| 00:06.06 | *** join/#asterisk x86 (~x86@i.am.leet.org) |
| 00:06.50 | x86 | I have a polycom IP601, and I'd like to be able to hit the "Messages" button for one-touch access to voicemail... where in the provisioning files can I change the behavior, and where can I tell it which number to dial for voicemail? |
| 00:11.03 | jblack | kuku: Test your network to see what sort of bandwidth you get from upstream (some providers lie). |
| 00:12.09 | jblack | Something is dropping packets between you and destination. |
| 00:12.42 | jblack | At least that's my belief. It could always be something else |
| 00:13.06 | *** join/#asterisk marl_scot (~marl_scot@cpc1-cowc4-0-0-cust381.14-2.cable.virginmedia.com) |
| 00:13.52 | marl_scot | anyone useing lcr for outbound calls from asterisk? need some advice on setting it up, please |
| 00:18.22 | *** join/#asterisk terracon (~tc@CPE001cf097a82c-CM0012254076d6.cpe.net.cable.rogers.com) |
| 00:23.52 | kuku | jblack: I have 50 down, 5 up |
| 00:23.59 | kuku | i should be able to handle 10 calls no problem. |
| 00:24.26 | kuku | So my original question was, is there a way to see that quality/delays on SIP on the CLI |
| 00:24.35 | kuku | IAX has something like this |
| 00:27.44 | *** join/#asterisk radic (~radic@dslb-094-216-225-110.pools.arcor-ip.net) |
| 00:44.19 | jdoe | depends on what info you want. |
| 00:44.38 | jdoe | ie sip show channelstats will show you jitter and loss... |
| 00:49.51 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 00:52.22 | *** join/#asterisk ariel_ (~chatzilla@c-98-254-138-7.hsd1.fl.comcast.net) |
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| 00:57.29 | WIMPy | marl_scot: your question, please :-) |
| 00:58.21 | marl_scot | hi WIMPy , i have setup lcr for * 1.4, and can dial in without any problem, but i cant work out how to dial out, via SIP -> LCR |
| 00:58.41 | marl_scot | im sure im missing something in my LCR file, but cant find it |
| 00:58.54 | marl_scot | do u want me to paste my configs to pastebin? |
| 00:58.58 | WIMPy | So what happens? |
| 00:59.29 | marl_scot | i try and dial out, i get a pause of approx 5 seconds, and get a chan unavailable |
| 00:59.29 | WIMPy | And why 1.4? |
| 01:00.07 | marl_scot | i hit big problems getting lcr running woith 1.8, was easier to fall back to 1.4 and get it all working |
| 01:00.40 | WIMPy | The git version has 1.8 support since the weekend. |
| 01:01.38 | marl_scot | this was installed before that, once i can get this working, i may make the next one 1.8 (i want to try some of the new features) but for now, am on 1.4 |
| 01:02.32 | WIMPy | Could be the power saving that strikes. |
| 01:02.57 | marl_scot | ? |
| 01:03.17 | WIMPy | Have you looked at the status screen? Does L1 go down on that port? |
| 01:03.39 | WIMPy | What kind of line is it? ptp? ptmp? |
| 01:04.06 | marl_scot | ptp (card connected to ISDN with telco) |
| 01:04.23 | marl_scot | whats the best way to get the status screen? |
| 01:04.51 | WIMPy | lcradmin state |
| 01:05.08 | WIMPy | But ptp should be permanently active. |
| 01:05.57 | marl_scot | L1 stays active, doesnt go down |
| 01:06.16 | marl_scot | i think im missing something int he lcr conf to allow the dialing out, but am getting really confused :( |
| 01:07.01 | WIMPy | You don't need something special. You select the port in the dial command. |
| 01:07.29 | WIMPy | Do you have more than one port? |
| 01:07.47 | marl_scot | it is a 4 port BRI card |
| 01:07.56 | marl_scot | the dial command in * is : exten => _9.,n,Dial(LCR/Ext/${EXTEN:1}) |
| 01:08.10 | marl_scot | i did have : exten => _9.,n,Dial(LCR/${EXTEN:1}) |
| 01:08.49 | WIMPy | And the port os called "Ext"? |
| 01:09.01 | marl_scot | yup |
| 01:09.17 | WIMPy | Well, you should think that the wrong name fails immediately. |
| 01:09.29 | WIMPy | Then I guess it's time for the lcr log. |
| 01:09.41 | marl_scot | it still fails when i take out the port name, and leave it to pick the next vailable |
| 01:10.57 | marl_scot | http://pastebin.ca/1992819 |
| 01:11.05 | marl_scot | has the lcr log for an attempt to dial out |
| 01:12.03 | WIMPy | Oh, much configuration. |
| 01:12.37 | marl_scot | the main config is for the DDI numbers inbound, which are all working |
| 01:12.56 | WIMPy | Wouldn't have been neccessary. |
| 01:13.37 | marl_scot | yup, i know i could do it as a general, all to * command, ended up putting it in as that to see if it made any differance to the outbound |
| 01:15.27 | WIMPy | Hmm. |
| 01:16.08 | WIMPy | Looks like the switch does not get the message. |
| 01:16.25 | marl_scot | switch as in asterisk? |
| 01:16.27 | *** join/#asterisk boch (~fseratti@190.220.65.19) |
| 01:16.37 | WIMPy | telco |
| 01:16.37 | marl_scot | or the telco |
| 01:16.40 | boch | hi |
| 01:17.11 | boch | im getting this error when trying to compile dahdi: You do not appear to have the sources for the 2.6.18-194.el5 kernel installed. Do you know what rpm i am missing to install ? |
| 01:17.48 | *** join/#asterisk Mhaddog_ (~Mhaddog@z65-50-116-17.ips.direcpath.com) |
| 01:18.00 | marl_scot | just spotted on the lcr status page, something about american tones, where would this be set? |
| 01:18.13 | lanning | ya, try installing "the sources for the 2.6.18-194.el5 kernel" |
| 01:18.56 | WIMPy | options.conf: tones_dir |
| 01:20.57 | marl_scot | is that part of the kernel tree rather than lcr? |
| 01:21.22 | WIMPy | That's lcr. |
| 01:21.37 | marl_scot | what is the string for UK? |
| 01:22.20 | WIMPy | There's only american, german and sume funny thing included. |
| 01:22.37 | WIMPy | At least when I last checked. |
| 01:22.43 | marl_scot | so how do u get it to work with the uk? |
| 01:23.04 | WIMPy | That's only tones, i.e. what you hear. |
| 01:23.13 | marl_scot | ah ok, dtmf? |
| 01:23.23 | marl_scot | and dial tones type thing? |
| 01:23.51 | WIMPy | No, dial and busy tones, etc and announcements. |
| 01:24.03 | marl_scot | ok, so that wouldnt stop us dialing out would it? |
| 01:24.33 | WIMPy | no |
| 01:24.59 | marl_scot | ok, so any ideas on how the telco isnt getting the messages? |
| 01:25.22 | WIMPy | Well, if the line is up, no. |
| 01:26.46 | marl_scot | http://pastebin.ca/1992826 now has a copy of the top section of the LCR status page |
| 01:27.27 | WIMPy | Oh. |
| 01:27.45 | marl_scot | that sounds bad! |
| 01:27.51 | WIMPy | I see port 0-2 is shown as ptmp. |
| 01:28.35 | marl_scot | ah ok i see what u mean, why is that as i have ptp in the config file |
| 01:28.37 | WIMPy | I haven't tried groups so far. |
| 01:28.51 | WIMPy | Yes, I sse it. |
| 01:28.55 | marl_scot | would i be better make 4 single entries? |
| 01:29.17 | marl_scot | or does the ptp have to be after each portnum? |
| 01:29.30 | WIMPy | I'm not sure what a group should look like. Trrying to find out. |
| 01:30.13 | marl_scot | yes, it does have to have ptp after each port def. |
| 01:30.30 | WIMPy | Ok. Good to know. |
| 01:30.31 | marl_scot | BINGOOOOOOOOOOOOOOOOOOOO |
| 01:30.52 | marl_scot | thats me able to dial out :) :) :) :) :) :) :) :) |
| 01:31.22 | WIMPy | cool |
| 01:32.14 | marl_scot | http://pastebin.ca/1992827 now has the correct working port def :) |
| 01:32.44 | marl_scot | i have been banging my head off the monitor for 4 hours to get that working! |
| 01:34.54 | mick_laptop | anyone have anything really cool that they do w/ asterisk that is a bit "out of the ordinary"? |
| 01:35.26 | mick_laptop | i'd like to maybe get some ideas on some things that I can do w/ my setup |
| 01:46.54 | boch | lanning, it is a just installed centos 5.5, there was a broken link |
| 02:14.40 | *** join/#asterisk binbash_ (~peter@ip4da5c213.direct-adsl.nl) |
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| 02:41.34 | gabdrach | hi, im with asterisk 1.8 on solaris. i want to add cdr -> mysql. i notice the addon directory under root sources, but gmake inside is not working. |
| 02:57.04 | *** join/#asterisk binbash_ (~peter@ip4da5c213.direct-adsl.nl) |
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| 03:11.38 | gabdrach | using asterisk 1.8 on solaris, i have no sound (calling form a soft phone and just executing Playback()) |
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| 03:28.13 | *** join/#asterisk raddy (~raddy@117.192.231.16) |
| 03:28.21 | raddy | Hello Everybody |
| 03:29.03 | raddy | I have an Motorola Wildcard based x100p clone fxo card, |
| 03:29.27 | raddy | the wcfxo driver is unable to detect the card. |
| 03:30.26 | raddy | But it is being recognized correctly, it uses an irq that is used by intel ata controler according to lspci -v |
| 03:31.21 | raddy | but according to /proc/interupts, the wcfxo module doesn't use any irqs. |
| 03:32.39 | raddy | i tried to follow some of the workarounds provided in http://www.x100p.com/support/doc/novavox-asterisk-card-installation-issues.pdf to resolve the irq conflicts issue indicated by lspci -v |
| 03:32.52 | raddy | But i couldn't. |
| 03:33.32 | raddy | Anybody looking at my messages? |
| 03:33.58 | *** join/#asterisk binbash_ (~peter@ip4da5c213.direct-adsl.nl) |
| 03:38.28 | *** join/#asterisk simplydrew (~simplydre@pool-96-238-59-82.prvdri.fios.verizon.net) |
| 03:38.45 | *** join/#asterisk sawgood (~sawgood@173-13-158-27-sfba.hfc.comcastbusiness.net) |
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| 03:41.42 | sawgood | from the Asterisk CLI, is there a way to 'dump' the contents of a sip set debug on (to a file rather than to the screen)? |
| 03:42.00 | sawgood | The concern is ... when I go back to view the messages, they have rolled off the screen |
| 03:44.19 | *** join/#asterisk JacksLivr (~root@208.251.149.4) |
| 03:45.14 | jbroome | sawgood: you can do an asterisk -rx 'sip debug' > debug.txt from the cli |
| 03:46.23 | sawgood | thank you |
| 03:46.36 | sawgood | by default, what is actually sent to /var/log/asterisk/full? |
| 03:46.48 | sawgood | I do not think SIP debug messages go there? |
| 03:46.53 | JacksLivr | hello. I am a n00b, but I have been reading and trying and I am stuck fairly early. I have a centos box with *1.6. I am following along with an * book and i cannot get my sip softphone to register. I have turned off selinux and iptables and matched the config for 2 different sip phones to sip.conf. I start it up and i never see the sip phone register and it keeps erroring out in the softphone side. what can i try? |
| 03:46.59 | JacksLivr | thanks |
| 03:47.27 | jbroome | sawgood: depends on your verbosity level |
| 03:47.42 | sawgood | core set verbose 5 |
| 03:49.27 | raddy | can anybody help me? |
| 03:50.04 | JacksLivr | at one point (briefly, and I am not sure whhy this happened for 10 seconds out of the hours i have tried) i get this |
| 03:50.07 | JacksLivr | Registration from '<sip:iphone@sip.dougstuff.com>' failed for '45.54.21.18' - No matching peer found |
| 03:50.13 | pabelanger | raddy: move it to a different slot |
| 03:54.41 | *** join/#asterisk iceman85 (~IceChat77@24.115.29.227.res-cmts.wtn.ptd.net) |
| 03:54.57 | iceman85 | hello |
| 03:55.20 | iceman85 | i have a question about sending DTMF Digits over PRI circuits in Asterisk |
| 03:56.15 | iceman85 | I have a phone system set up, and every time i dial an IVR that requires me to press # or * to confirm my entry, I must press that key twice in order to make the system send the DTMF tones for that key |
| 03:56.48 | iceman85 | is there any way that i can make asterisk send the DTMF tone for these keys immediately when pressed, just like if i was pressing 4 or 5? |
| 03:56.58 | raddy | @pabelanger : i have already tried that |
| 04:02.10 | sawgood | what 'if' you are already at the * CLI ... and you want to re-direct SIP debug messages to a file rather than the console (can you use any CLI command to do this) |
| 04:02.34 | sawgood | The reason is so you can stay at the CLI to watch other things, but have the SIP messages being logged at the same time |
| 04:03.55 | iceman85 | you may be able to compose a grep command to filter out the items that you need and run it on another console |
| 04:05.44 | sawgood | I sort of want to see the SIP debug messages come on the screen (not having to worrying about saving them) and at the same time they are being logged in a file somewhere |
| 04:06.29 | iceman85 | hrmm, I'm not sure how you would redirect SIP debug messages from the * CLI |
| 04:06.53 | sawgood | well, maybe a better way would be to ... see if there is a way to do this ... |
| 04:06.59 | iceman85 | perhaps |
| 04:07.01 | *** part/#asterisk nny (~Scott@cpe-174-107-201-103.sc.res.rr.com) |
| 04:07.11 | iceman85 | I would actually try running it off the logfile |
| 04:07.19 | sawgood | run sip set debug on (and have them fly by on the screen), but at the same time have anything on the console sent to a file in /var/log/asterisk |
| 04:07.38 | iceman85 | yep |
| 04:07.43 | JacksLivr | <PROTECTED> |
| 04:07.47 | iceman85 | and then grep out the stuff you want from the log. |
| 04:08.09 | sawgood | exactly ... but right now no SIP debug messages are going to any file I have |
| 04:08.21 | iceman85 | I see |
| 04:09.04 | iceman85 | Sorry i'm not sure how you would do that. |
| 04:10.39 | titter | You can edit logger.conf and add debug to the messages => line then issue a logger reload in the CLI and all debug messages will log to /var/log/asterisk/messages |
| 04:12.07 | sawgood | console => notice,warning,error,verbose |
| 04:12.32 | sawgood | oh I see it now |
| 04:12.58 | titter | yes messages => would log to the messages log |
| 04:13.03 | titter | so you can add debug to that line |
| 04:13.18 | titter | notice,warning,error,verbose,debug |
| 04:13.25 | sawgood | got it |
| 04:13.27 | sawgood | thank you! |
| 04:14.40 | sawgood | Why is it IF I have core set debug 0 set (but then I launch SIP set debug on) I see these debug messages? |
| 04:14.50 | sawgood | are these messages different than core debug messages? |
| 04:15.48 | titter | I don't recall off the top of my head, sip debugging turned on should only show sip debug |
| 04:16.19 | sawgood | it worked ... thank you! I now see the SIP messages in /var/log/asterisk/messages |
| 04:17.08 | sawgood | Now, when I have to send out a SIP message dump, I have a process in place |
| 04:17.13 | sawgood | this will work just fine! |
| 04:17.21 | titter | You can also log the CLI to a file, you can edit the console line to console => notice,warning,error,debug,verbose, then start the CLI with asterisk -gnrT | tee /tmp/cli-capture.txt and issue a logger reload then turn up verbosity and turn on debug, and the CLI will output to that txt file |
| 04:17.58 | titter | Either works, just if you use messages remember to remove the debug line because debugging can make logs grow pretty quick |
| 04:18.11 | iceman85 | oh yeah |
| 04:18.31 | iceman85 | titter, do you have any ideas about the question i asked? |
| 04:18.33 | sawgood | neat thank you |
| 04:19.54 | titter | iceman never seen that happen before, what version of *? |
| 04:20.01 | *** join/#asterisk fofware (~Fabian@host186.190-225-12.telecom.net.ar) |
| 04:20.22 | iceman85 | 1.2.6 |
| 04:21.04 | iceman85 | its a legacy system that i am being asked to support. |
| 04:21.45 | jbroome | wow |
| 04:22.09 | jbroome | i thought my 2.0.x was bad |
| 04:22.16 | iceman85 | ??? |
| 04:22.27 | titter | iceman85: eek, that one I honestly don't remember ... thats can old version of zaptel that I probably won't remember much of |
| 04:22.43 | titter | iceman85: is there a reason you cannot upgrade the system? |
| 04:23.01 | iceman85 | yeah I don't remember what version of zaptel it is. |
| 04:23.42 | iceman85 | upgrading is a possibility, but I'm concerned that an upgrade may break things. |
| 04:24.18 | iceman85 | unfortunately the people i'm working with rely more heavily on their phones than they do on their computers. |
| 04:24.38 | titter | iceman85: sip phones?> |
| 04:24.48 | iceman85 | yes. Aastra 9133i |
| 04:25.43 | iceman85 | the weird thing is, i can call my cell, and dial any number and hear the tones immediately as I press them |
| 04:25.50 | titter | iceman85: In the sip.conf what is dtmfmode set to? |
| 04:25.58 | jbroome | iceman85: that's why i'm building the replacement alongside the current grandpa server |
| 04:26.16 | titter | I also don't even know if dtmfmode is an option in that version of asterisk |
| 04:26.20 | iceman85 | let me look...@jbroome that's a good idea |
| 04:26.20 | jbroome | if things will stop breaking long enough for me to work on it |
| 04:26.27 | iceman85 | i think so.... |
| 04:26.35 | titter | Ya it was |
| 04:26.55 | sawgood | titter thanks again for the CLI logging tip (worked perfectly)! |
| 04:27.41 | titter | Ya the best bet for you is either build another server for them and just mimic the dialplan, or if hardware is an issue get some new drives and pull the old ones to rebuild the system off hours, and make sure everything works. If something fails you can pop in the old drives, and get them back up and then wait till off hours to figure out what was wrong |
| 04:28.04 | titter | sawgood: np |
| 04:28.21 | iceman85 | that may be |
| 04:28.31 | iceman85 | I'm thinking about converting them to a newer TB system |
| 04:28.48 | iceman85 | sip_additional.conf -- dtmfmode=rfc2833 |
| 04:29.08 | titter | iceman85: Are you using ulaw or alaw? |
| 04:29.20 | sawgood | is the CLI command: sip show channels (the best way) to determine if there are any 'active' calls on the Asterisk box? |
| 04:29.27 | raddy | Hello Everybody |
| 04:29.31 | iceman85 | ulaw i think |
| 04:29.43 | raddy | I have tried plugging the card in different slots too :) |
| 04:29.44 | titter | You can try setting it to inband and the device will send the dtmf |
| 04:30.04 | iceman85 | you mean the device will send the tones |
| 04:30.09 | raddy | It then shared the irq with the SMBus controller. |
| 04:30.10 | titter | yes |
| 04:30.11 | iceman85 | the device as being the phone |
| 04:30.16 | titter | correct |
| 04:30.19 | iceman85 | ok |
| 04:30.32 | iceman85 | I can try that, since that's a per extension thing. |
| 04:30.52 | iceman85 | however, i only have the problem with the * and the # keys |
| 04:31.09 | iceman85 | right now, I can dial numbers all day long and never miss a beat. |
| 04:31.20 | titter | sawgood: core show channels verbose |
| 04:32.01 | titter | iceman85: Ya I really don't know, it couldof been a bug, or something else .. that is a really old version of Asterisk. |
| 04:32.25 | iceman85 | because if i want to get the system to actually pass a * key or a # key to the PRi |
| 04:32.45 | iceman85 | I have to hit the key twice, and that's even a gamble as to whether or not it will pass it once, twice or at all... |
| 04:33.19 | titter | Try inband and let the phone do the dtmf ... might work, maybe zap had a bug or something is weird with the signalling |
| 04:33.32 | iceman85 | perhaps |
| 04:33.44 | raddy | Anybody listening? |
| 04:33.52 | iceman85 | I'm currently running a TB CE system in my office, |
| 04:34.02 | iceman85 | which is running asterisk 1.6 i think... |
| 04:34.08 | titter | raddy: what was the question? |
| 04:34.43 | iceman85 | I'll try that out and see if I get the same issue. |
| 04:34.52 | titter | ya its a quick test/fix |
| 04:35.05 | titter | I am checking out for a bit, bbl |
| 04:35.20 | iceman85 | upgrading is a daunting tastk, but it will correct some other issues like phantom voicemails |
| 04:35.24 | iceman85 | thanks titter! |
| 04:36.36 | raddy | titter : i have a motorola wildcard based x100p card, the wcfxo driver is not detecting the card correctly and failing to initialize it, lspci shows the device is sharing irq with intel ide controller, but /proc/interupts shows the wcfxo driver is not active. |
| 04:37.05 | iceman85 | are you running this on a dell box? |
| 04:37.10 | raddy | titter : i have tried installing the card in different slots and tried lot of kernel parameters to no avail. |
| 04:37.57 | iceman85 | PCI or PCIe? |
| 04:38.50 | iceman85 | I have a suspicion that you're already long past this troubleshooting step, however, I've had issues with IRQ sharing on the PCI bus on some older Dell servers. |
| 04:39.09 | iceman85 | Our workaround was to change from PCI to PCIe because of better IRQ handling. |
| 04:39.13 | *** join/#asterisk newmyth (~newmyth@68-117-102-144.dhcp.eucl.wi.charter.com) |
| 04:39.41 | *** join/#asterisk lost_soul (~shawn@cpe-67-249-141-227.twcny.res.rr.com) |
| 04:41.22 | raddy | iceman85 : is it directed to me :) |
| 04:41.37 | iceman85 | yeah |
| 04:41.48 | raddy | iceman85 : I'll let you know, what type of card we have. |
| 04:42.46 | iceman85 | I had issues with choppy audio due to IRQ sharing between the PCI bus, the NICs, and the HDD controllers. |
| 04:43.16 | iceman85 | in most cases we could custom configure them by disabling other junk like the serial controllers and printer port, but in some cases that wasn't available. |
| 04:44.03 | iceman85 | bbl |
| 04:44.03 | newmyth | I'm using the API to originate a call, but I don't know what determines the DTMF mode for the outgoing call. I assumed it's the outgoing entry in sip.conf. Correct? |
| 04:44.45 | JacksLivr | hello. I am a n00b, but I have been reading and trying and I am stuck fairly early. I have a centos box with *1.6. I am following along with an * book and i cannot get my sip softphone to register. I have turned off selinux and iptables and matched the config for 2 different sip phones to sip.conf. I start it up and i never see the sip phone register and it keeps erroring out in the softphone side. what can i try? |
| 04:46.01 | newmyth | JacksLivr, do you have tshark installed? do you have a firewall on? |
| 04:46.15 | JacksLivr | firewall off, selinux off |
| 04:46.26 | JacksLivr | thanks for responding |
| 04:47.02 | newmyth | open a terminal window ->> #tshark port 69 |
| 04:47.24 | JacksLivr | 69? isnt that tftp |
| 04:47.35 | JacksLivr | i have to find/install tshark |
| 04:47.37 | newmyth | :) - yep sorry |
| 04:47.45 | newmyth | <PROTECTED> |
| 04:48.10 | JacksLivr | installing wireshark now |
| 04:48.36 | newmyth | also enter the console - can you see your phone trying to register? |
| 04:48.50 | JacksLivr | i dont see it register |
| 04:48.57 | newmyth | do you see anything? |
| 04:48.57 | JacksLivr | or try |
| 04:49.00 | JacksLivr | no |
| 04:49.03 | JacksLivr | verbose 10 |
| 04:49.05 | JacksLivr | still nothing |
| 04:49.10 | newmyth | hummm...... |
| 04:49.17 | newmyth | going for wine, back in a sec. |
| 04:53.05 | newmyth | well, if your not seeing something, its either configuration or network blockage. are you NATed |
| 04:53.17 | JacksLivr | client is natted |
| 04:53.28 | JacksLivr | <PROTECTED> |
| 04:53.31 | JacksLivr | <PROTECTED> |
| 04:53.34 | JacksLivr | you have helped me see that |
| 04:54.31 | *** part/#asterisk raddy (~raddy@117.192.231.16) |
| 04:56.17 | JacksLivr | [Nov 16 06:18:17] NOTICE[3505]: chan_sip.c:21751 handle_request_register: Registration from '<sip:iphone@sip.myserver.com>' failed for '166.137.8.130' - No matching peer found |
| 04:57.23 | newmyth | So you have setup both sip.conf and extensions.conf to match your client info? what fields are populated in the registration section of your sortphone? |
| 04:58.51 | JacksLivr | i am working through a book. i dont have anything but a hello world extension in the extensions.conf yet |
| 04:59.06 | JacksLivr | smartphone |
| 05:00.45 | JacksLivr | username/password/domain/displayname are tall that is set |
| 05:01.09 | newmyth | ok, you need to setup an extenstion in there with a user (exten) and password, then set the softphone likewise. if you are using a default setup, then check to make sure extensiosn.conf ......... |
| 05:01.21 | newmyth | ok, what are your softphone settings? |
| 05:02.33 | JacksLivr | username and password match sip.conf and domain is * server |
| 05:03.10 | JacksLivr | its weird. only sometimes do i see any activity in the cli |
| 05:03.17 | newmyth | what does extension.conf have for your extension? |
| 05:04.18 | JacksLivr | well, it doesnt yet. i was gonna add something cause you suggested it. the book has that in the next part. im just trying to call hello wor;d right now |
| 05:05.29 | rbht | JacksLivr: why not use FreePBX instead? |
| 05:05.40 | JacksLivr | i want to learn * |
| 05:06.52 | newmyth | I don't have the book handy, and I don't know the orignal setup. getting started usually requires three changes. sip.conf, extensions.conf, and phone registration settings. |
| 05:12.34 | newmyth | anyone have experience with double DTMF using the API? |
| 05:20.45 | *** join/#asterisk atan (~atan@unaffiliated/atan) |
| 05:21.14 | *** part/#asterisk newmyth (~newmyth@68-117-102-144.dhcp.eucl.wi.charter.com) |
| 05:26.22 | *** part/#asterisk mick_laptop (~mick@clamwin/admin/mickhome) |
| 05:26.32 | *** join/#asterisk bmg505 (~leon@196-209-120-122.dynamic.isadsl.co.za) |
| 05:27.48 | *** join/#asterisk Denial- (Denial@drgi.co.uk) |
| 05:27.52 | *** join/#asterisk m_tadeu (~quassel@89.181.97.241) |
| 05:37.01 | *** join/#asterisk Defraz (~Defraz@gump.fuzecore.com) |
| 05:44.09 | *** join/#asterisk rbht (~kvs@c-71-201-93-166.hsd1.il.comcast.net) |
| 05:46.06 | *** join/#asterisk raddy (~raddy@117.192.231.16) |
| 05:46.15 | raddy | Hello Everybody |
| 05:46.50 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 05:49.38 | ChannelZ | Hi Dr. Nick! |
| 05:50.02 | *** join/#asterisk rbht (~kvs@c-71-201-93-166.hsd1.il.comcast.net) |
| 05:50.17 | *** join/#asterisk defswork (~andy@mx2.3gcomms.co.uk) |
| 06:03.48 | atan2 | If a VOIP phone includes "UK Power Supply" does this mean I can just get a US power supply to use it in North America? |
| 06:03.58 | atan2 | Or would the phone itself be somewhat different =\ |
| 06:07.48 | ChannelZ | well SIP is SIP whether you speak the Queen's english or American english, so probably yeah it would work if the whole power adapter thing did |
| 06:11.55 | atan2 | ChannelZ, do you know much about SIP video? =) |
| 06:18.44 | *** join/#asterisk magwas (~mag@mag.webhome.hu) |
| 06:19.30 | magwas | I have problem registering my asterisk to ekiga.net |
| 06:19.54 | magwas | It seems asterisk thinks I am registered |
| 06:20.14 | magwas | but when ekiga.net users try to call me, they recevie "unavailable" |
| 06:20.41 | magwas | what should I check with my asterisk config? |
| 06:21.53 | atan2 | magwas, wild guess here: extensions |
| 06:22.00 | atan2 | Did you setup some inbound route to connect the calls? |
| 06:22.07 | atan2 | If you didn't I would guess you're SOL. |
| 06:22.23 | magwas | yep. when others call my asterisk directly they can call me |
| 06:22.35 | ChannelZ | that doesn't mean anything |
| 06:22.48 | atan2 | magwas, yeah but it needs to detect ekiga calls |
| 06:22.51 | atan2 | Not just @yourasterisk |
| 06:23.03 | ChannelZ | If you have no matching sip peer for ekiga, or it's going to the wrong context... there can be a dozen things wrong |
| 06:23.24 | magwas | I have a matching sip peer, let's see whether is it correct |
| 06:23.26 | atan2 | Hell if I know what you need to change but I know Asterisk is picky for that stuff. But in its defense it always does what it's told!! |
| 06:23.33 | atan2 | Although I have one _strange_ ass thing happen to me earlier!! |
| 06:23.51 | atan2 | I called a long distance number in BC, and my Asterisk connected me to my outbound trunk to a number I called earlier in the day! |
| 06:23.57 | atan2 | I was so, so confused. |
| 06:24.02 | *** join/#asterisk espiceland (~erin@nat/digium/x-ghwmmnoadkybkwgj) |
| 06:25.15 | *** join/#asterisk seanjohn (~seanjohn@gateways.sheltoncomputers.com) |
| 06:25.20 | ChannelZ | logs or it didn't happen |
| 06:25.24 | magwas | my ekiga peer stance from sip.conf: http://pastebin.com/Qm28vx5S |
| 06:26.28 | ChannelZ | Do you see anything on the console (with verbose turned up a little) when someone tries to call or you get aboslutely nothing? |
| 06:27.02 | magwas | I see absolutely nothing. I guess the problem maybe before that, with registration |
| 06:27.36 | magwas | I have this for registration: register => magwas:mypass@ekiga.net |
| 06:27.42 | ChannelZ | well either they're not attempting to send you anything for whatever reason (their end) or it's going to the wrong place or you're firewalled off |
| 06:28.22 | magwas | I do have packet filter rules on my asterisk box, but nothing get caught by the packet filter either |
| 06:29.03 | magwas | sip show registry tells me this: |
| 06:29.05 | magwas | ekiga.net:5060 magwas 585 Registered Tue, 16 Nov 2010 01:10:21 |
| 06:29.33 | magwas | now you can try to call magwas@ekiga.net, and see what happens |
| 06:30.47 | magwas | interestingly I have another registration, to freecall.com, and I see these lines in the log: |
| 06:30.59 | magwas | <PROTECTED> |
| 06:31.13 | magwas | I do not see such lines for ekiga.net |
| 06:31.14 | ChannelZ | usually implies a communication error |
| 06:31.58 | magwas | so it should mean I do have communication error talking to freecall.com, but not with ekiga.net? |
| 06:32.20 | ChannelZ | yes and/or no |
| 06:33.09 | ChannelZ | It's trying to send a registration packet to freecall.com but is getting no response. Which could be your fault (the packet never makes it out of your network, or their reply never makes it back in) |
| 06:33.59 | ChannelZ | With ekiga the reverse could be true; You register OK but when they attempt to send you a call the packet never makes it to you |
| 06:34.32 | ChannelZ | I can't call that magwas address, they reject me like I'm supposed to auth |
| 06:36.31 | magwas | what about user= and secret= in my [ekiga] section. Maybe I do not need that? |
| 06:37.10 | magwas | I have commented them out and did a reload. can you call me now? |
| 06:37.49 | ChannelZ | I dunno how Ekiga works, if they try to auth as anyone or expect you to just accept. But you'd get an error if that was the case |
| 06:38.23 | ChannelZ | ok something just spoke to me in.. I have no idea what language |
| 06:38.24 | magwas | good, you could call me I guess.... |
| 06:39.14 | magwas | it is hungarian. 31 is the echo test, and 2001 is me. could you call extension 2001? |
| 06:39.30 | ChannelZ | yeah but I have no headset hooked up so I can't talk to you |
| 06:40.44 | magwas | it would be nice to check whether voice goes through to both directions, just to make sure |
| 06:40.56 | ChannelZ | is it an IVR? am I supposed to dial 2001? |
| 06:41.07 | ChannelZ | it called me a slut and hung up |
| 06:41.27 | *** join/#asterisk Sipster (~Sipster@modemcable045.5-200-24.mc.videotron.ca) |
| 06:41.30 | magwas | it is an asterisk, and you are supposed to dial 2001 to call me. |
| 06:43.03 | magwas | thank you very much! |
| 06:43.07 | ChannelZ | heh. Guess it defaulted to my webcam since my thingy is unplugged |
| 06:43.53 | ChannelZ | oh duh I guess I could have called you from my analog phone. Oh well |
| 06:44.44 | magwas | the point is that we could test it I am not very picky on sound quality. I just mentioned the noise because I thought you don't have mic. |
| 06:45.09 | ChannelZ | noisy computers on what was essentially a speakerphone |
| 06:45.34 | ChannelZ | I really need to move my server into the closet |
| 06:47.31 | magwas | and now it seems I cannot call out to ekiga.net... |
| 06:47.34 | magwas | chan_sip.c:12412 handle_response_invite: Failed to authenticate on INVITE to '"Magosányi Ãrpád" <sip:2001@188.36.152.83>;tag=as7b372d4e' |
| 06:49.12 | magwas | for outgoing calls I need username and secret, but for incoming calls no |
| 06:49.27 | magwas | is there a setting that says incoming calls should not be authenticated? |
| 06:50.04 | ChannelZ | generally you'd just make two peers |
| 06:50.40 | magwas | with the same host stanza? |
| 06:51.37 | ChannelZ | or actually you could do insecure=invite |
| 06:52.14 | magwas | I try this one first |
| 06:52.54 | magwas | outgoing call works... |
| 06:54.14 | magwas | could we test incoming? I have modified my dialplan so that after timeout it calls my extension. |
| 06:54.33 | magwas | so you either dial 2001 or just wait |
| 06:54.39 | *** join/#asterisk dlirit (~lirant@80.74.100.10) |
| 06:56.01 | magwas | no, this fallback does not work... |
| 06:59.34 | *** join/#asterisk seanjohn (~seanjohn@gateways.sheltoncomputers.com) |
| 07:03.32 | *** join/#asterisk seanjohn (~seanjohn@96.43.138.87) |
| 07:07.43 | *** join/#asterisk mintos (~mvaliyav@nat/redhat/x-ignnzjeohmlyjcrj) |
| 07:10.33 | *** join/#asterisk schmidts (~schmidts@lmlo.sil.at) |
| 07:10.35 | schmidts | good morning |
| 07:14.22 | *** join/#asterisk timahvo1 (~rogue@41.223.57.72) |
| 07:18.11 | magwas | ChannelZ could you try incoming call once again, by calling magwas@ekiga.net? |
| 07:21.01 | magwas | I am trying to use ztdummy so that I can create conference rooms. But got this: |
| 07:21.03 | magwas | ioctl32(asterisk:26825): Unknown cmd fd(29) cmd(80044a2f){t:'J';sz:4} arg(f542efc8) on /dev/zap/timer |
| 07:21.35 | magwas | even compiling the stuff was tricky because I am on sparc, and one of the modules said there is no big endian support. |
| 07:22.39 | magwas | anyone here who could use ztdummy on a big endian machine, or have a hint where it went wrong? |
| 07:23.54 | kaldemar | magwas: is there a particular reason for you to use zaptel instead of DAHDI? you're missing out on a few years' bug fixes. |
| 07:24.42 | magwas | only reason that I could find this solution first |
| 07:25.09 | *** join/#asterisk Get_The_Fish (~Get_The_F@c-24-8-50-199.hsd1.co.comcast.net) |
| 07:25.47 | Get_The_Fish | hello, I just learned that I could do :make in VIM and debug puppet manifests, and I wondered if I could do the same with extensions.conf... |
| 07:25.48 | magwas | what should I do to have a time source with dahdi? |
| 07:25.51 | *** join/#asterisk seanjohn (~seanjohn@96.44.175.237) |
| 07:26.10 | Get_The_Fish | magwas, install dahdi hardware or use dahdi dummy |
| 07:26.37 | Get_The_Fish | cause, you know, that would be kewl |
| 07:26.39 | magwas | I guess I am aiming at dahdi dummy |
| 07:27.05 | Get_The_Fish | then you just need to install dahdi then. Configure it to use dahdi dummy. |
| 07:27.12 | WIMPy | Or use a more recent Asterisk with confbridge instead of meetme. |
| 07:27.27 | magwas | will it work on sparc? |
| 07:27.47 | Get_The_Fish | you will still need a timing source, whether it be dahdi, pthread, timerfd, etc. |
| 07:27.58 | magwas | I have 1:1.4.21.2~dfsg-3+lenny1 maybe it does have confbridge? |
| 07:28.07 | WIMPy | No idea, but if any of the internal yiming sources wor, you should be fine. |
| 07:28.18 | kaldemar | magwas: newest dahdi doesn't have a separate dahdi_dummy module anymore, the dahdi module is all you need. so you won't waste your time looking for the separate dummy module. |
| 07:28.43 | WIMPy | No. Confbridge appeared somewhere alons 1.6.2.x. |
| 07:29.13 | Get_The_Fish | magwas http://lmgtfy.com/?q=dahdi+sparc |
| 07:29.21 | kaldemar | magwas: your version of asterisk most likely doesn't support DAHDI. the first one to support it was 1.4.22. so you're stuck with zaptel unless they've backported the changes in debian. |
| 07:30.07 | magwas | pbx_extension_helper: No application 'ConfBridge' for extension (user, 32, 1) |
| 07:30.36 | magwas | then I should upgrade... |
| 07:31.00 | WIMPy | In both cases, yes. |
| 07:32.52 | magwas | looks into lenny-backports for newer asterisk |
| 07:33.45 | Get_The_Fish | magwas, why not upgrade to 1.6.x? |
| 07:33.49 | magwas | 1:1.6.2.9-1~bpo50+3 I guess this should do |
| 07:34.46 | *** join/#asterisk timahvo1 (~rogue@41.223.57.76) |
| 07:34.47 | *** join/#asterisk TobSnyder (~schneider@dslb-088-073-204-138.pools.arcor-ip.net) |
| 07:34.55 | TobSnyder | hello |
| 07:36.36 | TobSnyder | numbers of incoming calls are without leading zeros, what makes it difficult to see whether its an international or national call, for example...are there any settings that affect, how the numbers of incoming calls are "delievered" (using ISDN BRI) |
| 07:36.37 | TobSnyder | <PROTECTED> |
| 07:37.59 | kaldemar | TobSnyder: you have no control over what your telco sends you. contact them. |
| 07:42.56 | *** join/#asterisk seanjohn (~seanjohn@173.242.118.91) |
| 07:43.19 | *** join/#asterisk Tim_Toady (~moi@77.49.252.191.dsl.dyn.forthnet.gr) |
| 07:50.57 | magwas | I have learned that installing from lenny-backports needs -t lenny-backports option to apt... |
| 07:53.14 | TobSnyder | kaldemar: even not with options like pridialplan, prilocaldialplan, nationalprefix, internationalprefix in chan_dahdi.conf? |
| 07:54.58 | *** join/#asterisk hehol (~hehol@buero-gw.dortmund.loca.net) |
| 07:55.01 | magwas | thank you all, it seems it working. |
| 07:55.05 | Get_The_Fish | so, there isn't anything that could be used as an "asterisk compiler", something to help debug extensions.conf |
| 07:55.58 | magwas | anyone willing to test my new shiny conference room? call magwas@ekiga.net and dial 32 |
| 07:57.10 | magwas | Get_The_Fish: it depends how you interpret it. maybe you just use asterisk -vvvvvvvvvvvvvvvvvvvr , and do a reload? |
| 07:57.29 | Get_The_Fish | I was looking for something that I could call from vim |
| 07:57.33 | Get_The_Fish | like :make |
| 07:57.43 | *** join/#asterisk iscario (~quassel@LMontsouris-156-24-36-77.w217-128.abo.wanadoo.fr) |
| 07:58.03 | magwas | you can call asterisk -r, maybe through a Makefile |
| 07:58.10 | Get_The_Fish | I did that with a puppet manifest for the first time today and thought "my god that was a hell of a lot easier, why dont we do that for asterisk" |
| 07:58.13 | *** join/#asterisk AlHafoudh (~AlHafoudh@adsl-dyn210.78-98-252.t-com.sk) |
| 07:58.36 | Get_The_Fish | it won't parse for syntax errors though |
| 08:00.11 | Get_The_Fish | I just thought it was frigging cool. And it would save me a butload of time. |
| 08:00.26 | Get_The_Fish | I think that I am going to bring it up tomorrow when there are ppl here |
| 08:00.31 | magwas | invites everyone to his new conference room. call magwas@ekiga.net dial 32, and help me to test it |
| 08:00.31 | Get_The_Fish | perhaps I could bounty it |
| 08:01.15 | magwas | the problem that either there is a config test tool for ekiga which you can use or there isn't |
| 08:01.26 | magwas | s/ekiga/asterisk/ |
| 08:01.41 | Get_The_Fish | magwas@ekiga.net was not reachable from here |
| 08:02.15 | Get_The_Fish | Im probably not a good candidate though, I am on a vpn atm |
| 08:02.16 | magwas | omg. not again! |
| 08:02.38 | magwas | ekiga.net:5060 N magwas 585 Registered Tue, 16 Nov 2010 02:48:55 |
| 08:02.58 | magwas | I guess this means asterisk thinks I am registered |
| 08:03.14 | *** join/#asterisk waschtl (~waschtl@3ed8a58a.d.d9tcloud.de) |
| 08:05.37 | magwas | Get_The_Fish: you could use a clever combination of reload and dialplan show to see whether you extensions.conf worked. Of course If it wasn't, you have already loaded it:) |
| 08:06.27 | Get_The_Fish | exactly. Someone just needs to write a syntax parser. Shouldn't be that hard. |
| 08:06.36 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 08:07.15 | Get_The_Fish | tzafrir, any talk of a syntax parser for extensions.conf, something that could be used with VIM? |
| 08:08.25 | magwas | Get_The_Fish: as that is already implemented in asterisk. maybe a new option which parses config and does nothing? |
| 08:09.04 | Get_The_Fish | or a seperate script even.... which sure would be easier.... |
| 08:10.28 | magwas | Get_The_Fish: if you reuse asterisk's own config parsing code, the probability that the parser and asterisk will differ on what the config means will drop under 1.0 |
| 08:11.25 | magwas | it will be a PITA to maintain a separate parsing script |
| 08:13.05 | *** join/#asterisk tamiel (~tamiel@213.30.183.226) |
| 08:14.53 | tzafrir | magwas, not really. Much of the syntax parsing is done as part of the execusion |
| 08:15.24 | tzafrir | Get_The_Fish, isn't there some existing vim syntax file for asterisk? |
| 08:18.16 | WIMPy | TobSnyder: You could try to see if CALLERID(ton) is set. |
| 08:23.47 | magwas | tzafrir: you could inhibit actual execution with a config flag. of course it would mean lots of small changes all around |
| 08:24.06 | *** join/#asterisk seanjohn (~seanjohn@gateways.sheltoncomputers.com) |
| 08:27.05 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 08:45.20 | *** join/#asterisk sjobeck (~sjobeck@67.136.135.137) |
| 08:49.02 | *** part/#asterisk sjobeck (~sjobeck@67.136.135.137) |
| 08:54.14 | ChannelZ | hiccups |
| 09:01.05 | magwas | maybe a newbie question: how should I call a given extension within a sip url? for example what would be the sip url for my conference room? it is reachable by calling magwas@ekiga.net and dialing 32 |
| 09:02.43 | *** join/#asterisk Faustov (user@gentoo/user/faustov) |
| 09:03.29 | WIMPy | magwas is your extension at ekiga.net. No further options there. |
| 09:04.39 | magwas | WIMPy: does it mean it is impossible to refer to my conference room directly in a sip url? |
| 09:05.21 | WIMPy | No. It means that you can't use a user account of another PBX to do you own extensions. |
| 09:06.14 | WIMPy | You can create as many extension@your.domain.com as you like. |
| 09:11.42 | magwas | My actual problem is that I am behind NAT. I have actually installed asterisk as a gateway. And there is a specific conference room on another pbx which I have to regularly use. It have a 7 digit room PIN, and it seems PIN@conferenceroom does not work. So I thought I would tell asterisk to call that room directly from extensions.conf |
| 09:12.24 | ChannelZ | well it depends on how you normally call it |
| 09:13.32 | magwas | I normally call it on the uri 1008@ip.address and then dial the conference pin |
| 09:14.23 | atan2 | Err, how do I call 020 8892 3030 from North America? What would my dialplan rule look like? |
| 09:14.39 | atan2 | never calls outside of where he lives apparently |
| 09:15.51 | ChannelZ | magwas: well then you'd have to 'script' it by Dial()ing the normal conf, and then using like SendDTMF() to enter your pin if that's what you're trying to do |
| 09:16.22 | magwas | ChannelZ: thx, I guess google will help with the rest... |
| 09:17.13 | *** join/#asterisk sekil (~sekil@80.93.247.26) |
| 09:17.22 | atan2 | Or perhaps there's a default catch-all I can use that will catch a number of any length? |
| 09:18.01 | ChannelZ | atan2: probably Dial(Whatever/01102088923030) |
| 09:18.33 | atan2 | Err, I think I want . to match any number right? |
| 09:18.40 | atan2 | like _011. would be 011 < whatever > |
| 09:18.43 | *** join/#asterisk OlafsenM (~mono@193.198.31.85) |
| 09:19.39 | ChannelZ | oh you want the exten pattern.. then yes |
| 09:19.47 | atan2 | <PROTECTED> |
| 09:19.55 | atan2 | what's that sucker mean? my trunk hates me? =) |
| 09:19.55 | ChannelZ | depending on how restricted you want it to be |
| 09:20.12 | ChannelZ | Sort of like "wrong number" in this case |
| 09:20.45 | atan2 | Everyone is busy/congested at this time (1:0/1/0) |
| 09:20.45 | atan2 | I so did NOT tell Asterisk I was congested... but okay |
| 09:20.48 | ChannelZ | your ITSP might have international restricted (mine does, you have to specifically turn it on if you want to use it) and/or might have a specific dial string they want you to use |
| 09:21.06 | atan2 | ChannelZ, I had mine enable it so I could call strange areas around the world |
| 09:21.10 | atan2 | What a pile of paperwork that was |
| 09:22.05 | *** join/#asterisk _zoom_ (~user@41.218.32.129) |
| 09:22.09 | _zoom_ | hi |
| 09:22.36 | ChannelZ | howdy |
| 09:22.36 | _zoom_ | in extension.conf i need to filter calls based on IP and extension? |
| 09:23.08 | ChannelZ | what do you mean based on IP? Are these 'anonymous' SIP calls? |
| 09:25.03 | _zoom_ | yes |
| 09:26.22 | ChannelZ | well you could look at the ip with the SIPCHANINFO function and then write GotoIf logic or whatever based on what you see |
| 09:26.43 | _zoom_ | ok |
| 09:27.24 | atan2 | is 0114402088923030 a valid number to call? =\ |
| 09:27.29 | atan2 | Did I leave off a 0 on there somewhere? |
| 09:27.38 | phix | hi gang! |
| 09:27.54 | WIMPy | CHANNEL(peerip)? |
| 09:27.58 | phix | atan2: that looks like a UK number to me |
| 09:28.11 | phix | atan2: you are dialing from the US |
| 09:28.12 | phix | ? |
| 09:28.13 | ChannelZ | I guess it looks right |
| 09:28.19 | magwas | solution: The D option to the Dial command transmits DTMF tones, with a 'w' causing a pause: |
| 09:28.26 | magwas | Dial(<num>@<peer>,180,D(w*w1w3)) |
| 09:28.37 | atan2 | phix, yes from us/Can |
| 09:28.51 | phix | atan2: :D |
| 09:28.55 | _zoom_ | ChannelZ: using item peerip |
| 09:28.59 | _zoom_ | ? |
| 09:29.02 | atan2 | phix, but I can't get the call to go through =( pfft |
| 09:29.07 | phix | :( |
| 09:29.25 | phix | atan2: what does sudo asterisk -rvvvvdddd say when you are making the call? |
| 09:29.31 | WIMPy | _zoom_: CHANNEL(peerip) |
| 09:30.10 | ChannelZ | what he said |
| 09:30.31 | ChannelZ | atan2: ask your ITSP what form they expect international to be in |
| 09:30.33 | phix | atan2: I see the problem |
| 09:30.43 | atan2 | phix: missing 0? |
| 09:30.46 | phix | atan2: try dialing 011442088923030 |
| 09:31.09 | atan2 | SUCCESS!! |
| 09:31.12 | atan2 | the zero, eh |
| 09:31.30 | phix | no, with UK (and AU an dprobably NZ and other countries) you drop the first 0 as it is defined a trunk only valid for calling from the area, not dialing in from internatiional lines |
| 09:31.35 | phix | :) |
| 09:32.44 | ChannelZ | Now you can get all the british phone sex you can afford! |
| 09:32.54 | phix | :D |
| 09:33.21 | atan2 | ^_^ |
| 09:33.30 | phix | British phone sex is more popular than porn, unless the "actors" don't open their mouth at all |
| 09:33.37 | atan2 | I think I called the wrong number there once |
| 09:33.52 | atan2 | Some poor sap in Twickenham must be so confused right now |
| 09:34.02 | phix | atan2: Give my Grandma a call for me and pretend it is me :P it will make her day :) |
| 09:34.18 | *** join/#asterisk ickmund (~ickmund@cli-5b7e85e4.bcn.adamo.es) |
| 09:34.33 | atan2 | Awww, the poor girl :P |
| 09:34.37 | phix | hehe |
| 09:34.39 | atan2 | beats you |
| 09:34.43 | atan2 | so spend time with your grandmother! |
| 09:34.44 | phix | :) |
| 09:34.51 | phix | I do every 2 yrs or so |
| 09:35.02 | atan2 | "Hi grandma! It's Phix!" |
| 09:35.07 | atan2 | "Who?" |
| 09:35.49 | phix | heh |
| 09:35.55 | phix | no no that is my real name :) |
| 09:36.10 | atan2 | I'm so sorry |
| 09:36.12 | phix | hehe |
| 09:36.19 | atan2 | So you're in the UK then? |
| 09:36.37 | phix | no, I live about 10,600 Miles from the UK |
| 09:37.07 | atan2 | I am confused why these folks don't answer the phone ( http://www.trustedgoods.com/aboutus.asp ) |
| 09:37.15 | atan2 | It's about 9am their time, no? almost 10? |
| 09:37.30 | aberrios_ | 9:37am |
| 09:37.46 | phix | ummm, daylight savings is over so they are +-0 UTC :) |
| 09:37.53 | ChannelZ | Maybe they're like government workers and don't open til 10 |
| 09:38.11 | phix | yeah 9:38AM |
| 09:38.14 | atan2 | Now I just need to find the suckers... http://maps.google.com/maps?f=q&source=s_q&hl=en&geocode=&q=30+Rugby+Road,+Twickenham,+TW1+1DG&sll=37.0625,-95.677068&sspn=49.844639,114.169922&ie=UTF8&hq=&hnear=30+Rugby+Rd,+Twickenham+TW1+1DG,+United+Kingdom&ll=51.455946,-0.339203&spn=0.009654,0.027874&z=16&layer=c&cbll=51.455824,-0.339071&panoid=MGW7Ao88NB-DP-XlM_00RQ&cbp=12,140.61,,0,18.58 |
| 09:38.19 | *** join/#asterisk dandate2 (~gtejkgjke@124.6.157.210) |
| 09:38.31 | dandate2 | hey anyone else having a problem of voicemail not being emailed? |
| 09:38.35 | phix | dandate2: no, it works great for me |
| 09:38.38 | aberrios_ | nope |
| 09:38.44 | dandate2 | damn what could that be |
| 09:38.46 | aberrios_ | checked your maillogs? |
| 09:38.52 | phix | dandate2: typo? |
| 09:39.23 | atan2 | What the hell... if I drove down a road in the UK like this I would be pulled over for being drunk http://tinyurl.com/23xkdcr |
| 09:39.29 | phix | firewall? mail server settings? air pressure? gravity? |
| 09:39.34 | dandate2 | u know it worked for me in the distant past, and in the not so distant past i saw it was blocked by hotmail etc. , what can i whitelist ? |
| 09:39.53 | phix | erh you get it delivered to a hotmail account? |
| 09:40.04 | dandate2 | no no that was in the not so distant past, right now trying to send to a godaddy account |
| 09:40.08 | phix | why not run your own mail server and get it delivered there? |
| 09:40.13 | ChannelZ | death to GoDaddy |
| 09:40.27 | phix | running stuff locally for the win |
| 09:40.32 | aberrios_ | get postfix installed |
| 09:40.40 | phix | yup |
| 09:40.44 | phix | or exim4 |
| 09:40.47 | dandate2 | i would have dedicated resources to a mailserver but it seems all major providers have blocked outgoing mail from them. and blocked all incomming mail from asterisk heh |
| 09:40.54 | phix | sudo apt-get install exim4-daemon-heavy done |
| 09:41.12 | ChannelZ | in any event it needs to call a delivery agent of some sort, whether that's sendmail, or msmtp, etc. so either something is not working on your end or it is and is being rejected elsewhere |
| 09:41.27 | phix | dandate2: I use a smart host (my ISP) for outgoing, it doesnt get blocked |
| 09:41.51 | aberrios_ | its probably using sendmail atm, have you checked whats in /var/log/maillog? |
| 09:42.32 | phix | sudo tail -f /var/log/{mail,syslog}* |
| 09:42.41 | magwas | if you want a lightweight send-only MTA, I suggest ssmtp |
| 09:42.44 | phix | then leave a voice mail |
| 09:42.57 | ChannelZ | revision: Death to GoDaddy and sendmail |
| 09:42.58 | phix | magwas: or exim4-daemon-light |
| 09:42.59 | Get_The_Fish | magwas, thanks for that |
| 09:43.04 | phix | ChannelZ: agreed |
| 09:43.22 | phix | Although sendmail was great in the 70s and early 80s |
| 09:43.32 | aberrios_ | anyone here taken RHCE and dCAP? If so which was harder? |
| 09:43.37 | phix | or was it the 80s and early 90s *shrugs* |
| 09:43.48 | ChannelZ | I think it was in the nevers |
| 09:44.03 | phix | aberrios_: nope, I am not qualified to use Linux :( |
| 09:45.26 | Maliuta_ | sendmail rocked well into the '90s ... it only failed once Postfix came along |
| 09:46.17 | phix | hmmm I think I used qmail before postfix |
| 09:46.36 | ChannelZ | I still use qmail |
| 09:46.43 | phix | then moved to exim due to work, now they have gone to postfix I need to start using that again :) |
| 09:46.47 | Maliuta_ | DJB blows goats |
| 09:46.58 | phix | Back in the slackware days :)| |
| 09:47.00 | ChannelZ | Even in it's clobbered-together state it beats the shit out of that trainwreck sendmail |
| 09:47.07 | phix | ah those were the days |
| 09:47.44 | magwas | phix: when programmers were real programmers. or was it the sixties? |
| 09:48.03 | phix | I think I used sendmail for about an hour, the fact that not only the config file was crypted but oyu had to compile it afterwards |
| 09:48.37 | ChannelZ | A config file written in a scripting language.. neither of which make sense to normal human beings |
| 09:49.10 | phix | where programming meant you had to use a jacks, valves, soildering irons and 3 phrase power :) |
| 09:49.12 | WIMPy | ChannelZ: What do you call your extensions.conf then? |
| 09:49.46 | phix | ChannelZ: erh the new grub annoys me with that shit |
| 09:49.48 | ChannelZ | or not really a scripting language.. whatever the hell it was |
| 09:49.49 | magwas | channelz: writing a config file in a script language is a good idea in itself. but you should choose a well-known one, and provide appropriate tools |
| 09:49.53 | phix | WIMPy: easy to follow |
| 09:50.25 | phix | nothing wrong with apache style config file or even xml :) |
| 09:50.45 | WIMPy | I never understood why people actually used that m4 shit. Far easier to just edit sendmail.cf yourself. |
| 09:51.01 | phix | WIMPy: wasnt it a requirement to do that? |
| 09:51.10 | ChannelZ | It was easier to buy stamps |
| 09:51.12 | WIMPy | Not at all. |
| 09:51.12 | phix | or was that just for optimisation? I forget now |
| 09:51.16 | phix | haha |
| 09:51.17 | phix | :) |
| 09:51.28 | WIMPy | It was just one of those frontends that "make things easier". |
| 09:51.29 | dandate2 | SN: Data format error |
| 09:52.07 | WIMPy | It just wasn't an X11 app like now. |
| 09:52.25 | WIMPy | But they often tent to "make things easier" as well. |
| 09:53.00 | WIMPy | m4 and sendmail are like freepbx and Asterisk. |
| 09:53.24 | ChannelZ | Rude |
| 09:53.36 | magwas | now. I have an ISDN NT, and a sun netra T1 running asterisk. I guess the only common interface between them right now is rs232. any suggestions to make them happy with each other? |
| 09:54.29 | WIMPy | magwas: You want to connect a BRI to your Sparc box? |
| 09:54.49 | magwas | WIMPy: I am thinking about that |
| 09:55.17 | WIMPy | magwas: That sounds like a real challange to me. |
| 09:55.33 | ChannelZ | Maybe try string. |
| 09:55.37 | magwas | maybe I could do something about fax and sms with only a rs232 cable |
| 09:55.54 | WIMPy | magwas: Sure. |
| 09:56.04 | phix | ah ok, well back to playing two worlds II now :D |
| 09:56.08 | magwas | but I guess I need some more hw to get voice |
| 09:56.26 | WIMPy | Well, if you've got the right TA. |
| 09:56.29 | ChannelZ | wanders off to bed |
| 09:57.07 | WIMPy | No you need the abandoned channel driver for ttys. |
| 09:57.08 | ChannelZ | oh did your insecure thing work for ekiga? I ran off. |
| 09:57.21 | WIMPy | Can't remember what it was called. |
| 09:58.01 | magwas | channelz: I _think_ it did, but no one have called me since that. and I have a shiny new conference room at extension 32. would you like to test it? |
| 09:58.45 | ChannelZ | it answered, I'm tired, bed. |
| 09:59.02 | magwas | channelz: thx |
| 10:01.38 | *** join/#asterisk mpe (~mpe@2.105.219.255) |
| 10:02.00 | magwas | now I am here with messages: Received message to <sip:0623555084@10.43.43.242> from <sip:2001@10.43.43.242>;tag=56e2e854-d6ef-df11-869c-0022fa951dc4, dropped it... |
| 10:03.36 | magwas | I guess first I have to figure out how sms works in asterisk at large... |
| 10:03.45 | dandate2 | so i enabled sendmail, now it goes to yahoo bulk folder regardless of whitelisting vm@asterisk, and does not appear at all in godaddy |
| 10:04.44 | dandate2 | i guess its only a matter of time it doesnt appear at all in yahoo also |
| 10:05.45 | dandate2 | considering how dangerous voicemail spam is... |
| 10:06.39 | WIMPy | spit via spam :-) |
| 10:08.39 | dandate2 | pornographic audio recordings via asterisk to the yahoo bulk folder. |
| 10:09.00 | dandate2 | mabye i can setup voicemail blasting for this |
| 10:13.55 | *** join/#asterisk krion (~seb@unaffiliated/krion) |
| 10:16.03 | magwas | any suggestions for RTFM regarding pushing faxes through asterisk between two sip endpoints? |
| 10:16.32 | magwas | any suggestions for fax-aware open source sip softphone? |
| 10:28.14 | *** join/#asterisk csnook (~chris@209-6-38-66.c3-0.smr-ubr1.sbo-smr.ma.cable.rcn.com) |
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| 10:58.02 | *** join/#asterisk np98ig (~Puppet@bl20-153-170.dsl.telepac.pt) |
| 10:58.52 | np98ig | hi there. can please someone help me with custom device states? |
| 11:04.05 | *** join/#asterisk bjornts (~Adium@eduroam-193-157-113-148.uio.no) |
| 11:08.31 | *** join/#asterisk gabdrach (~Adium@186.12.53.170) |
| 11:26.58 | magwas | np98ig: you might state your actual problem hoping that someone can help |
| 11:29.09 | np98ig | I have a cluster of Asterisk nodes and i want to replicate extension states. When a device changes the state (for example it is busy) i want to receive a notification of this event and replicate it to all the other Asterisk nodes |
| 11:30.23 | np98ig | I applied the func_devstate to the Asterisk and I use the AMI to receive notifications of these events. Then I send them to all the other Asterisk nodes |
| 11:30.36 | WIMPy | You should look into that XMPP stuff then. |
| 11:31.13 | np98ig | my problem is the extensions.conf configuration. I can change the state of an extension using the devstate change command |
| 11:31.40 | np98ig | but when i just use a softphone, i don't receive any notification about its state |
| 11:32.15 | *** join/#asterisk |Physis| (~|Physis|@187.60.42.210) |
| 11:36.50 | np98ig | what do i have to change on the extensions.conf file? |
| 11:56.34 | *** join/#asterisk oelewapperke (wapper@85-158-215-1.powered-by.benesol.be) |
| 11:56.42 | oelewapperke | how do I match a single digit phonenumber ? |
| 11:57.03 | oelewapperke | I've been trying to do this : exten => _X,1,Agi(agi://127.0.0.1:4848/DestinyIVR.agi) |
| 11:57.15 | oelewapperke | but it won't match (it won't even show up in "dialplan show" command) |
| 11:57.41 | *** join/#asterisk E-bola (~bola@188.120.76.228) |
| 12:00.35 | WIMPy | Yes, _X matches any single digit. |
| 12:00.56 | WIMPy | If your extension doesn't show, do a dialplan reload and watch for errors. |
| 12:31.46 | *** join/#asterisk z4nD4R (~zandar@pc-asnis9n5kqq6tftrd9hvs487uron66k.usr.iklub.sk) |
| 12:32.48 | z4nD4R | hi, i need connect my asterisk to pstn, wich devices recomment me to use? thx... |
| 12:38.50 | *** join/#asterisk puzzled (~foobar@puzzled.xs4all.nl) |
| 12:39.54 | E-bola | Can anybody think of a way to create my dialplan so i can dial 2 different extensions with 2 different callerid's for the same actual call |
| 12:40.17 | schmidts | E-Bola why not use local channels? |
| 12:40.35 | E-bola | schmidts: I thought of that, but doesnt things still happen serially? |
| 12:40.51 | E-bola | ohh or do u mean dial 2 local channels |
| 12:40.57 | E-bola | and have them behave in 2 different ways? |
| 12:40.58 | schmidts | yep ;) |
| 12:41.10 | E-bola | Cool, i allmost had the solution just needed the last input :P |
| 12:41.12 | schmidts | DIAL(LOCAL/123&LOCAL/456) |
| 12:41.35 | *** join/#asterisk Mhaddog_ (~Mhaddog@z65-50-116-17.ips.direcpath.com) |
| 12:42.46 | E-bola | How about for queues? Thats more tricky.... |
| 12:42.54 | E-bola | Have different agents have different callerid's |
| 12:43.00 | E-bola | (for incoming calls that is) |
| 12:43.41 | schmidts | that will not work, you have to split the queues up for each callerid |
| 12:43.51 | schmidts | or you will have 2 calls in the queue for all agents |
| 12:44.48 | z4nD4R | <PROTECTED> |
| 12:47.40 | *** join/#asterisk ruyo (~psantos@a83-132-152-91.cpe.netcabo.pt) |
| 12:54.09 | luckman212 | hey guys. this isn't really asterisk-specific but- you guys see the weird characters displayed in this screenshot? http://i.imgur.com/WcYNY.png |
| 12:54.29 | luckman212 | anyone know what settings I need in the terminal emulator to get those chars to display properly? |
| 12:54.46 | luckman212 | they should be those "DOS" style box characters I guess |
| 12:55.30 | Tim_Toady | centos? |
| 12:55.52 | luckman212 | yes the remote machine is centos- but I am connecting to it from a windows terminal |
| 12:55.55 | Tim_Toady | try running unset LANG |
| 12:56.34 | luckman212 | Tim_Toady: ah ha! that worked! |
| 12:56.54 | luckman212 | holy crap I banged my head for about 45 mins before giving up on that |
| 12:57.02 | luckman212 | cheers mate |
| 12:57.06 | Tim_Toady | :> |
| 12:57.50 | luckman212 | if I add SET= to my .bashrc that should "stick" right? |
| 12:57.58 | luckman212 | I meant LANG= |
| 12:58.15 | Tim_Toady | yeah |
| 12:58.23 | Tim_Toady | or .profile |
| 12:58.39 | luckman212 | ok thanks again |
| 13:00.08 | *** join/#asterisk n0cturnal (~quassel@piksta.net) |
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| 13:31.25 | Godfather_ | hi |
| 13:34.47 | *** join/#asterisk ik_5 (~ik@DSL212-143-222-125.bb.netvision.net.il) |
| 13:35.17 | Godfather_ | i'm trying to set up cdr, i installed asterisk addons 1.6.2.2, but i cant do a "cdr mysql status", any ideas? http://pastebin.com/hWVXhRnE |
| 13:43.28 | *** join/#asterisk Iamnacho (~Iamnacho@ip174-70-139-208.ks.ks.cox.net) |
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| 13:51.32 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 14:01.05 | z4nD4R | if i sniff srtp packets, what i shold see? |
| 14:02.07 | pabelanger | z4nD4R: encrypted packets? |
| 14:03.08 | z4nD4R | pabelanger: oc coufrsee.. but wireshrakt have no capabilty to SRTP... |
| 14:04.47 | z4nD4R | pabelanger: is this ok? http://www.upnito.sk/imggal.php?viewGallery=9805fd51ebb34cfae41b7ce1d8ad050e |
| 14:07.22 | pabelanger | z4nD4R: use wireshark to listen to the RTP stream, if you hear static, it is encrypted |
| 14:07.56 | z4nD4R | pabelanger: but how i can wireshark use to listen RTP stream? |
| 14:08.13 | Phlog | z4nD4R: might be a chance to show decode srtp traffic in wireshark |
| 14:08.19 | Phlog | just add the keys |
| 14:09.36 | pabelanger | z4nD4R: http://wiki.wireshark.org/RTP_statistics |
| 14:11.14 | Phlog | pabelanger: z4nD4R there is a simpley way to follow and listen to rtp in wireshark |
| 14:11.19 | Phlog | get the latest version |
| 14:11.27 | Phlog | then Telephony -> VoIP Calls |
| 14:11.30 | Phlog | Select a call |
| 14:11.35 | TobSnyder | someone knows if the ringtones are somewhere stored as file |
| 14:11.38 | Phlog | klick player -> decode |
| 14:11.52 | Phlog | you can listen to both directions directly from WS |
| 14:11.53 | TobSnyder | e.g. when initiating a call and waiting for the other party to take the phone |
| 14:13.09 | *** join/#asterisk Quintana (~sylvain@aghnar.doowan.net) |
| 14:19.53 | *** join/#asterisk lost_sou1 (~noymfb@cpe-67-249-141-227.twcny.res.rr.com) |
| 14:25.30 | *** join/#asterisk PMantis (~sswitzer@cpe-74-74-216-216.rochester.res.rr.com) |
| 14:27.08 | *** join/#asterisk ariel_ (~chatzilla@63.214.236.169) |
| 14:27.21 | PMantis | Hello, I have a customer that's in an old hard-to-wire building, and wants a phone system. Anyone know of a "digital" phone that supports paging (intercom) and works with Asterisk but uses existing 2-wire phone jacks? |
| 14:32.16 | trollasaurus | You could try to do 10BASE2 signalling over the wiring, but it may not work out well for you. |
| 14:33.39 | Katty | :> |
| 14:33.48 | Katty | HAPPY TUESDAY ALL YOU WONDERFUL PEOPLE! |
| 14:34.05 | Phlog | hi Katty the same to you |
| 14:35.00 | z4nD4R | Katty: u2 |
| 14:36.20 | trollasaurus | You could also probably do half-duplex 10BASE-T over it, which might actually be easier :-P |
| 14:36.31 | Katty | Phlog: :>>> |
| 14:36.38 | Katty | z4nD4R: <3 |
| 14:36.40 | PMantis | trollasaurus, I think it's highly subjective to what kind of cable is there. There's a 25 y/o phone system in there that failed |
| 14:36.50 | Katty | crittercam is absolutely buzzing this morning |
| 14:36.53 | Katty | infobot: crittercam |
| 14:36.53 | infobot | crittercam is, like, Katty's live broadcast of The Nut House at http://www.ustream.tv/channel-popup/squirrel-critter-cam |
| 14:37.06 | PMantis | trollasaurus, it might not be "twisted" pair at all. :( |
| 14:37.46 | trollasaurus | Indeed. it is likely that it is not for atleast part of the way, but you can't have everything... |
| 14:37.56 | PMantis | trollasaurus, I'm talking myself out of running IP over this, ESPECIALLY for a customer. |
| 14:38.12 | trollasaurus | You could do WiFi :-) |
| 14:38.30 | PMantis | trollasaurus, Hah, for an IP desk/wall phone? :) |
| 14:38.51 | trollasaurus | Yep, give each one of them a client AP (cheap Cisco APs are $40 USD) |
| 14:38.58 | trollasaurus | err, Linksys/Cisco |
| 14:39.09 | Linuturk | o/ |
| 14:39.33 | *** join/#asterisk ^^gurami^ (~gurami@173-13-161-26-sfba.hfc.comcastbusiness.net) |
| 14:40.03 | Godfather_ | i'm trying to set up cdr, i installed asterisk addons 1.6.2.2, but i cant do a "cdr mysql status", any ideas? http://pastebin.com/hWVXhRnE |
| 14:40.38 | PMantis | trollasaurus, This is a 7,200 sq ft house, and the existing wifi solution breaks up with the 3 XBox's they have, plus the 2 playstation 3's. I don't think I can trust voice over wifi in that house. :( |
| 14:41.27 | trollasaurus | You would need 3 network-side APs hard-wired, but they could be placed in more easily wired locations |
| 14:41.43 | trollasaurus | (Well, probably 3, no more than 5 depending on the actual configuration of the house) |
| 14:42.51 | trollasaurus | And it would only need 2 channels (for 3 APs), leaving the existing wifi network alone on its channel |
| 14:43.19 | PMantis | trollasaurus, I have a 4 AP WDS installed there right now. Unfortunately, the hard to wire area is also an area with lots of brick, and will likely be unreliable for wireless, too. |
| 14:43.48 | PMantis | trollasaurus, It's a shame that a 25 y/o phone system can offer all the features they need over this old wiring, but I can't replicate it without rewiring. |
| 14:44.04 | trollasaurus | I'm not a fan of WDS. I hard-wire my APs. |
| 14:44.28 | trollasaurus | I'm sure you can do some data over a 2-wire system :-P |
| 14:45.01 | *** join/#asterisk adyn (~adyn@pdpc/supporter/active/adyn) |
| 14:46.45 | trollasaurus | PMantis, There's also those "PowerLine" devices that do signalling over the electrical wiring |
| 14:48.07 | trollasaurus | It's more expensive than WiFi, but probably not as flaky as 10BASE-T over phone cabling. |
| 14:48.12 | PMantis | trollasaurus, I was considering that, too, but I think that'll require some testing, as their house has (I think) 4 different panel boxes with separate feeds from the curb. |
| 14:48.36 | trollasaurus | You can bridge them with WiFi :-) |
| 14:48.50 | PMantis | LOL |
| 14:49.11 | trollasaurus | Or just bridge them in attic/basement, since they'll all run along side each other up there :-P |
| 14:49.39 | trollasaurus | It'd just be 4x junction boxes and 4x of the devices, and a switch (so maybe 5x junction boxes) |
| 14:51.46 | PMantis | trollasaurus, I also suggested running new wire near the ceiling, then adding crown-molding to cover the wires. Or, cutting the drywall in areas where reaching the desired wall is not possible without it. |
| 14:52.28 | trollasaurus | I would probably just put hard-wired APs in the attic/basement :-P |
| 14:53.10 | PMantis | trollasaurus, Nothing is ideal, but I certainly need to provide IP phones in order to provide them with the paging they want. They currently have 14 extensions on this (key?) system. |
| 14:53.54 | trollasaurus | Client APs all over the place :-) |
| 14:54.38 | *** join/#asterisk lost_soul (~noymfb@cpe-67-249-141-227.twcny.res.rr.com) |
| 14:55.23 | PMantis | trollasaurus, Heh. Have you really ran IP phones over client APs? |
| 14:55.38 | trollasaurus | Yes |
| 14:55.39 | PMantis | (effectively a wireless bridge) |
| 14:56.41 | trollasaurus | It works out just fine |
| 14:57.02 | PMantis | trollasaurus, Oh... forgot about PoE. Hmmmmmm |
| 14:57.04 | *** join/#asterisk frigidzephyr (~rnewton@nat/digium/x-kjscwygedealgwnq) |
| 14:57.47 | trollasaurus | You can hide the client AP behind or even under a desk |
| 14:57.47 | *** join/#asterisk LemensTS (~matthew@adsl-70-238-158-17.dsl.stlsmo.sbcglobal.net) |
| 14:58.05 | *** part/#asterisk LemensTS (~matthew@adsl-70-238-158-17.dsl.stlsmo.sbcglobal.net) |
| 14:58.15 | trollasaurus | For the network-sided APs ? Yeah |
| 14:58.36 | trollasaurus | Heading to work, BRB |
| 15:01.51 | yonahw | anyone know where I can find the rules for what is valid in a dialplan label? I suspect that having an underscore in the label is what is causing my issue this morning. |
| 15:02.01 | PMantis | trollasaurus, Thanks for helping me to think through this. Other than Asterisk working with a proprietary digital phone set that works on a single pair cable, I think my only option is to use IP phones (what I want anyhow). However, placing a UPS on the PBX doesn't help if I don't use PoE phones - which isn't happening over wireless. The only solution to give us what we really want is cutting drywall and wiring, unless I'm miss |
| 15:02.02 | PMantis | ing something. |
| 15:04.20 | trollasaurus | I would have assumed the house had a generator |
| 15:04.57 | *** join/#asterisk BlackMaria (~Freischut@modemcable094.155-178-173.mc.videotron.ca) |
| 15:08.33 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 15:11.41 | *** join/#asterisk csnook (~chris@209-6-38-66.c3-0.smr-ubr1.sbo-smr.ma.cable.rcn.com) |
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| 15:19.18 | *** join/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 15:19.42 | jmkgreen | We have a bizarre problem that results in DTMF tones on some calls not being received. Where's the best place to ask about this one? |
| 15:27.28 | *** join/#asterisk krion (~seb@unaffiliated/krion) |
| 15:28.52 | *** join/#asterisk fabsoft (~chatzilla@host192-159-dynamic.245-95-r.retail.telecomitalia.it) |
| 15:29.15 | fabsoft | h iall |
| 15:32.11 | fabsoft | i would ask to this channel cause i cant understand why i get a warning message: Asked to transmit frame type ulaw, while native formats is 0x2 (gsm) read/write. this happen when a sip client use a different codec that uses the destination channel |
| 15:33.03 | fabsoft | the waring message disappers when the called peer answer the call |
| 15:33.34 | fabsoft | and the communication proceed with success.. |
| 15:36.18 | *** join/#asterisk aidinb (~Aidin@unaffiliated/aidinb) |
| 15:36.35 | Godfather_ | i'm trying to set up cdr, i installed asterisk addons 1.6.2.2, but i cant do a "cdr mysql status", any ideas? http://pastebin.com/hWVXhRnE |
| 15:37.29 | *** join/#asterisk drift- (ae3001a4@gateway/web/freenode/ip.174.48.1.164) |
| 15:41.10 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 15:41.10 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 15:41.56 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
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| 15:57.02 | *** join/#asterisk n0cturnal (~quassel@piksta.net) |
| 16:03.15 | ujjain | Do anybody have any idea why my iPhone and X-lite can connect to my Asterisk server, but PAP2T not anymore? I added port-forwarding. |
| 16:03.20 | ujjain | 5060, 10000-20000. |
| 16:03.23 | ujjain | TCP & UDP. |
| 16:07.38 | *** join/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 16:09.56 | *** join/#asterisk fofware (~Fabian@host186.190-225-12.telecom.net.ar) |
| 16:11.18 | *** join/#asterisk zerohalo (~zerohalo@173-13-92-27-NewEngland.hfc.comcastbusiness.net) |
| 16:16.54 | *** join/#asterisk cesar_CR (~cesar@201.192.86.30) |
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| 16:19.15 | *** join/#asterisk VaGoNeTaS (~nobody@200.111.138.170) |
| 16:19.20 | VaGoNeTaS | =) |
| 16:20.27 | *** join/#asterisk tuxxie (~Ryan@rrcs-70-63-90-226.midsouth.biz.rr.com) |
| 16:20.38 | tuxxie | can a sip call echo? |
| 16:20.53 | p3nguin_ | Not exactly. |
| 16:21.09 | p3nguin_ | You can hear echo on the handsets, but it's not SIP's fault. |
| 16:21.15 | tuxxie | or does it have to be headsets or when it is converted at our sip gatway |
| 16:21.23 | tuxxie | p3nguin_: understood. |
| 16:21.51 | tuxxie | how about when the call hits our trunk? |
| 16:23.01 | tuxxie | can it echo then? or is that very unlikely? |
| 16:23.44 | *** join/#asterisk krion (~seb@unaffiliated/krion) |
| 16:25.41 | *** join/#asterisk tris (~tristan@CAMEL.ETHEREAL.NET) |
| 16:26.28 | *** part/#asterisk sekil (~sekil@80.93.247.26) |
| 16:26.37 | Katty | hai |
| 16:26.47 | Katty | i would just like to say... |
| 16:26.48 | Katty | you all |
| 16:26.50 | Katty | are amazing |
| 16:29.39 | ChannelZ | at what? |
| 16:29.41 | drmessano | Thank you |
| 16:31.22 | thehar | yawns |
| 16:31.46 | p3nguin_ | tuxxie: Any time analog technology is introduced, there is a chance echo will occur. |
| 16:32.07 | drift- | p3nguin |
| 16:32.10 | drift- | i'm getting a manged switch |
| 16:32.15 | drift- | smc > managed switch |
| 16:32.28 | drift- | going to eliminate the linux router i have setup |
| 16:32.41 | ujjain | My iPhone and X-lite can connect to my Asterisk server, but PAP2T cannot anymore since I put asterisk server behind NAT? I added port-forwarding for TCP&UDP 5060, 10000-20000. |
| 16:33.13 | *** join/#asterisk coppice (~chatzilla@203.196.147.34) |
| 16:33.15 | drift- | http://tinyurl.com/23p96oo :D |
| 16:33.18 | Katty | ChannelZ: what do you mean at what? |
| 16:33.23 | Katty | ChannelZ: you're here aren't you |
| 16:33.39 | Katty | ChannelZ: and that's only the beginning of awesome |
| 16:34.05 | tuxxie | p3nguin_: thanks. |
| 16:35.49 | ChannelZ | I just thought maybe there were more qualifications |
| 16:35.53 | *** join/#asterisk Mhaddog__ (~Mhaddog_@c-98-203-21-151.hsd1.fl.comcast.net) |
| 16:40.59 | Katty | ChannelZ: you're awesome, just deal with it |
| 16:41.37 | *** join/#asterisk superflit (~superflit@140.226.49.160) |
| 16:42.33 | leifmadsen | \o/ |
| 16:43.36 | *** join/#asterisk drudge` (drudge@unaffiliated/drudge/x-837452) |
| 16:43.46 | Katty | leifmadsen: and you are awesome, too, sir |
| 16:43.55 | leifmadsen | w00t |
| 16:44.01 | leifmadsen | now I can retire! |
| 16:45.00 | Katty | (= |
| 16:46.55 | *** join/#asterisk slackytude (~slacky@2001:41b8:9bf:fe75:213:ceff:fef0:2369) |
| 16:47.10 | thehar | lol |
| 16:52.18 | *** join/#asterisk jimi_ (~jimi@unaffiliated/tuxguy) |
| 17:02.10 | *** join/#asterisk boch (c9d8f65d@gateway/web/freenode/ip.201.216.246.93) |
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| 17:02.58 | boch | is it possible to disable the # termination on Read() application? because i want to read ## for an especial option |
| 17:04.44 | *** join/#asterisk w4llnu55 (~w4llnu55@89.204.153.96) |
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| 18:06.46 | Katty | peeks in |
| 18:07.03 | *** join/#asterisk dopiwan (~chatzilla@216.75.133.83) |
| 18:12.11 | thehar | throws cookies over THERE |
| 18:12.42 | *** join/#asterisk Takapa (vegard@svanberg.no) |
| 18:12.57 | leifmadsen | jumps over THERE |
| 18:13.55 | *** join/#asterisk lesouvage (~lesouvage@62.140.137.136) |
| 18:14.25 | paulc | jumps on the cookie, with nothing better to do - borked desktop PC with the IT boys on the case = the gift of time, this morning :) |
| 18:17.58 | *** join/#asterisk Mhaddog___ (~Mhaddog_@173-109-121-46.pools.spcsdns.net) |
| 18:25.27 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 18:29.32 | jaytee | has anyone in here tried running asterisk on an Atom 330 cpu mini-ITX board? |
| 18:38.24 | citywok | that has more power than a lot of embedded systems do, so i imagine it will do fine |
| 18:42.32 | jaytee | citywok, thanks. was looking for some feedback from someone who had actually tried one |
| 18:43.03 | citywok | not it :P, i've used an aa50 which is a low power embedded device and it works pretty well, so i imagine a full blown PC can do much better |
| 18:43.46 | citywok | i have a fleet of atom 330 workstations, i suppose i could go nab one and test it out. what are you trying to find out? how many concurrent calls? |
| 18:44.26 | jaytee | more along the lines of compatibility issues with linux/asterisk |
| 18:44.39 | pabelanger | jaytee: Yes, I have it running on an ION 330 |
| 18:44.53 | pabelanger | Ubuntu 10.04 |
| 18:44.58 | citywok | oh, yea in that case it'll work perfectly fine on your favorite linux distro. |
| 18:45.11 | citywok | pabelanger: how many calls can it handle? |
| 18:45.29 | jaytee | I'd be targeting installations that would only have 6 to 20 concurrent calls so I'd imagine it could keep up fine. |
| 18:45.46 | citywok | oh yea, i would imagine it could handle 50 calls w/out translation |
| 18:45.55 | jaytee | cool beans! |
| 18:46.09 | pabelanger | citywok: only process 2-3 concurrent calls. I'm sure it could push much more. It's dual core |
| 18:46.23 | citywok | yea, i'm tempted to go nab one and bench it just to see |
| 18:47.33 | jaytee | citywok and pabelanger, thanks for the input. |
| 18:57.28 | *** part/#asterisk Get_The_Fish (~Get_The_F@c-24-8-50-199.hsd1.co.comcast.net) |
| 19:01.29 | *** join/#asterisk drudge` (drudge@unaffiliated/drudge/x-837452) |
| 19:02.04 | *** join/#asterisk bvierra (~bvierra@netblock-208-127-193-177.dslextreme.com) |
| 19:02.24 | bvierra | hey does anyone happen to know for the e911 law if you have to have it on every number in California |
| 19:04.17 | Corydon76-dig | bvierra: only your attorney can answer that question for you |
| 19:04.22 | bvierra | I figured :) |
| 19:04.45 | leifmadsen | you may or you may not need it, and it may depend on the phase of the moon |
| 19:04.50 | leifmadsen | or it may not |
| 19:05.22 | Corydon76-dig | Generally speaking, it is safer to have it and not need it than need it and not have it |
| 19:06.07 | p3nguin_ | If you're pinching pennies, it's an extra expense, though. |
| 19:06.46 | citywok | bvierra: your provider should be able to tell you the law |
| 19:06.54 | bvierra | awesome thanks :) |
| 19:07.14 | bvierra | it just doubles the price per DID, so its not a huge expense |
| 19:07.17 | bvierra | but was checking :) |
| 19:07.34 | citywok | bvierra: what? seriously? what provider? |
| 19:08.47 | p3nguin_ | I don't want to spend that extra dollar per month if I don't have to. |
| 19:08.54 | bvierra | flowroute |
| 19:09.05 | bvierra | but I think I can get them to waive it with so many numbers :) |
| 19:09.15 | citywok | ah, yea i use flowroute but i don't have e911 configured for any of those numbers |
| 19:09.19 | bvierra | basically its 1.39/mo for DID and 1.39 for e911 |
| 19:09.22 | citywok | i do my e911 with bandwidth.com |
| 19:09.29 | bvierra | since we a re a company we want to make sure :) |
| 19:09.43 | citywok | you only REALLY need 1 e911 number per physical address |
| 19:10.00 | citywok | have one number with each address on it, and when somebody calls from that location, pulse that DID as the CID, everything else justh appens |
| 19:10.03 | bvierra | that was my guess, just wanted to verify :) asking them and our attorney as well |
| 19:10.11 | bvierra | yea |
| 19:10.19 | citywok | that's how we do it, and bandwidth.com and qwest said i could do it. |
| 19:11.50 | *** join/#asterisk timholum (~chatzilla@68-117-120-138.static.eucl.wi.charter.com) |
| 19:16.30 | *** join/#asterisk gabdrach (~Adium@190.177.75.42) |
| 19:17.00 | *** join/#asterisk atan (~atan@unaffiliated/atan) |
| 19:18.44 | *** part/#asterisk ujjain (~ujjAIN@unaffiliated/ujjain) |
| 19:20.04 | *** join/#asterisk Hoi (~hoiung@host86-166-8-14.range86-166.btcentralplus.com) |
| 19:26.27 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 19:32.06 | luckman212 | how is the call quality with Flowroute? |
| 19:32.26 | thehar | great |
| 19:32.33 | magwas | My asterisk is being attacked right now |
| 19:32.50 | thehar | iptables is your friend |
| 19:33.28 | magwas | sip is not too firewall friendly... |
| 19:33.53 | magwas | the attacker is 211.151.79.149 |
| 19:34.02 | thehar | yes it is |
| 19:34.54 | thehar | iptables -A INPUT -s 211.151.79.149 -j DROP |
| 19:35.27 | magwas | have anyone encountered this attacker yet? |
| 19:35.50 | carrar | Welcome ti China |
| 19:35.52 | carrar | to |
| 19:36.02 | carrar | get used to it |
| 19:36.38 | thehar | yup |
| 19:36.52 | thehar | if you want to be an *** you could add /24 to it |
| 19:36.54 | thehar | or even /8 |
| 19:37.09 | carrar | 211.151.0.0/17 |
| 19:37.17 | thehar | hahahaha |
| 19:37.22 | thehar | 211.151.0.0/8 |
| 19:37.30 | carrar | route: 211.151.0.0/17 |
| 19:37.30 | carrar | descr: CHINA-ABITCOOL |
| 19:37.30 | carrar | descr: Abitcool(China) Inc. |
| 19:37.30 | carrar | country: CN |
| 19:37.35 | thehar | nice |
| 19:37.37 | carrar | <PROTECTED> |
| 19:37.45 | carrar | you will hit all over asia |
| 19:37.47 | thehar | yup |
| 19:37.49 | carrar | japan |
| 19:37.51 | thehar | asia sucks |
| 19:37.51 | carrar | korea |
| 19:37.58 | carrar | etc.. |
| 19:38.00 | thehar | asia likes sip |
| 19:39.19 | carrar | grep ": cn :" country-ipv4.lst | wc -l |
| 19:39.19 | carrar | 1002 |
| 19:39.29 | carrar | wget ftp://ftp.ripe.net/ripe/dbase/split/ripe.db.inetnum.gz |
| 19:39.52 | carrar | err |
| 19:39.52 | thehar | yes i know :] |
| 19:39.57 | carrar | not ripe I mean |
| 19:40.00 | thehar | i just like to span multiple countries when i block things |
| 19:40.09 | carrar | wget ftp://ftp.apnic.net/apnic/dbase/data/country-ipv4.lst |
| 19:40.13 | thehar | wget fail |
| 19:40.16 | carrar | wish ripe had that |
| 19:40.28 | magwas | any online blacklist I could use? |
| 19:40.31 | carrar | they both work |
| 19:40.33 | thehar | not really |
| 19:40.43 | thehar | there are a lot of honey pots out there |
| 19:41.00 | carrar | has nothing to do with that |
| 19:41.32 | carrar | never seen 211.151.79.149 specifical (just searched through all my logs) |
| 19:42.00 | carrar | I have months worth of blocked logs |
| 19:43.01 | carrar | Yesterdays Total Hits from China: 34187 |
| 19:43.35 | carrar | <PROTECTED> |
| 19:43.44 | carrar | 4,565 of those were attempting 5060 |
| 19:44.34 | carrar | Their fav port is 445:TCP |
| 19:45.24 | carrar | top hitting IP yesterday was 218.30.22.82 |
| 19:45.58 | magwas | I don't even log attempts at smb ports |
| 19:46.19 | thehar | logging is your friend |
| 19:46.27 | carrar | makes for interesting stats |
| 19:48.26 | magwas | if there are too many logs from kernel, they break. broken logs get logged on console. I am on a sparc, with a 9600 baud serial console. When I hit hard, it takes a looong time to write those logs to console. And it is from kernel, blocking everything:) |
| 19:49.33 | *** join/#asterisk JonnyD_work (~Jon@12.222.63.34) |
| 19:49.49 | magwas | and anyway ports scans are logged without smb ports. smb ports are just outside the universe known to my firewall:) |
| 19:49.50 | *** join/#asterisk JonnyD_work (~Jon@12.222.63.34) |
| 19:49.55 | carrar | This doesn't come from UNIX |
| 19:50.02 | carrar | never makes it to the unix boxes |
| 19:50.33 | drudge` | anyone have any exp with low priced sip phones? like cheaper than $100? |
| 19:50.43 | magwas | yep, it have a filtered out 445. and interestingly a 1521 |
| 19:50.55 | magwas | maybe that was the attack vector |
| 19:51.13 | carrar | top 6 yesterday |
| 19:51.14 | carrar | <PROTECTED> |
| 19:51.15 | carrar | <PROTECTED> |
| 19:51.15 | carrar | <PROTECTED> |
| 19:51.15 | carrar | <PROTECTED> |
| 19:51.15 | magwas | drudge: my ekiga is $0. cheap enough? |
| 19:51.17 | carrar | <PROTECTED> |
| 19:51.19 | carrar | <PROTECTED> |
| 19:51.22 | carrar | count / port |
| 19:51.59 | carrar | drudge, softphones are free |
| 19:52.09 | carrar | You can find used polycoms on ebay |
| 19:52.43 | carrar | http://cgi.ebay.com/Polycom-IP-430-/130455254191?pt=LH_DefaultDomain_0&hash=item1e5fbd34af |
| 19:53.32 | *** join/#asterisk yang (yang@freenode/sponsor/cacert.assurer.yang) |
| 19:54.56 | magwas | if size is not a concern, you can build a phone from just any used computer. You put a linux on it, add a headset, and it is done. |
| 19:55.57 | *** join/#asterisk SuPrSluG (~SuPrSluG@host-64-179-106-158.ind.choiceone.net) |
| 19:56.15 | magwas | not exactly a mobile phone unless you use some mobile equipment :) |
| 19:57.05 | WHYS | using API, I'm getting double (triple?) DTMF when I insert DTMF. Happens intermittently I've tried all DTMF modes. Where do I look? |
| 19:57.38 | WIMPy | What API? |
| 19:57.45 | Katty | hugs carrar |
| 19:57.48 | Katty | hugs WIMPy |
| 19:57.57 | WHYS | Ummm... perl AMI (?) I'm not on site now. |
| 19:58.00 | carrar | WOAH |
| 19:58.02 | carrar | Hi Katty |
| 19:58.07 | Katty | hi deary :> |
| 19:58.12 | WIMPy | Hi Katty |
| 19:58.14 | carrar | hugs the kattieanattor back |
| 19:59.19 | thehar | Katty: ! |
| 19:59.22 | thehar | hugs Katty |
| 19:59.25 | Katty | :>>>>>>>>>>>>>>>>>>>>>>>> |
| 19:59.28 | Katty | glomps thehar |
| 19:59.32 | thehar | ohyes |
| 19:59.33 | thehar | glomp me |
| 19:59.37 | Katty | i so did ^_^ |
| 19:59.42 | WHYS | Not that it should mater. I use it to originate calls, bridge a softphone and then insert DTMF. |
| 19:59.57 | magwas | do we have any known security issues with asterisk 1.6.2? |
| 20:00.05 | Katty | thehar: you seem like glomp worthy material |
| 20:00.10 | thehar | horray |
| 20:02.41 | drudge` | yea, i was lookin at the 301s or the linksys spa941s |
| 20:02.44 | magwas | have learnt a new word. glomp |
| 20:04.22 | drudge` | softphones are the right price but the end user is mortified at their ocmputers as it is |
| 20:05.06 | *** join/#asterisk nny (~Scott@174.107.201.103) |
| 20:05.09 | citywok | the aastra 30/31i phones are dirt cheap and pretty good |
| 20:06.31 | p3nguin_ | How about the Cisco SPA 504G? |
| 20:07.02 | p3nguin_ | It's a 4-line SIP phone. |
| 20:07.52 | nny | any fail2ban users here? Trying to implement it on a Centos derivitive called Clarkconnect. I have it installed, and see the line in iptables, but the init script for CC uses a custom firewall init script and not sure if fail2ban needs or cares about any of that, or only needs the line in iptables to work |
| 20:08.09 | thehar | >.> |
| 20:08.12 | thehar | <.< |
| 20:09.16 | nny | i have someone trying to brute force this box, but it doesn't seem to catch it. I assume I missed something |
| 20:09.49 | nny | aaand nm |
| 20:09.54 | p3nguin_ | The SPA 502G is a 1-line phone, if you don't need more lines. |
| 20:09.59 | nny | it wants /var/log/asterisk/full >< |
| 20:10.01 | nny | i fix |
| 20:10.02 | magwas | It is the day of brute forcers... |
| 20:10.10 | nny | oh yeah? |
| 20:10.13 | jbroome | also, age of aquarius |
| 20:10.20 | nny | here's my friend: 77.235.51.89 |
| 20:10.31 | nny | jbroome: it's not the dawning anymore? |
| 20:10.55 | nny | the nice thing is he is helping me test fail2ban |
| 20:10.57 | nny | what a nice guy |
| 20:11.09 | p3nguin_ | Greek |
| 20:11.17 | jbroome | let's see if i'm getting the same guy hitting me |
| 20:11.28 | magwas | my friend had end up in iptables like -A INPUT -s 211.151.0.0/17 -j DROP |
| 20:11.33 | nny | yes |
| 20:11.37 | p3nguin_ | Chances of that aren't very high. |
| 20:11.39 | nny | iptables -I INPUT -s 77.235.51.89 -j DROP |
| 20:11.58 | nny | gonan restart iptables once I fix fail2ban and see if it gets him automagically |
| 20:12.20 | WHYS | No takers on the multiple DTMF problem? not even a web page referral? |
| 20:12.38 | nny | WHYS: missed it, retype, IU LOVE DTMF issues |
| 20:12.40 | nny | I* |
| 20:12.53 | *** join/#asterisk thansen (~thansen@c-98-202-28-239.hsd1.ut.comcast.net) |
| 20:13.09 | WHYS | using API, I'm getting double (triple?) DTMF when I insert DTMF. Happens intermittently I've tried all DTMF modes. |
| 20:14.47 | thehar | what api |
| 20:14.52 | thehar | you mean ami? |
| 20:14.58 | nny | woot |
| 20:15.06 | nny | Chain fail2ban-ASTERISK (1 references) target prot opt source destination DROP all -- 77.235.51.89 anywhere |
| 20:15.10 | nny | and... thanks buddy! |
| 20:15.27 | *** join/#asterisk theron (~theron@ip244.scolloc.lh.net) |
| 20:15.30 | WHYS | sorry, ami - brain dead at work today. using perl AMI |
| 20:16.26 | nny | WHYS: hmm damn, not familiar with how AMI generates DTMF, i usually get stuck dealing with upstream lost in translation issues |
| 20:16.45 | WHYS | API = Am Pretty Ignorant |
| 20:17.25 | WHYS | this sounds like DTMF interupted and regenerated and back again. |
| 20:19.30 | WHYS | Oh, I should mention that sometimes I get no sound at all, sometime perfect DTMF, other times multiple tones rather than one. |
| 20:22.22 | nny | WHYS: where are you hearing the multiple tones? The far end? |
| 20:23.48 | WHYS | Yes, I'll call my home phone to check it, and I'll hear two or three tones in the time span I should have heard one. |
| 20:24.26 | nny | WHYS: any chance a device in between has DTMF detection turned on and is generating the tone additionally? |
| 20:25.00 | nny | WHYS: you can set a log to show dtmf and see if it's only sending one via asterisk. This sounds like an external issue |
| 20:25.06 | Katty | p3nguin_: looks like we got some rain down here, according to crittercam |
| 20:25.09 | Katty | p3nguin_: you guys get anything up north? |
| 20:25.21 | *** join/#asterisk Dovid (addc7f12@gateway/web/freenode/ip.173.220.127.18) |
| 20:26.28 | Katty | http://i.imgur.com/fjNl3.jpg <- who wants to go on vacation? |
| 20:26.48 | p3nguin_ | katty: Nothing yet, but the forecast shows showers are expected. |
| 20:27.11 | Katty | p3nguin_: i'm sure you guys could use some |
| 20:27.13 | WHYS | Could be I suppose. I use vitelity, and have tried info, auto RFCxxxx, and inband. some partially work others not. only once did I get clean DTMF. |
| 20:27.28 | WHYS | what do you mean by setting a log for DTFM. you mean specifically? |
| 20:27.33 | p3nguin_ | katty: We haven't had any significant rainfall in at least a month. |
| 20:28.05 | p3nguin_ | whys: You can include dtmf in asterisk's log. |
| 20:29.02 | WHYS | OK, I'll look it up and use it to try to help. Thanks - it's a path I can try. |
| 20:29.52 | p3nguin_ | You'd do well looking in logger.conf. |
| 20:30.10 | WHYS | is there any other way to inject DTFM into a call other then manually pressing keys (other than AMI)? |
| 20:30.45 | p3nguin_ | You could use SendDTMF(). |
| 20:31.18 | WHYS | but how would I call it extrenally without AMI? |
| 20:31.56 | p3nguin_ | It wouldn't be used quite that simply. It would need to be included in dial plan. |
| 20:32.44 | *** part/#asterisk ariel_ (~chatzilla@63.214.236.169) |
| 20:33.10 | WHYS | I am processing credit cards, - that would be one hell of a dial plan. |
| 20:35.16 | WHYS | hummmm - just found this: |
| 20:35.21 | WHYS | "Asterisk will consistently intercept, buffer and regenerate all DTMF tones it detects. This may be a problem" |
| 20:35.54 | WHYS | I've stumbled upon something. I love stumbling. |
| 20:39.41 | russellb | Asterisk is designed as a man in the middle attack. |
| 20:44.06 | *** join/#asterisk garymc (~chatzilla@host86-148-248-176.range86-148.btcentralplus.com) |
| 20:48.06 | *** join/#asterisk a1fa (~a1fa@unaffiliated/a1fa) |
| 20:48.17 | a1fa | i need a new voice talent :( |
| 20:48.26 | underdog | lays down the law...watches the wife wipe her feet on "the law" |
| 20:49.49 | Katty | why don't you ask her to help you write the law |
| 20:50.13 | Katty | we don't like being told what to do ;) |
| 20:50.52 | underdog | plays the "I'm a man" song |
| 20:51.44 | underdog | if Katty hasn't heard that...she needs to google it....she'll find it funny |
| 20:58.32 | WIMPy | I've got trouble with iax between a trunk and a 1.6.2.12-rc1. Sometimes I only get small fragemnts of voice in one direction. |
| 20:59.10 | WIMPy | This only happens on phone <-sip-> *1 <-iax-> *2 <-sip-> phone. |
| 20:59.32 | WIMPy | When I call a MOH extension on the remote server there is no problem. |
| 21:00.42 | WIMPy | But the ugly part starts when I dial phone <-sip-> *1 <-iax-> *2 <-iax-> *1 <-any-> phone. |
| 21:00.53 | WIMPy | Those two calls eat up >800 kbps. |
| 21:01.05 | WIMPy | How do I debug that issue? |
| 21:07.44 | *** join/#asterisk Olobola (~chatzilla@72.sub-75-210-25.myvzw.com) |
| 21:09.53 | *** join/#asterisk jhirley (~chatzilla@c-75-74-13-194.hsd1.fl.comcast.net) |
| 21:14.55 | *** join/#asterisk drudge` (drudge@unaffiliated/drudge/x-837452) |
| 21:15.05 | *** join/#asterisk tzafrir (~tzafrir@bzq-218-155-146.cablep.bezeqint.net) |
| 21:24.03 | *** join/#asterisk [canniballllera] (~cannibale@201-25-255-181.fnsce703.dsl.brasiltelecom.net.br) |
| 21:24.16 | *** part/#asterisk [canniballllera] (~cannibale@201-25-255-181.fnsce703.dsl.brasiltelecom.net.br) |
| 21:24.33 | bn-7bc | <PROTECTED> |
| 21:47.00 | *** join/#asterisk slacker775 (~dhollis@static-96-254-30-130.tampfl.fios.verizon.net) |
| 22:16.08 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 22:17.54 | WIMPy | Hmm. I replaced the 1.6.2.12-rc1 by 1.6.2.14 and the problem persisted. |
| 22:18.20 | WIMPy | The I replaced it by 1.8 trunk and the problems are gone. |
| 22:19.03 | WIMPy | Interestingly I don't have any trouble with a peer running 1.4.21.2. |
| 22:22.20 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 22:27.20 | carrar | Oh my |
| 22:27.30 | *** part/#asterisk slacker775 (~dhollis@static-96-254-30-130.tampfl.fios.verizon.net) |
| 22:28.56 | p3nguin_ | You probably won't have any troubles with 1.4.37, either. |
| 22:29.25 | timholum | hello, I am trying to run a script after a call is answered I added M(/path/to/script) to my Dial but it does not run? any idea's as to what I am doing wrong? |
| 22:29.46 | WIMPy | No idea, but why is 1.8.0 incompatible to 1.6.2.* but not 1.4? Sounds fishy to me. |
| 22:30.54 | WIMPy | But the bandwidht usage is more than fishy anyway :-( |
| 22:31.00 | _Corey_ | timholum: M() is for a macro, not an external program |
| 22:31.38 | timholum | is there a way to run an agi script when a Dial is answered? |
| 22:32.32 | timholum | I need to insert into a mysql db when a phone call is answered and by who, and it needs to be done in real time |
| 22:32.42 | WIMPy | Put it in that macro |
| 22:33.05 | timholum | ill have to look at macro's, I have never used them in asterisk |
| 22:33.26 | carrar | holy macro! |
| 22:33.39 | Nugget | Macroeconomics is pure voodoo. :) |
| 22:33.58 | _Corey_ | timholum: You're trying something that's a little exotic, so familiarize yourself with the macros and AGI |
| 22:34.01 | carrar | free & open market! |
| 22:34.22 | timholum | I know agi scripts fairly well, I just have never used macro's |
| 22:34.35 | timholum | I am looking at the voip-info.org page on them now |
| 22:34.40 | _Corey_ | you should have no problem figuring them out then |
| 22:34.55 | _Corey_ | :) |
| 22:35.23 | timholum | ok basicly a macro is just a context essencaly |
| 22:35.34 | timholum | from what I am seeing anyway |
| 22:36.28 | Nugget | yeah, with some convenient goofiness for passing arguments more seamlessly |
| 22:37.59 | *** join/#asterisk aidinb (~Aidin@unaffiliated/aidinb) |
| 22:38.01 | timholum | this should be pritty simple :) thanks |
| 22:44.23 | *** part/#asterisk nny (~Scott@174.107.201.103) |
| 22:47.16 | bvierra | hey guys when you get a: exited non-zero on... |
| 22:47.24 | bvierra | does it mean there was an error? |
| 22:47.29 | russellb | nope, just means the channel hun gup |
| 22:47.34 | bvierra | thought so |
| 22:51.48 | *** join/#asterisk aidinb (~Aidin@unaffiliated/aidinb) |
| 22:56.41 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 22:56.41 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.8.0 (2010/10/21), 1.6.2.14 (2010/11/11), 1.4.37 (2010/11/11), *-Addons 1.6.2.2, 1.4.12 (2010/09/22), dahdi-linux 2.4.0 (2010/09/01), dahdi-tools 2.4.0 (2010/09/01), libpri 1.4.11.4 (2010/09/01) -=- Visit the new official Asterisk wiki: wiki.asterisk.org |
| 22:56.46 | *** part/#asterisk Kinbote (~hogemoge@ec2-204-236-228-187.compute-1.amazonaws.com) |
| 23:04.23 | *** join/#asterisk JamesHarrison (~jharr@hometree.mmmetrics.co.uk) |
| 23:06.10 | *** join/#asterisk csnook (~chris@209-6-38-66.c3-0.smr-ubr1.sbo-smr.ma.cable.rcn.com) |
| 23:09.25 | *** join/#asterisk rrb3942 (~rbullock@cpe-67-242-215-62.rochester.res.rr.com) |
| 23:10.01 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 23:11.10 | *** join/#asterisk cccode (chris@astro.securespot.net) |
| 23:15.32 | bvierra | if this is in the logs: Executing [h@macro-dialout-trunk:1] Macro("SIP/500-000003fc", "hangupcall,") in new stack |
| 23:15.43 | bvierra | does that mean that someone hung it up on our end? |
| 23:16.45 | *** join/#asterisk mintee (~mintee@static-64-115-220-213.isp.broadviewnet.net) |
| 23:17.04 | mintee | what can i put in a dialplan to pull out all the variables? |
| 23:17.28 | citywok | mintee what are you trying to do? |
| 23:17.36 | citywok | what do you mean by "pull out" |
| 23:17.43 | mintee | i just wanna see what's defined |
| 23:17.51 | mintee | like a print_r in php |
| 23:18.04 | p3nguin_ | mintee: core show application DumpChan |
| 23:18.08 | mintee | so i'd use a NoOp(${GLOBAL}) |
| 23:18.34 | mintee | p3nguin_: ya, is there a way i can parse it out in the dialplan? |
| 23:18.49 | mintee | cause the channel closes the second it opens |
| 23:18.50 | p3nguin_ | mintee: Yes. DumpChan() |
| 23:18.55 | mintee | Oh, thanks |
| 23:18.58 | mintee | tries |
| 23:19.08 | JunK-Y | or type: core show globals in the CLI |
| 23:19.44 | WIMPy | dumpchan doesn't give you all variables. |
| 23:21.52 | mintee | i'm fighting with hylafax, iaxmodem and * to email to fax gateway |
| 23:21.55 | mintee | almost have it |
| 23:22.25 | mintee | just can't figure out the [fax-out] context |
| 23:37.57 | mintee | p3nguin_: DumpChan() was exactly what i was lookign for |
| 23:37.58 | mintee | thanks |
| 23:42.56 | *** join/#asterisk tzafrir_laptop (~tzafrir@212.179.75.202) |
| 23:56.59 | p3nguin_ | mintee: cool |
