural changes. it looks promising. ;-) |
| 02:43.51 | dlynes | robl^, so there's significant changes then, I'm guessing? |
| 02:44.06 | dlynes | robl^, like maybe the completion of the sip v3 stack? |
| 02:45.23 | robl^ | dlynes: honestly, I'm not sure what all has been changed, but skimming through some blog posts and change logs looks like its substantial. I haven't tried using it yet |
| 02:46.59 | dlynes | robl^, ah...yeah...I've tried 1.8 on a few machines, but had to roll them back to 1.6.2 series, all except for my home machine...they all needed pbx features, which had issues on 1.8 (transfers) |
| 02:47.34 | dlynes | robl^, I'm guessing the problems have been fixed by now, but my boss on those machines has been once bitten, twice shy :o |
| 02:53.32 | p3nguin | dijib: So putting the setting to softphone sends to s, while setting to IP PBX sends to the phone number? |
| 02:53.55 | p3nguin | ~asterisk10 |
| 02:53.55 | infobot | Asterisk 10 -- http://blogs.digium.com/2011/07/21/the-evolution-of-asterisk-or-how-we-arrived-at-asterisk-10/ |
| 02:54.01 | p3nguin | dlynes: ^^ |
| 02:54.40 | dijib | p3nguin, YES |
| 02:55.04 | dijib | ive almost perfected my current dialplan |
| 02:55.11 | dijib | of the day |
| 02:55.30 | p3nguin | karen_m: See this? |
| 02:55.41 | p3nguin | The problem is solved! |
| 02:55.52 | p3nguin | It's user error, and this time it isn't mine! |
| 02:56.25 | dijib | yeah path to switch the toggle is Account Manager --> Inbound Settings --> Device Settings |
| 02:56.33 | p3nguin | I'm familiar with the option. |
| 02:56.38 | dijib | in Voip.ms customer ports |
| 02:56.47 | p3nguin | I've just never changed it off IP PBX because I use an IP PBX. |
| 02:56.56 | dijib | no guff eh |
| 02:56.59 | p3nguin | I didn't know what it changed. |
| 02:57.06 | p3nguin | Never had a reason to check it. |
| 02:57.17 | dijib | after i do a few touchups here your going to have to see the whole dialplan |
| 02:57.48 | p3nguin | I hope you'll forgive me if I laugh. |
| 02:58.12 | p3nguin | I won't really laugh -- I'm just being mean. |
| 02:58.25 | p3nguin | But you can be sure I'll point out things I would change. |
| 02:59.37 | *** join/#asterisk nix8n82-phone (~AndChat@154.sub-174-253-178.myvzw.com) |
| 02:59.59 | p3nguin | like numbered priorities, for example. |
| 03:04.10 | WIMPy | Oh, wow. Looks like a busy night. |
| 03:19.25 | *** join/#asterisk godmachine-x6 (~godmachin@unaffiliated/godmachine-x6) |
| 03:25.22 | p3nguin | dijib: If you need the potential bandwidth savings of trunking, or if you need to get through bitchy firewalls, IAX2 certainly is an option. I use it for the trunking, but I don't know how much bandwidth I'm really saving with my low call volume. |
| 03:30.42 | ChannelZ | Yay, my first haxx0r! |
| 03:30.59 | ChannelZ | Call from '' (69.20.65.81:5060) to extension '011441214001365' rejected because extension not found |
| 03:33.18 | p3nguin | At least they aren't doing it on mine this time. |
| 03:33.22 | WIMPy | Oh, wow. A sensible prefix. |
| 03:33.46 | p3nguin | I started accepting all those calls and play Ringing() endlessly. |
| 03:34.25 | WIMPy | I've also seen quite some attempts to numbers containing 441214... The really interesting thing is to watch what they try in front of the 44. |
| 03:34.54 | WIMPy | Did it work? |
| 03:34.58 | *** join/#asterisk brdude (~brdude@c-24-5-194-184.hsd1.ca.comcast.net) |
| 03:35.19 | WIMPy | They never waited for a reply here. Those calls all ended with retransmission timeouts. |
| 03:35.21 | p3nguin | It works. I've even listened to some of them. They place the call, it rings, they wait for a bit, then hangup. |
| 03:35.53 | p3nguin | I think I need to redirect them somewhere else. |
| 03:36.38 | WIMPy | Dial the police and use option A to ecplain the reason for the call *ggg* |
| 03:36.53 | WIMPy | s/ecp/exp/ |
| 03:41.32 | dijib | so i need a third option on my Gotoif. :/ press 1, 2, or anything else |
| 03:41.41 | dijib | not possible is it? |
| 03:43.04 | dijib | without using that switch statement |
| 03:43.11 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 03:45.41 | p3nguin | GotoIf() only has two conditions: true and false. |
| 03:45.48 | dijib | i need 3 |
| 03:45.51 | dijib | or more |
| 03:46.01 | dijib | switch statement? |
| 03:46.22 | dijib | or if? |
| 03:46.24 | dijib | if() |
| 03:49.22 | WIMPy | switch is something completely different. |
| 03:49.29 | dijib | nevermind i think i know how to fix it. |
| 03:49.33 | *** join/#asterisk godmachine-x6 (~godmachin@unaffiliated/godmachine-x6) |
| 03:49.52 | WIMPy | You might be able to Goto a variable. |
| 03:50.25 | WIMPy | Goto(option-${digit},1) or the like. |
| 03:51.30 | *** join/#asterisk hesco_home (~hesco@c-76-109-144-184.hsd1.fl.comcast.net) |
| 03:52.21 | hesco_home | I just installed my first ever 1.8 asterisk. How would I do what in 1.6 was accomplished with sip show registry? |
| 03:52.35 | p3nguin | sip show registry |
| 03:53.09 | hesco_home | if sip is not available to me at the CLI> prompt, what is it I might be missing in my conifguration? |
| 03:53.17 | p3nguin | chan_sip |
| 03:53.26 | hesco_home | thanks, looking for that now |
| 03:53.35 | p3nguin | busted or missing sip.conf |
| 03:53.38 | p3nguin | is the cause |
| 03:55.39 | hesco_home | my /etc/asterisk/sip.conf looks fine. |
| 03:56.08 | WIMPy | 'module load chan_sip' |
| 03:56.34 | *** join/#asterisk tzafrir (~tzafrir@212.179.75.202) |
| 03:57.17 | p3nguin | And then figure out why it wasn't auto-loaded. |
| 03:57.45 | WIMPy | because of the errors that are displayed then. |
| 03:57.52 | dijib | nevermind, i fixed it with an include, in my inbound context |
| 03:57.55 | WIMPy | Or missing modules.conf. |
| 03:58.29 | hesco_home | thanks, the module load worked. I checked make menuselect again and chan_sip was enabled. |
| 03:59.03 | WIMPy | Then take a look at your modules.conf. |
| 03:59.28 | p3nguin | autoload should probably be on, and there should not be a noload for chan_sip.so. |
| 04:05.11 | dijib | p3nguin, if you care to have a look. heres my current config. im sure theres a mistake or three |
| 04:05.12 | dijib | http://pastebin.com/FJYDj9pt |
| 04:09.07 | *** join/#asterisk wonderworld (~ww@port-92-201-85-101.dynamic.qsc.de) |
| 04:11.15 | *** join/#asterisk seraphie (~erin@75.76.38.159) |
| 04:14.16 | *** join/#asterisk gladier (~gladier@pedobear.molested.me) |
| 04:14.21 | p3nguin | I dislike that you've used numbers like that for the names of the devices. |
| 04:15.44 | p3nguin | line 122 has an extraneous space. |
| 04:17.56 | p3nguin | so does 133, 139, 145, 151, 157, 163, 203, 209. and lines 185, 187, 189, 191 have several extraneous spaces. |
| 04:19.44 | p3nguin | dial1 isn't required in line 202. |
| 04:20.02 | p3nguin | dial1 label therefore isn't required in line 203. |
| 04:21.11 | p3nguin | And I still don't see how line 205 has a destination. What would be a possible value that the variable could contain if not 1? |
| 04:21.32 | p3nguin | Fix up those things, and I'd put it into production. |
| 04:21.41 | p3nguin | 20 minutes worth of work. |
| 04:21.54 | WIMPy | I like optimisation. Is the an automated version of yourself available? :-) |
| 04:22.15 | p3nguin | I wish. |
| 04:25.10 | gladier | hey guys - has anyone had a case where the default music on hold plays too fast (in ulaw format) - i haven't tried any other formats just yet |
| 04:25.56 | WIMPy | gladier: Yes. But that was many years ago and I don't remember anything about it. |
| 04:25.57 | p3nguin | What do you mean by default? Something included? |
| 04:26.15 | p3nguin | Or do you mean music you put into the default moh class? |
| 04:26.29 | gladier | default as in from asterisk. ie macroform-the_simplicity.ulaw |
| 04:26.48 | p3nguin | Could that be some sort of a timing issue? |
| 04:26.55 | gladier | yea trying to figure out from where |
| 04:27.12 | WIMPy | Try 'timing test'. |
| 04:27.15 | p3nguin | If it plays too fast, I'd think whatever provides your timing could be set too high. |
| 04:27.32 | gladier | It has been 1000 milliseconds, and we got 50 timer ticks |
| 04:27.34 | gladier | so right on the dot |
| 04:28.12 | p3nguin | This might be a USA daytime question. |
| 04:28.42 | WIMPy | Is that a virtual machine? |
| 04:28.42 | p3nguin | Not too much activity this time of night. |
| 04:28.47 | p3nguin | o.O |
| 04:28.54 | p3nguin | Didn't even think about that. |
| 04:28.55 | gladier | nope :) i've learnt the hard way about asterisk in VMs |
| 04:29.19 | gladier | figured it out ... had slin set in the musiconhold.conf and ulaw in the directory |
| 04:29.56 | ChannelZ | oops |
| 04:30.14 | gladier | no clue how that would even play since they're completely different formats |
| 04:30.45 | gladier | goes hunting to find some music on hold that the customer wont complain about |
| 04:36.13 | *** join/#asterisk X-Rob (~Rob@eth2083.qld.adsl.internode.on.net) |
| 04:39.26 | *** join/#asterisk irroot (~irroot@197.174.69.7) |
| 04:40.47 | dijib | 20 minutes of work.. hah that took me all day, with your handheld help #* |
| 04:41.41 | *** join/#asterisk paulc (~paulc@unaffiliated/paulc) |
| 04:42.50 | hesco_home | OK, I managed to get my desk phone to register to the new server, but an outgoing call gets an immediate fast busy. |
| 04:42.50 | hesco_home | sip show peers says: |
| 04:42.50 | hesco_home | 21/21 76.109.144.184 D N A 5060 UNREACHABLE |
| 04:42.50 | hesco_home | while on the legacy (working) server it shows, instead, this: |
| 04:42.51 | hesco_home | 21/21 76.109.144.184 D N A 5060 OK (86 ms) |
| 04:42.51 | hesco_home | any guidance on what might be missing here? |
| 04:43.03 | *** join/#asterisk Bryanstein (~Bryanstei@shellium/admin/bryanstein) |
| 04:43.05 | gladier | behind nat? |
| 04:43.34 | hesco_home | I'd imagine so. There is a router in the other room. |
| 04:43.53 | hesco_home | I got this dhcp lease from somewhere |
| 04:44.39 | dijib | hesco_home, lets see your extensions.conf |
| 04:45.02 | hesco_home | any preferred paste bin you like to use? |
| 04:45.05 | dijib | and sip.conf dont forget to hide your passwords |
| 04:45.12 | dijib | pastebin.com |
| 04:45.26 | dijib | who is your ITSP? |
| 04:52.04 | hesco_home | I think this one goes through Voicepulse. |
| 04:52.13 | hesco_home | Try this to start with: http://pastebin.com/uuzX2wjG |
| 04:52.30 | hesco_home | its a freepbx set up, ergo all the includes in the sip.conf. |
| 04:56.37 | p3nguin | We can't really support any FreePBX configurations at all here. |
| 04:57.11 | p3nguin | Usually someone will help in the more appropriate #freepbx channel, though. |
| 04:58.15 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 04:58.52 | *** join/#asterisk felimwhiteley (~quassel@46.7.101.58) |
| 04:58.53 | hesco_home | understand about the division of labor, but given the symptoms I report, what would you be looking for in a vanilla install that is not provided by my pastebin? |
| 05:03.34 | p3nguin | I haven't seen a debug yet. |
| 05:04.02 | p3nguin | If you're looking for nothing more than a guess, you probably don't have an appropriate extension for what you're calling. |
| 05:34.34 | *** join/#asterisk irroot (~irroot@pbx.distrotech.co.za) |
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| 05:39.10 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
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| 18:08.42 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 18:08.42 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 10.0.0-beta1 (2011/07/22), 1.8.5.0 (2011/07/11), dahdi-linux 2.4.1.2 (2011/04/11), dahdi-tools 2.4.1 (2011/03/03), libpri 1.4.12 (2011/07/06) -=- Visit the new official Asterisk wiki: wiki.asterisk.org -=- http://xkcd.com/806/ |
| 18:14.45 | tonsofpcs | azv4: do you not own the equipment? |
| 18:15.05 | azv4 | tonsofpcs: yes I own the equiptment |
| 18:15.17 | azv4 | to panasonic that is irrelavant |
| 18:15.26 | tonsofpcs | what was the model #? |
| 18:15.55 | azv4 | randy123 |
| 18:16.04 | azv4 | opps lol |
| 18:16.13 | azv4 | KX-TD500 |
| 18:16.31 | azv4 | good luck if you think google will come up with software heh |
| 18:16.53 | azv4 | Panasonic has done a great job forcing me to pay $75 an hour to have some moron make minor adjustments on our system |
| 18:17.01 | tonsofpcs | erm.... that's just the cabinet model # |
| 18:17.13 | azv4 | that is the phone system's model number |
| 18:21.24 | p3nguin | The cabinet is the system, and then you add in cards as needed. |
| 18:21.45 | tonsofpcs | I keep coming up with some third-party "programmator" software |
| 18:22.51 | tonsofpcs | found it. |
| 18:22.52 | tonsofpcs | http://web.archive.org/web/20060219071015/http://download.panasonic.co.uk/bts/Old_Products/Td/TD_Programming.htm |
| 18:23.04 | MrTelephone | Can you run asterisk for a facility based CLEC? |
| 18:23.09 | tonsofpcs | http://web.archive.org/web/20060208021232/http://download.panasonic.co.uk/bts/Old_Products/Td/TD.htm actually has a bunch more |
| 18:23.14 | *** join/#asterisk beek (~klinebl@pdpc/supporter/bronze/beek) |
| 18:23.18 | tonsofpcs | azv4: ^ |
| 18:23.39 | tonsofpcs | MrTelephone: don't you have service level requirements as a CLEC? |
| 18:23.47 | azv4 | tonsofpcs, those links are dead |
| 18:24.04 | MrTelephone | I'm pretty sure I hit 5 nines on my asterisk system |
| 18:24.06 | tonsofpcs | azv4: they work perfectly fine. |
| 18:24.06 | MrTelephone | haha |
| 18:24.17 | azv4 | tonsofpcs, I tried version 1, it was dead, trying version 6 now |
| 18:24.29 | tonsofpcs | 1) "five nines" is nothing. 2) there's a lot more requirements than "uptime" |
| 18:24.30 | SuperNull | Most clecs stick to soft switches .. we as a cable company have reviewed some .. cheapest in the $200-$300k range |
| 18:24.32 | azv4 | this isnt eht software |
| 18:24.43 | azv4 | tonsofpcs, this isnt even the programming software |
| 18:24.54 | tonsofpcs | http://web.archive.org/web/20060208024404/http://download.panasonic.co.uk/bts/Old_Products/Td/TD_Software.htm ? |
| 18:25.09 | tonsofpcs | click the second link I pasted, it links to other things. |
| 18:25.15 | azv4 | I have no idea what you have linked, but it isnt the software I need |
| 18:25.24 | azv4 | it is some 3rd party software |
| 18:25.33 | p3nguin | FreeSWITCH will cost you much less than $200,000. |
| 18:25.35 | *** join/#asterisk file (~file@asterisk/developer-and-muffin-lover/file) |
| 18:25.35 | *** mode/#asterisk [+o file] by ChanServ |
| 18:25.40 | tonsofpcs | azv4: those links are FROM PANASONIC. |
| 18:25.55 | azv4 | the software you linked is not the Panasonic software |
| 18:25.57 | azv4 | it is 3rd party |
| 18:26.02 | Kobaz | anyone want an 8 port and 4 port grandstream fxo |
| 18:26.21 | tonsofpcs | I don't know how much clearer I can be: That was on the Panasonic website until they took it down. |
| 18:26.40 | p3nguin | I'll give you the address for donation if you want to get rid of them. |
| 18:26.59 | azv4 | I found a couple softwares from TD series, but not 500 |
| 18:27.20 | tonsofpcs | azv4: how many lines/interfaces is your system populated with? |
| 18:27.44 | azv4 | I think it has something like 8 cards with 8 lines each |
| 18:27.57 | azv4 | 23 of which are using for ATA the rest for extensions |
| 18:28.29 | WIMPy | What does ATA mean in that case? |
| 18:28.40 | azv4 | incoming lines from PRI |
| 18:28.50 | azv4 | I might be using bad terminology |
| 18:29.38 | WIMPy | Oh, you convert a pri to 23 analog and connect that to the PBX? |
| 18:29.56 | azv4 | yes unfortunately |
| 18:30.17 | WIMPy | Interesting setup |
| 18:30.17 | azv4 | I didn't do it heh, was like that when I got here, I guess they didn't want to buy the PRI card for the PBX |
| 18:30.25 | tonsofpcs | WIMPy: that's how the PBX here is set up.... nitsuko |
| 18:30.27 | azv4 | would be nice if we had it so we could have caller ID and whatnot... |
| 18:30.55 | p3nguin | 23 individual ATAs to turn PRI into 23 2P2Cs? |
| 18:31.01 | azv4 | part of the reason why I want the programming software is so I can buy the card and set it up myself |
| 18:31.31 | WIMPy | p3nguin: They would need to be connected to something else again. |
| 18:31.50 | p3nguin | Why? |
| 18:31.51 | azv4 | the other part is because I am buying a voip gateway and I want to install it myself and not pay some idiot $75 an hour |
| 18:32.07 | WIMPy | 1st you need something that connects to the pri. |
| 18:32.29 | WIMPy | azv4: Replace the whole thing? |
| 18:32.31 | tonsofpcs | azv4: pm coming your way |
| 18:32.32 | azv4 | I think we have an Adtran device, I might be spelling it wrong, I cant remember |
| 18:32.58 | p3nguin | So the PRI can go to the Adtran, and that can turn it into SIP. |
| 18:33.03 | azv4 | WiMPy, I would love to, but to goto an IP system would require a complete rewire of 3 facilities |
| 18:33.23 | p3nguin | Then you have 23 ATAs connected between the Adtran and your PBX? |
| 18:33.34 | WIMPy | azv4: Then keep the phones. |
| 18:33.57 | WIMPy | That sounds even more interesting than I thought at first. |
| 18:34.12 | WIMPy | was thinking about some channelbank type thing. |
| 18:34.12 | azv4 | define ATA please |
| 18:34.22 | leifmadsen | ~ata |
| 18:34.22 | infobot | i heard ata is Analogue Terminal Adapter which provides an FXS and/or FXO and ethernet, see http://www.voip-info.org/wiki/view/ATA |
| 18:36.03 | azv4 | so I misused ATA |
| 18:36.07 | p3nguin | I'm just trying to get a mental image of the setup. |
| 18:36.13 | azv4 | we have no ATA yet |
| 18:36.28 | *** join/#asterisk hobodave (~hobodave@pdpc/supporter/professional/hobodave) |
| 18:36.31 | p3nguin | So the Adtran connects right to the PBX with some type of cabling? |
| 18:36.34 | azv4 | we have PRI into ADTRAN box, adtran is cross connected to PBX |
| 18:36.44 | azv4 | with 23 seperate pairs |
| 18:37.01 | *** join/#asterisk gnowen (~OwenMeany@adsl-74-242-247-147.rmo.bellsouth.net) |
| 18:37.03 | azv4 | I assumed the adtran box converted to analog |
| 18:37.07 | p3nguin | That PBX would accept 23 phone cables? |
| 18:37.17 | p3nguin | s/would/does/ |
| 18:37.28 | azv4 | it accepts many more |
| 18:37.43 | azv4 | I believe the cards support 8 each, with 8 cards |
| 18:37.46 | p3nguin | Okay, I can understand the physical configuration now. |
| 18:37.53 | azv4 | but each port can be incoming line or extension |
| 18:38.01 | azv4 | I THINK |
| 18:38.07 | WIMPy | Interesting again. |
| 18:38.09 | p3nguin | I wasn't trying to drag you off topic, just wanted to picture it. |
| 18:38.12 | azv4 | I am not sure yet, I would love to get the software to understand for sure |
| 18:38.17 | WIMPy | Sounds rather unlikely. |
| 18:38.25 | azv4 | but I can not see any cards that appear different than one another |
| 18:38.29 | azv4 | they all look the same |
| 18:39.02 | WIMPy | For analog, direction usually does matter. |
| 18:39.12 | WIMPy | For digital, usually not. |
| 18:39.16 | tonsofpcs | p3nguin: my PBX accepts what appears to be 8x 66 blocks... (RJ21x) |
| 18:39.50 | beek | Sounds like the Adtran is a channel bank to me. |
| 18:39.59 | p3nguin | tonsofpcs: Which PBX are you running? |
| 18:40.22 | tonsofpcs | from what I could tell by glancing at the TD### manuals, the extensions can support analog or digital or digital with an analog behind it! (4-wire system maybe?) |
| 18:40.41 | p3nguin | I personally like to build IP PBXs and skip all the phone wiring stuff. Where there's Ethernet, there's phone service. |
| 18:40.45 | azv4 | tonsofpcs, that is correct, each phone has an analog port that can be used at the same time |
| 18:40.52 | tonsofpcs | p3nguin: Nitsuko Onyx |
| 18:41.12 | azv4 | p3nguin, I would have to put a hub at every desk to go IP, |
| 18:41.16 | SuperNull | azv4 what series adtran ? i know some Adtran TA-900E series okay |
| 18:41.24 | p3nguin | azv4: Why? |
| 18:41.37 | azv4 | I would have to take a 20 minute walk to get the Adtran model |
| 18:41.51 | azv4 | p3nguin, becasue each desk in this company has only 1 ethernet |
| 18:41.58 | *** join/#asterisk drift- (ae3001a4@gateway/web/freenode/ip.174.48.1.164) |
| 18:42.03 | SuperNull | azv no telnet/ssh access? |
| 18:42.03 | Nugget | telnet is eeeeeeevil! |
| 18:42.15 | azv4 | SuperNull, that is correct |
| 18:42.19 | p3nguin | azv4: I use Cisco phones, and I only need one wall jack per office and no hub/switch to feed a phone and PC. The phones have a switch in them for that very reason. |
| 18:42.46 | Qwell | yeah nearly all phones have a switch port in them |
| 18:42.49 | azv4 | p3nguin, thank you for that information! I still doubt I can convince my CFO to upgrade phone systems at this time |
| 18:42.52 | SuperNull | is it a 1u with 1 centronix, 1 Ethernet and 1 T1 ? |
| 18:43.10 | p3nguin | If you have the right switches, you can run two VLANs on each port and divide voice data based on MAC address. |
| 18:43.19 | tonsofpcs | p3nguin: do the cisco phones have both PoE and local PSU support or do you have to buy a different model for different powering or are they all PoE? |
| 18:43.36 | azv4 | I would need a PoE switch also I guess... |
| 18:43.41 | WIMPy | azv4: I HAVEN'T SEEN IP PHONES WITHOUT A BUILT-IN 2-PORT SWITCH. |
| 18:43.48 | WIMPy | oops |
| 18:43.57 | p3nguin | tonsofpcs: They have power jacks and also work with PoE switches or injectors. |
| 18:44.37 | p3nguin | for example, I have PoE on mine, but the next one over has its own power cube. |
| 18:44.40 | tonsofpcs | p3nguin: nice. that's one thing that's bothered me about some of the phones we've been pricing. One or the other. Want to have both? best keep spares of both (and at that point, I'd buy all PoE and just buy a handful of local injectors) |
| 18:44.41 | WIMPy | Which seems to be true for at least most phones, as well. |
| 18:45.40 | azv4 | PoE switch isnt that expensive |
| 18:45.41 | p3nguin | If my PoE goes out, I can go grab a cube and plug it in and be back in service within minutes. |
| 18:45.54 | p3nguin | No, they aren't really that expensive if you need them. |
| 18:45.58 | azv4 | is PoE not reliable? |
| 18:46.08 | tonsofpcs | if your PoE goes out, doesn't your switch go out too? |
| 18:46.18 | p3nguin | It usually is reliable, but some things do go bad. |
| 18:46.35 | p3nguin | I'm using an injector rather than a PoE switch. |
| 18:47.19 | *** join/#asterisk Azrael808 (~peter@cpc17-walt12-2-0-cust657.13-2.cable.virginmedia.com) |
| 18:47.31 | tonsofpcs | I've also been pricing Cisco PoE switches, trying to figure out the difference between a few different models. SGE2010-[48 port gigabit poe] v 2960S-[whatever numbers mean 48 gigabit ports and PoE] |
| 18:47.52 | p3nguin | So technically there is a power cube for my phone, but it happens to be on the other end of the ehternet cable rather than on my desk attached to the phone. |
| 18:48.24 | WIMPy | p3nguin: So you get additional cable heating for the winter? |
| 18:48.35 | azv4 | I managed a PoE switch with a wireless network, and it was troublesome |
| 18:48.42 | *** join/#asterisk godmachine-x6 (~godmachin@unaffiliated/godmachine-x6) |
| 18:48.44 | p3nguin | Heh, yeah, don't want those cables freezing up. |
| 18:48.54 | azv4 | I never could figure out the exact point of failure, but I never ruled out PoE failure |
| 18:49.12 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 18:49.19 | p3nguin | PoE access points? |
| 18:49.30 | azv4 | it was 6 access points, Bountiful hardware (TRASH), and the access points were powered over ethernet |
| 18:49.41 | p3nguin | That's not uncommon. |
| 18:50.10 | azv4 | it was a headache, it was in a shipping warehouse, with 3 shifts, and if wireless went down, the whole warehouse stopped functioning |
| 18:50.27 | azv4 | so at 3am I am remoting in to reset switch, accesspoints, and so on |
| 18:50.37 | p3nguin | I think most admins prefer that so they don't have to get an electrician to put a power receptacle at each location they intend to deploy an AP. |
| 18:51.37 | azv4 | WIMPy, I believe he was referring to freezing up as in stop functioning, not literally freezing from tempature |
| 18:51.59 | p3nguin | I was talking about freezing as in ice. |
| 18:52.28 | p3nguin | He said the PoE heated the wiring, so I said it would keep them from freezing. |
| 18:53.24 | p3nguin | In the winter, we get water pipes freezing, so I'm familiar with applying heat to the conduits. |
| 18:57.30 | WIMPy | Yes, you don't want your data to freeze in the cable. |
| 18:57.36 | chuckf | The first job a buddy of mine got for wiring ethernet back in the late 90's was in an old house converted for a business. The first cold day the network went down. Turns out he ran all the cables under/behind a radiator and they melted into a big mess. |
| 18:57.50 | p3nguin | oh no! |
| 18:58.20 | chuckf | yeah, it was a good lauch |
| 18:58.27 | chuckf | s/lauch/laugh |
| 18:58.53 | p3nguin | mmm, lunch. |
| 18:59.53 | WIMPy | Melting data is definitely worse. |
| 19:00.24 | p3nguin | I can call a plumber if mine gets stuck in the tubes, but when it's melted, that's the end. |
| 19:00.47 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 19:09.04 | p3nguin | Just so you know, the setvar=_myVar=value thing works fine in the conf file. |
| 19:09.21 | p3nguin | At least so far that I have been testing it, it does. |
| 19:10.05 | WIMPy | Good to know. I might want that soon. |
| 19:11.01 | p3nguin | I kept losting a variable as channels were created, and I couldn't figure out what the heck was going on. |
| 19:11.26 | p3nguin | s/losting/losing/ |
| 19:11.41 | p3nguin | Easy fix. |
| 19:12.16 | *** part/#asterisk root52 (~root52@ip70-191-116-76.cl.ri.cox.net) |
| 19:16.03 | tzanger | hm, are there standard sip bruteforcing tools around? |
| 19:16.14 | tzanger | I have a client who wants me to prove that fail2ban works |
| 19:16.17 | WIMPy | Check your logs :-) |
| 19:16.21 | Qwell | tzanger: use sipp |
| 19:17.46 | WIMPy | Doesn't look like a technikal issue, but rather the definition of "works". |
| 19:17.48 | tzanger | Qwell: danke |
| 19:20.03 | *** join/#asterisk gogasca (~Adium@nat/cisco/x-nnlcovgqxsyrzyzb) |
| 19:20.57 | gogasca | hi ppl, upgrading from asterisk 1.8.2.2(freepbx 2.8.1) to 1.8.5.0, after upgrade is complete cannot place internal calls, nor internal to external callsâ¦but endpoints and trunk do register |
| 19:21.17 | gogasca | i just do in install folder of 1.8.5.0 .configure, make, make install and reboot |
| 19:21.29 | WIMPy | ~freepbx |
| 19:21.29 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there, or fpbx |
| 19:22.41 | gogasca | okâ¦im not looking for freepbx support nowâ¦some commands are missing after upgrade: |
| 19:22.43 | gogasca | example |
| 19:22.46 | gogasca | core show version |
| 19:23.34 | WIMPy | As far as I know it's impossible for 'core' commands to be missing. That's the idea of the core commands. |
| 19:23.53 | gogasca | maradona*CLI> core show version |
| 19:23.54 | gogasca | No such command 'core show version' (type 'core show help core' for other possible commands) |
| 19:24.03 | Qwell | What does core show help core show? |
| 19:24.08 | Qwell | giggles quietly to himself |
| 19:24.11 | gogasca | [root@maradona ~]# asterisk -V |
| 19:24.12 | gogasca | Asterisk 1.8.5.0 |
| 19:24.16 | Qwell | That was a serious question though. |
| 19:24.52 | gogasca | unicast |
| 19:25.08 | Qwell | Does it stop after 'f'? |
| 19:25.26 | gogasca | ahh looks like irc throttle the output |
| 19:25.31 | WIMPy | o.O |
| 19:25.44 | leifmadsen | well you shouldn't be pasting more than 2-3 lines anyways |
| 19:25.46 | leifmadsen | ~pb |
| 19:25.46 | 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. |
| 19:26.23 | gogasca | ok |
| 19:26.30 | gogasca | point i sthat core show version is missing |
| 19:26.37 | gogasca | just wanted to check if im running right procedure |
| 19:26.43 | gogasca | i stop my asterisk servic ein 1.8.2.2 |
| 19:26.49 | gogasca | then go to my 1.8.5.0 folder |
| 19:26.52 | gogasca | ./configure |
| 19:26.54 | gogasca | make |
| 19:26.56 | gogasca | make install |
| 19:26.57 | gogasca | reboot |
| 19:27.10 | WIMPy | Did you install it to the same place as it was before? |
| 19:27.16 | gogasca | yep |
| 19:27.27 | leifmadsen | ./configure && make && rm -f /usr/lib/asterisk/modules/* && make install && /etc/init.d/asterisk restart |
| 19:27.43 | leifmadsen | that's how I do it |
| 19:27.47 | gogasca | ok |
| 19:27.52 | gogasca | in fact cat /etc/asterisk/version |
| 19:27.52 | gogasca | Asterisk 1.8.2-rc1 |
| 19:27.56 | gogasca | shows old version |
| 19:28.07 | leifmadsen | well that file is certainly not installed by asterisk |
| 19:28.15 | leifmadsen | that's a freepbx thing I suspect |
| 19:28.17 | WIMPy | That file is not generated by Asterisk. |
| 19:28.21 | leifmadsen | ^^ :) |
| 19:28.27 | gogasca | yep |
| 19:28.30 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 19:28.31 | gogasca | in the full log |
| 19:29.04 | Qwell | you broke something, and Asterisk isn't completely starting. |
| 19:29.26 | gogasca | yep |
| 19:29.37 | gogasca | I see this in full log |
| 19:29.38 | gogasca | [2011-08-15 12:28:59] WARNING[3638] pbx.c: No application 'Macro' for extension (ext-local, 101, 1) |
| 19:29.39 | gogasca | [2011-08-15 12:28:59] DEBUG[3638] pbx.c: Spawn extension (ext-local,101,1) exited non-zero on 'SIP/105-00000000' |
| 19:29.47 | gogasca | im calling from 105 to 101 |
| 19:30.22 | gogasca | will be talking to freepbx ppl |
| 19:30.22 | WIMPy | Are you 100% sure, you didn;t end up with more than one Asterisk installation? Like one and a holf or something? |
| 19:30.42 | gogasca | this is what i basically did |
| 19:30.47 | gogasca | installed 1.8.2.2 |
| 19:31.01 | gogasca | i dowloaded the tar file un tar in /usr/local/src |
| 19:31.06 | gogasca | run ./configrue make and make install |
| 19:31.10 | gogasca | all good for 3 months |
| 19:31.27 | gogasca | i had some hold/resume issues and want to be in the latest 1.8.X train |
| 19:31.38 | gogasca | so i downloaded the tar file of 1.8.5 and un tar |
| 19:31.46 | gogasca | under /usr/local/src have 2 folder |
| 19:31.49 | gogasca | 1.8.2.2 and 1.8.5 |
| 19:31.54 | gogasca | then stop asterisk |
| 19:32.05 | gogasca | run ./conigure, make and make install from my 1.8.5 |
| 19:32.08 | gogasca | and reboot |
| 19:32.14 | gogasca | really straight forward |
| 19:33.48 | gogasca | will be upgrading to freepbx 2.9 |
| 19:33.55 | gogasca | im running http://www.freepbx.org/ 2.8.1 |
| 19:34.00 | gogasca | will let u know |
| 19:34.03 | gogasca | thanks guys |
| 19:34.12 | jaytee | good luck! |
| 19:35.32 | [sr] | anyone using video phones? |
| 19:36.47 | gogasca | i do |
| 19:36.56 | gogasca | i have tandberg e20s |
| 19:37.02 | [sr] | which codec do you use? |
| 19:37.13 | gogasca | H.264 |
| 19:37.29 | gogasca | butâ¦there is an issue in pre asterisk 10 |
| 19:37.38 | gogasca | asterisk dont support fmtp in SDP |
| 19:37.46 | gogasca | hence the quality negotiated in basic CIF |
| 19:37.49 | gogasca | not HD |
| 19:38.30 | [sr] | np |
| 19:38.34 | [sr] | i'm just testing for now |
| 19:39.03 | gogasca | yeah u should go to asterisk 10 |
| 19:39.08 | gogasca | for better experience |
| 19:44.22 | *** join/#asterisk oej (~olle@195.41.130.3) |
| 19:44.52 | *** join/#asterisk war9407 (war@c-71-62-61-74.hsd1.va.comcast.net) |
| 19:46.39 | *** join/#asterisk captiancrash (~jonmoo@64-233-236-58.static.evv.wideopenwest.com) |
| 19:46.42 | *** join/#asterisk Quintana (~sylvain@aghnar.doowan.net) |
| 19:55.24 | *** part/#asterisk Poincare (~jefffnode@2001:470:cb24:4::2) |
| 19:55.57 | gogasca | hi guys |
| 19:56.02 | gogasca | just installed freepbx 2.9.1 |
| 19:56.06 | gogasca | and reinstall 1.8.5 |
| 19:56.09 | gogasca | using ./configure && make && rm -f /usr/lib/asterisk/modules/* && make install && /etc/init.d/asterisk restart |
| 19:56.12 | gogasca | and working now |
| 19:56.13 | gogasca | :) |
| 19:56.15 | gogasca | arigato |
| 20:01.19 | *** join/#asterisk catphish_ (~charlie@2001:9d8:2005:2::3) |
| 20:06.40 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 20:06.41 | leifmadsen | heh, left behind and incompatible modules is almost always the reason for failed upgrade (in my experience) |
| 20:08.00 | WIMPy | Yes. It might be a good idea to clean the modules directory on install. |
| 20:08.03 | pdtpatrick | Question .. im trying to load a module but keep seeing undefined symbol: ast_speech_write |
| 20:08.27 | WIMPy | Either just warn about ones that won;t be replaced or maybe have an extra 3rd party directory. |
| 20:09.47 | catphish_ | the install script warns you anyway |
| 20:10.34 | WIMPy | Yes, but why always do it by hand? The chances that things go wrong if you don't are high. |
| 20:19.10 | *** join/#asterisk tully` (~tully@66.76.60.154) |
| 20:21.45 | tully` | Is there any way to kill an agi script from asterisk without using signals? |
| 20:22.41 | tully` | because in all of my testing the dialplan will hang on an agi script and refust to run the hangup section of the dialplan |
| 20:22.47 | tully` | refuse* |
| 20:23.31 | *** join/#asterisk chasing`Sol (~cS@smtp.master-zone.net) |
| 20:28.11 | leifmadsen | WIMPy: well that already happens (the warning) |
| 20:28.46 | WIMPy | Yes, but you always have to do it manually. |
| 20:29.00 | leifmadsen | the reason to do it by hand is that if you just implictly wipe out the directory, there is a not-zero chance of removing a third party module or precompiled module (like codec_g729) |
| 20:29.18 | leifmadsen | adding "rm -f /usr/lib/asterisk/modules/*" to my list of ocmmands to run is not difficult |
| 20:29.31 | WIMPy | And the chance that that will keep Asterisk from working. |
| 20:29.34 | leifmadsen | and adding it to "make install" has a high chance of hurting rather than helping |
| 20:30.31 | WIMPy | I'd rather have a missing module than an Asterisk installation that doesn't work. |
| 20:31.18 | leifmadsen | I'd rather people learn how to help themselves |
| 20:31.42 | leifmadsen | plus you can't know where the actual modules live anyways |
| 20:31.51 | leifmadsen | it's possible someone previously installed asterisk into a different location |
| 20:31.58 | leifmadsen | only the administrator can know where the modules actually live |
| 20:32.12 | WIMPy | Yes, that can be a lot of fun. |
| 20:32.20 | leifmadsen | I doubt you'll ever get the developers to permit such a change to the Makefile |
| 20:33.46 | WIMPy | I fuess it would be best if asterisk reliably refused to load modules that weren;t made for the current version. |
| 20:33.48 | WIMPy | guess |
| 20:33.58 | *** join/#asterisk bpgoldsb (~bpgoldsb@dominii-2-pt.tunnel.tserv4.nyc4.ipv6.he.net) |
| 20:34.47 | bpgoldsb | Does anyone know if there's publicly available testing numbers that I can use that will 1) Do nothing 2) Echo back what I say? I'm trying to troubleshoot some long distance related Echo |
| 20:35.04 | *** join/#asterisk navaismo (~navaismo@fixed-203-100-27.iusacell.net) |
| 20:35.17 | WIMPy | You want to troubleshoot echo with echo? |
| 20:35.32 | navaismo | hello |
| 20:35.59 | bpgoldsb | Possibly. I guess MoH or general noise will work too. |
| 20:36.47 | catphish_ | i'd provide a public echo server, but i don't have anything production secured in place right now |
| 20:37.34 | bpgoldsb | Or, option #2, does anyone know of a place that will give me a trial account for free? |
| 20:37.36 | navaismo | I have a problem with long distance calls using asterisk 1.8.5 the call is dropped after 5 seconds |
| 20:37.47 | bpgoldsb | I can host my own asterisk instance in a vm to trunk that number to |
| 20:38.02 | navaismo | and the cli show me "Re-invite to non-existing call leg on other UA" |
| 20:38.21 | catphish_ | there are voip providers you can sign up to without depositing anything i think |
| 20:38.44 | *** join/#asterisk oej (~olle@195.41.130.3) |
| 20:38.54 | catphish_ | i think voiptalk in the uk provide free numbers like echo, time |
| 20:39.02 | catphish_ | not sure though |
| 20:39.14 | *** join/#asterisk gogasca (~Adium@nat/cisco/x-ndryvpkctmiokqvu) |
| 20:42.07 | navaismo | any tips? |
| 20:44.31 | catphish_ | reinvite is nothing more than an evil curse sent to upset people with NAT :) |
| 20:44.43 | catphish_ | but unfortunately i don't know anything useful about it |
| 20:44.49 | *** part/#asterisk captiancrash (~jonmoo@64-233-236-58.static.evv.wideopenwest.com) |
| 20:48.14 | [sr] | going to sleep |
| 20:48.17 | [sr] | bye |
| 20:48.24 | catphish_ | bye |
| 20:48.40 | navaismo | its a weird problem because with local numbers we dont have problems |
| 20:54.52 | *** join/#asterisk nix8n82-phone (~AndChat@24.143.28.16) |
| 20:55.07 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 21:00.44 | *** join/#asterisk TimeRider (steve@188-220-33-123.zone11.bethere.co.uk) |
| 21:01.10 | gogasca | sorry naviamso can u repaste ur question just joined |
| 21:03.17 | *** part/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
| 21:04.34 | *** join/#asterisk nix8n82-phone (~AndChat@192.sub-174-253-160.myvzw.com) |
| 21:05.14 | navaismo | <PROTECTED> |
| 21:05.23 | navaismo | <PROTECTED> |
| 21:05.59 | gogasca | u have sip trunk to pstn or analog cards? |
| 21:06.47 | navaismo | sip trunk |
| 21:07.31 | gogasca | at some point call connects ? |
| 21:07.36 | gogasca | or just rings for 5 secs ? |
| 21:07.41 | navaismo | this is only for long distnace local and cellphones work great |
| 21:08.04 | navaismo | yes, ring then answer hear voice about 5 secs and dropped |
| 21:09.33 | *** join/#asterisk rneese (~rneese@cpe-72-184-189-80.tampabay.res.rr.com) |
| 21:09.50 | gogasca | I c, do u have any nat in the middle, i had something similar with hold/resume |
| 21:09.53 | rneese | what lang is the digium asterisk gui 2.0 written in |
| 21:10.11 | gogasca | the internal IP was announced in reinvite |
| 21:10.15 | gogasca | and remote end disconnected the call |
| 21:10.27 | gogasca | only during initial setup ip was rewritten properly |
| 21:11.04 | gogasca | I had javascript:void(null) configured |
| 21:11.13 | gogasca | under sip_general_custom.conf |
| 21:11.21 | navaismo | Yes i have a Nat but in the sip debug the Contact IP is written with my public IP |
| 21:11.38 | gogasca | what about the SDP info ? |
| 21:12.28 | gogasca | for the working call what is the difference? if u have both sip debugs i can take a look |
| 21:13.08 | navaismo | this is the hangupcause in the debug '-Asterisk-HangupCause: No user responding' |
| 21:13.22 | navaismo | i only have the long distance |
| 21:13.26 | *** part/#asterisk rneese (~rneese@cpe-72-184-189-80.tampabay.res.rr.com) |
| 21:14.58 | WIMPy | You get 'no user responding' after it is already ringing? Or is that self generated fake ringing? |
| 21:15.34 | WIMPy | Oh, even after connect. That doesn't make sense. |
| 21:16.56 | navaismo | after connect |
| 21:17.20 | navaismo | i can hear at least 3 or 5 seconds of remote voice then the call is dropped |
| 21:22.31 | *** join/#asterisk mozart_ar (~mozart_ar@host54.190-230-144.telecom.net.ar) |
| 21:23.47 | navaismo | here is the log http://pastebin.com/rvLfuWKK |
| 21:25.29 | *** join/#asterisk BenC[UK] (~bcummins@cpc2-lock3-2-0-cust77.6-1.cable.virginmedia.com) |
| 21:26.25 | mozart_ar | Hello, is there a way to restart the busy mode ? |
| 21:26.55 | mozart_ar | my extension is rejecting calls with the busy message |
| 21:27.35 | mozart_ar | and I have hang down my extension, is not busy |
| 21:27.50 | mozart_ar | some clue to try ? |
| 21:28.30 | mozart_ar | I have try to search in CLI , but I dont sure wich key I should to see |
| 21:30.11 | *** join/#asterisk Hanumaan (~Hanumaan@dslb-088-066-149-183.pools.arcor-ip.net) |
| 21:33.51 | BenC[UK] | mozart_ar: sip show channels - you should be able to see any calls |
| 21:34.33 | mozart_ar | let me see |
| 21:35.01 | mozart_ar | 0 active SIP dialogs |
| 21:35.55 | *** join/#asterisk penguin (GreenWolf@cpe-74-77-221-5.buffalo.res.rr.com) |
| 21:36.25 | BenC[UK] | channel request hangup all |
| 21:36.47 | BenC[UK] | are you sure its busy though, not an error.. |
| 21:36.52 | BenC[UK] | if you check the cli |
| 21:36.54 | BenC[UK] | and make a call |
| 21:36.58 | *** join/#asterisk Azrael808 (~peter@cpc17-walt12-2-0-cust657.13-2.cable.virginmedia.com) |
| 21:36.59 | BenC[UK] | you should see any errors |
| 21:37.48 | penguin | does anyone know how how to set a time for the sip registry |
| 21:38.14 | penguin | so its not registering with my provider every 5 seconds? |
| 21:38.39 | penguin | <PROTECTED> |
| 21:38.42 | mozart_ar | BenC[UK] |
| 21:38.58 | *** join/#asterisk nix8n82-phone (~AndChat@24.143.28.16) |
| 21:39.05 | mozart_ar | I got this in a verbose CLI: "Got SIP response 486 "Busy Here" back from 10.10.0.6" |
| 21:39.26 | *** join/#asterisk ClintGoudie-Nice (~clint@smtp.callware.com) |
| 21:39.27 | mozart_ar | "10.10.0.6" is IP from is registered my extension |
| 21:39.44 | ClintGoudie-Nice | heya all, is there any way to disable SDP for a given sip trunk? |
| 21:40.00 | BenC[UK] | mozart_ar: I am not sure then :( |
| 21:40.04 | BenC[UK] | what SIP device is it? |
| 21:40.17 | ClintGoudie-Nice | or even all sip? this is just a jimmy rig to duplicate an issue we're having with a switch |
| 21:40.40 | mozart_ar | thanks anyway BenC[UK] |
| 21:42.10 | BenC[UK] | penguin: registertimeout=X |
| 21:43.36 | BenC[UK] | penguin: goes in global section |
| 21:45.46 | *** join/#asterisk mygfiscontradict (~mygfiscon@pool-173-68-96-49.nycmny.fios.verizon.net) |
| 21:46.00 | mygfiscontradict | i'm running ubuntu server |
| 21:46.11 | mygfiscontradict | and asterisk installed on it |
| 21:46.27 | mygfiscontradict | but each time i try to make a call, it ends |
| 21:49.12 | penguin | BenC[UK]: thanks so much |
| 21:49.32 | BenC[UK] | mygfiscontradict: check the asterisk logs/cli |
| 21:49.57 | mygfiscontradict | looking, can i show you my log? |
| 21:50.29 | mygfiscontradict | BenC[UK], can i show you my logs* |
| 21:52.45 | mygfiscontradict | if anyone else is interested |
| 21:52.45 | *** join/#asterisk Alric (~alric@64.6.54.218) |
| 21:52.46 | mygfiscontradict | http://pastebin.com/U3KdeFPv |
| 21:52.46 | ChannelZ | Asterisk ends, or the call ends? |
| 21:53.09 | mygfiscontradict | from what the logs say ChannelZ |
| 21:53.18 | mygfiscontradict | chan_sip.c: Registration from '"tinu" <sip:tinu@192.168.1.16>' failed for '192.168.1.6:52688' - No matching peer found |
| 21:53.19 | ChannelZ | Did you configure this at ALL? |
| 21:53.51 | mygfiscontradict | i did so |
| 21:54.02 | mygfiscontradict | i followed these steps |
| 21:54.02 | mygfiscontradict | http://letitknow.wordpress.com/2011/05/05/how-to-install-asterisk-1-8-on-ubuntu-server-11-04/ |
| 21:54.43 | mygfiscontradict | am I missing anything? |
| 21:55.31 | Alric | Regarding SIP messaging, does anyone know where Asterisk gets the extension to call from? It seems to be using the request line instead of the To header, but that seems odd to me. |
| 21:56.28 | ChannelZ | mygfiscontradict: yeah a lot it seems like |
| 21:56.43 | mygfiscontradict | mind pointing me in the right direction? |
| 21:56.58 | ChannelZ | mygfiscontradict: You're trying to register as "tinu" but the instructions sure don't have a peer named that, and apparently you don't either |
| 21:57.54 | mygfiscontradict | so let me understand this |
| 21:58.00 | p3nguin | Please do. |
| 21:58.16 | ChannelZ | Also they only setup 2 test extensions, 1001 and 1002 so the attempts to dial real numbers I see won't work even once you fix your SIP peers |
| 21:58.47 | mygfiscontradict | i see my mistake |
| 21:58.55 | mygfiscontradict | i changed the 1001 to tinu |
| 21:58.58 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-hqoccdooyhrtolof) |
| 21:59.02 | ChannelZ | mygfiscontradict: in sip.conf they setup 2 peers called [1001] and [1002] (which is a "mistake" in and of its self but we'll ignore that for now) |
| 21:59.45 | mygfiscontradict | ok |
| 22:00.56 | mygfiscontradict | go on |
| 22:01.05 | ChannelZ | I'm done |
| 22:01.21 | mygfiscontradict | ChannelZ: |
| 22:01.22 | mygfiscontradict | http://pastebin.com/pZHiXGNP |
| 22:01.25 | mygfiscontradict | better? |
| 22:02.04 | ChannelZ | Almost. Now you have them as 101 and 102, but the rest of the example you (presumably) used from the website is going to try and dial 1001 and 1002 |
| 22:02.29 | ChannelZ | (see the Dial statements in extensions.conf) |
| 22:02.33 | mygfiscontradict | i'll change that too |
| 22:03.21 | mygfiscontradict | btw, thank you |
| 22:03.26 | mygfiscontradict | big time, this is saving my job! |
| 22:03.35 | mygfiscontradict | fixed :) |
| 22:03.38 | mygfiscontradict | lets try it now |
| 22:04.39 | *** join/#asterisk grantm (~grant@68.142.138.4) |
| 22:13.03 | mygfiscontradict | ChannelZ: why is call heading to 192.168.1.6? |
| 22:13.12 | mygfiscontradict | when my server's ip is 192.168.1.16? |
| 22:15.35 | mygfiscontradict | or anyone else please |
| 22:15.57 | BenC[UK] | 1.6 is the sip phone I guess? |
| 22:16.08 | BenC[UK] | the one thats calling |
| 22:19.48 | ChannelZ | How do you think the call is "heading to" 192.168.1.16? |
| 22:21.24 | *** join/#asterisk tzafrir (~tzafrir@bzq-218-155-145.cablep.bezeqint.net) |
| 22:31.47 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 22:38.08 | *** join/#asterisk Brixius (~Brixius@PDN-VBA.OnvoyInc.fw.onvoy.net) |
| 22:47.41 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 22:50.34 | mygfiscontradict | BenC[UK]: good point |
| 22:50.36 | mygfiscontradict | thanks ChannelZ |
| 22:52.18 | *** join/#asterisk corretico (~luis@201.201.44.82) |
| 22:54.05 | *** join/#asterisk navaismo (~navaismo@fixed-203-100-27.iusacell.net) |
| 22:56.03 | *** join/#asterisk Suikwan (~administr@76.76.200.29) |
| 22:56.51 | Suikwan | Can anyone tell me if a timing device is required when using SIP trunks? |
| 22:57.31 | *** join/#asterisk caveat- (~false@newshell1.bshellz.net) |
| 23:00.50 | ChannelZ | not generally no |
| 23:01.29 | Suikwan | I thought so, but had my doubts...only when using E1, T1, etc, right? |
| 23:01.48 | WIMPy | Device as in hardware: no. Device as in external software: In theory no, because of some bug, maybe yes. |
| 23:02.22 | WIMPy | Digital interfaces will provide timing. |
| 23:03.13 | Suikwan | thanks ChannelZ, WIMPy |
| 23:11.18 | *** join/#asterisk TenJack (~chatzilla@174-24-187-154.tukw.qwest.net) |
| 23:11.42 | *** part/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 23:12.29 | TenJack | Hey, Im trying to setup Adhearsion in production and I need to open port 5038. I added the line "-A INPUT -p tcp --dport 5038 -j ACCEPT" to my iptables.up.rules file, but this does not seem to do the trick. |
| 23:12.53 | TenJack | anyone know what rule will open this port up for adhearsion to connect to asterisk? |
| 23:13.17 | *** join/#asterisk SantiamTech (alex@173-164-96-37-Oregon.hfc.comcastbusiness.net) |
| 23:13.32 | mygfiscontradict | can i use google voice + AsteriskNOW together? |
| 23:13.37 | WIMPy | You need to put the rule at the right position. |
| 23:13.50 | mygfiscontradict | would it work as good as 1.8? |
| 23:14.17 | mygfiscontradict | BenC[UK]: plus, do i need a sip service to use google voice? |
| 23:14.31 | TenJack | WIMPy: it's after other rules just like it, like: "-A INPUT -p tcp --dport 80 -j ACCEPT" |
| 23:15.00 | TenJack | but when i do "netstat -atn | grep 5038" nothing happens |
| 23:15.08 | ChannelZ | Google Voice is its own thing so you don't need an ITSP if that's what you mean |
| 23:15.11 | WIMPy | If you have some reject or drop rule before, that won;t do anything. |
| 23:15.17 | mygfiscontradict | ChannelZ: thank you |
| 23:15.31 | ChannelZ | TenJack: Did you reload your firewall? |
| 23:15.42 | mygfiscontradict | ChannelZ: do you know if google voice is dependable for long term uses? |
| 23:15.56 | mygfiscontradict | 2 years* |
| 23:16.13 | ChannelZ | no idea |
| 23:16.21 | mygfiscontradict | if not, which sip provider would you recommend for the us |
| 23:16.34 | ChannelZ | I'm using it more as an amusement (for Google Talk, not Voice specifically) |
| 23:16.46 | TenJack | ChannelZ: I believe so, i did "/sbin/iptables -F" then "/sbin/iptables-restore < /etc/iptables.up.rules" |
| 23:17.03 | ChannelZ | Google Voice is a little bit of a PITA because of how it works, you have to program Asterisk to answer the call and send DTMF to accept the call... |
| 23:17.15 | mygfiscontradict | i see |
| 23:17.26 | mygfiscontradict | mhmmm, i think its best then to go with sip provider |
| 23:17.34 | ChannelZ | ~itsplist-us |
| 23:17.34 | 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 |
| 23:17.41 | mygfiscontradict | thanks infobot |
| 23:18.25 | TenJack | ChannelZ: is that the right way to write the rule? |
| 23:18.27 | p3nguin | tenjack: Allowing connection through the firewall does not make the service listen. netstat shows port status. |
| 23:18.35 | ChannelZ | Also Google is still fiddling with the protocol it seems (Talk behaves differently than Voice) so if it was your primary means of communication it's entirely possible it might get broken at some point for days/weeks until fixed on the Asterisk side |
| 23:19.14 | TenJack | p3nguin: what do you mean by service exactly? |
| 23:19.22 | p3nguin | tenjack: a daemon |
| 23:19.26 | ChannelZ | TenJack: I dunno sorry, I don't use the scripts you use. In general the syntax is right AFAIK |
| 23:19.27 | p3nguin | tenjack: something listening |
| 23:19.52 | TenJack | p3nguin: like Adhearsion? |
| 23:20.01 | ChannelZ | Are you sure it's the firewall and not Manager not being configured on Asterisk? |
| 23:20.08 | p3nguin | tenjack: Like if you start httpd, it listens on port 80. Just because you allow port 80 through the firewall does not start an httpd for you. |
| 23:21.20 | TenJack | p3nguin: yea, i know. ive tried starting Adhearsion and it won't connect to Asterisk. I get a connection refused error. |
| 23:21.42 | ChannelZ | RE: is Manager setup on Asterisk in the first place? |
| 23:21.43 | TenJack | so I thought it was b/c the port that it was trying to connect through wasn't open |
| 23:21.43 | p3nguin | tenjack: Is it on the same host as asterisk? |
| 23:22.00 | TenJack | p3nguin: yes |
| 23:22.07 | p3nguin | How does it connect to asterisk? |
| 23:22.10 | TenJack | ChannelZ: yes |
| 23:22.18 | TenJack | I have it connected and working in development |
| 23:22.52 | ChannelZ | uhm... |
| 23:23.07 | ChannelZ | that seems contradictory? |
| 23:23.11 | TenJack | theres a adhearsion.yml file and it has ami: host: 127.0.0.1 |
| 23:23.32 | TenJack | ChannelZ: its working on my local development machine but not on my production slice |
| 23:24.00 | p3nguin | I don't know how it connects to asterisk, so I can't tell you what to look for. |
| 23:24.05 | TenJack | ChannelZ: I have iptables configured on my production slice so I was assuming it was something to do w that |
| 23:24.18 | ChannelZ | On your production machine can you "telnet localhost 5038" and get something? |
| 23:25.30 | TenJack | ChannelZ: I'm not sure what you mean exactly. Somewhat new to this... |
| 23:25.39 | ChannelZ | in a shell |
| 23:25.41 | TenJack | ChannelZ: if I do "netstat -atn | grep 5038" nothing happens |
| 23:26.04 | TenJack | ChannelZ: i dont have telnet installed |
| 23:26.07 | ChannelZ | then Asterisk isn't even listening on port 5038 |
| 23:26.20 | p3nguin | It's supposed to listen on TCP 5038? |
| 23:26.22 | *** join/#asterisk mozart_ar (~mozart_ar@host54.190-230-144.telecom.net.ar) |
| 23:26.22 | ChannelZ | so manager isn't configured |
| 23:26.32 | ChannelZ | (or loaded) |
| 23:26.45 | ChannelZ | p3nguin: AMI |
| 23:26.59 | p3nguin | o i c |
| 23:27.01 | ChannelZ | At least that's what I assume Adhearsion is trying to talk to |
| 23:27.18 | *** join/#asterisk Cesare (~Adium@creati59.lnk.telstra.net) |
| 23:27.19 | TenJack | yes thats true |
| 23:27.22 | TenJack | it uses AMI |
| 23:27.23 | p3nguin | I've tried to find out a couple times, but he doesn't want to tell me. |
| 23:27.34 | p3nguin | FINALLY AN ANSWER! |
| 23:27.52 | TenJack | p3nguin: oh sorry! i didnt understand what you were asking exactly |
| 23:28.18 | ChannelZ | Well again TenJack if nothing is listening on port 5038 then Manager is misconfigured/not running |
| 23:28.28 | p3nguin | Better go fix that. |
| 23:29.10 | TenJack | Yea, you're right. ok thanks |
| 23:34.24 | ChannelZ | YAY! |
| 23:35.09 | p3nguin | channelz: Did you happen to pick up on where the problem with voipms and that pesky s extension turned out to be? |
| 23:35.23 | ChannelZ | Not me, no. |
| 23:35.43 | ChannelZ | I don't use them. I thought maybe it was that Contact: string but that seemed not to be the case last I saw |
| 23:36.36 | p3nguin | There's a setting in the portal for the end user to select if he is using: ATA device, IP Phone or Softphone or Asterisk, IP PBX, Gateway or VoIP Switch |
| 23:36.48 | p3nguin | I use Asterisk, so I've always had it set as such. |
| 23:37.07 | ChannelZ | Ah. Figured it had to be something on their end |
| 23:37.10 | p3nguin | Change it to ATA/softphone, and apparently all calls go to 's'. |
| 23:37.27 | ChannelZ | Interesting |
| 23:37.48 | ChannelZ | When set to Asterisk, does it pay attention to the registration extension also? |
| 23:38.16 | ChannelZ | I'm sort of assuming not |
| 23:38.16 | p3nguin | I'd have to try it. I don't register with an extension because I have multiple DIDs. |
| 23:38.48 | p3nguin | The first guy that told me about the problem is the one who figured it out. He said calls magically started going to his phone number instead of s, and he realized he had changed that setting. |
| 23:39.01 | p3nguin | dijib or whatever |
| 23:40.01 | ChannelZ | yah |
| 23:40.10 | p3nguin | I'd guess if I registered with /exten it would ignore that and send to the DID number, since it ignores the s (lack of /exten in registration). |
| 23:40.13 | ChannelZ | well that's cool. Was the other one around to see the solution? |
| 23:40.23 | ChannelZ | (I forget who it even was) |
| 23:40.36 | p3nguin | I told karen_m, but got no response. |
| 23:41.10 | p3nguin | I'm happy to know they didn't screw up something so simple and necessary. |
| 23:41.31 | p3nguin | I'm wondering if the default setting is to ATA/softphone, though. |
| 23:42.33 | p3nguin | Now if I could only figure out why chan_sccp-b breaks the caller ID on a forwarded call, I'd be a happy guy for a while. |
| 23:42.47 | ChannelZ | shrugs on that one |
| 23:43.44 | p3nguin | When using SIP on my phone, if I enabled CFwdAll and a call came in, it would "bounce" off the phone and go back out to the number specified. The caller ID number would make it from end to end without being altered, so I would know who was actually calling. |
| 23:44.15 | p3nguin | Using SCCP, the call is somehow deflected and it takes on the caller id value of the phone. |
| 23:44.23 | BenC[UK] | Guys, I am trying to use the AMI to originate a call, send it into a dialplan, which then calls a number, and if its answered, send the call into a queue.. |
| 23:44.37 | p3nguin | Screws me right up, thinking someone is calling from the office when it's really someone calling through. |
| 23:44.42 | ChannelZ | Has the channel been Answer()ed yet? |
| 23:45.13 | *** join/#asterisk saxa (~sasa@189.26.255.43) |
| 23:45.29 | p3nguin | Let me look. I don't think so, but I need to check. |
| 23:45.40 | pdtpatrick | Question .. if im seeing stub_ast_key_get: Crypto support not loaded.. what module do i need to load in asterisk ? |
| 23:45.54 | ChannelZ | although that shouldn't matter, if the phone actually issues a redirect... |
| 23:46.21 | p3nguin | Nope, no answer and no apps that answer. |
| 23:46.29 | ChannelZ | if you call forward it to another SIP extension does it show the right or wrong CID? |
| 23:47.03 | p3nguin | I'd imagine it will show the internal cid number just like if I picked up the handset and called the other phone. |
| 23:47.17 | ChannelZ | tests |
| 23:47.21 | p3nguin | I can test it. |
| 23:48.14 | ChannelZ | Mine gives me the original CID |
| 23:48.15 | p3nguin | It shows that the IP phone called the other IP phone. |
| 23:48.24 | p3nguin | Yeah, when I used SIP it worked fine. |
| 23:48.45 | ChannelZ | So it's something with the sccp channel then I guess |
| 23:48.56 | p3nguin | There was a redirect and the original caller CID stayed attached. |
| 23:49.18 | p3nguin | But SCCP breaks it somehow, and I don't know where to look. |
| 23:49.36 | p3nguin | Do you happen to have any hard-wired callerid value on your sip peer? |
| 23:49.47 | ChannelZ | Yes |
| 23:50.03 | ChannelZ | well... I mean the peer in sip.conf has CID associated with it, my name and local extension |
| 23:50.05 | p3nguin | I guess it's just the difference between SIP and SCCP. |
| 23:50.11 | p3nguin | oh, hmm. |
| 23:50.22 | p3nguin | I'm talking about callerid=something <number> |
| 23:50.29 | ChannelZ | yeah.. that |
| 23:50.35 | p3nguin | Okay, that's good. |
| 23:50.48 | ChannelZ | It's Bob <210> which is my exten |
| 23:50.54 | p3nguin | That means the redirect doesn't care about that setting and still retains the original CID anyway. |
| 23:51.03 | ChannelZ | Right |
| 23:51.34 | ChannelZ | Because it's not actually originating a call. That's what I was wondering, if your particular phone was maybe doing something goofy other than a redirect, but it seems not |
| 23:52.02 | ChannelZ | So it's the sccp channel picking up the info elsewhere, it seems |
| 23:52.23 | p3nguin | I'm removing the cid_num line in my sccp.conf on that phone and I'll try again. |
| 23:52.27 | ChannelZ | (I haven't actually ever looked at what the SIP redirect even looks like..) |
| 23:52.48 | p3nguin | Maybe in sccp, that setting is absolute, where in sip it is almost absolute. |
| 23:54.06 | p3nguin | Hmph. Without cid_num, the callerID num field is empty. Empty! |
| 23:54.18 | p3nguin | So it's just chan_sccp being screwy. |
| 23:54.51 | p3nguin | I can't believe no one ever complained to the devs about that. |
| 23:55.02 | *** join/#asterisk Korolev (~SPKorolev@204.88.28.115) |
| 23:55.09 | ChannelZ | Yeah they're just not paying attention to the channel CID it sounds like. If you specifically set CallerID(num) to something and then dial out your sccp channel, does it get the CID you set? |
| 23:55.44 | p3nguin | like... where? |
| 23:56.05 | *** join/#asterisk xpot-mobile (~james@166-70-100-198.ip.xmission.com) |
| 23:56.42 | ChannelZ | like make an extension 5555 that does a Set(CALLERID(num)=5555) and then Dial through sccp wherever it is you're dialing to. Then call 5555 from your SIP phone (which presumably has different callerid= set in sip.conf) |
| 23:57.58 | ChannelZ | Does the CID come up then as 5555 or the CID of your SIP phone? (just wondering if it's somehow looking up the peer and getting the information on its own, or if it's getting passed through the channel like normal) |
| 23:58.52 | p3nguin | I guess I wasn't clear with what I'm doing. I have only SCCP phones on Asterisk. If I set callfwd on my phone to my cell number, and then you call me at 762 from anywhere, my SCCP phone forwards your call to my cell number with the cid_num value I have set for the SCCP phone. |
| 23:59.30 | p3nguin | So it my SCCP phone's cid_num is 8004444444, and you call me from 3145551212, when the call is forwarded to my cell phone, I see a call from 8004444444. |
| 23:59.39 | p3nguin | s/it/if/ |