| 00:06.50 | *** join/#asterisk TehRabbitt (~TehRabbit@pool-71-172-89-155.nwrknj.fios.verizon.net) |
| 00:08.02 | TehRabbitt | p3nguin: hey man |
| 00:08.40 | TehRabbitt | still having that weird context issue with chan_sccp_b.... upgraded both asterisk and chan_sccp_b to the latest versions and still I can't seem to dial out from the sccp phones |
| 00:09.10 | E-bola | Hmm weird. If i try to dial out via my SIP provider i get no warnings, if i try to dial out via my IAX provider i get a warning that AES encryption is disabled... |
| 00:09.21 | EmleyMoor | Intermittently, IAX calls are not (or, more correctly *are*) getting through |
| 00:11.19 | *** join/#asterisk [netman] (~netman@59.Red-83-54-216.dynamicIP.rima-tde.net) |
| 00:15.11 | drmessano | EmleyMoor, which 1.6 branch and which release? |
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| 00:25.02 | paulc | I want NOANSWER to be a valid thing I can stick in GOSUB_RESULT when calling Dial. Even though I don't know shit about C etc, I hacked it up in app_dial.c - what's the best way for me to get that back into production code - Mantis ticket + diff + disclaimer? |
| 00:25.55 | EmleyMoor | 1.6.2.9-2, present in Debian squeeze |
| 00:26.39 | EmleyMoor | Have temporarily switched to SIP at provider end for now - not sure where IAX fault lies |
| 00:28.33 | EmleyMoor | Will resume this in the morning - if it is a provider fault maybe they will have fixed it by then |
| 00:40.31 | weta | Hoping someone might be willing to look at my config for a TDM400P & analog phone and point me in the right direction. Thanks in advance. http://pastebin.com/VzMTDMth Incoming calls from POTS work fine. Outgoing/Analog to SIP don't work at the moment. |
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| 00:48.13 | ChannelZ | your dahdi config is a little messed up it looks like |
| 00:48.43 | ChannelZ | Is that your WHOLE chan_dahdi.conf or did you just paste in a couple of lines? |
| 00:50.13 | weta | no just a few lines. I took the one produced by default and modified it very slightly. |
| 00:50.27 | weta | I can post the whole thing in a separate pastebin |
| 00:50.37 | ChannelZ | Yes |
| 00:51.37 | ChannelZ | chan_dahdi.conf is the only one used. dahdi-channels.conf is generated by one of the config tools but doesn't actually do anything unless you specifically #include it |
| 00:54.57 | weta | Here it is: http://pastebin.com/FPdZe2HX |
| 00:55.07 | weta | Thanks again. I really appreciate the help. |
| 00:56.02 | ChannelZ | let's just scrap this and make one from scratch |
| 00:56.20 | weta | OK, I'm not surprised that's a good idea. |
| 00:57.21 | weta | I have /etc/asterisk in git so quite comfortable doing what ever. |
| 01:03.43 | ChannelZ | .... |
| 01:04.38 | ChannelZ | Here's a minimal one. http://pastebin.com/W9xWbh7g |
| 01:06.06 | ChannelZ | The thing to note in your current config is how the config file options actually work. Essentially you were assigning channel 1 and 4 with all the same options |
| 01:06.45 | ChannelZ | and not defining the signalling on either, which apparently was defaulting to fxs_ks if your incoming line was working |
| 01:07.46 | ChannelZ | But basically in chan_dahdi.conf, when you say "channel => X" that channel/channels are assigned all of the config options that appeared above it. So the whole thing reads very sequentially. |
| 01:10.07 | ChannelZ | Now your dialplan (extensions.conf) is another matter. |
| 01:11.07 | ChannelZ | You didn't show the whole thing there either so I dunno if it's necessarily going to do what you want either |
| 01:12.06 | weta | Yes, my dialplan is a bit messy at the moment - I know I can make it cleaner. |
| 01:12.27 | ChannelZ | Generally your FXO (POTS) and FXS (handset) channels will be in different contexts so people calling in can't just start dialing internal extensions (unless you really want them to) |
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| 01:14.36 | weta | Sorry to ask a basic question. I gave the new chan_dahdi.conf a quick try. It behaved the same as before in that the analog phone doesn't get a dial tone. If I call the extension (202), and pick up about a second after, I do get a dial tone. Is this expected? |
| 01:14.45 | ChannelZ | IE channel 1 would maybe be in an 'incoming' context which is your IVR or just dials a device, goes to voicemail, whatever. Then your channel 4 or other phones are in an 'outgoing' or 'internal' context which has extensions for dialing out, to other extensions, accessing voicemail, etc. |
| 01:15.18 | weta | I see regarding incoming vs. outgoing contexts. That makes a lot of sense. |
| 01:15.23 | ChannelZ | Not seeing your entire dialplan I have no idea. Are you sure each channel is what you think it is? |
| 01:16.12 | weta | I used cat /proc/dahdi/1 with the pots trunk connected and not and was able to confirm channel 1. |
| 01:16.31 | weta | From there, it was easy since I know my card has two fxs and two fxo's. |
| 01:17.25 | weta | I'm not expecting they'd be physically out of sequence on the card.... 1 is on top, thus 4 would be the bottom. |
| 01:17.27 | ChannelZ | So you have 2 dual channel modules; 2 FXO and 2 FXS |
| 01:17.33 | weta | Yes that's right |
| 01:17.42 | weta | red and green as I recall |
| 01:18.19 | ChannelZ | ok. So what does your /etc/dahdi/system.conf look like? |
| 01:19.42 | weta | Here it is: http://pastebin.com/hh4ewr9F |
| 01:22.21 | ChannelZ | change it to this: http://pastebin.com/41gnpFN4 |
| 01:22.36 | ChannelZ | Then stop asterisk, and restart the dahdi drivers and start asterisk again |
| 01:25.24 | weta | Done |
| 01:27.30 | weta | Just an FYI, dahdi show channel(s) looks the same, a quick test shows it is functioning the same - you probably expected as much though. |
| 01:27.40 | ChannelZ | not really no |
| 01:28.05 | ChannelZ | do 'dahdi show channel 4' and pastebin that |
| 01:30.15 | weta | http://pastebin.com/5MU4pmaR |
| 01:30.16 | ChannelZ | brb |
| 01:31.21 | ChannelZ | dunno why you're not getting a dialtone when you pick up the phone connected to channel 4. Is the phone broke? :) You have the molex connector plugged in on the card? |
| 01:33.36 | weta | The phone should be good. The funny thing is if I dial 202, and then pick up a second later, I get a dial tone |
| 01:33.49 | weta | I'll try the other 3 ports just for giggles. |
| 01:34.14 | ChannelZ | pick what up a second later |
| 01:35.16 | weta | call 202, and I notice the line indicator stops flashing on the analog phone. Then pick up the handset or hit speaker and I get tone. |
| 01:35.50 | ChannelZ | But calling from what? |
| 01:36.08 | weta | I have a bunch of SIP phones set up for this asterisk instance. |
| 01:36.30 | weta | Just happens to be one on my desk right beside the analog |
| 01:36.54 | ChannelZ | ok |
| 01:37.11 | ChannelZ | I thought you meant you were picking up the analog phone, getting nothing, and dialing 202 and then getting a dialtone.. |
| 01:37.59 | ChannelZ | Are you still getting "Unable to enable echo cancellation on channel 4 (No such device)" on the console when you call it? |
| 01:39.26 | weta | No, that message is gone. |
| 01:39.54 | weta | I tried the other 2 ports, same behavior |
| 01:40.15 | weta | Actually, I didn't see the line indicator disappear for the other 2 |
| 01:41.18 | ChannelZ | one thing I dunno what is going on in your dialplan is it's dialing "dahdi/4/s" not sure what the /s is all about - but that shouldn't have anything to do with your dialtone not being there when you pick up the phone. |
| 01:41.51 | weta | That's interesting. That time, when I picked up the analog phone, I got a two way connection. Could speak/hear both ways. |
| 01:41.53 | ChannelZ | So when you call it and pick up the analog handset, you get a dialtone rather than bridged to the phone you're dialing from? |
| 01:42.44 | weta | When I call 202 from a SIP phone, and pick up on the analog, I get bi-directional voice. |
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| 01:42.55 | weta | Before it was dialtone. |
| 01:43.23 | ChannelZ | But when you just pickup the analog, you still get nothing? |
| 01:43.49 | weta | Yes, that's right |
| 01:44.21 | weta | If I don't pick up on the analog right away, it I lose the connection |
| 01:44.31 | ChannelZ | And you're SURE you connected power to the TDM board? |
| 01:44.45 | weta | I can double check to be certain. |
| 01:45.14 | weta | back in a few. (I'm on the same system) |
| 01:45.15 | ChannelZ | It sounds like maybe not. Or that the card is defective. |
| 01:45.37 | ChannelZ | (or that your PSU is defective and not giving good juice to the molex connector you're using) |
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| 01:59.47 | weta | ChannelZ: Power seems fine. Noticed for the first time the card has faintly etched 1,2,3,4 on the back... never noticed that before as it needs the right light. |
| 01:59.58 | weta | Swapped analog phones to control. And it does the same thing. |
| 02:00.23 | weta | Interestingly, if I hang up after talking between SIP/analog, and pick up again right away, I get dial tone. |
| 02:02.47 | ChannelZ | but not if you wait? |
| 02:03.09 | weta | That's right. Wait a second or two more and it's back to no dialtone when you pick up the analog |
| 02:03.33 | ChannelZ | I dunno everything you're describing sounds a bit fishy like the card is wonky or something (or at the least your FXS module) |
| 02:03.52 | weta | I'll configure port 3 and see if it does the same thing. |
| 02:04.26 | weta | And see if someone in town wouldn't mind me borrowing a card for an hour or two in exchange for a case of beer :-) |
| 02:04.59 | weta | Thank you so much for your help. I'm taking nodes (and git managing my code) so I'll share what I learn later. Hopefully it might benefit others. |
| 02:05.06 | weta | s/nodes/notes/ |
| 02:06.34 | ChannelZ | I think we have your hardware config sorted out at least so I'm not sure what else could be wrong. I have the TDM800P which is basically the 4-module/8-port version of that card. What version of * by the way? |
| 02:06.57 | atan | All my setup just went all to hell. For some reason if the SIP line says "while" the other person hears "wi", if anything at all |
| 02:07.24 | weta | 1.6.2.11 |
| 02:07.37 | atan | I swear something have voice canceling and I am going to throw it out the window when I find what |
| 02:11.51 | WIMPy | chan_iax contantly moans about "Max retries exceeded to host" whilt the host ist actually reachable and calls can be placed to and from that peer. |
| 02:12.20 | WIMPy | Anyone have an idea wat's happening or how to investigate? |
| 02:14.14 | weta | ChannelZ: ooo, interesting. Switched to port 3 and suddenly my analog phone's time and date were set and a message displaying 1 new call appeared. |
| 02:15.16 | WIMPy | Can someone point me to a listing of IAX types and subclasses? |
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| 03:14.39 | p3nguin | switches MoH to Christmas mode |
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| 03:34.43 | ChannelZ | weta: Hmm. Did you buy the card new? |
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| 03:41.49 | WIMPy | Somehow it looks like it moaning about more timed out retries than it's actually transmitting. |
| 04:03.41 | chuckf | exit |
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| 04:22.35 | weta | ChannelZ: I bought it new, though it's been a year or more. I came back and saw this in dmesg: http://pastebin.com/uCQ2P5eH |
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| 05:11.52 | ChannelZ | weta: hmmmm.. seems not good.. |
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| 06:20.19 | smeet2002 | hey guys! |
| 06:21.27 | smeet2002 | apprently I managed to compile h323 drivers for Asterisk 1.8 :) |
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| 06:27.24 | ChannelZ | makes a note of it |
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| 06:35.03 | shamelessn00b | Hi all |
| 06:35.19 | shamelessn00b | idk, but is any of you familiar with JSGF grammars? |
| 06:43.19 | smeet2002 | me not |
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| 07:18.00 | `mc | hello |
| 07:28.52 | EmleyMoor | I've done a rebuild and upgrade, and now I'm having problems with incoming IAX calls not working - they usually, but not always, fail after processing the first line of the dialplan. I have already turned off the requirement for CallTokens |
| 07:29.14 | p3nguin | core set verbose 4 |
| 07:29.18 | p3nguin | Make a call. |
| 07:29.24 | p3nguin | Pastebin the output. |
| 07:31.08 | EmleyMoor | Damn - two successes in a row (!) |
| 07:31.39 | p3nguin | Yeah, hate those successful calls. |
| 07:32.57 | EmleyMoor | Output no more interesting at verbose 4 than it was at 3 - will pastebin anyway... |
| 07:33.23 | p3nguin | Had you pasted the verbose 3, I wouldn't have asked for verbose 4. |
| 07:34.15 | p3nguin | 3 and 4 are likely to be EXACTLY the same. |
| 07:35.26 | EmleyMoor | http://paste.debian.net/100870/ anyway |
| 07:36.42 | p3nguin | And I'd like to see your dialplan, too. |
| 07:37.11 | p3nguin | We can start with just the fromiax context if your extensions.conf is huge. |
| 07:38.13 | p3nguin | I'm assuming you realize the call is from a peer which has not authenticated. |
| 07:38.31 | EmleyMoor | p3nguin: Yes - that is normal |
| 07:38.46 | p3nguin | Normal for whom? |
| 07:39.05 | EmleyMoor | This provider, to my system |
| 07:41.03 | EmleyMoor | Just that extension about to be pastebinned - feel free to ask me for more if you need it but a failed call never gets to the NoOp... |
| 07:41.22 | EmleyMoor | http://paste.debian.net/100871/ |
| 07:41.54 | EmleyMoor | (using that extension partly because it's not a very important one) |
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| 07:46.16 | p3nguin | Now I need to see defauly context, extension 4819. |
| 07:46.58 | EmleyMoor | OK - will paste, even though that condition is false |
| 07:48.09 | p3nguin | I realize that the GotoIf() is returning 0, which should make the current extension continue, but I'd still like to verify that the conditional branching is valid... if you don't mind sharing it. |
| 07:49.09 | p3nguin | I've encountered things more odd breaking dialplan, so let's eliminate it. |
| 07:49.19 | p3nguin | s/more/less/ |
| 07:49.25 | EmleyMoor | http://paste.debian.net/100872/ |
| 07:50.41 | p3nguin | And variables ${PHIL} & ${DAVID} & ${HALL} all validate to valid channel technology and peer names? |
| 07:51.01 | `mc | anyone know of a good resource for askterisk development with C#/.NET? |
| 07:51.03 | EmleyMoor | Can show them if you like, just to be sure |
| 07:52.17 | EmleyMoor | http://paste.debian.net/100873/ |
| 07:53.13 | p3nguin | Looks good enough. |
| 07:54.31 | p3nguin | Did you look at iax debug? |
| 07:54.51 | EmleyMoor | No - that was going to be my next guess |
| 07:55.14 | `mc | wheres thanks? |
| 07:55.16 | `mc | that? |
| 07:55.55 | p3nguin | I hate that the verbose output just shows the GotoIf() running and then a Hangup(). There's not a lot to go on. |
| 08:01.49 | EmleyMoor | http://paste.debian.net/100874/ |
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| 08:08.45 | BugKhaM | is there a way to block an ip address from accesing SIP in asterisk? my iptables doesn't work properly |
| 08:09.16 | `mc | anyone know of a good resource for askterisk development with C#/.NET? |
| 08:09.16 | BugKhaM | need to block that first before lloking for holes in my iptables |
| 08:09.31 | EmleyMoor | I'm temporarily takisg my incoming calls by SIP because IAX won't work |
| 08:09.37 | EmleyMoor | taking |
| 08:10.56 | greezmunkey | EmleyMoor: Still having trouble with that? I thought you were going to tkat it up with your provider. Did you find something? |
| 08:11.19 | greezmunkey | s/tkat/take |
| 08:11.55 | EmleyMoor | greezmunkey: I may well try to take it up with them once they are open - I could not see anything wrong with the IAX debug |
| 08:14.14 | greezmunkey | EmleyMoor: Who is your provider? |
| 08:20.26 | EmleyMoor | vouptalk.org |
| 08:20.32 | EmleyMoor | voiptalk.org |
| 08:21.08 | EmleyMoor | Have raised a ticket with them - could well be that between us we can sort it |
| 08:23.49 | greezmunkey | Heh, maybe your data has been re-routed through China Telecom! |
| 08:31.12 | EmleyMoor | I had ropy call quality (outbound) the other week |
| 08:34.04 | greezmunkey | EmleyMoor: All of your routing gear is alright? |
| 08:34.42 | EmleyMoor | greezmunkey: My own router is fine - C&W's seem to be too |
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| 09:07.09 | greezmunkey | Is there a mechanism that would allow an extension to always use a particular port on a gateway. Where any calls from ExtnA are directed to Port1, and then whatever digits were originally dialed are dumped to that port? |
| 09:07.14 | EmleyMoor | Can anyone explain to me Dial's "U" option - where can the routine be? |
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| 09:22.25 | greezmunkey | EmleyMoor: http://www.asteriskguru.com/tutorials/dialplan_applications.html has some good information |
| 09:23.59 | EmleyMoor | Can't see where it might mention that, apart from one place, where it doesn't |
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| 09:24.47 | sam_affable | hey |
| 09:25.19 | EmleyMoor | will investigate - might be able to get rid of Macros altogether |
| 09:27.37 | sam_affable | hey im trying to to install asterisk freepbx ....i followed this guide http://www.corenetworks.com.au/wiki/doku.php?id=install_voip_server and after finishing install of freepbx ..seems that apache is not working ..everything starts but when i try to access any site on my web server ..it dosent load |
| 09:27.53 | sam_affable | http://120.0.0.1/html dosent load |
| 09:28.20 | sam_affable | i meant http://127.0.0.1/html |
| 09:28.25 | greezmunkey | sam_affable: try 127.x.x.x |
| 09:28.32 | greezmunkey | ok |
| 09:28.50 | EmleyMoor | That's more FreePBX than Asterisk... |
| 09:29.01 | sam_affable | oh okie |
| 09:29.13 | sam_affable | will try to reconfigure and see |
| 09:29.42 | sam_affable | maybe some permission problem ..as im not able to access any site on my web server |
| 09:29.43 | sam_affable | ty |
| 09:30.18 | sam_affable | also try freepbx channel ty |
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| 09:42.16 | EmleyMoor | will let people know if/when he gets IAX fixed, just in case it's something others may need to know |
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| 10:57.24 | EmleyMoor | Found the cause of my IAX problem :-) |
| 11:00.38 | EmleyMoor | Default route clash |
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| 11:17.24 | EmleyMoor | I've got a warning about using | rather than , as the application delimiter - but I thought I was using it in the sense of "logical OR"... have I done it wrong in this line? |
| 11:17.36 | EmleyMoor | exten => 02070434817,n,GotoIf(${DB_EXISTS(calloutweek/1)} | ${DB_EXISTS(nightshift/1)}?workcall) |
| 11:18.43 | EmleyMoor | I think I may have spotted a mistake |
| 11:21.03 | EmleyMoor | Yes - se |
| 11:21.18 | EmleyMoor | Needed to $[] it |
| 11:26.09 | ChannelZ | yes |
| 11:26.56 | *** join/#asterisk Ad-Hoc (~nimbus@62.1.141.242.dsl.dyn.forthnet.gr) |
| 11:27.00 | EmleyMoor | Can SIP listen to both public and private interfaces? |
| 11:27.45 | ChannelZ | It's all or one |
| 11:28.52 | *** join/#asterisk angryuser_laptop (~angryuser@90-156-167-83.reverse.alphalink.fr) |
| 11:32.03 | EmleyMoor | How do I verify whether it's doing all or one? |
| 11:32.35 | ChannelZ | netstat -alpn |grep aster |
| 11:33.06 | ChannelZ | or udpbindaddr in sip.conf - 0.0.0.0 is everything else you can specify an IP (and port) to listen on |
| 11:33.28 | ChannelZ | If you have multiple interfaces but don't want it on all of them, firewall them off |
| 11:33.40 | EmleyMoor | 0.0.0.0 means all - OK then |
| 11:34.08 | EmleyMoor | (when I add desk phones they will probably be "inside") |
| 11:38.22 | *** join/#asterisk sawgood (~sawgood@173-13-158-27-sfba.hfc.comcastbusiness.net) |
| 11:40.31 | ChannelZ | So is sawgood like phoenetic slang for "it's all good"? |
| 11:44.03 | sawgood | yes that is what it stands for |
| 11:44.30 | ChannelZ | Cool. Pretty good for almost 5am |
| 11:44.46 | sawgood | you are on CST? |
| 11:45.04 | ChannelZ | Mountain |
| 11:49.15 | WIMPy | chan_iax contantly moans about "Max retries exceeded to host" whilt the host ist actually reachable and calls can be placed to and from that peer. |
| 11:49.27 | WIMPy | Anyone have an idea what's happening or how to investigate? |
| 11:53.54 | ChannelZ | Well in general it's trying to re-send a frame and apparently not getting a response. Intermittent connectivity, errant registration to someplace that doesn't exist anymore... |
| 11:54.38 | WIMPy | The exist no registrations and the peer is fully working as I said. |
| 11:55.08 | WIMPy | Actually setting iax2 debug on I'm unsure it really tried to transmit anything. |
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| 12:01.53 | WIMPy | Looks a little fishy to me. |
| 12:02.47 | ChannelZ | well barring it being a bug, it's re-transmitting frames 4 times and failing for *some* reason |
| 12:05.22 | WIMPy | In the debug I only see POKE frames that do get answered, no retransmissions. |
| 12:05.29 | *** join/#asterisk sam_affable (sam@202.53.10.123) |
| 12:05.32 | sam_affable | hi |
| 12:05.39 | WIMPy | The warnings seem to come out of thin air. |
| 12:06.34 | sam_affable | hey when ever i try to stop or restart asterisk amportal stop i get Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) |
| 12:06.44 | sam_affable | asterisk CLI is working . |
| 12:06.52 | sam_affable | any idea on how to fix that |
| 12:10.10 | ChannelZ | I don't see in the source that it triggers any logging when it retransmits that would specifically indicate re-transmission |
| 12:12.31 | WIMPy | __attempt_transmit: Max retries exceeded |
| 12:13.24 | ChannelZ | ....yeah.... |
| 12:14.02 | WIMPy | http://wimpy.yeti.dk/pastebin |
| 12:15.08 | ChannelZ | how high is your iax debug turned up? |
| 12:15.41 | WIMPy | Err, are there levels? I did iax2 set debug on. |
| 12:16.28 | ChannelZ | I assumed so from the source but maybe those are the 3 kinds.. |
| 12:17.12 | ChannelZ | or actually this one line I'm looking at.. I think it outputs to general debug but if iax debug is on, that's it. |
| 12:17.22 | WIMPy | Turning up core debug does not give any more information |
| 12:18.16 | ChannelZ | well I see a message at level 3 in send_packet but it might not actually help |
| 12:20.56 | WIMPy | Where can I find a list of the iax types and subclasses? |
| 12:23.09 | WIMPy | Are they POKE/ACK messages? |
| 12:23.29 | ChannelZ | probably iax2.h |
| 12:25.46 | WIMPy | Ja, PING/LAGRQ |
| 12:28.17 | ChannelZ | What is 10.9.42.1? Another * box or a softphone or.. |
| 12:28.31 | WIMPy | Another * |
| 12:29.25 | ChannelZ | well you could tcpdump on both ends and see if packets are being transmitted but not received on the other end |
| 12:30.07 | ChannelZ | which would at least tell you if it's a networking problem or not |
| 12:30.22 | ChannelZ | if the frames really are being lost |
| 12:31.52 | WIMPy | I have more of a feeling they aren't being sent. Not the ones it moans about at least. Both boxes show the other as OK and calls in both directions work just fine. |
| 12:32.19 | WIMPy | No, tcpdump seems to show the same as iax debug. |
| 12:34.53 | WIMPy | It's always three packets. Always alternating TxRxTx and RxTxRx, both qualifying each other. |
| 12:35.14 | *** join/#asterisk ickmund (~ickmund@cli-5b7ee407.bcn.adamo.es) |
| 12:35.46 | WIMPy | Interesting may be that iax debug shows TxRxTx and RxRx without an Tx inbetween, even though it actually is sent. |
| 12:36.53 | *** join/#asterisk qjb (~qjb@a83-163-158-168.adsl.xs4all.nl) |
| 12:36.55 | WIMPy | But the warnings come much more fequent. |
| 12:37.01 | ChannelZ | Are they running dissimilar versions? |
| 12:37.36 | WIMPy | They are both on svn from about a week ago. |
| 12:38.12 | WIMPy | I tried 1.8.0 here but had to go svn because of IAX. |
| 12:39.14 | WIMPy | At that time the one you see moaning now was 1.6.2.10-RCsomething which went bad on the IAX part. Updating to 1.6.2.14 didn't help so I used svn there too. |
| 12:39.44 | ChannelZ | what do you mean 'went bad' |
| 12:40.05 | WIMPy | IAX consuming vast amounts of bandwidth. |
| 12:40.50 | WIMPy | With two concurrent calls voce quality was miserable. Took soem time before I found out that my 1MBit uplink was the problem. |
| 12:41.07 | WIMPy | One single call was eating about 260kbit. |
| 12:41.46 | WIMPy | I did (and do) not have issues to an old 1.4 version or when using zoiper. |
| 12:46.32 | WIMPy | That's why I replaced the 1.6 by trunk instead of downgrading here. |
| 12:47.22 | WIMPy | .1 being here |
| 12:57.22 | *** join/#asterisk AlHafoudh (~AlHafoudh@adsl-dyn175.78-98-181.t-com.sk) |
| 13:01.49 | WIMPy | Oh, actually I could not place a call from .1 to .2 now. I could before. |
| 13:01.58 | WIMPy | I have restarted it now. |
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| 14:26.06 | robl^laptop | ael reload |
| 14:26.12 | robl^laptop | oops. wrong window! |
| 14:26.42 | WIMPy | He, an ael user. |
| 14:28.03 | robl^laptop | WIMPy: yup! I have 5 lines in extensions.conf. I have prolly ~900 lines in ael ;-) |
| 14:29.52 | robl^laptop | hrmmm wc -l says I have a total of 1,344 lines in ael. |
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| 15:25.37 | *** join/#asterisk gregd (~gregd@188-220-38-34.zone11.bethere.co.uk) |
| 15:26.10 | gregd | guys, is i use sipdtmfmode(inband), how can increase/deacrease the output level of a tone? |
| 15:30.48 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 15:32.03 | smeet2002 | I wish I knew |
| 15:33.41 | gregd | so i guess it is not possible straight out of box |
| 15:34.46 | smeet2002 | I don't know, sorry..I had a hard time with H323 recent days... |
| 15:35.40 | gregd | unless... does the tone come from a file, or is it generated? as if it comes out of the file i could change the file |
| 15:37.18 | smeet2002 | I am not so deep.. |
| 15:39.38 | *** join/#asterisk Sipster_ (~Sipster@modemcable045.5-200-24.mc.videotron.ca) |
| 15:43.09 | kaldemar | gregd: you can try to amplify the whole call with func VOLUME if the audio goes through asterisk. it will amplify everything though. the tones do not come from a file. |
| 15:43.50 | gregd | kaldemar: can I change the VOLUME throughout the call? |
| 15:44.02 | kaldemar | gregd: what sends the tones? |
| 15:44.11 | gregd | sendDTMF() |
| 15:44.52 | gregd | other option that I guess will work, is to create my own playback that contains simply the tone i'm interested in with correct volume |
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| 16:28.05 | gregd | guys, i do not get it - musiconhold - i've got a few groups defined in musiconhold.conf ... how do i allocated them, and actually to what are they allocated?! |
| 16:28.40 | p3nguin | What are you trying to do? |
| 16:29.40 | *** join/#asterisk polemon (mcp@polemon.org) |
| 16:29.55 | polemon | hey, Asterisk people |
| 16:30.16 | gregd | p3nguin: play my own file when musiconhold |
| 16:30.51 | polemon | I was wondering, why doesn't asterisk use a small scripting language for dialplans (like Lua), instead it uses its own language? |
| 16:31.47 | p3nguin | gregd: Did you create a class in musiconhold.conf for this purpose? |
| 16:32.14 | gregd | p3nguin: yes, i have my own class called "USER" |
| 16:32.32 | p3nguin | gregd: Show the section to me. |
| 16:32.46 | p3nguin | pastebin if it's more than a few lines. |
| 16:33.21 | gregd | p3nguin: http://pastebin.com/VDDrHy1U |
| 16:33.26 | smeet2002 | guys, who is interested in testing Asteriks interconnection right now? |
| 16:34.02 | p3nguin | gregd: ls -dl /var/lib/asterisk/moh/User/ |
| 16:34.28 | gregd | p3nguin: -rw-r--r-- 1 asterisk asterisk 4448260 Nov 28 07:04 world.wav |
| 16:34.40 | p3nguin | fail |
| 16:34.42 | p3nguin | Try again. |
| 16:34.58 | gregd | p3nguin: http://pastebin.com/btxQSNux |
| 16:35.16 | p3nguin | Now run the command I asked for and show me the output. |
| 16:35.17 | gregd | sorry |
| 16:35.25 | gregd | p3nguin: drwxr-xr-x 2 asterisk asterisk 4096 Nov 28 07:12 /var/lib/asterisk/moh/User/ |
| 16:35.32 | smeet2002 | I have started Asterisk 1.8 , tested two sip phones , they are working , now I would like to make a "long" call..is anybody interested? |
| 16:35.33 | p3nguin | okay |
| 16:35.56 | p3nguin | gregd: Did you reload the moh? "moh reload" should do it. |
| 16:36.01 | *** join/#asterisk sawgood (~sawgood@173-13-158-27-sfba.hfc.comcastbusiness.net) |
| 16:36.05 | gregd | i did reload the whole asterisk |
| 16:36.43 | p3nguin | gregd: Does "moh show classes" show your new User class? |
| 16:37.03 | *** join/#asterisk AdvoWork (~AdvoWork@unaffiliated/advowork) |
| 16:38.13 | p3nguin | gregd: And if so, does "moh show files" show your world.wav file under the User class? |
| 16:38.19 | gregd | p3nguin: it does show |
| 16:38.37 | gregd | p3nguin: it does not show my wav file though |
| 16:38.47 | *** join/#asterisk eject_ck (~eject_ck@haitek.cosmonova.net.ua) |
| 16:39.03 | p3nguin | gregd: Do other moh classes work correctly? |
| 16:39.32 | gregd | p3nguin: http://pastebin.com/xhjcfRMg |
| 16:39.49 | gregd | p3nguin: the default one work.. it plays a file.. but god know which one |
| 16:40.07 | p3nguin | It's easy to find out what file is playing. |
| 16:40.25 | p3nguin | gregd: Can I see "moh show files" too? |
| 16:40.26 | gregd | hm.. i see |
| 16:41.01 | gregd | p3nguin: http://pastebin.com/FbnWzg6y |
| 16:41.23 | p3nguin | gregd: Your "world" file is listed. |
| 16:41.29 | p3nguin | That's good. |
| 16:41.45 | gregd | it is indeed... but now how do I make my USER class being used? |
| 16:42.01 | gregd | it always plays default class on hold |
| 16:42.02 | p3nguin | gregd: Did you set a music class in the extension? |
| 16:42.08 | gregd | no, i havent |
| 16:42.23 | gregd | can i set it up globally somehow? |
| 16:42.52 | p3nguin | If you want it to be the global default, use the default class rather than a class called User. |
| 16:43.59 | gregd | fair enough. So i guess, an extension has to have "musiconhold=User" line, is that correct? |
| 16:46.19 | gregd | i moved everything to default class and it works :) |
| 16:46.22 | gregd | p3nguin: thanks! |
| 16:46.27 | p3nguin | No, an extension would have SetMusicOnHold(User). |
| 16:46.55 | p3nguin | extensions are in extensions.conf, by the way. |
| 16:47.58 | *** join/#asterisk timahvo1 (~rogue@41.223.57.73) |
| 16:48.15 | p3nguin | SetMusicOnHold(User) or Set(CHANNEL(musicclass)=User) |
| 16:48.21 | *** join/#asterisk hehol (~hehol@buero-gw.dortmund.loca.net) |
| 16:55.27 | smeet2002 | anybody wants to test my asterisk? to make a call to me ? ^-)) |
| 17:18.00 | *** join/#asterisk ali536 (~ali536@c-98-223-181-243.hsd1.in.comcast.net) |
| 17:18.03 | ali536 | test |
| 17:18.13 | ali536 | sorry...wrong window |
| 17:18.43 | ali536 | i'm trying to setup an sip server that simply echos phone calls |
| 17:18.47 | ali536 | will i need zaptel? |
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| 17:19.08 | WIMPy | zaptel has been replaced by dahdi. |
| 17:19.17 | ali536 | oh |
| 17:19.25 | WIMPy | How do you plan to receive calls? |
| 17:19.28 | ali536 | do i need it ? i have 1.8 |
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| 17:19.50 | ali536 | a client will initiate an INVITE |
| 17:20.00 | ali536 | i'll just reply back with whatever the client says |
| 17:20.12 | ali536 | so i'll basically have one extension |
| 17:20.29 | ali536 | i'm editing the extensions.conf and sip.conf files |
| 17:20.35 | ali536 | do i need to update any other file? |
| 17:24.42 | *** join/#asterisk `mc (~mc@69.37.245.215) |
| 17:25.53 | `mc | anyone know a good resource for asterick development in .NET? |
| 17:28.41 | ali536 | isn't asterisk made in c? |
| 17:29.30 | `mc | C# |
| 17:29.39 | `mc | yea, its made in C |
| 17:29.44 | ali536 | no, i mean asterisk is made in c |
| 17:29.49 | `mc | yes |
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| 17:30.50 | ali536 | so how's you develp in .net? |
| 17:33.01 | `mc | libraries/API that connects to the server |
| 17:40.53 | `mc | thank [TK]D-Fender , you're my hero |
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| 19:26.16 | carbocalm | good morning. I'm trying have asterisk dial an outbound number + |
| 19:26.16 | carbocalm | <PROTECTED> |
| 19:28.52 | p3nguin | What have you done so far? |
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| 19:39.39 | carbocalm | I already have changed my dialplan to accept 11-19 as speed dial |
| 19:39.39 | carbocalm | <PROTECTED> |
| 19:39.39 | carbocalm | <PROTECTED> |
| 19:39.50 | carbocalm | sorry, lost connection |
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| 19:43.00 | p3nguin | Show me extension 11 so I have something to work off of. |
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| 19:49.40 | p3nguin | carbocalm: Show me your extension 11. |
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| 19:54.34 | carbocalm | exten => 11,1,Goto(macro-outdial,4165551234,1) |
| 19:55.38 | carbocalm | [macro-outdial] |
| 19:55.39 | carbocalm | exten => s,1,Set(CALLERPRES()=allowed) |
| 19:55.39 | carbocalm | exten => s,n,Dial(IAX2/unlimitel/${ARG1},60) |
| 19:55.39 | carbocalm | exten => s,n,Goto(out-${DIALSTATUS},1) |
| 19:55.40 | p3nguin | carbocalm: Can you also show me the outdial macro? Had you not been using a macro, we could easily modify your Dial() command to add some wait time and then additional digits. |
| 19:55.42 | carbocalm | exten => s,n,Hangup |
| 19:55.45 | carbocalm | exten => out-BUSY,1,Hangup(17) |
| 19:55.48 | carbocalm | exten => out-CONGESTION,1,Hangup(3) |
| 19:55.51 | carbocalm | exten => out-CHANUNAVAIL,1,Hangup(3) |
| 19:55.54 | carbocalm | exten => out-NOANSWER,1,Hangup(16) |
| 19:55.57 | carbocalm | exten => _out-.,1,Hangup(16) |
| 19:56.00 | carbocalm | <PROTECTED> |
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| 20:04.58 | carbocalm | thanks p3nguin, I've got it ("we could easily modify Dial()")...thanks again |
| 20:05.02 | p3nguin | carbocalm: If it were mine, I would change extension 11 to not use the macro. I would change it to exten => _11.,1,Dial(IAX2/unlimitel/4165551234,60,D(www${EXTEN:2})) and then I would dial 11 123. |
| 20:05.49 | p3nguin | Dialing 11 123 would call 4165551234, wait 1.5 seconds, then send 123, then bridge the call. |
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| 21:01.40 | greezmunkey | p3nguin: the EXTEN:2 piece baffles me a bit, from your discussion above. What does the "2" signify, does that contain the "123" as dialed above? |
| 21:02.04 | p3nguin | That was my intention. |
| 21:02.53 | p3nguin | It would parse the EXTEN variable, starting with an offset of 2 (skipping the first two digits). |
| 21:03.18 | greezmunkey | p3nguin: thanks, so functionay at the phone, the user dials 11, pauses, and then dials 123. Are they waiting for secondary dialtone or something -- ah, I see, cool. |
| 21:03.34 | p3nguin | Dialing 11123456789 would end up with 123456789. |
| 21:03.43 | greezmunkey | p3nguin: I see. |
| 21:03.54 | p3nguin | no literal pause while pressing the keys |
| 21:04.09 | greezmunkey | p3nguin: got that, the offset does the trick ;) |
| 21:04.25 | p3nguin | The D(www...) creates the pause when sending the tones. |
| 21:04.40 | p3nguin | Each w is .5 seconds of wait time. |
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| 21:05.05 | greezmunkey | p3nguin: so if $EXTEN can use an offset, I assume that you can also prepend, postpend, etc. |
| 21:05.25 | p3nguin | Sure. |
| 21:05.26 | greezmunkey | p3nguin: $EXTEN+123 maybe |
| 21:05.40 | greezmunkey | p3nguin: I'll read up on that one ;) |
| 21:06.13 | greezmunkey | p3nguin: that's about where I am in the book right now anyway. |
| 21:06.22 | p3nguin | If I had dialed 1100 in that above example, ${EXTEN:2}123 would be 00123. |
| 21:06.41 | greezmunkey | p3nguin: nice |
| 21:08.13 | greezmunkey | p3nguin: How would one dedicate a port on a fxo gateway to a particular sip device, for purposes of haveing consistant CID at the PSTN, like exten 123 always uses port #1 on gateway x.x.x.1? |
| 21:08.58 | greezmunkey | p3nguin: I should modify that to say analog fox type gate. |
| 21:08.58 | greezmunkey | s/fox/fxo |
| 21:09.06 | p3nguin | That's not an asterisk dilemma. |
| 21:09.25 | ChannelZ | Just Dial() the appropriate thing. |
| 21:09.26 | p3nguin | That's a networking problem. |
| 21:10.02 | p3nguin | You dial what you want to dial, but you'd have to configure your switch or router to feed traffic out a specific port. |
| 21:10.41 | ChannelZ | I don't think he means port as in tcpip but physical FXO port. |
| 21:10.56 | p3nguin | He also said SIP device, which didn't add up well. |
| 21:11.12 | greezmunkey | p3nguin: I see, so the gate would have to parse the incoming sip connection, determine which device is making the call, then point that at the proper port. |
| 21:11.46 | p3nguin | VLANs are typically used for directing network traffic out/in specific ports in Ethernet. |
| 21:12.24 | p3nguin | Whatever you're actually trying to do doesn't sound well thought out. |
| 21:12.26 | ChannelZ | greezmunkey: You want it so when certain phones are used they all go out a specific POTS line, right? |
| 21:12.45 | greezmunkey | ChannelZ: exactly, like I have three or four analog CO lines, each has their own "number" in the PSTN, user "A" always wants to dial out on line #1 so that ppl he calls see the number on his business card, for example. |
| 21:13.19 | greezmunkey | ChannelZ: I spent a while last night trying to crack that one. |
| 21:14.00 | p3nguin | Just dial using whichever channel/port you want to use. |
| 21:14.01 | ChannelZ | You just have to Dial() the appropriate technology from your dialplan depending on who it is. There's a bunch of different ways you could go about it |
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| 21:14.47 | ChannelZ | A bunch of GotoIf logic to look at whose calling, or you could just duplicate your outgoing extensions in different contexts for each person and hard-code the destination in that way... |
| 21:15.16 | ChannelZ | Or set a variable for each SIP device of the outgoing trunk they should use, and you use that variable in your common Dial()'s.. |
| 21:16.06 | greezmunkey | ChannelZ: yeah, That's what I've found. But, the $EXTEN with offsets, pre, and postpending may just be a big part of the solution as well. I was looking at having the same "phone" in differant contexts last night. |
| 21:16.53 | ChannelZ | setvar=OUTTRUNK=DAHDI/4 in sip.conf for a peer - then you Dial(${OUTTRUNK}/1234567890) |
| 21:17.03 | p3nguin | You can only assign one context per device. Subsequently, you can include other contexts within the assigned contexts to add more contexts to the device's assigned context. |
| 21:17.06 | ChannelZ | you'd just have to make sure it's set to something valid for everyone |
| 21:17.17 | greezmunkey | That last post was to your 2nd to last. The variable thing looks promising. Like I said earlier, that's about where I am in the book. |
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| 21:18.10 | greezmunkey | ChannelZ: I'll get back to it here in a bit. Thanks again ChannelZ and p3nguin |
| 21:20.33 | ChannelZ | sho |
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| 23:33.50 | smeet2002 | who knows why "www.gnophone.com" and "www.iaxtel.com" are dead??? |
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| 23:45.00 | ChannelZ | no clue |
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| 23:48.37 | smeet2002 | who wants to connect to my asterisk ? |