| 00:01.48 | jdoe | anyone else here trying 1.8? |
| 00:04.46 | logicwrath | im required to register each line on a seperate trunk |
| 00:04.50 | logicwrath | its a broadvoice thing |
| 00:05.33 | logicwrath | i have the correct number of trunks setup (7) |
| 00:06.36 | logicwrath | did you see the lines in teh config? 83-97, 101-111, then 300-314 |
| 00:07.24 | logicwrath | is there something besides lines 101-111 i should be getting back from broadvoice to keep * frmo retransmitting as seen in lines 300-314 |
| 00:14.12 | jdoe | ah |
| 00:14.12 | jdoe | haha |
| 00:14.22 | jdoe | fyi it wasn't forking because I'm an idiot. |
| 00:14.30 | jdoe | if you launch the server with -v it won't background. |
| 00:29.12 | *** join/#asterisk seanstickle (~seanstick@c-98-218-2-48.hsd1.dc.comcast.net) |
| 00:35.55 | *** join/#asterisk csnook (~chris@209-6-38-66.c3-0.smr-ubr1.sbo-smr.ma.cable.rcn.com) |
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| 00:57.31 | *** join/#asterisk Gibby (~gibby@204.118.10.244) |
| 00:57.57 | Gibby | what does asterisk recognize as a pause for programming speed dial on a phone? |
| 00:58.31 | p3nguin | What channel technology? |
| 00:59.35 | Gibby | hih? |
| 01:00.25 | p3nguin | Unless you're talking about an analog channel technology, Asterisk doesn't "recognize" a pause at all. |
| 01:00.43 | p3nguin | If you're using SIP, for example, the phone creates pause. |
| 01:00.56 | p3nguin | And Asterisk won't wait for it. |
| 01:01.07 | Gibby | ahh ok, that is what i thought, but i read some where the the phone send whatever it gets |
| 01:01.27 | p3nguin | If you're writing Dial() commands, you can use w for wait. |
| 01:01.47 | Gibby | tried, i get call can not be completed as dial... from asterisk |
| 01:01.51 | p3nguin | I believe each w is a half second. |
| 01:04.08 | jdoe | hrm. |
| 01:04.25 | Gibby | i have tried , . : p w ] |
| 01:04.32 | Gibby | not the last 1 sorry type |
| 01:04.36 | Gibby | typo |
| 01:04.40 | jdoe | super bloated, I wonder if that's a leak or just something strange.. |
| 01:14.19 | Gibby | could it be something wrong with my outbound rules? |
| 01:14.31 | *** part/#asterisk seanstickle (~seanstick@c-98-218-2-48.hsd1.dc.comcast.net) |
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| 01:21.17 | *** join/#asterisk KingDavidNYC (~Chris1232@pool-96-224-34-135.nycmny.east.verizon.net) |
| 01:21.27 | KingDavidNYC | Hello |
| 01:24.10 | *** join/#asterisk [cannibalera] (~cannibale@200-180-6-131.fnsce703.dsl.brasiltelecom.net.br) |
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| 02:19.47 | KingDavidNYC | Hello |
| 02:22.17 | KingDavidNYC | anybody here? |
| 02:30.24 | p3nguin | ~ask |
| 02:30.24 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 02:36.52 | *** join/#asterisk Besticles (~Besticles@ip68-104-111-21.lv.lv.cox.net) |
| 02:37.06 | KingDavidNYC | Can anybody help me figure out why I can not here the prompts on my side of the call?... I know it is a nat issue, and I know it is because the people have the ports blocked, but I dont know what ports to tell them to open |
| 02:37.26 | KingDavidNYC | They are very stringent in security |
| 02:38.17 | KingDavidNYC | so I open 5060 and rtp from 10000 to 20000, I can make calls... but I just can't play a simple prompt |
| 02:38.30 | Besticles | With AMI Originate, how do I Originate on a specific dahdi channel? I mean I kinda got it working but CLI is reporting an error, even though the call is going thorugh. |
| 02:38.52 | Besticles | [Oct 10 19:32:47] WARNING[9990]: chan_dahdi.c:11160 dahdi_request: Unknown option '-' in '1-1/702400XXXX' |
| 02:48.13 | pabelanger | ~sipnat |
| 02:48.13 | infobot | extra, extra, read all about it, sipnat is Quick guide on configuring Asterisk + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 02:48.18 | pabelanger | KingDavidNYC: ^^ |
| 02:48.37 | jdoe | why might 1.8 be using 10 times as much ram as a comparable setup on 1.6.2? I figured it might be a debugging build but I don't see anything to that effect in config.log etc. |
| 02:49.47 | ChannelZ | Besticles: There's a whole Channel: you need to specify.... |
| 02:51.56 | Besticles | I am sending Channel: DAHDI/1-1/702400XXXX |
| 02:52.05 | Besticles | Unless I am missing something, that's pretty specific. |
| 02:52.28 | ChannelZ | 1-1 is not a correct channel |
| 02:52.34 | Besticles | hrm |
| 02:52.47 | Besticles | dahdi/1/702400XXX? |
| 02:52.49 | ChannelZ | just use 1 |
| 02:52.57 | Besticles | alright thanks |
| 02:55.04 | ChannelZ | KingDavidNYC: You're not really providing enough info |
| 02:56.48 | Besticles | ChannelZ, does asterisk try to make the call anyways if Channel 1 is use? |
| 02:56.54 | Besticles | Like, next channel available? |
| 02:57.33 | Besticles | Sometimes I get the call coming in on a diff channel than expected. I dont know if that's Asterisk doing that, or maybe crappy code on my part. |
| 03:01.07 | *** join/#asterisk darkskiez (~mhb@darkskiez.ipv6.darkskiez.co.uk) |
| 03:02.59 | KingDavidNYC | ChannelZ: The prompts play but I can not hear them from my xlite phone connected outside the network where the asterisk box resides |
| 03:03.54 | KingDavidNYC | ChannelZ: Now, I had this same problem once, and I was able to fix it by setting externip to the external ip address of the the box and nat=yes |
| 03:04.35 | drmessano | ~sipnat |
| 03:04.35 | infobot | sipnat is, like, Quick guide on configuring Asterisk + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 03:05.28 | KingDavidNYC | ChannelZ: But in this new box is different, the company I am doing this for is a bank and they really dont want to open any ports except the ones I tell them and I dont know what port would that be |
| 03:05.56 | drmessano | 5060 UDP and the RTP ports |
| 03:06.16 | drmessano | 10000-20000 UDP or whatever you set them to in rtp.conf |
| 03:07.17 | KingDavidNYC | drmessano: yes, right, I did that, but again, the phone call works... I dont have any problems with the phone call.... I am trying to figure out why I can not hear the prompts play? |
| 03:08.05 | ChannelZ | Besticles: no.. you need to use groups for that |
| 03:08.14 | KingDavidNYC | specifically: what ports should I open to hear a prompt play? Asterisk can not hear my dtmfs either |
| 03:08.47 | logicwrath | Are you suing xlite to send the dtmf tones? |
| 03:09.00 | ChannelZ | There are 2 sides to NAT, yours and the softphone's |
| 03:09.10 | ChannelZ | (I'm assuming they're both behind firewalls) |
| 03:09.16 | drmessano | KingDavidNYC, the prompts dont go over different ports. |
| 03:09.47 | drmessano | KingDavidNYC, the call is setup using specific ports and from there, you dont change RTP ports because you're listening to something different |
| 03:09.52 | ChannelZ | the ports you set in rtp.conf tell Asterisk to ask the remote side to send their media streams to it on those ports. The opposite is true of the other side. |
| 03:11.00 | KingDavidNYC | ChannelZ: why is it that when I set externip=the external ip of the server, it fixes it on my other asterisk box? |
| 03:11.05 | ChannelZ | If the remote side can't send to those ports, you get no audio in. Conversely, the remote side (phone in this case) requests of asterisk where to send its RTP stream. If you can't send out to those ports, they'll get no audio from you |
| 03:12.38 | KingDavidNYC | ChannelZ:But I have set rtp range from 10000 to 20000 on the rtp.conf.... and I have also open rtp 10000 to 20000 on the firewall.... why would it still not play? |
| 03:12.41 | ChannelZ | KingDavidNYC: Because there are two issues. The IPs each side tells the other to contact them at, and the port numbers being open on either side for that communication to take place. |
| 03:13.13 | ChannelZ | KingDavidNYC: Because rtp.conf only controls where the OTHER SIDE SENDS THEIR MEDIA. |
| 03:13.29 | ChannelZ | The phone its self tells asterisk where it to send its media. |
| 03:13.52 | ChannelZ | s/where it/where/ |
| 03:13.56 | KingDavidNYC | ChannelZ: the only thing I can think of, is to also set an equal rtp range from xlite, but I find no option to do that on x-lite |
| 03:14.07 | ChannelZ | Softphones typically only have 1 port |
| 03:14.11 | logicwrath | there is an option |
| 03:14.23 | ChannelZ | xlite specifically I dunno |
| 03:14.47 | KingDavidNYC | logicwrath: I looked everywhere on x-lite, I cant find it |
| 03:14.56 | logicwrath | under sip account settings/properties, then topology, then manually specify range |
| 03:17.03 | KingDavidNYC | SHIT, IT FIXED IT!!!!! |
| 03:17.23 | KingDavidNYC | You guys are geniouses!!!!1 |
| 03:18.00 | ChannelZ | You'd be surprised how little that pays. |
| 03:20.01 | KingDavidNYC | ChannelZ: why would it pay little man?... if you are good, you can be making over 100K,.... right? |
| 03:20.43 | ChannelZ | I was being facetious |
| 03:21.56 | logicwrath | i still have no resolution: My server keeps retransmitting the SIP registrations as seen in this log file. http://pastebin.com/PJJC5DDr - Lines 83-97 it retransmits, lines 101-111 i get back a 200 OK from the server, lines 300-314 i retransmit again. |
| 03:22.19 | logicwrath | should i be able to see my server sending back an ACK |
| 03:23.02 | KingDavidNYC | ok guys, please one more... on the same asterisk box, I can not connect to my outbound carrier...question is: why in the world the sip debug says <--- Reliably Transmitting (NAT) to X.X.X.X:22600 ---> |
| 03:23.02 | KingDavidNYC | SIP/2.0 401 Unauthorized |
| 03:23.54 | KingDavidNYC | why is it trying to go out on port 22600, if I have only range from 10000 to 20000 on rtp.conf? |
| 03:24.51 | logicwrath | close xlite and re-open it |
| 03:25.20 | logicwrath | try sip show peers in CLI to check your ports |
| 03:28.48 | *** join/#asterisk darkskiez (~mhb@darkskiez.ipv6.darkskiez.co.uk) |
| 03:29.11 | KingDavidNYC | logicwrath: it says my IP, port 15548 status=UNREACHABLE |
| 03:30.56 | logicwrath | is your firewall setup to forward 10000-20000 on the softphone side? |
| 03:31.11 | logicwrath | as well as 5060 |
| 03:33.48 | logicwrath | aha, it looks like my * box should be sending ACK after SIP200OK and it is nto according to sip debug |
| 03:34.40 | KingDavidNYC | logiwrath: I suspect something like that is the problem.. please explain.. the softphone is outside the nat, I suspect is that when the asterisk box sends the requests to the router, the router tries to reach the service provider on a different port |
| 03:35.07 | logicwrath | both sides needs firewall setup |
| 03:35.16 | logicwrath | port forward 5060 and 10000-20000 |
| 03:42.38 | *** join/#asterisk mintos (~mvaliyav@nat/redhat/x-jucfibwpafkrwnht) |
| 04:30.23 | p3nguin | The client side rarely needs ports forwarded. |
| 04:31.05 | p3nguin | Configure NAT correctly for the peer on the server side, and forward the ports on the server side. |
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| 04:45.06 | *** join/#asterisk trelane (~trelane@funtoo/staff/trelane) |
| 04:45.12 | trelane | any suggestions for a console softphone? |
| 04:45.20 | trelane | linux console |
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| 04:52.05 | squidly | trelane: i didnt know there was a linux console softphone |
| 04:56.01 | WIMPy | The good old ohphone. |
| 04:56.29 | WIMPy | There's also that demo client of that sip library, but I can't remember the name. |
| 05:06.11 | jdoe | erm. |
| 05:06.25 | jdoe | for the g.729 codecs, is there a 64bit 'register' bin that's actually 64bit? |
| 05:12.06 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 05:14.11 | ChannelZ | probably not |
| 05:15.33 | jdoe | haha. |
| 05:15.45 | jdoe | I like how there's an x86-64 dir though. |
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| 05:53.19 | rayno_b | Hi there everybody. I need some assistance. I have a Digium b410p card that I can't make work. I installed the DAHDI drivers and all went well, but the alarms remain RED (and the actual light at the back also. |
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| 06:43.36 | luvmyh0nda | can i use an analog phone line for accepting calls using asterisk/freepbx? |
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| 06:46.31 | Maliuta | luvmyh0nda: with the right hardware, yes |
| 06:46.50 | Maliuta | luvmyh0nda: needs an ATA/SIP adapter |
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| 07:30.44 | schmidts | good morning |
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| 09:01.05 | mufasis | what is asterisk |
| 09:01.22 | WIMPy | * |
| 09:01.27 | WIMPy | ^ such a thing |
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| 09:08.33 | eMBee | :-) |
| 09:18.11 | [sr] | :) |
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| 09:40.21 | aimka | Hi, is there a way to disable the option crc4 when running dahdi_genconf with the configuration file /etc/dahdi/genconf_parameters ? |
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| 09:56.05 | schmidts | i hate these stupid hackers, or no not the hackers just my stupid customers which doesnt ensoure their system to be safe |
| 09:56.19 | schmidts | s/ensoure/ensure/ |
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| 10:09.39 | hrhrhr | schmidts: isn't that a value add for you tho? :P |
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| 10:17.33 | barbacha | hi all |
| 10:17.43 | schmidts | i just can make sure if they got hack the wont loose a big amount of money ;) |
| 10:17.50 | barbacha | I have a question. I'm settuping an asterisk with digium in the nederlands (amsterdam) |
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| 10:18.32 | schmidts | barbacha 1.) lay down the joint, 2.) everything will be much clearer then :D |
| 10:18.48 | barbacha | after some time (a few minutes) of innactivity, my T0 level1 link goes down (Dutch Telecom puts it down). I have changed options in misdn.conf to "force it up" upon outgoing call but it doen't work |
| 10:19.18 | barbacha | incomming call (from cellphone) wakes up the link from Dutch Telecom and after this outgoing call is possible for some minutes but soon after it goes down again.... |
| 10:19.37 | WIMPy | misdn? That has been discontinued over two years ago. |
| 10:20.10 | WIMPy | And since when can digium hardware be used with misdn? |
| 10:20.21 | aimka | q |
| 10:21.08 | barbacha | WIMPy: well sorry if I say things wrong I'm relativly new to all this |
| 10:21.22 | barbacha | WIMPy: but the symptom is the one I told |
| 10:21.41 | WIMPy | Tell us exactely what you're using. |
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| 10:25.54 | barbacha | WIMPy: what's the quickest way for me to gather this information (I did not install the machine) |
| 10:27.57 | *** join/#asterisk [sr] (~Unknowned@pa8-84-91-197-8.netvisao.pt) |
| 10:28.05 | [sr] | hi WIMPy |
| 10:28.20 | WIMPy | Hi [sr] |
| 10:28.38 | [sr] | WIMPy: i'm still waiting for the guy who handles MISDN.ORG to create the ALIAS for BUGS.MISDN.ORG |
| 10:28.46 | WIMPy | barbacha: First thing would be the type of hardware. |
| 10:30.01 | WIMPy | barbacha: Asterisk and Linux versions: asterisk -rx "core show version" ; cat /proc/version |
| 10:30.34 | WIMPy | Is it a 'normal' linux install or some Asterisk Distro? |
| 10:31.22 | WIMPy | [sr]: misdn.org doesn't see much activity, does it? |
| 10:33.04 | barbacha | it's a "normal distro" debian |
| 10:33.04 | barbacha | Asterisk 1.6.1.12 built by root @ asterisk on a i686 running Linux on 2010-09-15 15:18:39 UTC |
| 10:33.17 | barbacha | Linux version 2.6.26-2-686 (Debian 2.6.26-25) |
| 10:33.35 | barbacha | 04:00.0 ISDN controller: Digium, Inc. Wildcard B410 quad-BRI card (rev 01) |
| 10:33.39 | barbacha | ^ from lspci |
| 10:33.46 | barbacha | and I *do* confirm we use misdn |
| 10:33.56 | barbacha | (be it deprecated or not) |
| 10:34.44 | andylockran | I'm getting a weird problem where my phones are registering, but they are unable to make or receive calls. The debug http://dpaste.com/256232/ |
| 10:35.25 | andylockran | it registers fine, we see the register messages, but then nothing gets through to the server post that - not even debugs :( |
| 10:35.30 | andylockran | this was all working before the weekend |
| 10:36.01 | andylockran | any recommendations? |
| 10:36.15 | *** join/#asterisk sgimeno (~chatzilla@163.117.211.10) |
| 10:36.36 | [sr] | WIMPy: no idea.. |
| 10:37.57 | WIMPy | barbacha: With a digium interface card, you should be using the dahdi drivers from digium. That's what they're made for. |
| 10:38.15 | WIMPy | But make sure to have the latest version of libpri. |
| 10:39.09 | barbacha | yes I know about dahdi, but the guy who did this setup wanted to do "the same as our other (quite old) servers" |
| 10:39.16 | barbacha | and I wont do the switch right now |
| 10:39.20 | WIMPy | I didn't know their hardware was supported by misdn. Wonder it it would work with misdn2 as well. |
| 10:39.38 | barbacha | libpri is libpri-1.4.10.2 |
| 10:40.17 | WIMPy | Hmm. Right. the old chan_misd uses libpri itsel, I guess. |
| 10:41.03 | WIMPy | So just installing the latest libpri and re-compiling chan_misdn might so the trick. |
| 10:41.31 | barbacha | for information I already had exactly the same problem/bug and I fixed it with "l1watcher_timeout=5" (as opposed by the default 0) in /etc/asterisk/misdn.conf |
| 10:41.53 | barbacha | but this seems to be not working in the nederlands |
| 10:43.22 | WIMPy | Cool. Found an old misdn.conf. Yes, got the same there. |
| 10:43.49 | WIMPy | But there has been a problem with libpri, line deactivation and TEI management lately. |
| 10:46.07 | WIMPy | Can you make outbound calls again after an inbound call succeeded? |
| 10:54.46 | *** join/#asterisk rayno_b (~chatzilla@41.219.94.194) |
| 10:58.20 | andylockran | http://dpaste.com/256238/ - I have a problem and can't work it out... |
| 10:58.36 | andylockran | phones register ok - but can't make or receive calls |
| 10:58.53 | rayno_b | hi there everybody - newbie to Asterisk. I've got my system configured with a Digium b410p card and dahci_tool reports the first channel is OK (Only 1 channel plugged in). However, I can't make calls and when I dail in, CLI does not report anything. I think it's because chan_dahdi.so is not loaded, but when I try to load this, it says Unable to load module chan_dahdi.so |
| 10:58.55 | rayno_b | Command 'module load chan_dahdi.so' failed. |
| 11:00.08 | WIMPy | rayno_b: Set verbose and debug to 9 and try again. It should tell you, why. |
| 11:00.37 | rayno_b | WIMPy - do you mind helping me to set verbose and debut to 9? |
| 11:00.46 | rayno_b | dubut = debug. sorry. |
| 11:00.59 | WIMPy | core set verbose 9 |
| 11:01.05 | WIMPy | core set debug 9 |
| 11:01.24 | rayno_b | okay cool, stdby |
| 11:02.32 | rayno_b | WIMPy - I changed both levels and it was successful, but loading the module returns the same: |
| 11:02.35 | rayno_b | Unable to load module chan_dahdi.so |
| 11:02.37 | rayno_b | Command 'module load chan_dahdi.so' failed. |
| 11:03.15 | WIMPy | Do you have a chan_dahdi.so at all? |
| 11:03.36 | WIMPy | Did you install dahdi before compiling Asterisk? |
| 11:03.59 | WIMPy | module show like dahdi |
| 11:04.13 | rayno_b | I downloaded the Asterisk ISO and installed. Then after that installed DAHDI. |
| 11:05.09 | rayno_b | http://pastebin.com/hQncJXvN |
| 11:05.47 | WIMPy | Ok, it exists. |
| 11:06.31 | WIMPy | Try to module unload chan_dahdi.so then load it again. |
| 11:07.09 | rayno_b | It comes back with: Unable to unload resource chan_dahdi.so Command 'module unload chan_dahdi.so' failed. |
| 11:07.24 | rayno_b | If I log onto the web interface, I do not see the module listed there. |
| 11:07.31 | WIMPy | Nothing else? |
| 11:08.44 | rayno_b | I'll paste what I see as enabled. |
| 11:08.46 | rayno_b | stdby |
| 11:10.36 | rayno_b | The web interface does not copy so nicely. |
| 11:10.56 | rayno_b | Under basic, the only DAHDi module I have enabled there, is DAHDi Config |
| 11:10.59 | WIMPy | neer knew that the b410p is also just a HFC-4S design. interesting. |
| 11:11.21 | rayno_b | No further module by the name of DAHDi is enabled on the web interface. |
| 11:11.32 | WIMPy | rayno_b: You use the shell via web? |
| 11:12.02 | rayno_b | I use the shell via ssh access (if I understand your question correctly) |
| 11:12.23 | WIMPy | Sorry, but I can't comment on the GUI. |
| 11:12.28 | WIMPy | Ok. |
| 11:12.54 | WIMPy | But you get no additional messages there when trying to load chan_dahdi? |
| 11:13.03 | rayno_b | Let me try once again |
| 11:13.28 | WIMPy | Maybe the configutation is completely missing? |
| 11:13.52 | rayno_b | This is what I get: http://pastebin.com/ZextYLUq |
| 11:14.46 | hrhrhr | i used misdn when i set up a b410 some time ago. pretty sure it was the only option at the time... |
| 11:15.11 | WIMPy | First it fails to load, then it reads it's configuration? *scratch head* |
| 11:15.18 | hrhrhr | nice to know that dahdi supports it natively now tho |
| 11:15.40 | rayno_b | I used the card's manual document which kinda suggested to go with DAHDi. |
| 11:16.07 | WIMPy | hrhrhr: Yes, the hardware seems to be the same as the others. |
| 11:17.05 | rayno_b | Is the order I did it okay? First install Asterisk ISO and then only DAHDi? |
| 11:17.06 | WIMPy | Just more expensive. |
| 11:17.52 | WIMPy | rayno_b: I have no idea how to use that distribution, sorry. |
| 11:18.56 | *** join/#asterisk ukine_work (~ukine@14-145.97-97.tampabay.res.rr.com) |
| 11:20.37 | rayno_b | WIMPy - Which distribution do you recommend with The Digium b410p card? |
| 11:21.18 | WIMPy | Use whatever you're familiar with. |
| 11:23.33 | rayno_b | WIMPy - I've never used any, I'm only starting with this to see if I can get it working correctly - so I am unfamilier with the entire open source PBX scenario. Okay, I've downloaded the AsteriskNOW iso which installed CentOS and Asterisk NOW. And then I followed the DAHDI instructions and I think my only problem now is to get the card drivers to interface correctly with Asterisk. |
| 11:25.03 | WIMPy | I guess it needs to generate some configuration, but that's hard to say. |
| 11:25.12 | WIMPy | You've got no Linux experience then? |
| 11:26.16 | rayno_b | No no, I'm clued up on Linux. It's the the PBX part that's new to me. |
| 11:26.34 | creativx | man |
| 11:26.52 | creativx | im being haunted by choppy sip audio.. i thought i had fixed it by changing from monitor to mixmonitor.. but to no avail |
| 11:28.15 | WIMPy | rayno_b: Unless you just want some basic config and not find out about the real strengths of Asterisk, I'd suggest you use your favourite Linux distro and install a pure Asterisk there. |
| 11:28.25 | WIMPy | And read the book. |
| 11:28.28 | WIMPy | ~book |
| 11:28.28 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 11:29.14 | *** join/#asterisk deonv (~adium@pixfirewall.itn.com.na) |
| 11:31.44 | creativx | ok.. what the h can cause this? sip audio.. softphone.. if both parties talk at the same time, the audio gets very choppy for either one end or both.. any ideas? |
| 11:32.09 | creativx | can it be the softphone? |
| 11:32.50 | WIMPy | Shitty echo cancellation on the softphone? Shitty sound system? |
| 11:33.06 | creativx | hmm |
| 11:33.14 | creativx | im gonna try another softphone |
| 11:33.17 | creativx | easiest to test that first |
| 11:33.27 | creativx | the weird thing is that the problem seems to have risen lately |
| 11:33.45 | WIMPy | Is it all local? |
| 11:34.08 | creativx | i just tested sip softphone -> local asterisk -> sip itsp -> mobile phone |
| 11:34.19 | WIMPy | Maybe you can disable EC? |
| 11:34.39 | creativx | gonna try that now |
| 11:40.20 | barbacha | back |
| 11:40.26 | barbacha | 12:46 < WIMPy> Can you make outbound calls again after an inbound call succeeded? |
| 11:40.29 | barbacha | ^ yes |
| 11:41.21 | *** part/#asterisk andylockran (~andylockr@genesis.zrmt.com) |
| 11:45.10 | *** join/#asterisk odenkos (~odenkos@ip-212-081-019-170.static.nextra.sk) |
| 11:46.52 | barbacha | once the link is up (activated by incoming call) it stays up and works normally for a few minutes |
| 11:48.19 | *** join/#asterisk sekil (~sekil@80.93.247.26) |
| 11:51.42 | WIMPy | barbacha: Ok, that's not the TEI management related one then. But I'd still try a libpri update. |
| 11:51.57 | WIMPy | Or you compare the version of that other system. |
| 11:53.40 | *** join/#asterisk rrb3942 (~rbullock@208.34.105.161) |
| 11:56.29 | barbacha | WIMPy: could you define "TEI" for the noob I am ? |
| 11:59.28 | WIMPy | Terminal Equipment Identifier. |
| 11:59.59 | WIMPy | It's used to address physical devices on the ISDN bus. |
| 12:08.22 | *** part/#asterisk bsaxon (~bsaxon@68-113-127-34.dhcp.leds.al.charter.com) |
| 12:09.13 | *** join/#asterisk angryuser (~angryuser@LPuteaux-151-42-27-99.w193-251.abo.wanadoo.fr) |
| 12:10.54 | hrhrhr | creativx: look at broadcast silence options too |
| 12:15.32 | barbacha | ok I *think* I have found the solution |
| 12:16.11 | barbacha | it *count* be to call the misdn_check_l2l1(g:extern,2) asterisk application for each outgoing call |
| 12:16.26 | barbacha | this application will try to up the port when it's down before passing the call |
| 12:16.57 | creativx | hrhrhr: it became apparant that when everyone got their new hp pc's nobody bothered to image the proper x-lite settings.. |
| 12:17.32 | *** join/#asterisk deonv (~adium@pixfirewall.itn.com.na) |
| 12:19.25 | *** join/#asterisk ariel_ (~chatzilla@63.214.236.169) |
| 12:20.12 | creativx | hmm |
| 12:20.27 | creativx | how can i remove all members from a queue, without specifying each |
| 12:20.41 | angryuser | creativx, with a little script |
| 12:20.48 | angryuser | and asterisk -x |
| 12:22.02 | creativx | well there seems to be some script that has caused a mess.. |
| 12:22.22 | creativx | cause removing this member ' SIP/<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.01//EN""http://www.w3.org/TR/html (dynamic) (Invalid) has taken no calls yet |
| 12:22.31 | creativx | is looking difficult |
| 12:22.32 | creativx | hehe |
| 12:27.50 | *** join/#asterisk underdog_ (~whyareyou@abel.33ad.org) |
| 12:28.28 | creativx | theres been some cocking up, thats for sure |
| 12:39.30 | *** join/#asterisk t_dot_zilla (~chatzilla@firewall-a.buf.ny.i-evolve.net) |
| 12:42.47 | *** join/#asterisk metiu (~chatzilla@85-18-228-185.ip.fastwebnet.it) |
| 12:47.00 | metiu | how do I start a call from the dialplan, wait for some action, then bridge the two halves of the call? I tried by using originate(), but when originate exits it jumps to the given extension and/or app... I don't know what app to use |
| 12:47.14 | metiu | I tried with an extension using "bridge" |
| 12:47.38 | metiu | and it connects, then drops the call mysteriously |
| 12:48.11 | metiu | because it exits to the next priority, which is Hangup as usual |
| 12:49.19 | *** join/#asterisk SuPrSluG (~SuPrSluG@host-64-179-106-158.ind.choiceone.net) |
| 12:51.24 | *** join/#asterisk coppice (~chatzilla@m121-202-107-246.smartone-vodafone.com) |
| 12:54.55 | creativx | what besides monitor() can call soxmix ? |
| 12:55.22 | creativx | from asterisk internally that is |
| 12:59.02 | logicwrath | im gonna be reloading my * box to 1.4 from 1.6 but i figured before i do I would post this one more time in case someone wants to help |
| 12:59.04 | logicwrath | My server keeps retransmitting the SIP registrations as seen in this log file. http://pastebin.com/PJJC5DDr - Lines 83-97 it retransmits, lines 101-111 i get back a 200 OK from the server, lines 300-314 i retransmit again. It also appears that my server is not sending an ACK after it recieves the 200 OK. I used grep and found 0 ACK datagrams transmitted. |
| 12:59.26 | *** join/#asterisk fofware (~Fabian@host199.190-31-51.telecom.net.ar) |
| 13:08.33 | metiu | do I need to use a confbridge/meetme to do what I said? |
| 13:09.22 | *** join/#asterisk bsaxon (~bsaxon@12.107.149.61) |
| 13:10.25 | *** join/#asterisk ManxPower (~manxpower@user-24-214-153-32.knology.net) |
| 13:10.37 | *** part/#asterisk ManxPower (~manxpower@user-24-214-153-32.knology.net) |
| 13:10.57 | *** join/#asterisk timahvo1 (~rogue@41.223.57.72) |
| 13:17.24 | *** join/#asterisk SaiSoma (~chatzilla@client105.jdcc.edu) |
| 13:19.42 | *** part/#asterisk barbacha (~nico@chez.nicolas.bouthors.org) |
| 13:22.02 | fors1 | hm. so after upgrading to 1.6, all my IVR's times out after Background() has completed, totally ignoring my TIMEOUT(response) value. Has anything changed with TIMEOUT since 1.4? (couldn't find any docs that would explain why it times out) |
| 13:22.47 | *** join/#asterisk BeeBuu (~chatzilla@183.28.2.165) |
| 13:22.54 | BeeBuu | hello,all |
| 13:25.33 | BeeBuu | when i try mixmonitor(${},b,/home/user/a_shell.sh ${mixmonitor_filename}), that command doesn't work,any help? |
| 13:26.36 | WIMPy | BeeBuu: core show application mixmonitor |
| 13:27.23 | phix | hi BeeBuu! |
| 13:28.25 | BeeBuu | yes,i did. |
| 13:29.13 | BeeBuu | full command is -->mixmonitor(${mixmonitor_filename},b,/home/user/a_shell.sh ${mixmonitor_filename}) |
| 13:29.45 | BeeBuu | i want to move the monitor file to somewhere via a shell command file |
| 13:29.57 | BeeBuu | but it seem don't work |
| 13:30.09 | BeeBuu | anything i missed? |
| 13:30.12 | WIMPy | As it's mixmonitor() that sets mixmonitor_filename, it will still be empty at the time you call it. |
| 13:31.03 | Gibby | The following ports need forwarded if asterisk is behind a firewall right? udp 5060 and udp 10000-20000 |
| 13:31.20 | WIMPy | Yes, it tells you how to parse the parameters at time of execution of the command parameter. |
| 13:31.50 | BeeBuu | hm........ |
| 13:32.03 | WIMPy | Gibby: That would be the safe way for the standard configuration. |
| 13:33.56 | Gibby | that is what I thought, i can make outgoing calls find, incoming is still an issue, all i see on the asterisk console is http://pastebin.com/t6MTX5yQ |
| 13:34.07 | BeeBuu | WIMPy: how can i know my shell file had work or not? |
| 13:34.27 | WIMPy | ~sipnat |
| 13:34.27 | infobot | i guess sipnat is Quick guide on configuring Asterisk + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 13:34.32 | *** join/#asterisk nunners (~chatzilla@host81-130-94-234.in-addr.btopenworld.com) |
| 13:34.45 | BeeBuu | i get the message under CLI: ==Executins ..... |
| 13:34.57 | Gibby | ty WIMPy |
| 13:34.58 | WIMPy | BeeBuu: With an empty parameter, it won't work. |
| 13:35.39 | WIMPy | Otherwise you can write a log from your script. |
| 13:36.09 | BeeBuu | WIMPy: Executing [/home/user01/cpsh 1002_234_2009-12-05-18-08-50_1260007730.2.gsm] |
| 13:36.21 | BeeBuu | that's come from CLI |
| 13:36.38 | BeeBuu | it mean correct? |
| 13:38.10 | WIMPy | Looks like a date from last year. Otherwise it could be correct. |
| 13:39.24 | BeeBuu | that's a example |
| 13:40.20 | BeeBuu | how can know the shell command file is run correctly? |
| 13:40.41 | WIMPy | ^^ Otherwise you can write a log from your script. |
| 13:45.41 | *** join/#asterisk af_ (~getsmart@78.134.21.149) |
| 13:46.20 | *** join/#asterisk suvir (~suvir@ppp-124-120-130-177.revip2.asianet.co.th) |
| 13:46.50 | *** part/#asterisk suvir (~suvir@ppp-124-120-130-177.revip2.asianet.co.th) |
| 13:49.02 | *** join/#asterisk pigpen (~mark@fw.seamans.cc) |
| 13:51.09 | *** join/#asterisk frigidzephyr (~rnewton@nat/digium/x-nzkxhponxszdqgck) |
| 13:51.28 | *** join/#asterisk raden (~J@66-168-15-100.dhcp.stpt.wi.charter.com) |
| 13:51.53 | raden | how can i make it so a user can access asterisk CLI ? instead of just root ? |
| 13:58.42 | *** join/#asterisk metiu_ (~chatzilla@85-18-228-185.ip.fastwebnet.it) |
| 13:58.55 | pigpen | You know, I have never even considered that. |
| 14:00.53 | pigpen | raden, well, I quickly added a user to the asterisk group, with no preval. |
| 14:01.08 | pigpen | my guess is it largely depends on the user running the asterisk processes. |
| 14:01.33 | pigpen | Please emphasize "my guess" |
| 14:02.10 | pigpen | Users shouldn't be able to touch the CLI anyway...too dangerous. |
| 14:02.27 | pigpen | write an app which access the ami or such. |
| 14:05.26 | *** part/#asterisk nunners (~chatzilla@host81-130-94-234.in-addr.btopenworld.com) |
| 14:09.36 | *** join/#asterisk nunners (~chatzilla@host81-130-94-234.in-addr.btopenworld.com) |
| 14:10.22 | nunners | Asterisk 1.6/Dahdi 2.4 Can someone tell me if modprobe wctdm24xxp should actually produce output? I type it and it doesn't, so does that mean it's not working? |
| 14:11.12 | tzanger | nunners: do you see the card in lspci output? |
| 14:11.48 | nunners | tzanger: I'm trying to work out why it isn't working, and am therefore going through everything I can think of! so yes if lspci helps, then yes! |
| 14:12.03 | nunners | I suppose I'd better explain the problem first though.... |
| 14:12.11 | *** join/#asterisk ManxPower (~manxpower@user-24-214-153-32.knology.net) |
| 14:12.28 | tzanger | nunners: modprobe doesn't generally give output. you see the output in the kernel system log, and you see that by running "dmesg" and looking at the last few lines |
| 14:12.48 | tzanger | nunners: if there is nothing there about actually initializing the card, see if the card is listed in the lspci (list pci) output |
| 14:13.17 | nunners | tzanger: output from dmesg is.... [12771.985142] dahdi: Registered tone zone 0 (United States / North America) |
| 14:13.19 | nunners | [13172.570669] dahdi_echocan_mg2: Registered echo canceler 'MG2' |
| 14:13.21 | nunners | [13172.570669] dahdi: Registered tone zone 4 (United Kingdom) |
| 14:13.22 | nunners | which I guess is fine.... |
| 14:13.33 | tzanger | nunners: no, that doesn't specifically say it did anything for a tdm2400 |
| 14:13.37 | tzanger | all that is is the normal dahdi output |
| 14:14.03 | nunners | [ 10.608285] wctdm24xxp 0000:03:01.0: Port 1: Installed -- AUTO FXO (UK mode) |
| 14:14.05 | nunners | [ 11.440228] wctdm24xxp 0000:03:01.0: Port 2: Installed -- AUTO FXO (UK mode) |
| 14:14.06 | nunners | [ 11.968228] wctdm24xxp 0000:03:01.0: Port 3: Installed -- AUTO FXO (UK mode) |
| 14:14.08 | nunners | [ 12.496228] wctdm24xxp 0000:03:01.0: Port 4: Installed -- AUTO FXO (UK mode) |
| 14:14.10 | nunners | [ 12.508232] wctdm24xxp 0000:03:01.0: Found a Wildcard TDM: Wildcard TDM410P ( 0 BRI spans, 4 analog channels) |
| 14:14.12 | tzanger | nunners: don't flood the channel |
| 14:14.13 | nunners | Sorry - should have pastebin'd that but... it shows its working! |
| 14:14.25 | tzanger | but it appears that it found a tdm400 with four FXO modules |
| 14:14.32 | tzanger | so yes, it looks like the card and driver are happy |
| 14:14.43 | nunners | which is correct... so why can't I dial out, dahdi channels always busy! |
| 14:14.44 | *** join/#asterisk nicola_pav (~chatzilla@mail2.tikalnetworks.com) |
| 14:14.59 | tzanger | busy or unavailable |
| 14:15.15 | *** join/#asterisk mog (~mog@c-68-62-169-225.hsd1.al.comcast.net) |
| 14:15.16 | tzanger | I don't know the UK tones, but in north america there is a difference between fast and normal busy tones |
| 14:15.17 | nicola_pav | hello. I have ubuntu connected to ther internet. I attached a usb-to-ethernet and to this usb ethernet a switch where i added an ip phone |
| 14:15.22 | nunners | busy |
| 14:15.46 | nicola_pav | how can i let the ip phone take internet from the ubuntu and register to asterisk server |
| 14:15.48 | nicola_pav | ? |
| 14:16.11 | ManxPower | nunners, have you provided the output of dahdi_cfg -vvv? |
| 14:16.20 | *** join/#asterisk eugeneoden (~goden@rrcs-97-77-255-86.sw.biz.rr.com) |
| 14:16.57 | nunners | http://pastebin.com/ZpU3n8GW |
| 14:17.32 | nunners | Manxpower: so again, I think that's correct! |
| 14:18.04 | ManxPower | nunners, also cat /proc/dahdi/1 |
| 14:18.25 | *** join/#asterisk bluOxigen (~sfds@unaffiliated/bluOxigen) |
| 14:18.39 | logicwrath | can someone verify that I should be seeing an ACK after the SIP 200 OK in the sip debug logs? I am new to sip debugging and i want to know for future reference |
| 14:19.12 | nunners | Manxpower: http://pastebin.com/8sbAQLcs |
| 14:19.35 | ManxPower | nunners, that RED in the list means "NO LINE VOLTAGE DETECTED" |
| 14:19.41 | nunners | manxpower: that shows in use... but they aren't - there's actually only one plugged in! |
| 14:20.02 | schmidts | logicwrath yes there should be an ACK to an 200 OK |
| 14:20.04 | nunners | manxpower: which means what (sorry)... the lines work, I can't see why there would be no voltage? |
| 14:20.05 | ManxPower | nunners, In use means "asterisk is running" it does not mean" active call" |
| 14:20.17 | ManxPower | nunners, the server is not detecting the lines plugged into the system |
| 14:20.39 | nunners | manxpower: ok so that's the problem.... :cheekily: .... how do I solve it? |
| 14:20.43 | ManxPower | until you clear the RED alarm you will not be able to make calls. |
| 14:20.53 | ManxPower | nunners, connect lines to the server and it will be solved. |
| 14:21.26 | nunners | manxpower: the line on channel 1 is connected! the others aren't live yet with BT! Do all four need to be connected for any to work? |
| 14:22.17 | ManxPower | nunners, the server is not detecting lines connected to the server. There is nothing complicated to this. Someone screwed up the wiring to your server. |
| 14:22.56 | ManxPower | as you can see port 1 is in red alarm -- the server is not detecting a line on that port. Are you sure you didn't wire up ports 21-24 and not 1.4 |
| 14:23.32 | nunners | manxpower: as I'm the one who's built it... must be me... what could I have screwed up? The card is plugged in correct, the lights are on the card, and there's only four ports... so not sure what else I could have done? |
| 14:23.33 | ManxPower | nunners, What color are the modules on the card? red or green? |
| 14:23.44 | nunners | manxpower: all red... which i think is correct fxo... |
| 14:24.12 | ManxPower | nunners, you do not have a software issue, you have a wiring issue. |
| 14:24.20 | nunners | manxpower: or am I having a very blond moment? and I've bought the wrong modules? |
| 14:24.31 | ManxPower | nunners, you did not buy the wrong modules. |
| 14:24.40 | ManxPower | You have a problem with the wiring. |
| 14:24.49 | nunners | manxpower: good... blond moment over... |
| 14:24.56 | *** join/#asterisk rgsteele (~rgsteele@apo.aweber.com) |
| 14:25.08 | nunners | manxpower: when you say the wiring, do you mean the wires from the BT line to server, or internally within the server? |
| 14:25.22 | ManxPower | from the telco to the server |
| 14:25.44 | nunners | manxpower: ok so possible cable problem! |
| 14:25.55 | *** join/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 14:27.02 | nunners | manxpower: thanks for the help - I'm glad I'm not going mad, as I though I was for a while!!!! |
| 14:32.28 | logicwrath | my server is not responding ACK to SIP 200 OK. I have SIP debug logs to show this. I was going to reload with asterisk 1.4 and get away from 1.6.2.12. Is there anything else I should try? |
| 14:32.43 | logicwrath | i just retransmit registrations over and over till my IP is blocked |
| 14:33.21 | logicwrath | http://pastebin.com/ypKBDYqk - line 402, 434, 464, and 496 show retransmissions and no ACK transmissions. |
| 14:34.16 | *** join/#asterisk UQlev (~Yuriy@212.50.99.8) |
| 14:35.13 | metiu | can I join an existing Page MeetMe conference? Does Page tell me back which conf# it has created? |
| 14:36.22 | ManxPower | logicwrath, fix your NAT problems |
| 14:36.26 | ManxPower | ~sipnat |
| 14:36.26 | infobot | sipnat is, like, Quick guide on configuring Asterisk + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 14:37.08 | logicwrath | i have NAT setup properly.. what makes you think there is a NAT problem |
| 14:37.36 | logicwrath | ive even used 2 diff firewalls and tried it in a DMZ |
| 14:37.48 | ManxPower | because Asterisk is telling the remote server the IP to respond to is 10.1.10.22 |
| 14:37.58 | logicwrath | what line |
| 14:38.18 | ManxPower | what are your localnet= and externip= options in sip.conf set to |
| 14:39.45 | logicwrath | http://pastebin.com/DBBwRvaa |
| 14:39.53 | logicwrath | thats not even a valid subnet for me |
| 14:39.59 | logicwrath | i have no idea how that even shows up |
| 14:40.39 | ManxPower | logicwrath, fix your NAT problems |
| 14:40.42 | logicwrath | that is an eyebeam client IP anyways |
| 14:40.46 | logicwrath | not asterisk |
| 14:42.07 | ManxPower | logicwrath, Ah, I can't help then. |
| 14:42.22 | ManxPower | But once you do fix your NAT issues you will start to make progress. |
| 14:42.35 | nunners | manxpower: hurray... found a cable that doesn't cause the line to be red!!!! thanks for the help... just need to work out why I can't dial out still though....! |
| 14:42.46 | logicwrath | i need proof there is NAT problems |
| 14:43.06 | ManxPower | logicwrath, what is the IP of your Asterisk server |
| 14:43.11 | logicwrath | 10.10.0.14 |
| 14:43.26 | logicwrath | 99.56.133.1 outside |
| 14:43.40 | ManxPower | and the IP of your service provider? |
| 14:43.50 | logicwrath | 147.135.0.128 |
| 14:45.23 | ManxPower | NAT ISSUE ===> [Oct 11 08:48:51] VERBOSE[2747] chan_sip.c: Retransmitting #2 (NAT) to 147.135.0.128:5060: |
| 14:45.36 | ManxPower | it might not be a nat issue, but it sure looks like it |
| 14:45.45 | nicola_pav | i am trying to connect an ip phone to an ubuntu laptop so it can share its internet connection to register to an asterisk server |
| 14:45.54 | nunners | sorry guys, me again... still trying to solve the problems, getting closer though! If I check the status of a dahdi channel (1 in this case) and it shows Hookstate: offhook and signalling type: FXS Kewlstart, does this show the correct info for an fxo channel? |
| 14:46.06 | nicola_pav | i did interet sharing on ubuntu but no luck |
| 14:46.12 | nicola_pav | any hint? |
| 14:46.17 | ManxPower | nunners, does cat /proc/dahdi/1 still show all ports RED? |
| 14:46.41 | nunners | manxpower: 1 WCTDM/0/0 FXSKS (In use) (SWEC: MG2) |
| 14:46.43 | nunners | No red! |
| 14:46.51 | ManxPower | nicola_pav, when you have a specific Asterisk related question people might start helping |
| 14:47.15 | ManxPower | nunners, good. now we will need a pastebin of a failed call (not an OLD pastebin, tou plugged in a line so you need a new pastebin) |
| 14:47.19 | nicola_pav | manxpower, since its ip phone |
| 14:47.22 | logicwrath | ManxPower: http://pastebin.com/buX5JnFY |
| 14:47.31 | nicola_pav | thought maybe one had similar issue |
| 14:47.34 | nicola_pav | and can help |
| 14:47.50 | ManxPower | nicola_pav, I've connected hundreds of IP phones to Asterisk. there is nothing special about it. |
| 14:48.36 | nicola_pav | i want to connect it to my laptop to share its internet to regsiter to an asterisk server. I am doing this in order to debug via wireshark |
| 14:48.36 | ManxPower | logicwrath, that register debug has nothing to do with making a call |
| 14:48.48 | logicwrath | i dont have problems making calls |
| 14:48.53 | pigpen | ManxPower, you told me it was thousands of phones.... |
| 14:48.54 | ManxPower | nicola_pav, "internet sharing" != Asterisk |
| 14:49.12 | logicwrath | inbound/outbound calling work fine, until my IP gets blocked for registering over and over |
| 14:49.20 | ManxPower | pigpen, I was using internet inches then |
| 14:49.26 | nicola_pav | manxpower, thank u |
| 14:49.56 | nunners | manxpower: Failed call... http://pastebin.com/3wG7GsUf |
| 14:50.29 | nunners | manxpower: unfortunately that's all I can get out of it, as what I understand as verbose isn't verbose!!! |
| 14:51.06 | pigpen | ManxPower, thanks for the clarification. |
| 14:51.13 | ManxPower | nunners, pastebin the output of "dahdi show channels" |
| 14:51.50 | nunners | manxpower: dahdi show channels -> http://pastebin.com/YV6Eiqp9 |
| 14:52.21 | *** join/#asterisk aidinb (~Aidin@unaffiliated/aidinb) |
| 14:52.32 | ManxPower | nunners, are you connecing to asterisk by using "asterisk -rvvv" ? |
| 14:52.49 | nunners | no just -r.... |
| 14:53.03 | nunners | manxpower: ah... see what that does now... trying call again! |
| 14:54.03 | nunners | manxpower: http://pastebin.com/6JBNcJuL I'm just having a look as well, so my find the problem myself....! |
| 14:54.17 | pigpen | ManxPower, in 1.4 there was a limit of the number of phones that could be included in a single page group before the system segfaulted. Largely an issue with command length, but even if that were made larger, it would segfault. |
| 14:54.20 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 14:54.34 | pigpen | This was worked around by having a page group call other page groups. |
| 14:54.43 | nunners | manxpower: ok, so it's unavailable, not busy.... |
| 14:54.48 | pigpen | Do you know if this has been address in the 1.6.1.x or 1.6.2.x ? |
| 14:55.18 | pigpen | ie: I to pages routinely to 75 - 500 sip phones. |
| 14:55.50 | ManxPower | nunners, I cannot help you further, you are using FreePBX. |
| 14:56.06 | *** part/#asterisk ManxPower (~manxpower@user-24-214-153-32.knology.net) |
| 14:56.09 | nunners | ok... sorry... unfortunately no-one on freepbx was willing... thanks anyway! |
| 14:56.38 | pigpen | nunners, yeah, they stick so much in there, it is difficult to figure out what is going on. |
| 14:57.29 | pigpen | Remember that is a free product to entice you to purchase their $$ product. |
| 14:57.51 | *** join/#asterisk anonymouz666 (~anonymouz@187-28-37-118.poolip.RJO.embratel.net.br) |
| 15:00.48 | *** part/#asterisk nunners (~chatzilla@host81-130-94-234.in-addr.btopenworld.com) |
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| 15:34.58 | *** join/#asterisk telnettech (~telnettec@216.49.139.56) |
| 15:35.08 | telnettech | good morning everyone |
| 15:35.42 | telnettech | how do I force an incoming call on a SIP trunk to only accept RFC2833? |
| 15:36.43 | pigpen | telnettech, I think you place dtmfmode=rfc2833 in the peer,user,friend definition in the sip.conf |
| 15:36.54 | *** join/#asterisk SeTTleR (~bernd@p5DDECD15.dip.t-dialin.net) |
| 15:37.31 | telnettech | pigpen. I have that and it works IP to IP but when a call is forwarded from my SIP provider that is coming in from PSTN, it doesnt work |
| 15:37.46 | SeTTleR | hi |
| 15:38.16 | telnettech | the m option is RTP/AVP 0 18 and the Asterisk PBX accepts that and doesnt try to negotiate to RTP/AVP 101, which would signify Out of Band |
| 15:38.34 | pigpen | so your sip provider forwards to pstn? So what you are really asking is how to I set RFC2833 on my xyz DAHDI device? |
| 15:39.49 | telnettech | pigpen: no i am connected to provider with a SIP trunk......when they receive a call that is not an IP device thru the PSTN, the dtmf doesnt work with IVR on Asterisk PBX |
| 15:40.08 | telnettech | If a call is forwarded from SIP provider that is an IP phone, it works fine |
| 15:40.25 | telnettech | there is no Dahdi trunks in this scenario |
| 15:42.01 | pigpen | so let me make sure I understand: |
| 15:42.43 | pigpen | pstn ---->your provider------>sip trunk------>Your Asterisk |
| 15:43.16 | telnettech | pigpen: correct |
| 15:43.54 | telnettech | also IP --------> my provider ------------------> sip trunk ---------------> my asterisk |
| 15:44.11 | telnettech | http://pastebin.com/pDKd8bPc Here is my settings for the sip trunk |
| 15:44.20 | pigpen | So in the above diagram (textgram?) dtmf is not happy when it gets "inside" your asterisk, however, in the one you posted just now, it works fine? |
| 15:44.31 | telnettech | correct |
| 15:44.44 | telnettech | if it is PSTN call using SIP trunk, it doesnt work |
| 15:46.49 | pigpen | either way it is passing your sip trunk. make sure the rfc setting as I noted above is in your trunk into to the provider. |
| 15:47.01 | pigpen | currently your dtmfmode is set to auto. |
| 15:47.07 | pigpen | hardset it. |
| 15:47.21 | pigpen | if you still have issues, call your provider and tell them to fix their end. |
| 15:47.46 | telnettech | hahaha... I am the provider.....trying work with customer that owns asterisk |
| 15:47.53 | pigpen | I have had several echo cancelers completely screw up dtmf. |
| 15:48.03 | pigpen | haha....FIX YOUR SHIT!! haha |
| 15:48.16 | pigpen | ^^^thanks for the laugh. I needed that. |
| 15:49.07 | pigpen | yeah, try adding that line in their sip.conf. As it is set now is not right. Ditch the "&rfc2833" at the end of the ulaw defiinition too. |
| 15:50.27 | telnettech | this is weird cause it was working last week but today is a different thing.........we dont change anything on the SIP message |
| 15:50.40 | telnettech | I can use an IP phone in our network and the IVR works fine |
| 15:51.05 | pigpen | are you using an echo canceler? I refuse to use them after the hell they put me through. |
| 15:51.13 | pigpen | ^^^hardware that is. software is fine. |
| 15:51.14 | telnettech | no |
| 15:51.34 | telnettech | it is Asterisk 1.4.22 as the version |
| 15:51.46 | *** join/#asterisk rayno_b (~chatzilla@41.182.12.133) |
| 15:52.15 | rayno_b | Hi everybody. I need some help. I have a trixbox CE 2.8.0.4 system with Digium B410p card. Incoming calls work perfectly, but when trying to make outgoing calls, I get: "All circuits are busy". I'm obviously missing something. Can anyone please help get me in the right direction? |
| 15:52.30 | *** join/#asterisk Tim_Toady (~moi@193.92.224.201.dsl.dyn.forthnet.gr) |
| 15:52.31 | pigpen | telnettech, yeah, dtmf issues are a pain. |
| 15:52.49 | pigpen | I provide service in a similar fashion but using iax trunks with no issues for years. |
| 15:53.50 | pigpen | rayno_b, before you get the boot, you will want to get with the people in the trixbox channel. |
| 15:54.42 | pigpen | this channel is for "un-packaged" versions of asterisk. <<< someone correct me if there is a better term |
| 15:54.51 | rayno_b | pigpen -> OOPS! Sorry, I thought it would be okay because the systems have the same underlying stuff. But thanks for pointing that out. |
| 15:55.20 | pigpen | yeah the stuff they put on top of asterisk really changes the whole shoot'n match |
| 15:55.58 | rayno_b | pigpen -> okay thanks man. |
| 15:56.07 | pigpen | if you want to learn asterisk, ditch trixbox, and start learning. this way you can really support it. |
| 15:56.59 | rayno_b | pigpen -> do you think Asterisk is a better product than Trixbox? |
| 15:58.12 | pigpen | if you don't want to go through hell yes. |
| 15:58.50 | pigpen | when you have trix, sure, you can point/click your way though setting it up, but if something goes wrong.....hold on to your hat. |
| 15:59.06 | SeTTleR | hi, i have a question: has anybody of you used the System() app the last days? I can't get it to work. I am using asterisk 1.6.2.13 and only try to execute a shell script with this command. The SYSTEMSTATUS, I get is APPERROR all the time. I changed the shellscript to a shebang line followed by exit 0 but nothing changed |
| 15:59.31 | rayno_b | Which version of Asterisk? I've tried out AsteriskNOW but had a hell of a time getting my digium b410p card to work on it. It simply didn't want to accept calls, and also did not want to make outgoing calls. |
| 15:59.41 | pigpen | SeTTleR, I use it daily on 1.6.1.13 |
| 15:59.57 | SeTTleR | everything works fine here, except this System app... |
| 16:00.03 | pigpen | primarily email line usage alerts. |
| 16:00.27 | SeTTleR | 1.6.1.13 or 1.6.2.13? :-) |
| 16:01.21 | pigpen | Sorry, I have some on 1.6.1.12 and 1.6.1.13. |
| 16:01.41 | pigpen | I have reported a bug that -REALLY- affects my operations on anything 1.6.2.+ |
| 16:02.08 | SeTTleR | I don't know what's wrong with this statement. I even see in the logs, that this dialplan statement is executed, but the script never runs and i get APPERROR. Is there a way to get more error messages? |
| 16:02.12 | pigpen | So I patiently wait for a dev to pickup my bug. |
| 16:02.19 | SeTTleR | aha? what bug? |
| 16:02.42 | pigpen | https://issues.asterisk.org/view.php?id=18105 |
| 16:02.53 | pigpen | At least I think it is a bug. |
| 16:03.15 | pigpen | But I am happy to have anyone correct me (which means either way the issue is resolved) |
| 16:03.22 | SeTTleR | i even search the bug tracker, the forums, the mailing lists, but haven't found anything.. |
| 16:03.36 | pigpen | what user is your asterisk running as? |
| 16:04.41 | SeTTleR | mmh maybe i should check the voicemail here :-) |
| 16:04.59 | SeTTleR | it was running as nobody, now it is running as root and nothing changed |
| 16:05.41 | pigpen | k. Mine is running as root as well, although it is not a good idea. Make sure your box is dedicated and secure. |
| 16:05.42 | *** join/#asterisk ferdna (~yup@cpe-24-92-114-97.elp.res.rr.com) |
| 16:05.43 | SeTTleR | it is in a chroot and I have all executables needed there. I think there might be a problem with chroot.. |
| 16:06.02 | pigpen | yeah..dunno. brings in quite a few more items to check. |
| 16:06.14 | pigpen | We run gentoo with hardened sources. |
| 16:06.15 | SeTTleR | it was only to test, if that was the problem |
| 16:07.54 | pigpen | I use the system cmd to notify me via email if a customers designated number of concurrent calls attempt to be exceeded. |
| 16:08.01 | SeTTleR | mmh i think i will start checking everything again... it's really annoying, that there are now usable error messages. SUCCESS, FAILURE and APPERROR is not sufficient, because you don't know _what_ went wrong.. |
| 16:08.30 | pigpen | yeah, try turning on some debug and check out your logs. |
| 16:08.47 | pigpen | you can also run asterisk on the console...I have seen it produce a bit more. |
| 16:08.53 | *** join/#asterisk csnook (~chris@va-76-1-132-194.dhcp.embarqhsd.net) |
| 16:09.12 | SeTTleR | hehe i tried that.. set verbose and debug levels to 24 or something.. could not see anything, except that the statement is executed and I get APPERROR |
| 16:09.57 | SeTTleR | I mean, APPERROR says, the app is executed and the return value is >0. but if I only do a exit 0 in the shell script, I even get that APPERROR |
| 16:10.30 | SeTTleR | of course, I do a dialplan reload everytime... |
| 16:11.05 | pigpen | start with something easy. |
| 16:11.09 | *** join/#asterisk rrb3942 (~rbullock@208.34.105.161) |
| 16:11.16 | SeTTleR | maybe I just have to sleep and think about that :-) |
| 16:12.03 | pigpen | exten => _NXXNXXXXXX,n,System(echo -e "ACME has ${ACMELINES} defined. There was an attempt to exceed this." | /bin/mail -s "PBX - ACME Used All IAX Channels" ${ADMINNOTIFY}) |
| 16:12.08 | pigpen | ^^^an example. |
| 16:12.26 | pigpen | corny but helpful. |
| 16:12.39 | SeTTleR | mmh ok, i'll give it a try, thx |
| 16:25.15 | SeTTleR | oh no, I think i've found the problem.. the initial implementation made use of binaries from the chroot. then for debugging purposes i changed that to a simple shell script, but the main problem was always, that there was no shell in the chroot. the system() app simply does a sh -c which obviously doesn't work without a shell |
| 16:25.19 | SeTTleR | doh |
| 16:25.41 | *** join/#asterisk frigidzephyr (~rnewton@nat/digium/x-keozipsbvfkfvcpt) |
| 16:26.27 | *** join/#asterisk angryuser (~angryuser@LPuteaux-151-42-27-99.w193-251.abo.wanadoo.fr) |
| 16:27.10 | *** join/#asterisk espiceland (~erin@nat/digium/x-zfoczztsbusudihf) |
| 16:28.09 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 16:28.35 | pigpen | heh. yeah. |
| 16:28.42 | pigpen | use vm's instead of chroots |
| 16:30.03 | *** join/#asterisk phobosd (~phobosd@icwydt.com) |
| 16:30.17 | phobosd | whats the name of the channel where i could potentially contract out a coder? :x |
| 16:30.31 | pigpen | we are in the process of moving several customers to vm's as 1.6 has the timing issues resolved. |
| 16:31.30 | *** join/#asterisk JuStIcIa_ (~justicia@190.52.236.133) |
| 16:31.42 | pigpen | phobosd, you may want to post what kinda of coding work to get a better response. |
| 16:32.30 | phobosd | i just need someone to come in and write a 'on hold' script to monitor calls that are on hold, and if they're on hold for > 5 minutes, to shoot out an email |
| 16:32.54 | phobosd | pretty simple, if there's software out there that will do that already, great, otherwise will pay for foo :) |
| 16:33.59 | SeTTleR | thx for your help pigpen |
| 16:34.31 | SeTTleR | i will fix the thing the next days and see if this really solves the problem |
| 16:34.54 | SeTTleR | cu later |
| 16:35.01 | *** join/#asterisk Alagar (~Administr@122.164.30.104) |
| 16:36.19 | *** join/#asterisk bluOxigen (~sfds@unaffiliated/bluOxigen) |
| 16:37.05 | pigpen | phobosd, you may want to look into dumping the calls into a queue rather than putting them on hold. |
| 16:37.14 | pigpen | depending on the nature of the call. |
| 16:38.58 | phobosd | right, that's what i mean, sorry |
| 16:39.33 | pigpen | there are several packages that will monitor queues. |
| 16:39.35 | pigpen | live. |
| 16:39.40 | pigpen | and produce reporting |
| 16:39.42 | phobosd | right, but will they notify? |
| 16:39.47 | phobosd | i need instant-notification sort of thing |
| 16:39.53 | pigpen | dunno... |
| 16:40.44 | phobosd | right now we use 'monast', which does live monitoring |
| 16:45.12 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 16:45.12 | *** mode/#asterisk [+o malcolmd] by ChanServ |
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| 16:58.48 | citywok | i wrote my own tool to watch the queue logfile, it isn't very difficult and will provid you instant reporting if you'd like. we use it to put a HUD up in our call centers about who is on the phone, etc. |
| 16:59.19 | bougyman | we're using orderlystatsSE here. |
| 16:59.26 | bougyman | it uses AMI for the real-time |
| 17:00.46 | citywok | ah i didn't consider watching the AMI, but the logfile has all the info in it and is very easy to tail. |
| 17:02.53 | phobosd | citywok: oh yeah? mind sharing? :) |
| 17:06.25 | *** join/#asterisk timahvo1 (~rogue@41.223.57.78) |
| 17:07.33 | *** join/#asterisk lost_soul (~noymfb@cpe-67-249-141-227.twcny.res.rr.com) |
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| 17:16.16 | citywok | phobosd: i went with the descriptions of everything from the voip-info page: http://www.voip-info.org/wiki/view/Asterisk+log+queue_log |
| 17:16.26 | *** part/#asterisk mbowie (~mbowie@99-7-126-96.lightspeed.simica.sbcglobal.net) |
| 17:17.44 | *** join/#asterisk Poincare (~jefffnode@v74.ampersant.be) |
| 17:18.15 | *** join/#asterisk guax (~guax@unaffiliated/guaxinim) |
| 17:18.46 | guax | im with a problem in my asterisk, i keep getting "undefined symbol: cap_set_proc" but libcap and its dependencies are installed |
| 17:28.48 | *** join/#asterisk Benwa (~Schnitzel@unaffiliated/benwa) |
| 17:30.08 | *** join/#asterisk Fruchthoernschen (~Fruchthoe@trir-4d0ba860.pool.mediaWays.net) |
| 17:30.24 | Fruchthoernschen | Hey ho |
| 17:34.03 | Benwa | Ho hey |
| 17:34.25 | thehar | Katty: i showed up for dinner but you gave me the wrong address! |
| 17:34.26 | thehar | snickers |
| 17:36.20 | MuskyHusky | butterfingers |
| 17:37.43 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 17:38.07 | *** join/#asterisk Alagar (~Administr@122.164.30.104) |
| 17:39.09 | Katty | thehar: :> |
| 17:39.14 | Katty | thehar: they turned out amazing. |
| 17:39.19 | thehar | hehe |
| 17:39.22 | beek | hugs Katty |
| 17:39.44 | Katty | hugs Benwa |
| 17:39.45 | Katty | oh |
| 17:39.47 | Katty | hugs beek |
| 17:39.50 | Katty | Benwa: please disregard. |
| 17:39.56 | Benwa | :) |
| 17:40.00 | beek | Maybe s/he would like hugged! |
| 17:40.14 | Katty | it's not polite to hug strangers :< |
| 17:40.36 | Benwa | you always hug me and always says " please disregard." |
| 17:40.39 | Benwa | :( |
| 17:40.56 | Benwa | *say |
| 17:43.28 | Katty | well i suppose we could be friends. |
| 17:43.47 | carrar | Thats what BARNEY says |
| 17:43.52 | Katty | well i'm not barney. |
| 17:43.56 | Katty | hugs carrar |
| 17:44.03 | Katty | but we can't be friends if you have a contagious disease. |
| 17:44.04 | Katty | like mono. |
| 17:44.05 | carrar | barneyhuggles katty back!! |
| 17:44.48 | Katty | so Salvation Army had a donation of bread from Schnucks |
| 17:44.58 | Katty | got 3 loaves. gonna make stuffing using this recipe |
| 17:45.09 | Katty | http://www.food.com/recipe/crock-pot-stuffing-49609 <- and i'm gonna add raisins |
| 17:45.18 | Katty | and then...once that's done |
| 17:45.25 | Katty | gonna make this minus the stuffing part http://www.food.com/recipe/crock-pot-chicken-gravy-and-stuffing-3470 |
| 17:45.37 | carrar | YOU CAN'T ADD RAISINS!! |
| 17:45.40 | Katty | shred the chicken, and stir them together...sprinkle a little stuffing on top and then bake it |
| 17:45.43 | Katty | till bubbly |
| 17:45.46 | Katty | carrar: watch me. |
| 17:45.50 | carrar | WOAH |
| 17:45.55 | thehar | i have a delicious stuffing for you |
| 17:46.01 | thehar | it's a cornbread sausage stuffing |
| 17:46.02 | Katty | ....that sounds.... |
| 17:46.05 | Katty | oh |
| 17:46.08 | Katty | do tell |
| 17:46.14 | thehar | it is so moist |
| 17:46.16 | thehar | and so delicious |
| 17:46.28 | Katty | >.< |
| 17:46.37 | Katty | drags self out of the gutter |
| 17:46.43 | Katty | recipe please? (= |
| 17:46.48 | thehar | LOL |
| 17:46.50 | thehar | gross |
| 17:47.05 | Katty | sorry, bad week :< |
| 17:47.25 | thehar | http://www.williams-sonoma.com/recipe/sausage-corn-bread-and-chestnut-dressing.html |
| 17:47.53 | Katty | mmm that sounds good |
| 17:48.18 | *** join/#asterisk drudge` (tacos@unaffiliated/drudge/x-837452) |
| 17:48.32 | Katty | i wonder if the stores are selling chestnuts yet |
| 17:50.03 | thehar | get canned/bottled |
| 17:50.52 | bmoraca_work | bottled pigs' feet? |
| 17:51.34 | Katty | ^_- |
| 17:51.43 | Katty | that defeats the purpose of making homemade stuffing |
| 17:51.51 | thehar | hehe |
| 17:51.56 | thehar | not really |
| 17:54.44 | *** join/#asterisk t0n1 (~paolo@ip-62-143-224-210.unitymediagroup.de) |
| 17:56.29 | Katty | hmm, so cooking the chicken, in the crock pot, with brown gravy |
| 17:56.43 | Katty | and THEN having stuffing with that |
| 17:57.01 | Katty | apparently my other half thinks it sounds better than the cheesy soupy getup |
| 17:59.29 | *** join/#asterisk wcselby (~wcselby@pdpc/supporter/professional/wcselby) |
| 17:59.49 | wcselby | o/ |
| 18:00.04 | wcselby | anyone have any agent-paused / agent-unpaused sound files ? |
| 18:00.07 | tzanger | what cheesy goupy getup? I don't normally associate thanksgiving with cheese |
| 18:00.43 | Katty | tzanger: http://www.food.com/recipe/crock-pot-chicken-gravy-and-stuffing-3470 |
| 18:00.49 | Katty | hugs wcselby |
| 18:01.04 | wcselby | o/ Katty :) |
| 18:01.20 | Katty | how do you get chicken stock |
| 18:01.26 | Katty | if you're just using chicken breasts |
| 18:01.37 | wcselby | from the can? |
| 18:01.48 | wcselby | a can of chicken stock, I mean |
| 18:01.51 | Katty | seems like fresh would be better. |
| 18:01.56 | wcselby | i'm sure |
| 18:01.56 | Katty | but that would probably work |
| 18:04.09 | *** join/#asterisk Gibby (~gibby@204.118.10.244) |
| 18:04.17 | Katty | ahh, i see |
| 18:04.22 | Katty | they're boiling a whole chicken |
| 18:05.05 | Gibby | I am using asterisk now, I login into the server via ssh, trying to use the CLI commands, i issue asterisk -r, it connects but no further commands work |
| 18:05.11 | thehar | your'e going to MAKE your own chicken stock? |
| 18:05.16 | thehar | if you're going to do that |
| 18:05.22 | thehar | be particular about where you get your chicken |
| 18:05.25 | thehar | buy it locally from a farm |
| 18:05.31 | thehar | kill it and pluck it |
| 18:05.32 | Katty | dunno if i have that kind of energy |
| 18:05.36 | Katty | but i will for sure in november. |
| 18:05.47 | Katty | unless i do the ham in cherry dr pepper |
| 18:05.58 | Qwell | do what now? |
| 18:06.09 | thehar | i've done dr pepper ham |
| 18:06.10 | thehar | i hated it |
| 18:06.12 | Katty | Qwell: nigella lawson has a recipe for ham in coca cola |
| 18:06.18 | thehar | brown sugar honey ham = best |
| 18:06.20 | Katty | Qwell: but i do it in cherry dr pepper |
| 18:06.27 | Qwell | Katty: WTB |
| 18:06.51 | thehar | honeybakedham.com ftw |
| 18:07.24 | wcselby | my wife gets a ham every year that has some kind of honey glaze, she adds brown sugar and some kind of mustard = yummy! |
| 18:07.35 | Katty | Qwell: http://www.cookstr.com/recipes/ham-in-coca-cola |
| 18:07.39 | Katty | Qwell: it's really not hard at all |
| 18:07.40 | wcselby | i obviously don't know all the details |
| 18:07.51 | wcselby | but I do know it's yummy |
| 18:08.11 | Katty | wcselby: it's probably regular mustard and brown sugar |
| 18:08.14 | wcselby | she does a nice turkey, also. but i think that's just your basic turkey stuff, but I don't know, I usually only partake in the eating part of thanksgiving |
| 18:08.30 | thehar | Katty: for as much as you cook you should get a cooksillustrated.com account.. it's SO worth it |
| 18:09.03 | Katty | thehar: eh idk, i've tried some things from cooks illustrated |
| 18:09.10 | Katty | thehar: generally speaking, i tend to prefer taste of home more |
| 18:09.16 | thehar | they are my only source for cooking supplies |
| 18:09.18 | Katty | thehar: but it might just be the area i was raised in |
| 18:09.46 | Katty | thehar: and food.com you can sort by rating, which is extremely handy |
| 18:09.57 | Katty | thehar: it used to be recipezaar |
| 18:10.01 | thehar | i'm a food blog frequenter |
| 18:10.09 | thehar | i like that people actually cook it and show it |
| 18:10.10 | thehar | and review it |
| 18:10.14 | Katty | like mine? |
| 18:10.18 | thehar | you have one! |
| 18:10.22 | Katty | tho i don't review |
| 18:10.33 | Katty | it's just a personal reference, really |
| 18:10.59 | jdoe | why would 1.8.0-rc3 be using ~10 times as much ram as 1.6.2.9? |
| 18:11.28 | thehar | simplyrecipes is one of my favs |
| 18:14.08 | Katty | they don't have any ratings |
| 18:14.22 | Katty | :< |
| 18:14.29 | thehar | they are all good |
| 18:15.03 | WIMPy | Eat shit. Billions of flies worldwide can't be wrong. |
| 18:15.20 | Katty | gives WIMPy a cookie |
| 18:15.31 | WIMPy | What colour? |
| 18:15.50 | Katty | a snowflake sugar cookie, with orange sprinkles and it's drizzled with white icing |
| 18:16.20 | WIMPy | Doesn't sound bad. |
| 18:17.14 | Gibby | is this is cooking/baking channel? |
| 18:17.21 | thehar | lol |
| 18:17.30 | thehar | Dial(SIP/food) |
| 18:17.52 | Katty | yes. anymore silly questions. |
| 18:18.22 | WIMPy | CTCCP |
| 18:18.34 | WIMPy | Client to client cooke protocol |
| 18:18.39 | Gibby | why isn't asterisk now recognizing "w" as a wait |
| 18:18.39 | WIMPy | +i |
| 18:19.20 | carrar | goes into +i mode to eat his cookie |
| 18:19.45 | thehar | +m |
| 18:26.04 | Katty | it can be naptime now please? |
| 18:26.08 | thehar | yes |
| 18:26.11 | Gibby | not yet |
| 18:26.12 | *** join/#asterisk bluOxigen (~sfds@unaffiliated/bluOxigen) |
| 18:26.34 | tzanger | I had a jimmyjohn's cookie for lunch today |
| 18:26.39 | tzanger | well for dessert after my #17 sub |
| 18:26.46 | Katty | what kind of cookie |
| 18:26.49 | thehar | mmmmJJ~! |
| 18:26.50 | tzanger | chocolate chunk |
| 18:26.57 | Katty | sounds nomable. |
| 18:26.59 | tzanger | I was going to go for the oatmeal rasin but felt like chocolate |
| 18:27.02 | tzanger | exceedingly so |
| 18:27.07 | thehar | they are so amazing |
| 18:27.09 | Katty | mm oatmeal raisin |
| 18:27.11 | Katty | i am a fan of those. |
| 18:27.16 | thehar | JJ delivers to our office -5 minutes from hitting submit on the web site |
| 18:27.34 | tzanger | me too |
| 18:27.46 | Katty | you know...i have oatmeal and raisins at home |
| 18:27.46 | tzanger | thehar: yeah I go there to pick it up becuase I go to the dog park to eat |
| 18:27.56 | tzanger | Katty: sure but you're nowhere near me, so what good does it do for me |
| 18:29.23 | Katty | i have other people to feed. |
| 18:29.25 | tzanger | you know, this is not an insurmountable problem |
| 18:29.28 | tzanger | you could mail it to me |
| 18:29.31 | Katty | i could. |
| 18:29.38 | thehar | fedex that |
| 18:29.42 | Katty | and if i packed it with a bit of bread, they wouldn't get stale. |
| 18:29.51 | tzanger | hmm |
| 18:29.56 | tzanger | I smell a lucrative business venture for you |
| 18:30.00 | Katty | has mailed cookies before. |
| 18:30.09 | thehar | mail me some of your delicious enchiladas |
| 18:30.19 | Katty | there are only 3 left |
| 18:30.22 | Katty | and we plan on having those for dinner |
| 18:30.28 | thehar | make more |
| 18:30.32 | Katty | :< |
| 18:31.03 | wcselby | i'd like a cookie mailed to me |
| 18:31.06 | wcselby | please |
| 18:32.46 | Katty | me too. |
| 18:32.46 | Gibby | i will send a cookie to whoever helps me :)~ |
| 18:33.05 | wcselby | Gibby - what's your question? |
| 18:33.07 | thehar | what do you need |
| 18:33.08 | thehar | hahahaha |
| 18:34.18 | Gibby | thought that might work :) |
| 18:34.46 | logicwrath | ~sipnat pls send my cookie to ..... |
| 18:35.04 | Gibby | I have an inbound issue, but I am working with my provider on that..... however, when I program speed dial on my phone, and add a w to the number for a wait, asterisk rejects it, even thought the phone passes it just as it is |
| 18:35.46 | wcselby | are you dialing out over a dahdi trunk or a sip trunk? |
| 18:35.51 | Gibby | sip |
| 18:36.00 | wcselby | afaik, w only works for dahdi calls |
| 18:36.25 | wcselby | i could be way off though |
| 18:36.34 | wcselby | usually when I am, p3nguin or someone jumps in and lets everyone know |
| 18:36.34 | Gibby | ahh ok, getting closer, i have tried p w , . : none work |
| 18:37.18 | *** join/#asterisk jason^ (~x@unaffiliated/jason/x-0000002) |
| 18:37.37 | jdoe | why would 1.8.0-rc3 be using ~10 times as much ram as 1.6.2.9? |
| 18:37.39 | p3nguin | You can use w when dialing SIP if you put it inside the D() option. |
| 18:38.15 | Gibby | not sure i follow you p3nguin |
| 18:38.46 | jason^ | what's a cheap way to get analog phone calls into a server |
| 18:38.55 | p3nguin | Dial(SIP/peer/phonenumber,30,D(wwwwwwwww567)) |
| 18:39.10 | jason^ | hardware wise |
| 18:39.23 | Gibby | that is not where the speed dial is, i am programing it on the phone |
| 18:39.50 | p3nguin | Dial phonenumber, wait 4.5 seconds, send DTMF 567. |
| 18:40.12 | p3nguin | I told you yesterday that you cannot insert pause from the phone when using SIP. |
| 18:40.17 | p3nguin | THe answer is no different today. |
| 18:41.02 | p3nguin | You _could_ possibly get the phone to dial the number, then wait, then dial more numbers. |
| 18:41.28 | p3nguin | I don't know if your phone is capable of that, though. |
| 18:42.14 | Gibby | i went an re-read what you told me yesterday, i mis-interpreted it |
| 18:42.49 | p3nguin | The problem is that when the phone dials a phone number using SIP, it sends the INVITE right away. |
| 18:43.18 | p3nguin | When using Dahdi or some other analog technology, it is interpreting each keypress. |
| 18:44.04 | p3nguin | So if you can make your phone dial the entire phone number, wait, then send more numbers... it could be possible to achieve what you wanted. |
| 18:44.40 | Gibby | that is what I am looking for now, so i could define it in asterisk and just have my phone dial the asterisk number? |
| 18:44.59 | *** part/#asterisk guax (~guax@unaffiliated/guaxinim) |
| 18:45.56 | p3nguin | Asterisk won't interpret the pause like you asked yesterday. It will interpret the extension you dial as the phone number. If your phone later dials more numbers, asterisk will handle the number like regular DTMF. |
| 18:46.38 | p3nguin | I think what you need on your phone will be known as two-stage dialing. |
| 18:46.51 | p3nguin | aka, send a phone number first, then dial more numbers later. |
| 18:48.08 | Gibby | ok, that might help me google better, but to go back to the Dial(Sip.... you mentioned earlier, that would be programed on the asterisk server right? |
| 18:48.15 | *** join/#asterisk baddemanax (~baddemana@host-85-27-42-254.brutele.be) |
| 18:48.16 | p3nguin | yes |
| 18:49.10 | Gibby | and that would be assigned a * code? |
| 18:49.23 | p3nguin | The way you were wanting to do it... if the phone does not send the call, then wait, then send more digits later, Asterisk will never be able to process the call. |
| 18:49.50 | p3nguin | That's a regular extension. You can make the extension match whatever you feel like dialing. |
| 18:50.21 | p3nguin | If you want to dial "1" and make it match, exten => 1,1,Dial(stuff) would be fine. |
| 18:50.39 | Gibby | ahh ok, that seems the easiest, let me try that |
| 18:50.44 | *** join/#asterisk csnook (~chris@va-76-1-132-194.dhcp.embarqhsd.net) |
| 18:52.21 | Gibby | So it would be a custom extenstoin and then put the options under device options right? |
| 18:54.32 | p3nguin | I have no idea what that means. |
| 18:55.02 | Gibby | in the freepbx web interface |
| 18:55.30 | p3nguin | ~freepbx |
| 18:55.30 | 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 |
| 18:56.08 | p3nguin | Any information I have given you has absolutely nothing to do with FreePBX. |
| 18:56.30 | Gibby | hmm but all i did was install asterisknow |
| 18:56.54 | p3nguin | I've provided you with "vanilla" asterisk information. |
| 18:57.03 | Gibby | now i am more confused |
| 18:57.29 | telnettech | good afternoon all....working on DTMF issue and have a question......If I have disallow=all and allow=ulaw with dtmfmode=auto, will the disallow/allow parameters cause issues with the DTMF? |
| 18:57.39 | telnettech | this is for a SIP trunk |
| 19:00.06 | *** join/#asterisk ccesario (~ccesario@189-29-53-173-ac.cpe.vivax.com.br) |
| 19:01.30 | wcselby | Gibby - asterisknow comes with both freepbx and asterisknow gui's, however, those sit on top of "vanilla" asterisk. there are specific channels on irc for freepbx and asterisk-gui |
| 19:01.34 | *** join/#asterisk doolittlework (doolittlew@41-134-22-11.dsl.mweb.co.za) |
| 19:03.22 | doolittlework | hi there i am using asterisk application mysql on one of my servers to do abount 15000 queries per day has, today if has crashed still trying to figure out why, has anyone here used mysql app with great results? |
| 19:07.15 | *** join/#asterisk oDesk (~f@77.30.231.214) |
| 19:07.40 | *** join/#asterisk ManxPower (~manxpower@user-24-214-153-32.knology.net) |
| 19:07.45 | oDesk | hello, i want to set extension that run bash script |
| 19:08.03 | ManxPower | oDesk, Are you running FreePBX or Trixbox? |
| 19:08.09 | oDesk | FreePBX |
| 19:08.16 | ManxPower | can't help you then |
| 19:08.40 | oDesk | i can set the extension manually into extensions_custom.conf |
| 19:09.00 | ManxPower | exten => 667,1,System(/path/to/your/script.sh) |
| 19:09.15 | oDesk | ManxPower: but i don't know what to write, oh that looks what i needed |
| 19:09.56 | oDesk | ManxPower: would the file be in Root group or asterisk one ? |
| 19:10.05 | ManxPower | "core show applications" and "core show application system" is your friend |
| 19:10.14 | ManxPower | it will run as whatever user Asterisk is running as |
| 19:10.27 | oDesk | ManxPower: great |
| 19:10.38 | oDesk | ManxPower: thank you very much |
| 19:11.42 | oDesk | ManxPower: the script basically i wrote to restart my Internet Router to refresh the internet if it went bad and i'm away for any reason |
| 19:14.13 | oDesk | ManxPower: ok i'm going to lose connection to try it, thank you and have a good night |
| 19:14.24 | *** join/#asterisk Hband (~Hband@178.sub-97-212-79.myvzw.com) |
| 19:33.29 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 19:34.31 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 19:38.32 | Gibby | wcselby: thanks cleared that up, i haven't been able to access the asterisnow gui tho |
| 19:38.55 | wcselby | Gibby - it should be an installation choice, if you installed AN1.7 |
| 19:39.19 | Gibby | ahh ok |
| 19:39.28 | Gibby | i let it accept the default |
| 19:46.28 | p3nguin | I thought you had to select one: Asterisk-GUI or FreePBX |
| 19:46.40 | p3nguin | I didn't know there was a default. |
| 19:46.43 | wcselby | p3nguin - you can also select no GUI |
| 19:46.48 | wcselby | i did not know there was a default |
| 19:46.53 | p3nguin | Right, but I don't know there is a default. |
| 19:47.02 | wcselby | i haven't used it, just heard about all the new features and stuff |
| 19:47.49 | *** join/#asterisk [cannibalera] (~cannibale@201-41-194-22.fnsce703.dsl.brasiltelecom.net.br) |
| 19:47.50 | p3nguin | AsteriskNOW is a nice system, but I'm not really a fan of any GUI. |
| 19:47.53 | Gibby | it popped up to choose but had a timeout and that it chose the default |
| 19:48.19 | p3nguin | I see. |
| 19:51.29 | Gibby | p3nguin, what file would i put Dial(SIP/peer/phonenumber,30,D(wwwwwwwww567)) in? |
| 19:51.47 | carrar | <PROTECTED> |
| 19:52.00 | p3nguin | extensions.conf |
| 19:52.36 | carrar | ~book |
| 19:52.36 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 19:52.42 | jdoe | whoa |
| 19:52.43 | jdoe | haha |
| 19:53.23 | Gibby | that is what I thought p3nguin, thanks |
| 19:54.53 | *** join/#asterisk fullstop (~fullstop@static-173-210-91-3.ngn.onecommunications.net) |
| 19:55.12 | fullstop | Hi all. Is there a way to record a new voice mail message and have it recorded to a folder other than INBOX ? |
| 19:56.36 | doolittlework | hi guys i am stuck on cdr i want to add some more fields in the database came accross this link http://www.voip-info.org/wiki/view/Asterisk+cdr+mysql the extending cdr section i added a field "custom_reference" to the database and all calls end to the h extension but it does not add the data to mysql database what am i missing? |
| 19:56.54 | doolittlework | using asterisk v 1.6 |
| 19:57.01 | fullstop | I think you are missing punctuation. |
| 19:57.55 | doolittlework | fullstop: refering to me? |
| 19:58.14 | fullstop | Yes. I am just poking fun at your run-on sentence. |
| 19:58.23 | *** join/#asterisk danboid (~iatn@88-104-7-86.dynamic.dsl.as9105.com) |
| 19:58.26 | telnettech | <PROTECTED> |
| 19:58.49 | telnettech | using version 1.4.22 |
| 19:59.14 | *** join/#asterisk Hband (~Hband@159.sub-97-224-96.myvzw.com) |
| 19:59.20 | beek | telnettech: I wouldn't think so. |
| 19:59.34 | doolittlework | telnettech: dtmfmode=inband |
| 19:59.35 | doolittlework | Choices are inband, rfc2833, info or auto |
| 19:59.49 | doolittlework | use rfc2833 |
| 20:00.00 | danboid | I'm based in the UK and looking for a SIP account/ provider - who to choose? |
| 20:00.09 | fullstop | telnettech: does your SIP provider send the dtmf inband or as RTP info packets? |
| 20:00.20 | telnettech | doolittle: i have auto set as dtmfmode, per my posting above ^^^^^^^^ |
| 20:00.48 | fullstop | telnettech: RFC2833 is my preference. auto should work with your settings. Are you having problems? |
| 20:00.53 | ManxPower | telnettech, set it to what you want, not auto |
| 20:01.11 | telnettech | i am the provider.......I have a customer that is having this issue....I have another Asterisk box that doesnt have the allow/disallow parameters on the sip peer and it works fine |
| 20:01.25 | telnettech | i have dtmfmode=auto and that is it |
| 20:01.50 | fullstop | ManxPower: You mentioned a few days ago that you have sip lines for ~$7 a month. Who do you have them through? |
| 20:01.53 | *** join/#asterisk anonymouz666 (~anonymouz@187-28-37-118.poolip.RJO.embratel.net.br) |
| 20:02.26 | *** join/#asterisk eugeneoden (~goden@rrcs-97-77-255-86.sw.biz.rr.com) |
| 20:02.33 | ManxPower | fullstop, no, I get them for $0/month (no DID) or $1.75/month (DID) plus usage, of course. MY usage ends up being $5 - $6 /month |
| 20:02.48 | fullstop | ManxPower: Aah.. I see. |
| 20:02.56 | *** join/#asterisk aidinb (~Aidin@unaffiliated/aidinb) |
| 20:03.01 | telnettech | Manxpower: dtmf doesnt work with any other settings as we are not the endpoint. These are calls that are passing thru our network down the sip trunk to the asterisk box which has an IVR and it doesnt work |
| 20:03.01 | ManxPower | Vitelity.net |
| 20:03.11 | fullstop | My daughter would crush the $5-$6 just by calling Grandma. |
| 20:03.28 | ManxPower | telnettech, I wish you the best of luck. |
| 20:04.01 | doolittlework | has any have any luck adding extra fields in the cdr-database with success please help me i am desperate |
| 20:04.03 | danboid | Otherwise, what the best site to use to pick a SIP provider? |
| 20:04.04 | telnettech | Manxpower: Im just trying to help this customer so that the next 1 that comes along and says that it is our problem, i can point them to the right direction |
| 20:04.21 | ManxPower | you can point them to using RFC2833 DTMF |
| 20:04.32 | *** join/#asterisk KavanS (~KavanS@unaffiliated/kavans) |
| 20:04.57 | telnettech | We have about 20 SIP trunks setup so far and this is the only 1 that doesnt work? He says that he has tried 2833 and that didnt work....that is why he called and opened a ticket with us |
| 20:05.04 | ManxPower | he lied. |
| 20:05.14 | ManxPower | Well, I don't know that since you have audo |
| 20:05.15 | ManxPower | auto |
| 20:05.37 | oDesk | ManxPower: i'm getting this error Auto fallthrough after this line Executing [s@custom-restart-router:1] System("SIP/100-00000192", "/root/sh.sh") in new stack |
| 20:05.49 | ManxPower | oDesk, Good! |
| 20:06.15 | fullstop | telnettech: I would run a trace with wireshark and see what is going on. |
| 20:06.37 | telnettech | I have a box with same version and we have it set to auto cause that seemed to work best with our PSTN gateway |
| 20:06.39 | oDesk | ManxPower: Auto fallthrough and i don't see effect for the executed script |
| 20:07.00 | fullstop | oDesk: /root/sh.sh has execute bit set? |
| 20:07.20 | telnettech | fullstop: I would like to but I cant seem to get him to do 1 on his side at the same time. |
| 20:07.27 | danboid | Don't tell me you guys don't know the answer. If I'm being OT asking about SIP providers, what channel shoud I ask in? |
| 20:07.36 | ManxPower | oDesk, what do you have set as the next priority for that extension? |
| 20:07.42 | citywok | ~itsp |
| 20:07.42 | infobot | [~itsp] An ITSP is an Internet Telephony Service Provider (or VoIP telephone company). They allow you to either SEND calls to the PSTN (this is called termination), RECEIVE calls from the PSTN (called origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs. |
| 20:07.47 | oDesk | fullstop: yes, it's just echo text.. and i've tried another script that i know working to restart the router but it doesn't work |
| 20:08.03 | fullstop | danboid: you are asking for the UK.. it is possible that there is no one else here from the UK. |
| 20:08.44 | oDesk | ManxPower: i've just set custom destination, then i put the line you wrote into custom context into extensions_custom.conf |
| 20:09.02 | ManxPower | one of our customers had such terible service from every UK SIP provider they tried that the actually purchased a PRI line from London to NYC to connect to SIP service |
| 20:09.24 | ManxPower | oDesk, so the call falls off the end of the dialplan (aka auto fallthru) |
| 20:09.40 | danboid | fullstop, its possible, yes. I still thought there may be an online sip provider resource internationally regarded as being the best for comparisons? |
| 20:10.06 | danboid | fullstop, Although I do think it unlikely I'm the only Brit in here |
| 20:10.40 | fullstop | danboid: I'm not aware of any.. but maybe a fellow Brit will check in. |
| 20:11.00 | danboid | One question you will be able to answer though- whats the best (FOSS, pref.) SIP client for Windows? |
| 20:11.32 | Gibby | p3nguin, would it just be 1 exten line? |
| 20:11.35 | ManxPower | I don't think many people here that are serious about VoIP use software clients. |
| 20:11.54 | KavanS | I'm pretty damned serious about VoIP |
| 20:12.11 | Katty | i am serious about cookis. |
| 20:12.16 | Katty | and soon soft house booties. |
| 20:12.17 | ManxPower | KavanS, how have those softphones been working out for you? |
| 20:12.30 | danboid | I would expect most people here are pretty serios about VoIP |
| 20:12.43 | KavanS | ManxPower, only use 1 soft phone actually it runs on the iphone...works pretty well actually |
| 20:12.59 | KavanS | I wasn't contradicting your statement, I don't think running soft phones would be a successful solution |
| 20:13.09 | ManxPower | KavanS, Try it with 50, all on windows machines |
| 20:13.22 | danboid | and can tell m,e what the best clients are for Linux and Win. I was going to try Empathy under Lin first just because it comes with buntu- no? |
| 20:13.37 | KavanS | danboid, linux use ekiga - that is if you can get sound working |
| 20:13.45 | oDesk | ManxPower: i'm only having 1 line into the context, that is s,1,System(/root/sh.sh) .. what should be the end of the dialplan to avoid fallthru ? |
| 20:14.04 | ManxPower | danboid, from reports I've seen on this channel, all softphones suck. Xten/eyeBeam/BRIA or whatever they are calling it this week seems to be the most popular |
| 20:14.11 | ManxPower | oDesk, add Hangup as the next priority |
| 20:14.24 | ManxPower | then the call won't fall off the edge of the dialplan |
| 20:14.38 | ManxPower | Since you are running freepbx nothing is sure. |
| 20:15.22 | Katty | well, 1 thing is for sure. |
| 20:15.30 | Katty | you're going to be tugging some of your hair out before it's all said and done. |
| 20:17.34 | thehar | came in at the wrnog time |
| 20:17.39 | thehar | TUG IT |
| 20:17.40 | thehar | pull it out! |
| 20:19.16 | danboid | KavanS, Well I lied :) I did try ekiga first actually but it gave me some connection error. I failed to find a solution within 15m so I scrapped that and Empathy is next |
| 20:19.28 | danboid | ManxPower, X-Lite seems to be there free one now |
| 20:19.47 | danboid | s/there/their - sorry grammar cops |
| 20:20.01 | KavanS | yeah ekiga/softphones in general - suck |
| 20:21.11 | danboid | I always support open standards where poss and I'm sick of hearing about skype so I was hoping someone here might point me towards to best (as good as softphones go) alternative |
| 20:21.17 | Katty | thehar: i have a knot in my shoulder :< |
| 20:21.22 | thehar | ohnoes |
| 20:21.26 | thehar | i have knots everywhere |
| 20:21.30 | thehar | i haven't had a massage in 2 months! |
| 20:22.53 | danboid | Whats the consensus on skype in here? Is that as good as softphones get? Can SIP softphones not compete? |
| 20:23.06 | *** join/#asterisk Nwab (~Schnitzel@unaffiliated/benwa) |
| 20:23.41 | thehar | blink and x-lite are pretty good |
| 20:24.43 | Katty | thehar: unfortunate :< |
| 20:26.11 | ManxPower | danboid, people that use skype don't generally come to this channel. |
| 20:27.54 | danboid | ManxPower, Well I don't use skype and I wanted to talk to people who know what they're on about with FOSS voip so this seemed as good a channel as any |
| 20:28.01 | *** join/#asterisk Poincare (~jefffnode@v74.ampersant.be) |
| 20:28.34 | riscphree | I had the unfortunate job of integrating skype with an asterisk box |
| 20:28.39 | riscphree | didn't care for it |
| 20:28.45 | *** join/#asterisk csnook (~chris@va-76-1-132-194.dhcp.embarqhsd.net) |
| 20:28.53 | thehar | SkypeForAsterisk |
| 20:28.56 | *** part/#asterisk [cannibalera] (~cannibale@201-41-194-22.fnsce703.dsl.brasiltelecom.net.br) |
| 20:28.58 | riscphree | ya |
| 20:29.20 | danboid | ManxPower, So are you not a Manx resident then? |
| 20:29.28 | ManxPower | riscphree, did you get help for it on this channel? |
| 20:29.37 | riscphree | nope |
| 20:29.50 | ManxPower | danboid, no, I do not live in or near the Isle of Man |
| 20:30.15 | danboid | ManxPower, You're a biker then? |
| 20:30.24 | oDesk | ManxPower: i still can see this Executing [s@custom-restart-router:1] System("SIP/102-00000196", "/root/sh.sh") in new stack , but the script not actually executed |
| 20:30.45 | ManxPower | If you take the name Homer Simpson used when he changed his name (to "Max Power") and change the Max to Manx (a breed of cat) then you get my nick |
| 20:31.16 | ManxPower | oDesk, what user does Asterisk run as on your system? |
| 20:31.39 | oDesk | ManxPower: root, even the script i chown to asterisk |
| 20:31.51 | ANurmi | You didn't get it off of a hair dryer? |
| 20:32.09 | ManxPower | oDesk, is this a production server? |
| 20:32.22 | oDesk | ManxPower: personal one |
| 20:33.31 | ManxPower | oDesk, stop asterisk. Start it as "asterisk -cvvvddd" and see if you get any better debugging into on the asterisk console |
| 20:33.38 | ManxPower | (when you try the call) |
| 20:34.55 | ManxPower | oDesk, starting asterisk with -c (run in foreground) you will see and output the script sends to STDERR. |
| 20:34.55 | oDesk | ManxPower: i think you missed the r ? |
| 20:35.06 | oDesk | ManxPower: oh ok |
| 20:35.09 | citywok | odesk no, d is daemon mode |
| 20:35.15 | ManxPower | oDesk, no. "r" means "connect to existing asterisk process) |
| 20:35.19 | ManxPower | no, "d" is debug! |
| 20:35.33 | citywok | errr, lolol, duh. just typing asterisk = daemon mode. soryr. |
| 20:36.08 | citywok | sorry, fighting another outage here and in bmc remedy land |
| 20:36.30 | Gibby | we have had tons of bmc remedy issues lately |
| 20:36.51 | citywok | yea... 3 hours ago ours died for no reason middle of the day. we have them on the phone trying to fix it for 2 hours now. |
| 20:37.00 | citywok | we just sent the majority of our agents home for the day. |
| 20:39.19 | *** join/#asterisk eugeneoden (~goden@rrcs-97-77-255-86.sw.biz.rr.com) |
| 20:41.34 | *** join/#asterisk oDesk (~f@2.89.12.98) |
| 20:41.41 | oDesk | ManxPower: back |
| 20:41.56 | oDesk | ManxPower: it works into the daemon mode |
| 20:42.21 | oDesk | ManxPower: so what could be the problem ? |
| 20:45.01 | *** part/#asterisk oDesk (~f@2.89.12.98) |
| 20:45.11 | *** join/#asterisk oDesk (~f@2.89.12.98) |
| 20:45.14 | *** part/#asterisk danboid (~iatn@88-104-7-86.dynamic.dsl.as9105.com) |
| 20:47.29 | oDesk | ManxPower: can you see my typing ? |
| 20:47.54 | jdoe | no, what does it say? |
| 20:47.56 | Letoric | oDesk: you weren't typing in here until just now, after you rejoined a couple minutes ago |
| 20:48.17 | oDesk | ManxPower: Oh,sorry.. the script works into the daemon mode |
| 20:49.17 | oDesk | ManxPower: but after i exit from daemon mode, i've tried amportal start and again it doesn't work.. so what could be the problem ? |
| 20:52.06 | oDesk | i hope my previous typing is already shown ? |
| 20:53.00 | Letoric | oDesk: connect using asterisk -r |
| 20:53.09 | Letoric | oDesk: then set debug to 3 and verbose 3 |
| 20:53.16 | Letoric | oDesk: watch the call, see what it says |
| 20:54.25 | telnettech | check out time....everyone have a good night!!!!!!! |
| 20:55.05 | oDesk | Letoric: Executing [s@custom-restart-router:1] System("SIP/102-00000001", "/root/telnetscript.sh") in new stack .. this is the line i'm getting |
| 20:56.34 | Letoric | oDesk: Ok, so it looks like it's calling the script? |
| 20:56.37 | oDesk | Letoric: I haven't changed anything, so it doesn't work in the production mode , but works into the daemon mode ? strange |
| 20:57.00 | Letoric | oDesk: daemon mode IS production mode, to my understanding. Are you referring to console mode perhaps? |
| 20:57.09 | oDesk | Letoric: yes, but i'll strongly feel the call because the script restart my internet connection |
| 20:57.29 | oDesk | Letoric: i'm referring to amportal start mode |
| 20:57.58 | Letoric | oDesk: That must be one of the non vanilla installation options, I'm not familiar with that |
| 20:58.33 | oDesk | Letoric: um.. |
| 20:58.36 | Letoric | oDesk: Does your script have debugging? That's where I would look |
| 20:58.56 | doolittlework | persistent me got it working |
| 20:58.59 | Letoric | oDesk: It sounds as though Asterisk is launching it, and that's the extent of what you're going to see from Asterisk point of view |
| 20:59.00 | doolittlework | horay |
| 20:59.25 | Katty | thehar: i have one less knot :> |
| 20:59.28 | Letoric | oDesk: if you pastebin your script, I can take a look to see if I can help. I'm no expert, but I've learned a lot from others in here. |
| 20:59.39 | oDesk | Letoric: yes i've done sendmail debug, it doesn't reach .. but if i ran the script directly it sends the mail |
| 20:59.43 | thehar | yay! |
| 20:59.53 | Letoric | oDesk: are you using sudo in the script at all? |
| 21:00.25 | Letoric | oDesk: one scripting issue I ran into was forgetting that when using sudo, you have to give it a switch to tell it it's being called in script, not manually. |
| 21:00.33 | oDesk | Letoric: no i'm using expect |
| 21:00.55 | oDesk | Letoric: i'll pastebin the code |
| 21:00.59 | Katty | thehar: nevermind, just found a new one :/ |
| 21:01.10 | thehar | ohnoes Katty |
| 21:01.18 | Katty | lower back this time >.< |
| 21:01.19 | Letoric | oDesk: sounds good. Is Asterisk running as root, asterisk, or some other user? |
| 21:02.25 | doolittlework | exten => _.,n,GotoIf($[${call_credit} < 2]?nocredit,1) if callcredit is bigger that 2 will this call just continue? |
| 21:02.30 | oDesk | Letoric: http://pastebin.com/8h83gR8Y yes asterisk running as root |
| 21:03.17 | Katty | thehar: 3 in my lower back |
| 21:03.30 | thehar | Katty: we are not loved |
| 21:03.34 | thehar | who will massage us |
| 21:03.40 | Katty | actually i am loved. |
| 21:03.42 | Katty | and massaged. |
| 21:03.44 | thehar | me too :4 |
| 21:03.45 | thehar | haha |
| 21:03.46 | Katty | but apparently not enough. |
| 21:03.50 | thehar | indeed |
| 21:03.51 | JamesHarrison | Would it be possible to set up asterisk so that you could call an extension and then have a MP3 file play on the extension being called rather than connecting a call through? |
| 21:03.52 | thehar | neglect! |
| 21:03.58 | Katty | i agree. |
| 21:04.02 | Katty | let's file a complaint. |
| 21:04.09 | thehar | files a complaint to the BOARD |
| 21:04.12 | thehar | oh yes |
| 21:04.13 | Letoric | JamesHarrison: yes. |
| 21:04.13 | thehar | the board |
| 21:04.20 | p3nguin | gibby: Unless you need to do other things, one line is enough. |
| 21:04.32 | JamesHarrison | Letoric: Okay, thanks. Just checking feasability quickly :) cheers. |
| 21:04.33 | Katty | thehar: is your back as speed bumpy as mine? |
| 21:04.34 | Letoric | oDesk: Add debugging to your script, I'm fairly certain the challenge is in the scripting |
| 21:04.40 | thehar | yes |
| 21:05.48 | Letoric | oDesk: Wish I could be more help, but I'm past what I can offer - if you change the script to something simple, like writing to a text file, you'll see that Asterisk is executing it ok, it's just the actions/shell that you are working with that are presenting challenges |
| 21:06.04 | oDesk | Letoric: it works, i've other script that sends mail, and it doesn't work too, but calling it directly will run it correctly, here is the other script |
| 21:06.17 | Gibby | so exten => 286,1,Dial(SIP/trunkname/phonenumber,30,D(wwwwwwwww567))2ndnumbersequense? |
| 21:06.31 | p3nguin | gibby: No. |
| 21:06.37 | oDesk | Letoric: writing to file will be good debugging option |
| 21:06.40 | Letoric | oDesk: I understand it works when you execute it manually, but does it work if you have the system call it? |
| 21:06.42 | oDesk | Letoric: i'll try it |
| 21:06.44 | Katty | thehar: so i have one knot on the back right side, that makes my right calf muscle twitch when i rub it >.< |
| 21:06.47 | Katty | thehar: what's up with that |
| 21:06.53 | p3nguin | (1338.54) <p3nguin> Dial(SIP/peer/phonenumber,30,D(wwwwwwwww567)) |
| 21:06.55 | p3nguin | (1339.44) <p3nguin> Dial phonenumber, wait 4.5 seconds, send DTMF 567. |
| 21:07.09 | Letoric | oDesk: set a crontab to run your script, have it run and see if it works ok, that will give you a decent idea if the script works in batch mode, vs manually executed mode |
| 21:07.13 | oDesk | Letoric: yes it does work when i was running asterisk on -cvvvddd |
| 21:07.18 | thehar | Katty: lol it's all connected |
| 21:07.26 | Letoric | oDesk: that's console mode, not daemon mode ;) |
| 21:07.47 | Letoric | oDesk: c is console, vvv is verbosity 3, ddd is debug 3 |
| 21:07.48 | Katty | thehar: oh, well that's good to know. |
| 21:07.55 | Katty | thehar: i thought my internal wiring got snickerdoodled up |
| 21:08.01 | oDesk | Letoric: i'll try to write to file now |
| 21:08.09 | oDesk | Letoric: will let you know |
| 21:08.13 | thehar | Katty: mmm i should make snickerdoodles tonight |
| 21:08.29 | Katty | thehar: but then you'd just eat them |
| 21:08.31 | thehar | I made the most devine nutty chocolate chip zucchini bread this weekend. |
| 21:08.35 | thehar | oh god it was so good |
| 21:08.58 | Gibby | p3nguin, I don't have to put exten in front of it? |
| 21:09.11 | p3nguin | gibby: Yes, you do. |
| 21:09.20 | Katty | did you save me any |
| 21:09.27 | thehar | the bf took the 2nd loaf to work |
| 21:09.43 | p3nguin | gibby: I was giving you the command part of the exten. |
| 21:09.57 | Gibby | ok, so where does the 2nd number sequence go then? |
| 21:10.22 | *** join/#asterisk [cannibalera] (~cannibale@201-41-194-22.fnsce703.dsl.brasiltelecom.net.br) |
| 21:10.44 | p3nguin | I shouldn't need to keep repeating the example and the description of what it does. |
| 21:11.13 | Gibby | you never said where the 2nd sequence goes |
| 21:11.18 | p3nguin | Dial(SIP/peer/phonenumber,30,D(wwwwwwwww567)) <--- THIS means THIS ---> Dial phonenumber, wait 4.5 seconds, send DTMF 567. |
| 21:11.28 | p3nguin | I've told you THREE TIMES now. |
| 21:12.17 | thehar | giggles |
| 21:13.42 | oDesk | Letoric: http://pastebin.com/y6U10KRG |
| 21:14.32 | oDesk | Letoric: this should write /root/sh.log with hello into it |
| 21:14.40 | oDesk | Letoric: but that doesn't happen |
| 21:14.48 | citywok | p3nguin, stop responding. that's what i do when people keep asking the same questino over and over. |
| 21:15.23 | citywok | what's the most popular windows IRC client? I dislike xchat&mirc b/c they aren't free, and Bersirc doesn't do tab completion or let me ctrl-delete full words. |
| 21:15.35 | *** join/#asterisk darksk1ez (~mhb@darkskiez.ipv6.darkskiez.co.uk) |
| 21:15.40 | p3nguin | kvirc |
| 21:15.49 | Letoric | oDesk: add a step to your dial plan, before hangup |
| 21:16.10 | oDesk | Letoric: delay for ex. ? |
| 21:16.11 | *** part/#asterisk exothermc (~miles@74.85.89.146) |
| 21:16.21 | Letoric | oDesk: set some random variable to the value of ${SYSTEMSTATUS} |
| 21:16.29 | Letoric | see what the call reports when you run it |
| 21:16.34 | Letoric | should show failure or success |
| 21:17.07 | citywok | p3nguin thanks, does it do macros too? /msg nickserv identify, /join #asterisk ? |
| 21:17.09 | fullstop | citywok: there is a free win32 xchat... not the one which expires.. |
| 21:17.12 | Letoric | oDesk: I still think it's your script, based on what you showed from the dialplan earlier |
| 21:17.18 | fullstop | someone builds from source. |
| 21:17.55 | citywok | oh, yea i see that xchat 2 actually |
| 21:18.22 | *** join/#asterisk darksk1ez (~mhb@darkskiez.ipv6.darkskiez.co.uk) |
| 21:18.29 | Letoric | also, I would suggest keeping it simple and having system only call the script/pass variables to it, then let the script do the work |
| 21:18.59 | Letoric | oDesk: instead of calling the shell in the system command (not sure it matters, but as I said, I'm not an expert, can only pass what I've learned) |
| 21:19.10 | fullstop | are there any known bugs with specifying the emailsubject in voicemail.conf? It doesn't appear to be taking, even while the other options are. |
| 21:19.11 | oDesk | Letoric: then how can you explain it when it works into the console mode ? |
| 21:19.17 | p3nguin | citywok: I don't know about macros, but it does all of the standard IRC client commands. It's not a broken implementation of IRC. |
| 21:20.03 | Letoric | oDesk: console mode is running vanilla asterisk. As mentioned earlier, I don't recognize what you are running as far as the aaportal or whatever it was. I use plain vanilla asterisk |
| 21:20.21 | oDesk | Letoric: the systemstatus echos this Executing [s@custom-restart-router:2] NoOp("SIP/102-00000004", "APPERROR") in new stack |
| 21:20.22 | p3nguin | It may or may not support /nickserv, but you could always create an alias for /quote nickserv if it doesn't. |
| 21:20.52 | *** join/#asterisk citywok (~kvirc@67-134-194-33.dia.static.qwest.net) |
| 21:20.57 | oDesk | Letoric: i'm on 1.6 |
| 21:21.00 | oDesk | Letoric: i'm on 1.6.2 |
| 21:21.25 | *** join/#asterisk eugeneoden (~goden@rrcs-97-77-255-86.sw.biz.rr.com) |
| 21:21.35 | Letoric | oDesk: me too, but I don't have the portal you speak of. I either launch asterisk service, or asterisk console. |
| 21:22.05 | Letoric | oDesk: maybe p3nguin can help - he's helped me with a lot of the scripting |
| 21:22.06 | oDesk | Letoric: will try that into the console mode and see what SYSTEMSTATUS will have for me |
| 21:22.33 | oDesk | Letoric: thank you very much for your time, i appreciate it |
| 21:23.39 | Letoric | oDesk: Sure. Sorry I haven't been very helpful ;( |
| 21:23.42 | oDesk | now on console mode Executing [s@custom-restart-router:2] NoOp("SIP/102-00000000", "SUCCESS") in new stack |
| 21:24.01 | Letoric | oDesk: ok, start asterisk as a service. None of this portal stuff you speak of! |
| 21:24.09 | Letoric | oDesk: just simple old service asterisk start |
| 21:24.20 | Letoric | then do asterisk -r to connect and watch it go |
| 21:24.33 | oDesk | Letoric: that will take care of dahdi and other services too ? |
| 21:25.02 | Letoric | oDesk: I don't use dahdi, you may have to start that separately, but I thought it was just a module |
| 21:25.19 | Letoric | dahdi became unnecessary for timing with asterisk 1.6.2 |
| 21:25.59 | oDesk | Letoric: great, i'll have that setup |
| 21:30.52 | Katty | thehar: i found the knot making my left arm hurt :> |
| 21:31.49 | thehar | did you kill it? |
| 21:32.05 | Qwell | untie it? |
| 21:32.20 | Katty | i'm working on it |
| 21:32.23 | Katty | tho i'm starting to worry it's not a knot |
| 21:33.19 | Katty | but Qwell knows how that goes |
| 21:37.43 | Qwell | I know nothing! |
| 21:39.05 | Letoric | Qwell: nobody questions that ;P |
| 21:39.22 | Letoric | Qwell: joking, of course ;) |
| 21:39.30 | *** join/#asterisk alascap (~mjcalaska@170-157-165-209.nac.static.gci.net) |
| 21:39.39 | Katty | Qwell: i'm sure you can guess where i'll end up if i dont' think it's a knot |
| 21:40.50 | *** join/#asterisk cjk (~cjk@vodsl-11065.vo.lu) |
| 21:41.07 | Katty | Qwell: luckily it seems to be going away |
| 21:41.26 | oDesk | Letoric: lol, just to let you know ... i've descovered it |
| 21:41.43 | *** join/#asterisk csnook (~chris@138.210.3.1) |
| 21:41.45 | oDesk | Letoric: only butting & at the end of the command make it run |
| 21:42.35 | cjk | hi, i am testing the latest asterisk 1.8 version and I notice that quite a lot of open udp sockets between and 4000 and 5000. there are no calls, so its not RTP traffic. Any idea why asterisk opens those ports? |
| 21:43.37 | oDesk | ManxPower: so putting & at end of the command fixed the Error, just FYI |
| 21:43.56 | Letoric | oDesk: Nice ;) |
| 21:44.44 | *** join/#asterisk Hband (~Hband@ip68-2-140-46.ph.ph.cox.net) |
| 21:47.45 | Gibby | p3nguin, i got it mostly working, with peer after SIP/ i would only get a busy signal, put my outgoing trunk and it dials, just just send the 2nd number sequence, read somewhere shouldn't use wait since that makes it wait for user input |
| 21:57.17 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
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