| 00:00.19 | WIMPy | Ah, it works if I remoce that control char at the end. |
| 00:00.51 | qakhan | Nix414s here is my config |
| 00:00.52 | WIMPy | Looks like Freepbx shit. And more importantly it has no timestamps. |
| 00:00.53 | qakhan | http://pastebin.com/Y1RDrByg |
| 00:01.04 | qakhan | and description |
| 00:01.51 | tm1000 | pulled right from asterisk -rvvvv me thinks |
| 00:02.07 | WIMPy | And enabling debug for chan_dongle might also shed some light. |
| 00:02.31 | WIMPy | No reason not to have timestamps. |
| 00:02.46 | wacomcito | its a asterisk -rvvvvvvvvvvvvvvvvvvvvvv registry. |
| 00:02.58 | wacomcito | no /var/log file. |
| 00:04.11 | WIMPy | Enable timestamps in your asterisk.conf. |
| 00:04.33 | *** join/#asterisk kresp0 (~kresp0@109.Red-79-144-71.dynamicIP.rima-tde.net) |
| 00:04.41 | WIMPy | And/or enable debugging. Don;t know the command off head. I haven't installed chan_dongle ATM. |
| 00:04.42 | wacomcito | wich file debug or log? is better from pastebin |
| 00:05.33 | WIMPy | did not understand that question. Please rephrase. |
| 00:08.34 | wacomcito | http://pastebin.com/sMFV0eHS |
| 00:08.46 | wacomcito | the full debug with timestamps |
| 00:09.35 | WIMPy | Ok, so 3s seem to be a realistic time for a call attempt. |
| 00:11.02 | Nix414s | qakhan: thats pretty cool. it works when a call comes in from the PRI and is redirected to Swift? |
| 00:12.24 | wacomcito | i cant understand HANGUPCAUSE = 17. |
| 00:12.47 | WIMPy | 17 = "User busy" |
| 00:12.55 | wacomcito | ah.. ok. |
| 00:13.34 | Nix414s | wacomcito: go to a phone book and look up Mr. Lonely McNobodyloves me, he's bound to not be busy and can take your call. Also, sorry i'm north american so i have to ask this: are you certain the number format is correct for your trunk? |
| 00:13.54 | qakhan | yes |
| 00:14.24 | wacomcito | xDDDD |
| 00:14.33 | wacomcito | yes numbers are ok |
| 00:14.34 | qakhan | Nix414s if you dial the number which i mentioned. you can hear the choppy vocie. |
| 00:14.37 | wacomcito | dngle status says: |
| 00:14.59 | wacomcito | Location area code : CBD |
| 00:14.59 | wacomcito | Cell ID : 8B7 |
| 00:14.59 | wacomcito | Subscriber Number : +34696675349 |
| 00:14.59 | wacomcito | SMS Service Center : +34609090909 |
| 00:15.12 | qakhan | but if i dial through ext then voice the very clear |
| 00:15.43 | *** join/#asterisk nam3l3zz (~quassel@86-46-207-104-dynamic.b-ras1.pgs.portlaoise.eircom.net) |
| 00:15.55 | WIMPy | wacomcito: Looks ok so far. Try to enable dongle debug. |
| 00:16.28 | wacomcito | with set? |
| 00:17.24 | WIMPy | Don't know the command off head. I haven't installed chan_dongle ATM. |
| 00:17.27 | Nix414s | qakhan: my guess at the choppiness is g729. its an evil vile beast when working with a PRI. For testing purposes have everything negotiate in g711 and see if the choppiness is still there. |
| 00:18.59 | qakhan | i tested with ulaw and alaw, i heard only static. i even could not hear any single word |
| 00:20.41 | Nix414s | qakhan: weird ... point the phone number to a phone with only ulaw, is the audio still choppy? |
| 00:21.24 | Nix414s | qakhan: have the phone connected directly to serverA. no need to muddy up the test with that iax bridge. |
| 00:22.11 | *** part/#asterisk Nix414s (~jubiejank@vpn.bctconsulting.com) |
| 00:24.23 | qakhan | yes PRI is connected server A, and all calls are fine from PSTN |
| 00:24.33 | qakhan | while i use g729 |
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| 00:45.48 | qakhan | guys i fixed it |
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| 01:31.23 | *** join/#asterisk mmourani (~Adium@bas1-montreal23-1176231882.dsl.bell.ca) |
| 01:31.58 | mmourani | salut tout le monde |
| 01:32.01 | mmourani | hi everyone |
| 01:32.08 | mmourani | hope your doing good today |
| 01:40.03 | *** join/#asterisk longday24 (~longday@71-91-82-23.dhcp.leds.al.charter.com) |
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| 01:50.28 | mmourani | i have a mac osx lion and wireshark is not working on my machine |
| 01:50.34 | mmourani | any idea ? |
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| 02:04.15 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
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| 02:32.36 | cstk421 | anyone ever use whatsup gold to monitor ? |
| 02:32.43 | cstk421 | asterisk* |
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| 03:19.27 | mattc-ucwb | hi, my name is Matt. I am looking for some help in regards to Snom 320 and expansion panels |
| 03:19.51 | mattc-ucwb | i work for a not for profit organisation in Australia |
| 03:23.07 | mattc-ucwb | the exp panels are being used as switchboard/speed dial, and 13xxxx numbers are not being recognised |
| 03:25.15 | mattc-ucwb | hello? |
| 03:30.42 | mattc-ucwb | is anyone here? |
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| 03:37.06 | mattc-ucwb | hello |
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| 03:41.25 | mattc-ucwb | @rumbles Hi! |
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| 05:04.04 | *** join/#asterisk pensmit (~pensmit@unaffiliated/pensmit) |
| 05:04.28 | pensmit | What do you have to do to get a blf light to light up for voicemail? |
| 05:09.50 | pensmit | hints? |
| 05:11.40 | *** join/#asterisk usrb1n (55ba790f@gateway/web/freenode/ip.85.186.121.15) |
| 05:13.03 | usrb1n | Hello guys. I used to install always only Asterisk server and FreePBX. Now I'm trying to fix something on a server but it has a2billing installed. This has removed all the configs from /etc/asterisk. There's no sip.conf, extensions.conf...etc. |
| 05:13.12 | usrb1n | And a2billing.conf only contains the database info and few more things |
| 05:13.24 | usrb1n | I need to bind he IP address to asterisk because it's not listening on eth0 |
| 05:13.42 | usrb1n | And I can't find any config to do that. Do you guys have any idea where can I configure this ? |
| 05:18.07 | ChannelZ | bind the IP address to asterisk for what? |
| 05:18.26 | usrb1n | Because it's not working on public ip address |
| 05:18.37 | *** join/#asterisk Vutral (ss@mirbsd/special/Vutral) |
| 05:18.48 | ChannelZ | but for what? are you talking about SIP or what? |
| 05:18.49 | usrb1n | The port remains closed. I've stoped iptables and there's no other firewalls on the server |
| 05:20.25 | *** join/#asterisk mintos (mvaliyav@nat/redhat/x-pndqssyebaicuzhp) |
| 05:20.47 | kaldemar | usrb1n: how did you judge it to be closed? |
| 05:20.49 | ChannelZ | ..the port for WHAT... |
| 05:21.10 | usrb1n | Port 5060 so asterisk can get the calls. Isn't that normal ? |
| 05:21.41 | ChannelZ | not without sip.conf (and thus chan_sip) running |
| 05:21.53 | kaldemar | port 5060 what? |
| 05:21.56 | ChannelZ | And it's UDP so if you are trying to telnet or something as a test, that won't work |
| 05:22.15 | pensmit | What do you have to do to get a blf light to light up for voicemail? |
| 05:22.31 | ChannelZ | configure the mailbox for the peer |
| 05:22.49 | pensmit | can you give me an example |
| 05:22.54 | pensmit | for instance i was using freepbx and |
| 05:22.58 | ChannelZ | mailbox=200 |
| 05:23.11 | ChannelZ | oh.. FPBX I have no idea |
| 05:23.17 | pensmit | no base asterisk |
| 05:23.21 | pensmit | was using freepbx |
| 05:23.23 | ChannelZ | buried as some option somewhere no doubt |
| 05:23.33 | pensmit | so freepbx it was working |
| 05:23.40 | pensmit | but now in base asterisk |
| 05:23.49 | pensmit | the key associated with a *97 |
| 05:23.55 | pensmit | will fetch the voicemail |
| 05:24.03 | pensmit | but i can't figure out how to get it tolight up |
| 05:24.27 | pensmit | lets say yeah 200 is an extension |
| 05:24.35 | pensmit | and the mailbox is 200 |
| 05:24.44 | ChannelZ | Well not sure what *97 has to do with MWI but in sip.conf give set the mailbox for the peer. |
| 05:24.49 | ChannelZ | Asterisk will send notifications. |
| 05:24.59 | ChannelZ | Or you can get the phone to subscribe to the mailbox |
| 05:25.15 | pensmit | could you please give me an example |
| 05:25.55 | ChannelZ | I did |
| 05:26.05 | pensmit | just in the device section of sip.conf for device lets say 200 type mailbox=200 |
| 05:26.09 | ChannelZ | [whatever]mailbox=200 |
| 05:26.10 | pensmit | and that's it |
| 05:26.17 | pensmit | ok |
| 05:26.19 | pensmit | thanks a lot |
| 05:26.28 | ChannelZ | under whatever peer |
| 05:27.18 | pensmit | hah |
| 05:27.20 | pensmit | that worked |
| 05:27.24 | pensmit | you're the man |
| 05:27.45 | ChannelZ | writes that down |
| 05:28.21 | pensmit | mwi |
| 05:28.30 | pensmit | message waiting indicator? |
| 05:28.47 | pensmit | by the way how do you get the phone to subscribe |
| 05:28.50 | pensmit | hints? |
| 05:28.51 | ChannelZ | Yes'm |
| 05:29.06 | ChannelZ | Well no somewhere in the phone's config |
| 05:29.38 | pensmit | sip show subscriptions |
| 05:29.39 | pensmit | ? |
| 05:29.50 | pensmit | ok thanks |
| 05:29.53 | pensmit | enough for tonight |
| 05:29.54 | ChannelZ | That would show you if any peers were subscribing to any events |
| 05:29.59 | pensmit | oh |
| 05:30.08 | ChannelZ | But it's a device-side thing, not an asterisk thing. |
| 05:30.13 | pensmit | ok |
| 05:30.15 | pensmit | thank you |
| 05:30.44 | *** join/#asterisk Vutral (ss@mirbsd/special/Vutral) |
| 05:31.42 | ChannelZ | There's no need to subscribe if it's working |
| 05:31.50 | pensmit | yes..ok |
| 05:31.52 | pensmit | thank you |
| 05:32.00 | pensmit | just trying to understand more |
| 05:32.40 | ChannelZ | But subscription is basically the device asking asterisk "please tell me when something happens". Setting the mailbox is making asterisk tell the device whether it asked or not. |
| 05:33.05 | ChannelZ | s/tell the device/tell the device anyway/ |
| 05:33.49 | pensmit | thanks a lot |
| 05:41.10 | *** join/#asterisk aruntomar (~Thunderbi@117.219.116.157) |
| 05:41.20 | pensmit | Well that's nuts |
| 05:41.39 | pensmit | the light comes on but now when I press the message button it says |
| 05:41.49 | pensmit | Not Found: asterisk |
| 05:41.55 | pensmit | and gives me a busy signal |
| 05:42.42 | pensmit | before the light wouldn't come on but when I pressed the button it would dial *97 and go to voicemail |
| 05:43.18 | pensmit | exten => *97,1,VoiceMailMain(${CALLERID(num)}@default) |
| 05:43.26 | ChannelZ | The device has a config for what extension it's supposed to dial when you hit the voicemail button |
| 05:43.40 | pensmit | yes |
| 05:43.47 | ChannelZ | look at the console and see what it's actually dialing. |
| 05:43.54 | ChannelZ | Or what is failing. All will be revealed. |
| 05:43.58 | ChannelZ | core set verbose 3 |
| 05:44.10 | pensmit | it's never hitting asterisk |
| 05:44.21 | pensmit | i started teh console with asterisk -rvvvvvv |
| 05:44.35 | pensmit | that message is on the phone |
| 05:44.37 | pensmit | weird |
| 05:44.44 | *** join/#asterisk Vutral (~ss@mirbsd/special/Vutral) |
| 05:44.46 | pensmit | snom 720 |
| 05:45.32 | ChannelZ | then it's busted |
| 05:47.42 | pensmit | no it works fine on freepbx |
| 05:49.42 | ChannelZ | So say you the phone is not sending anything to asterisk. |
| 05:49.58 | ChannelZ | does the math |
| 05:50.58 | pensmit | i set sip debug on and i see stuff |
| 05:51.22 | pensmit | hey do you have to put a nat=yes in the mailbox too? |
| 05:51.44 | pensmit | voicemail.conf |
| 05:51.52 | ChannelZ | that's a peer option not a voicemail thing |
| 05:52.16 | *** part/#asterisk Defraz (~Defraz@gump.fuzecore.com) |
| 05:53.39 | pensmit | Looking for asterisk in employees |
| 05:53.41 | pensmit | hmm |
| 05:53.53 | *** join/#asterisk j4jackj (~jack@99.199.11.127) |
| 05:53.56 | pensmit | a context i have is employees |
| 05:54.01 | pensmit | what is this asterisk crap |
| 05:54.01 | j4jackj | Hello |
| 05:54.04 | pensmit | it's looking for |
| 05:54.23 | pensmit | why doesn't the bastard just dial *97 |
| 05:54.26 | pensmit | the bastard |
| 05:54.27 | pensmit | lol |
| 05:54.39 | j4jackj | pensmit: hello? |
| 05:54.42 | ChannelZ | I'm not clairvoyant. No idea what you're looking at. |
| 05:54.49 | j4jackj | Same here. |
| 05:55.17 | WIMPy | pensmit: Because that's what your sip.conf says. |
| 05:55.38 | pensmit | no the sip.conf doesn't mention asterisk |
| 05:56.05 | j4jackj | ?!?!?! |
| 05:56.05 | WIMPy | Then it's the default if not configured. |
| 05:56.20 | pensmit | default what? |
| 05:56.38 | pensmit | the phone is programmed to just dial *97 when i press the messages button |
| 05:56.50 | WIMPy | vmexten |
| 05:56.53 | pensmit | but after i put mailbox=200 on the sip device |
| 05:56.58 | *** join/#asterisk Vutral (~ss@mirbsd/special/Vutral) |
| 05:57.03 | pensmit | it does something weird |
| 05:57.06 | ChannelZ | This is all meaningless not seeing any console output, error, configs... |
| 05:57.10 | WIMPy | The phone doesn't need to be programmed if it's clever. |
| 05:57.30 | pensmit | what's the vmexten? |
| 05:58.05 | WIMPy | The extension to retrieve vm. |
| 05:58.14 | WIMPy | Just as the name suggests. |
| 05:58.22 | pensmit | ahh |
| 05:58.23 | pensmit | hmm |
| 05:58.33 | pensmit | maybe the phone is looking for an extension called asterisk |
| 05:58.38 | pensmit | for vm stuff |
| 05:58.41 | pensmit | let me put that in there |
| 05:59.17 | WIMPy | I just told you that it's the default. |
| 05:59.29 | pensmit | don't know what you mean by that |
| 05:59.37 | pensmit | the phone is asking for that |
| 05:59.51 | j4jackj | pensmit: then it is not our problem |
| 05:59.53 | WIMPy | vmexten=asterisk |
| 06:00.07 | pensmit | ok.. |
| 06:00.13 | pensmit | i'm confused |
| 06:00.15 | WIMPy | No, it's not asking. Asterisk tells it to call there when it notifies the phone about messages. |
| 06:00.17 | pensmit | the phonen is sending that |
| 06:00.20 | pensmit | oh |
| 06:00.22 | pensmit | shit |
| 06:00.24 | pensmit | ok |
| 06:00.50 | pensmit | can you define vmexten in the general section |
| 06:00.55 | WIMPy | That's why I said the phone does't need to be configured. |
| 06:01.00 | pensmit | yeah |
| 06:01.01 | pensmit | ok |
| 06:01.02 | pensmit | gotcha |
| 06:01.12 | pensmit | based on what Channelz told me earlier |
| 06:01.12 | WIMPy | ... If it speaks an Asterisk compatible version of SIP. |
| 06:01.19 | pensmit | that makes perfect sense |
| 06:02.03 | WIMPy | The funny thing is that this is not true for the Digium phones. |
| 06:02.23 | j4jackj | What? They don't speak AsterSIP? |
| 06:03.07 | pensmit | Worked like a fucking champ |
| 06:03.22 | pensmit | WIMPY and CHANNELZ are awesome! |
| 06:03.24 | pensmit | thanks guys |
| 06:03.28 | ChannelZ | Fucking champs? You have different TV channels than me |
| 06:04.43 | pensmit | should i put vmexten=blah blah in the general section? |
| 06:04.47 | pensmit | of sip.conf |
| 06:04.58 | pensmit | if i want to change that |
| 06:05.05 | pensmit | last question |
| 06:05.11 | pensmit | then i'll leave you guys alone |
| 06:05.17 | pensmit | i tell you what |
| 06:05.20 | pensmit | why don't i try it |
| 06:05.23 | pensmit | and leave you alone anyhow |
| 06:05.24 | pensmit | lol |
| 06:05.27 | pensmit | thanks guys |
| 06:05.29 | WIMPy | yes |
| 06:05.39 | pensmit | I just want you guys to know |
| 06:05.50 | pensmit | Your milkshake truly does bring all the boys to the yard. |
| 06:07.09 | ChannelZ | Things in [general] are auto-inherited as defaults to any peers below. |
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| 06:51.45 | jeev | if ping between two iax2 servers is 25ms, what would cause sip show peers to show the trunk at over 1000ms? it keeps dropping. |
| 06:52.15 | j4jackj | jeev: no such thing as a sip trunk |
| 06:52.27 | jeev | sorry, iax2 show peers |
| 06:52.49 | jeev | it's fluctuating between 34 and 1034 ms even though icmp is 25ms and 0 packet loss |
| 06:52.56 | j4jackj | it's probably to connection problem. |
| 06:52.56 | jeev | woops |
| 06:53.09 | jeev | would i not see the same conneciton problem with ping? |
| 06:53.19 | j4jackj | it's probably thaat the two servers are on a slow line that has low latency, like ISDN |
| 06:53.50 | jeev | so the fact that it's fine for 3 months and today is not, no packet loss anywhere, no bw usage.. |
| 06:53.58 | j4jackj | strnge |
| 06:54.20 | j4jackj | Maybe it's Asterisk being arsey. |
| 06:54.32 | *** join/#asterisk mirela666 (~mirko.bra@iecommailer.itaf.eu) |
| 06:55.10 | jeev | i have two WAN, i took it off the 50 meg symmetrical to the 10/1 dsl. |
| 06:55.34 | jeev | lets see how att to att looks |
| 06:56.20 | *** join/#asterisk atcmmi (~atcmmi@119.122.155.145) |
| 06:56.52 | j4jackj | Put it on the 50/50 <-> 50/50, or the 10/1 <-> 10/1 if the 50/50 is problem |
| 06:56.54 | jeev | ah, same. |
| 06:57.06 | jeev | it happened again. |
| 06:57.14 | jeev | must not be a problem with office A |
| 06:58.20 | jeev | just as a test, i am rebooting the linux router and asterisk box at office B. |
| 06:58.55 | jeev | there is an openvpn connection also between the two locations and that was not fluctuating. |
| 07:09.58 | jeev | lets see how long it lasts before it pms's |
| 07:10.00 | jeev | gona pass out |
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| 12:55.08 | pensmit | I found a way to dial voicemail whether you're busy or unavailable online and all was working well. Here is the code. |
| 12:55.12 | pensmit | http://hastebin.com/juricigiju.coffee |
| 12:55.50 | pensmit | Until this morning. It went to voicemail for someone when they dialed a certain areacode and number. |
| 12:56.02 | pensmit | Yeah...the area code started with 202 |
| 12:56.03 | pensmit | lol |
| 12:56.19 | pensmit | Can anyone show be a better way to do this to avoid this problem? |
| 13:00.56 | pensmit | hmm |
| 13:01.00 | pensmit | gotoif |
| 13:03.39 | kaldemar | someone going to voicemail by dialing something that starts with 202 did not use that extension. unless they managed to dial something that starts with 202-. |
| 13:04.46 | *** join/#asterisk dash_ (~d45h@unaffiliated/dash-/x-7576607) |
| 13:04.58 | kaldemar | GotoIf($["${DIALSTATUS}" = "BUSY"]?busy) etc. and you can use labels instead of separate extensions. |
| 13:11.46 | pensmit | thanks guys |
| 13:12.32 | pensmit | I think the phone itself is sending that - character |
| 13:12.44 | pensmit | when you dial 2027896542 |
| 13:12.50 | pensmit | the goto is perfect |
| 13:14.23 | Katty | good morning |
| 13:14.42 | [TK]D-Fender | Katty: Mew. |
| 13:15.10 | Katty | [TK]D-Fender: how's your back? |
| 13:15.24 | pensmit | Hopefully it's hurting him a lot. |
| 13:15.34 | Katty | frowns |
| 13:15.41 | *** join/#asterisk himura_x_ (~himurax@pc-34-179-160-190.cm.vtr.net) |
| 13:15.47 | pensmit | ...as he's too petty to answer questions now. Sad D-Fender...holding grudges. |
| 13:15.50 | Katty | hopefully it's not quite so knotty. |
| 13:16.09 | [TK]D-Fender | Katty: normalizing. Tendons from the neck and shoulder are kinda tense constantly though. having done a little extra lifting lately I think I can start phasing in the weights slowly |
| 13:16.27 | Katty | nods |
| 13:16.31 | Katty | improvement is improvement! |
| 13:16.38 | Katty | no more clavical problems either? |
| 13:16.59 | pensmit | The universe trying to tell him something maybe? |
| 13:17.27 | Katty | pensmit: would you please go drink some coffee and stop ruining my happy morning? |
| 13:17.37 | [TK]D-Fender | Katty: Clavicle fused weeks ago and I haven't put any strain on it really. minor loss of range of motion, but nothing really inportant |
| 13:17.53 | Katty | pensmit: there's enough crap going on in everyones lives (tho pathetic they may be) without adding to the drama in here. |
| 13:17.56 | pensmit | yes, that's it' kiss his butt so you can get your answer |
| 13:18.10 | pensmit | Katty...D-Fender was the jerk...to me |
| 13:18.14 | pensmit | multiple times |
| 13:18.16 | Katty | pensmit: i have been mister fenders friend for 7 years |
| 13:18.21 | Katty | pensmit: and i will continue to ask how he is doing. |
| 13:18.24 | pensmit | well you should know then |
| 13:18.31 | Katty | pensmit: yes, i generally do. |
| 13:18.35 | Katty | pensmit: why are you so cranky this morning? |
| 13:18.43 | pensmit | ok i'll tellyou why |
| 13:18.48 | pensmit | i have real users with real problems |
| 13:18.49 | Katty | please do. |
| 13:18.51 | pensmit | i started with freepbx |
| 13:18.59 | pensmit | d-fender and his ass clones |
| 13:19.04 | pensmit | wouldn't help me |
| 13:19.08 | pensmit | switched to asterisk |
| 13:19.23 | pensmit | then d-fender helps but in the most obnoxious way |
| 13:19.25 | pensmit | then |
| 13:19.34 | pensmit | refuses to help cause i called him on it |
| 13:19.42 | pensmit | so he's only reaping what he sowed |
| 13:19.42 | Katty | nods |
| 13:19.47 | pensmit | so...fuck him |
| 13:19.52 | Katty | i do understand the frustration of your situation. |
| 13:19.57 | Katty | but please keep in mind we are all volunteers in here |
| 13:20.07 | pensmit | like i said...fuck him |
| 13:20.15 | Katty | we are not paid to sit around and fix your problems for you |
| 13:20.28 | Katty | if you need help, in a bad way, asap..i would recommend a consultant. |
| 13:20.35 | pensmit | I say, if you can't be a normal, nice human being |
| 13:20.37 | Katty | fender can be a bit...blunt at times. |
| 13:20.40 | pensmit | that is considerate of others |
| 13:20.55 | pensmit | don't spend your time harassing people in a channel like this to get your jollies and ego stroked |
| 13:20.59 | pensmit | that's all it is really |
| 13:21.12 | Qwell | pensmit: Go away. |
| 13:21.26 | pensmit | looks like an ass clone |
| 13:21.28 | Katty | but i enjoy harassing people. |
| 13:21.34 | pensmit | well do it |
| 13:21.37 | pensmit | just expect blowback |
| 13:21.39 | Katty | i...do? |
| 13:21.43 | himura_x_ | hello |
| 13:21.44 | *** kick/#asterisk [pensmit!~north@pdpc/sponsor/digium/Qwell] by Qwell (Go. Away.) |
| 13:21.44 | *** join/#asterisk pensmit (~pensmit@unaffiliated/pensmit) |
| 13:22.03 | Katty | good morning, himura_x_ |
| 13:22.22 | pensmit | Qwell if you kick me...it only proves that point |
| 13:22.36 | pensmit | so...no biggie to me as it's only the new users here that actually help |
| 13:22.47 | pensmit | I've seen channels get corrupted this way |
| 13:22.55 | *** mode/#asterisk [+b *!*pensmit@*unaffiliated/pensmit] by Qwell |
| 13:22.56 | *** kick/#asterisk [pensmit!~north@pdpc/sponsor/digium/Qwell] by Qwell (Past logs agree with me. Bye.) |
| 13:22.57 | ickmund | pensmit: I've felt like you do now at a point, but in the end, if you don't want his expert advice, go ask someone else |
| 13:23.13 | himura_x_ | i need your help |
| 13:23.39 | [TK]D-Fender | Qwell: *sigh* |
| 13:24.02 | Qwell | and of course he messages me. Why did I have to get "one of those" this morning? |
| 13:24.11 | himura_x_ | i have a problem with pickup call in asterisk 1.4 |
| 13:24.37 | himura_x_ | i need rescue the callerid in my sip phone |
| 13:24.46 | Katty | hugs Qwell |
| 13:24.47 | Qwell | himura_x_: ...rescue? |
| 13:24.53 | sruffell | hugs all around! |
| 13:24.59 | Katty | hugs sruffell |
| 13:25.08 | [TK]D-Fender | himura_x_: Please note that no branch below 1.8 is actively supported. |
| 13:25.30 | WIMPy | himura_x_: Upgrade and it will magically work. |
| 13:25.51 | Katty | Qwell: i think we need to go find a little coffee shop hole in the wall and try This Morning all over again |
| 13:26.31 | himura_x_ | caller id know who's calling |
| 13:27.06 | himura_x_ | upgrade is oly options |
| 13:27.12 | himura_x_ | only option |
| 13:27.39 | [TK]D-Fender | himura_x_: 1.4 does not have that option. When you do a pickup you a re placing a call, not "receiving" one |
| 13:29.43 | [TK]D-Fender | himura_x_: Current branches have support for pushing the callerID to the phone. |
| 13:29.48 | [TK]D-Fender | himura_x_: UPGRADE <_ |
| 13:30.05 | *** join/#asterisk gaps (48344182@gateway/web/freenode/ip.72.52.65.130) |
| 13:30.20 | himura_x_ | ok thanks |
| 13:30.28 | gaps | can i use hangup handlers for dial status... ? |
| 13:30.50 | [TK]D-Fender | gaps: What is your goal? |
| 13:31.40 | gaps | actually sometimes if my dialed number is busy via gsm network, asterisk is restarting.. inspite of that.. thought of handling the hangup... status |
| 13:32.07 | [TK]D-Fender | gaps: By "asterisk restarting" do you mean actually CRASHING and the process reloading? |
| 13:32.23 | gaps | [TK]D-Fender: yes |
| 13:32.56 | [TK]D-Fender | gaps: then I'm not sure what you meant by "hangup handlers". An Asterisk crash is just a crash.... somehting is unstable. What version are you running? |
| 13:33.14 | gaps | [TK]D-Fender: Asterisk 11.1.2 |
| 13:33.50 | [TK]D-Fender | gaps: 11.5.1 is the current release. Please upgrade and see if the problem persists |
| 13:33.58 | *** join/#asterisk emk (~emk@unaffiliated/emk) |
| 13:34.38 | gaps | [TK]D-Fender: I think I can't upgrade because am using Sangoma GSM card... and they have given patch for Asterisk 11.1.2 |
| 13:35.20 | [TK]D-Fender | gaps: Sangoma does not care about Asterisk. They interface with DAHDI. |
| 13:35.27 | [TK]D-Fender | gaps: Whose version you DON'T have to change |
| 13:35.55 | gaps | [TK]D-Fender: Ok |
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| 13:42.22 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 13:43.51 | gaps | [TK]D-Fender: So no use if I check the dial status and if it is busy I can play some message, so that I can avoid Asterisk service restart... ? |
| 13:44.31 | *** join/#asterisk kresp0 (~kresp0@109.Red-79-144-71.dynamicIP.rima-tde.net) |
| 13:45.04 | [TK]D-Fender | gaps: Asterisk should not be crashing. Crash = game over. No dialplan app is going to save you from that. |
| 13:47.30 | gaps | [TK]D-Fender: Ok... fine.. |
| 13:47.34 | gaps | [TK]D-Fender: Thanks |
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| 14:14.39 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:15.55 | Katty | waves to putnopvut |
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| 14:16.52 | putnopvut | spins in a circle and waves, figuring that one of those directions is where Katty is |
| 14:18.17 | Katty | ^_^ |
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| 14:28.10 | anonymouz666 | I am back |
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| 16:12.32 | Katty | looks in |
| 16:13.11 | [TK]D-Fender | looks out |
| 16:13.30 | newtonr | looks between |
| 16:13.43 | *** join/#asterisk vittorio88 (~vittorio@net-2-34-114-99.cust.dsl.vodafone.it) |
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| 16:15.05 | Katty | prepositional phrases for everyone! |
| 16:15.37 | [TK]D-Fender | it isn't it, is it? |
| 16:20.52 | vittorio88 | Hi, everbody! I've been using Asterisk for a while, and I have a nasty problem I can't fix. I am a SW engineer. |
| 16:20.55 | vittorio88 | I am trying to set ip sip/tls on asterisk 11.5 with blink on win7 and Ubuntu raring. Neither works. |
| 16:20.56 | vittorio88 | I successfully authenticate and can dial, but get NO AUDIO. Same config with no tls works just fine. |
| 16:20.56 | vittorio88 | Asterisk prints: |
| 16:20.57 | vittorio88 | <PROTECTED> |
| 16:20.58 | vittorio88 | [Sep 19 14:40:20] WARNING[3965]: tcptls.c:261 handle_tcptls_connection: FILE * open failed! |
| 16:20.58 | vittorio88 | [Sep 19 14:40:52] NOTICE[3803]: chan_sip.c:27543 handle_request_subscribe: Failed to authenticate device "vitto" <sip:vitto@sip.promaq.mx>;tag=gyrBSLiINm5odzuoicYskTta2IE-edB3 for SUBSCRIBE |
| 16:20.59 | vittorio88 | Any ideas as to the cause or possible debugging steps? I have been tearing my hair out on this one for a couple of weeks now. |
| 16:21.01 | vittorio88 | I set up another Asterisk with a similar config, and it works with the softphone locally. |
| 16:21.01 | vittorio88 | Is it possible the no audio problem is a networking issue, and the not caused by the issue from the warning? |
| 16:21.03 | *** join/#asterisk jasonwert (~w3rt@75-134-81-98.static.aldl.mi.charter.com) |
| 16:21.07 | *** kick/#asterisk [vittorio88!~north@pdpc/sponsor/digium/Qwell] by Qwell (pastebin) |
| 16:23.25 | Faustov | Qwell: he hasn't really broken the rule of 4 lines |
| 16:23.35 | Faustov | just saying... |
| 16:23.48 | Qwell | Last I checked, 13 was bigger than 4. |
| 16:25.42 | [TK]D-Fender | Faustov: note the line doubling and time-stamps that was a giant mangled copy/paste |
| 16:26.10 | Faustov | yeah you're right |
| 16:26.18 | Faustov | just defends people from time to time |
| 16:26.26 | *** join/#asterisk vittorio88 (~vittorio@net-2-34-114-99.cust.dsl.vodafone.it) |
| 16:28.12 | vittorio88 | Hi. I have an issue with Asterisk, sip and tls. It is decribed here: http://pastebin.com/ZFW7NyVe Can anyone gimme a hand? |
| 16:28.13 | *** join/#asterisk Katty (~Katty@97-91-121-70.static.stls.mo.charter.com) |
| 16:28.37 | [TK]D-Fender | vittorio88: [Sep 19 14:40:52] NOTICE[3803]: chan_sip.c:27543 handle_request_subscribe: Failed to authenticate device "vitto" <sip:vitto@sip.promaq.mx>;tag=gyrBSLiINm5odzuoicYskTta2IE-edB3 for SUBSCRIBE <- this is a SUBSCRIBE, not a CALL. |
| 16:28.47 | [TK]D-Fender | vittorio88: You are not showing relevant debug |
| 16:31.02 | vittorio88 | ok. one sec plz. |
| 16:31.49 | vittorio88 | ok. so it is likely that that message is unrelated to the call itself? |
| 16:32.04 | vittorio88 | because the softphone shows authenticated |
| 16:33.40 | vittorio88 | I receive those messages continuously with asterisk running. |
| 16:33.54 | [TK]D-Fender | vittorio88: SUBSCRIBE is not a CALL. |
| 16:34.00 | [TK]D-Fender | \vittREGISTER is not a call. |
| 16:37.26 | vittorio88 | ok. that's clear now. what could be the causes for FILE * open failed ? |
| 16:38.28 | navaismo | not ca installed correctly |
| 16:38.39 | vittorio88 | on client or server end? |
| 16:41.39 | anonymouz666 | is it possible to define the directory of "make samples" on command line? |
| 16:41.46 | anonymouz666 | to not overwrite the default /etc/asterisk |
| 16:44.51 | *** part/#asterisk volga629 (~bendersky@CPE085b0e07d3f2-CM7cb21b15b251.cpe.net.cable.rogers.com) |
| 16:46.37 | vittorio88 | in makeopts, the variable ASTETCDIR defines where make samples puts the samples |
| 16:47.12 | vittorio88 | I would modify it there run make samples and then modify it back because it may serve other purposes |
| 16:47.35 | vittorio88 | alternatively you can temporarily rename /etc/asterisk to /etc/asteriskBACKUP |
| 16:47.38 | vittorio88 | then make samples |
| 16:47.52 | [TK]D-Fender | or just never "make samples" |
| 16:48.37 | [TK]D-Fender | all it does is copy the folder that's right there. It's a practically worthless command. |
| 16:48.58 | *** join/#asterisk k610 (~K610@cable-78.29.241.186.coditel.net) |
| 16:51.31 | vittorio88 | yeah. in configs. |
| 16:53.33 | vittorio88 | is there a proper way to install the ca cert other than just reference it in sip.conf ? |
| 16:55.47 | dongola7 | vittorio88: might be a dumb question, but you have the right permissions set on your certs and keys, right? |
| 16:56.02 | dongola7 | vittorio88: because i've goofed that up regularly |
| 16:56.02 | vittorio88 | yeah, thanks |
| 16:56.07 | vittorio88 | me too. |
| 16:56.19 | vittorio88 | 666 and owner is asterisk |
| 16:57.25 | j4jackj | 666 |
| 17:01.17 | vittorio88 | is there any way to know what file tcptls.c: FILE * open failed! is referring to? |
| 17:03.18 | *** join/#asterisk protocoldoug (~doug@unaffiliated/protocoldoug) |
| 17:03.42 | dongola7 | vittorio88: suppose you could look through the asterisk source (the line number is right there, after all), but it doesn't appear to be obvious at first glance. did you have this working first _withou_ TLS enabled? Might be worth a shot that way. |
| 17:05.24 | vittorio88 | <PROTECTED> |
| 17:05.51 | vittorio88 | i have 2 parallel configs that are identical except for tls |
| 17:05.59 | vittorio88 | udp works, tls doesn't |
| 17:06.04 | dongola7 | vittorio88: looks like it's something with the SSL socket, which traces back to the SSL context, and that's as far as i got. |
| 17:06.41 | vittorio88 | ok, so I guess it's not referring to a file on my system or even my certs |
| 17:06.59 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 17:08.39 | vittorio88 | the strange thing is the sip client authenticates, so that mean SSL is working, correct? I can dial, I just don't have audio. |
| 17:11.35 | dongola7 | vittorio88: can you post your sip.conf file? (hiding the secrets) |
| 17:12.37 | dongola7 | vittorio88: as well as a copy of the asterisk console output when you try to make a call? |
| 17:13.01 | vittorio88 | ok. I need to prepare it. gimme a min |
| 17:26.03 | vittorio88 | here is sip.conf |
| 17:26.04 | vittorio88 | http://pastebin.com/ncMemS4Z |
| 17:26.13 | vittorio88 | here is a test call |
| 17:26.26 | vittorio88 | http://pastebin.com/vSz3tzbH |
| 17:26.35 | vittorio88 | there is no audio for that call |
| 17:27.05 | [TK]D-Fender | vittorio88: TLS is also only the SIP negotiation. |
| 17:27.15 | [TK]D-Fender | viiit has precisely nothing to do with the AUDIO |
| 17:28.51 | vittorio88 | yeah, i know. i'm just mentioning it. |
| 17:29.45 | [TK]D-Fender | vittorio88: You should instead... be looking at the actual full call. |
| 17:30.02 | *** join/#asterisk luke_ftw (~luke_ftw@unaffiliated/luke-ftw/x-9245329) |
| 17:30.06 | vittorio88 | want the sip debug? |
| 17:32.12 | *** join/#asterisk nam3l3zz (~quassel@86-46-207-104-dynamic.b-ras1.pgs.portlaoise.eircom.net) |
| 17:37.14 | *** join/#asterisk eXcAliBuR (~eXcAliBuR@206.162.174.6) |
| 17:37.25 | eXcAliBuR | I need a pastebin |
| 17:37.54 | [TK]D-Fender | Go get one |
| 17:38.07 | eXcAliBuR | http://pastebin.com/6GsrRrTS |
| 17:38.14 | eXcAliBuR | i see lots of them in my cli |
| 17:38.35 | eXcAliBuR | is somone trying to use my box? |
| 17:38.39 | [TK]D-Fender | eXcAliBuR: clearly |
| 17:38.59 | [TK]D-Fender | eXcAliBuR: And you are allowing anonymous SIP calls to hit your dialplan. |
| 17:39.05 | SuperNull | i know this is bad practice but i have an 'emergency' legacy box i need to build.. how do i get the addons to compile ? i do make in the folder and it bitches |
| 17:39.16 | eXcAliBuR | how do i fix it? |
| 17:39.40 | [TK]D-Fender | eXcAliBuR: "allowguest=no" in [general] |
| 17:41.18 | eXcAliBuR | i did thaqt |
| 17:41.21 | eXcAliBuR | am i safe now? |
| 17:41.36 | eXcAliBuR | did i stop al-quada from using my box? |
| 17:42.08 | [TK]D-Fender | depends on your definition of "safe" and "from whom" |
| 17:42.55 | SuperNull | nm got the problem was missign dev files. |
| 17:47.52 | ChannelZ-Wk | use fail2ban to block them |
| 17:50.28 | dongola7 | vittorio88: sorry, i got nothing. i'm not seeing anything weird in the sip.conf, but those SSL errors bother me. |
| 17:52.46 | vittorio88 | thanks for taking a look at it! they are ugly errors. They are the only ones present on my box and they are recurring. |
| 17:54.46 | vittorio88 | i'm going afk, but i'll be back later. please let me know if anybody find anything! |
| 17:59.57 | *** join/#asterisk jpoz (~jpoz@c-24-22-121-42.hsd1.or.comcast.net) |
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| 18:14.12 | anonymouz666 | who is going to astricon 10? |
| 18:14.24 | ChannelZ-Wk | I"m going to the bathroom in 10 |
| 18:14.28 | Qwell | anonymouz666: #astricon |
| 18:15.16 | _Corey_ | There's an #astricon ? ;) |
| 18:15.41 | Qwell | _Corey_: every year |
| 18:15.52 | *** join/#asterisk MLNoah (~chatzilla@noc.metalink.net) |
| 18:15.55 | *** join/#asterisk d45h (~d45h@unaffiliated/dash-/x-7576607) |
| 18:18.00 | _Corey_ | I hope someone checked the hotel this year to make sure there was actually cell coverage... ;) |
| 18:19.56 | mjordan | WiFi and cell coverage at conferences is like snipe hunting. It sounds plausible, but is really just an exercise in futility. |
| 18:20.26 | newtonr | Yeah I was about to say. Your choice of building is constrained by much more than cell coverage availability. |
| 18:20.35 | _Corey_ | AT&T was particularly problematic last year on the ground floor... |
| 18:21.40 | newtonr | _Corey_, getting good wifi and cell coverage was high on everyones list, so there is that. No guarantees. :D |
| 18:24.17 | *** join/#asterisk g_r_eek (~g_r_eek@ppp-2-85-214-57.home.otenet.gr) |
| 18:28.07 | *** join/#asterisk gaps (48344182@gateway/web/freenode/ip.72.52.65.130) |
| 18:30.34 | navaismo | My exboss will talk in the astricon somone please make a stream |
| 18:31.28 | _Corey_ | newtonr: I'm kidding around slightly... obviously it's a challenge. I think the total lack of signal was confined to AT&T |
| 18:31.51 | mjordan | navaismo: talks are typically recorded |
| 18:31.59 | mjordan | navaismo: live streams are a bit more challenging |
| 18:32.23 | navaismo | :( |
| 18:32.39 | newtonr | _Corey_, Yeah I had AT&T and had to go all the way outside the building to get any signal last year. I think they had a Faraday cage in there |
| 18:32.53 | leifmadsen | #astricon is now open for business! |
| 18:33.01 | MLNoah | I have an Asterisk 11 system where some peers are configured so that their MWI light checks multiple mailboxes (e.g. SIPPEER(peerinfo,mailbox) returns mailbox@context,mailbox2@context). I want users who only have one mailbox that they're monitoring to automatically get connected to their mailbox with VoiceMailMain, but users with multiple to be prompted for which # to check |
| 18:33.02 | leifmadsen | Qwell: oh you already mentioned it |
| 18:33.05 | leifmadsen | nice moves |
| 18:33.06 | Qwell | leifmadsen: so late |
| 18:33.09 | leifmadsen | Qwell: ikr? |
| 18:33.56 | MLNoah | is there a performance difference between using REGEX() or $["VAL" = FILTER("VAL")] to see if the user has two mailboxes? or is there another, better way to do it that i'm not thinking of? |
| 18:36.23 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 18:36.43 | *** join/#asterisk giovani (~giovani@unaffiliated/giovani) |
| 18:36.55 | giovani | anyone here used voxbone? |
| 18:44.30 | ChannelZ-Wk | Regex is probably slower than most but unless you're doing it thousands of times a second I'm not sure it could possibly matter |
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| 19:53.48 | *** join/#asterisk foamz (~foamy@pool-173-56-99-246.nycmny.fios.verizon.net) |
| 19:53.49 | foamz | hi |
| 19:53.56 | foamz | can anyone recommend a voip softswitch? |
| 19:54.01 | foamz | i was looking at A2Billing |
| 19:54.11 | foamz | not sure if that is a recommended solution or not though |
| 19:55.44 | [TK]D-Fender | a2billing is not a "voip softswitch" |
| 19:55.50 | pabelanger | not sure if trolling or serious question |
| 19:56.12 | [TK]D-Fender | pabelanger: "Powered By Ignorance" (tm) |
| 19:56.14 | [TK]D-Fender | ;) |
| 19:56.14 | pabelanger | foamz, what do you want to do? |
| 19:57.10 | *** join/#asterisk ketas- (ketas@ketas6-sixxs.si.pri.ee) |
| 19:59.18 | *** join/#asterisk ppc (ppc@cloudy.cmang.org) |
| 20:02.36 | foamz | uhm sorry |
| 20:02.52 | foamz | i was looking at this http://www.asterisk2billing.org/ |
| 20:02.52 | *** join/#asterisk g_r_eek (~g_r_eek@ppp-2-85-214-57.home.otenet.gr) |
| 20:03.02 | foamz | calls itself a voip billing softswitch |
| 20:03.17 | foamz | basically terminating VOIP calls to a GSM gateway from a call originator |
| 20:04.51 | foamz | sorry for ignorance |
| 20:05.07 | foamz | is what im saying making sense? |
| 20:08.10 | pabelanger | foamz, well, using asterisk you can build it yourself |
| 20:08.24 | pabelanger | if you are looking for an out of the box solution, there are many different ones |
| 20:08.56 | drmessano | A2Billing is just a GUI using Asterisk. You don't need A2Billing to terminate voip calls to a GSM gateway |
| 20:09.17 | drmessano | Its overkill, actually.. |
| 20:09.45 | [TK]D-Fender | foamz: that is a set of scripts for BILLING based on Asterisk CDR's |
| 20:09.59 | [TK]D-Fender | foamz: In and of itself A2Billing doesn't do "voip" .... or anything |
| 20:10.19 | [TK]D-Fender | foamz: No more than an iPhone case lets you place calls. |
| 20:12.32 | ppc | [TK]D-Fender: hey whats up |
| 20:14.55 | foamz | ok but will Asterisk do what I need? |
| 20:15.29 | [TK]D-Fender | foamz: What does this GSM gateway speak? |
| 20:16.25 | foamz | what do you mean speak? like SIP? |
| 20:16.57 | foamz | or like G.711 G.729 |
| 20:17.09 | [TK]D-Fender | foamz: Well * can talk "sip" to that calling end... it's a question of what it would have to speak to this other piece of equipment you're referring to |
| 20:18.01 | [TK]D-Fender | [16:16]foamzwhat do you mean speak? like SIP? [16:16]foamzor like G.711 G.729 <- this is not an "or" question. SIP is a protocol, the other 2 are voice codecs |
| 20:18.22 | foamz | yeah |
| 20:18.31 | foamz | i was asking if he was asking for protocol or codec |
| 20:18.44 | foamz | sorry if that wasnt clear |
| 20:18.52 | foamz | noob trying to learn |
| 20:19.08 | [TK]D-Fender | foamz: You need to understand what each side is speaking. |
| 20:19.29 | foamz | yeah but what do you mean by that? what exactly do I need to match up? |
| 20:19.32 | [TK]D-Fender | foamz: If it is a "GSM Gateway", then one end is obviously GSM ... and the other end is SOMETHING ELSE |
| 20:19.43 | *** join/#asterisk g_r_eek (~g_r_eek@ppp-2-85-214-57.home.otenet.gr) |
| 20:19.48 | [TK]D-Fender | foamz: It could be a box with POTS lines out the back for all we know |
| 20:20.06 | [TK]D-Fender | foamz: What that box does will change what you need for * to talk to it |
| 20:20.46 | *** join/#asterisk gumati (~gumati@197.215.128.2) |
| 20:20.50 | foamz | oh |
| 20:21.01 | foamz | it changes VoIP-SIP or PRI-ISDN to GSM |
| 20:21.06 | gumati | any expert in sip and dahdi small job but urgent |
| 20:21.26 | [TK]D-Fender | gumati: Does it really require an "expert"? |
| 20:21.32 | [TK]D-Fender | gumati: How much of one? |
| 20:21.38 | ppc | lol |
| 20:21.47 | gumati | to extension '218925474843' rejected because extension not found in context 'default'. |
| 20:21.52 | gumati | this my error |
| 20:21.52 | [TK]D-Fender | foamz: Will if it can speak SIP ... and your OTHER end speaks SIP .... the sure |
| 20:22.17 | [TK]D-Fender | gumati: It is clearly sending your call into [default] based on your configs and you don't have a dilplan match for the number |
| 20:22.31 | [TK]D-Fender | gumati: it is VERY specific about what it is looking for, and where. |
| 20:22.47 | [TK]D-Fender | gumati: You shouldn't have to guess why it isn't working. |
| 20:22.57 | gumati | i know |
| 20:23.02 | [TK]D-Fender | gumati: Go look at your extensions and see why it is you don't have amatch there for it |
| 20:23.07 | gumati | but am not asterisk and linux |
| 20:23.12 | gumati | and it's urgent |
| 20:23.29 | gumati | i know it's in extension.conf |
| 20:24.23 | *** join/#asterisk vaneck (~vaneck@96-38-5-189.static.jcsn.tn.charter.com) |
| 20:25.57 | ppc | I have a feeling that was urgent |
| 20:28.37 | navaismo | i can charge 300USD for that fix |
| 20:28.50 | navaismo | hehe :D |
| 20:29.14 | [TK]D-Fender | checkout time, BBIAB |
| 20:29.36 | *** join/#asterisk gumati (~gumati@197.215.128.2) |
| 20:30.03 | gumati | sorry got disconnected |
| 20:30.20 | gumati | anyone who willing help me my skype: gumati_m |
| 20:30.53 | Penguin | Now why would I want to go use skype when I'm right here on IRC? |
| 20:31.27 | ppc | so a while ago I setup PIAF to just mess around, tried installing it again last night, what a mess |
| 20:31.53 | ppc | gonna try out elastix |
| 20:31.55 | Penguin | It's horrible. If you want something nice and also quick, see AsteriskNOW. |
| 20:32.06 | Penguin | PiaF and Elastix are both horrible. |
| 20:32.20 | ppc | yeah? |
| 20:33.10 | ppc | Penguin: Whats so good about AsterisKNOW? |
| 20:33.18 | navaismo | gumati: oh you miss my bid of 300USD |
| 20:33.24 | ppc | lol |
| 20:33.32 | ppc | navaismo: IT IS URGENT |
| 20:33.41 | navaismo | yes thats why are 300USD |
| 20:33.51 | Penguin | The best thing is that is doesn't have a bunch of bullshit that you have to wade through to use it, AND we can support it fairly well. |
| 20:33.52 | navaismo | per hour* |
| 20:33.54 | ppc | I have to get my production VOIP system working RIGHT NOW, so I hop onto any IRC channel I can find! |
| 20:34.22 | Penguin | We can support it even better if you don't mess it up by installing the FreePBX option on it. If you do that, you'll have to get support from the FreePBX channel. |
| 20:34.26 | navaismo | Penguin: wrong if you mentioned freepbx here the asterisk guys jump and slap your head woth the ~freepbx |
| 20:34.39 | Penguin | I said AsteriskNOW. |
| 20:34.43 | Penguin | Not FreePBX. |
| 20:35.04 | navaismo | Usually people installing asteriskNow intall it with freepbx :P |
| 20:35.05 | Penguin | But then I mentioned how if he chooses the FreePBX option, he has to go to the FreePBX channel for help. |
| 20:35.12 | Penguin | They're not all there in the head. |
| 20:35.39 | *** join/#asterisk gumati (~gumati@197.215.128.2) |
| 20:35.42 | gumati | any one here instersted to make money from small and easy job please added me in skype : gumati_m |
| 20:35.48 | navaismo | I guess my bid was to high for an urgent job |
| 20:35.57 | ppc | navaismo: theres your sign! |
| 20:35.58 | gumati | didnt see it |
| 20:36.05 | navaismo | :D |
| 20:36.25 | navaismo | 300 per hour :P |
| 20:36.25 | gumati | what was ur bid? |
| 20:36.53 | gumati | dont mind at all |
| 20:37.00 | gumati | it's one hour job |
| 20:37.05 | gumati | even less |
| 20:37.08 | gumati | add me |
| 20:37.33 | ppc | damn |
| 20:37.45 | ppc | I need to get into irc tech support 1099 work |
| 20:37.52 | gumati | :D |
| 20:39.06 | navaismo | gumati: i dont use skype |
| 20:39.15 | gumati | gumati@gmail.com |
| 20:41.11 | gumati | are you in? |
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| 20:55.21 | anonymouz666 | simple question: Using Ast 11, the output of "logger show channels" is "Console Enabled - DEBUG NOTICE WARNING ERROR VERBOSE DTMF FAX" core set verbose 3 and even with that I can't see [VERBOSE] stuff in CONSOLE |
| 20:55.29 | anonymouz666 | why? |
| 20:56.51 | anonymouz666 | - /var/log/asterisk/full File Enabled - DEBUG NOTICE WARNING ERROR VERBOSE DTMF FAX - here outputs correctly |
| 20:58.14 | navaismo | you mean the formated log in the cli? |
| 21:02.10 | elguero | anonymouz666: Checkout https://issues.asterisk.org/jira/browse/ASTERISK-21921... that might be what you are seeing |
| 21:03.22 | anonymouz666 | Yeah... I am running 11.6-rc1 |
| 21:04.49 | mjordan | except that should be fixed in that version. |
| 21:04.54 | mjordan | So if you're seeing something else, it's new. |
| 21:05.33 | mjordan | or at least, a variation on the theme of verbose messages having problems in the console. |
| 21:08.17 | anonymouz666 | I am trying to see, but verbose doesn't show up :-) |
| 21:09.09 | newtonr | anonymouz666, what levels does "core show settings" show your debug and verbose at? |
| 21:09.33 | anonymouz666 | debug 0, verbose 3 |
| 21:09.36 | newtonr | anonymouz666, and are you on a -c console or a -r console ? |
| 21:09.41 | anonymouz666 | asterisk -rv |
| 21:10.07 | anonymouz666 | I mean asterisk -rvvv :-P |
| 21:11.29 | newtonr | anonymouz666, when you say verbose doesn't show up, as you looking for items prefixed with VERBOSE, or are you looking for lines that don't have a prefix (which is what verbose messages look like on the console) |
| 21:12.23 | anonymouz666 | yes I am looking for items prefixed with [VERBOSE], I want to show the tech team the call-id stuff. |
| 21:13.10 | Penguin | If you want it to say VERBOSE, change all your Verbose() lines in dial plan to include those letters in that configuration. |
| 21:13.25 | newtonr | anonymouz666, on the console, the VERBOSE items don't have a VERBOSE prefix. |
| 21:14.20 | anonymouz666 | so the logger option verbose for file 'full' is behave different than verbose option for console? |
| 21:14.54 | mjordan | yup |
| 21:15.10 | mjordan | verbose to console is a different beast. In fact, it has some specific variants specifically for various consoles |
| 21:15.24 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 21:16.37 | anonymouz666 | So in the past wasn't that way. Things change. Hehe. Is there a way to [VERBOSE] everything on CLI instead of spreading Verbose() into dialplan? |
| 21:17.55 | [TK]D-Fender | anonymouz666: What is this "everything" you're speaking of? |
| 21:18.59 | anonymouz666 | [Sep 26 18:18:11] VERBOSE[14202][C-0000000f] netsock2.c: == Using SIP RTP TOS bits 184 ... [Sep 26 18:18:11] VERBOSE[14202][C-0000000f] app_dial.c: -- SIP/300-00000017 is ringing |
| 21:19.22 | anonymouz666 | I want to make it to possible for the tech team to see this kind of verbose inside asterisk -r |
| 21:19.37 | navaismo | thats what i talking about^ |
| 21:20.04 | [TK]D-Fender | You already always see that stuff at CLI.... |
| 21:20.12 | anonymouz666 | Not in Ast 11. |
| 21:20.16 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.124.249) |
| 21:20.19 | anonymouz666 | Are you using Asterisk 11? |
| 21:20.26 | *** join/#asterisk g_r_eek (~g_r_eek@ppp-2-85-214-57.home.otenet.gr) |
| 21:20.28 | [TK]D-Fender | Then you aren't at the right CLI verbose level |
| 21:20.33 | [TK]D-Fender | and yes. |
| 21:20.40 | anonymouz666 | I am at verbose 3. |
| 21:20.45 | [TK]D-Fender | core set verbose 10 |
| 21:20.57 | anonymouz666 | doesn't work with 10 either. |
| 21:21.06 | [TK]D-Fender | <PROTECTED> |
| 21:21.17 | [TK]D-Fender | You're doing somethng wrong... |
| 21:21.24 | anonymouz666 | with [VERBOSE] prefix? |
| 21:21.36 | anonymouz666 | with CALL-ID stuff just like I pasted ? |
| 21:21.45 | [TK]D-Fender | That... would be another matter |
| 21:22.12 | [TK]D-Fender | And where do those ID's come from? |
| 21:22.59 | anonymouz666 | - /var/log/asterisk/full |
| 21:23.50 | anonymouz666 | navaismo: are you fighting with this also? |
| 21:24.22 | navaismo | nope |
| 21:25.16 | navaismo | a workaround for you is: give this commadn to your tech --> tail -f /var/log/asterisk/full |
| 21:26.14 | newtonr | anonymouz666, if you want to see verbose messages with a VERBOSE prefix, use the logs. If you don't want the verbose prefix, use the console. That is as simple as it gets. Go beyond that and you are going to complicate things. |
| 21:26.27 | navaismo | +100 |
| 21:27.16 | anonymouz666 | But using logs I lost the colors... :-) |
| 21:27.23 | drmessano | deskfaces |
| 21:27.25 | anonymouz666 | alright guys, I understood. |
| 21:27.33 | anonymouz666 | thanks for the explanation |
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| 21:59.03 | ppc | Penguin: I got it working, kind of |
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| 22:06.24 | ppc | I can't hear anything on my 7940 but my voice does make it to the caller |
| 22:09.52 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-nggznpxedvxdhbvo) |
| 22:12.45 | *** join/#asterisk snadge (snadge@unaffiliated/snadge) |
| 22:14.13 | Penguin | Sounds like a NAT problem. |
| 22:15.06 | ppc | dude |
| 22:15.12 | ppc | how did i not see that, yes that was it |
| 22:15.17 | snadge | anyone done much work on the number of simultaneous calls or call connection attempts per second with asterisk? |
| 22:15.38 | snadge | basically finding the limits of any individual asterisk node |
| 22:15.53 | pabelanger | snadge, load question, really depends on what you are doing |
| 22:15.58 | pabelanger | eg: transcoding |
| 22:16.18 | pabelanger | if you need load, drop a proxy in front of asterisk to off-load a bunch of the SIP traffic |
| 22:16.35 | snadge | yeah.. i will be looking at netscaler for that |
| 22:17.00 | snadge | but im still curious about how easy it will be to get asterisk to fall over |
| 22:17.14 | pabelanger | that is a loaded question |
| 22:17.14 | snadge | apparently the limitation isn't cpu, disk, memory or anything like that |
| 22:17.18 | snadge | but ports |
| 22:17.29 | snadge | running out of available ports |
| 22:17.33 | snadge | in the kernel or whatever |
| 22:17.56 | snadge | so clearly the solution to that is to load balance like you say |
| 22:17.58 | pabelanger | snadge, what are you trying to do |
| 22:18.14 | snadge | well its a voip telco.. so route lots of calls i guess |
| 22:18.40 | snadge | i have only just started, so im unfamiliar with the specific configuration details.. ie.. whether transcoding is used etc, im not up to that yet |
| 22:19.05 | snadge | my task is to set up a test environment, to replicate what they're using in production |
| 22:19.09 | pabelanger | well, you need to map it out, because you're in for some work |
| 22:19.21 | snadge | yeah i very much realised that when i signed up for it |
| 22:19.49 | pabelanger | start with 1 proxy (kamailio) and 1 asterisk box. Get it working, then add another asterisk box |
| 22:19.51 | snadge | my level of experience with asterisk is using asterisk now in a small office, with an incoming call directory.. and one outgoing trunk to a single sip provider |
| 22:19.53 | pabelanger | then another proxy |
| 22:20.00 | pabelanger | rinse and repeat |
| 22:20.11 | snadge | and using the freepbx web frontend to do that |
| 22:20.33 | pabelanger | ya, not going to work using freepbx, it is only designed for a single box |
| 22:20.44 | snadge | going from that.. to voip telco, with hundreds of thousands of calls per day |
| 22:20.57 | snadge | and of course manual asterisk configs etc |
| 22:21.44 | snadge | its a bit of a stretch i know.. but im not a complete noob, so it should be achievable |
| 22:22.12 | pabelanger | Yup, I'm in the middle of doing it right now. |
| 22:22.30 | snadge | telco or internal office type situation? |
| 22:22.38 | snadge | the latter is much less stressful ;) |
| 22:23.14 | pabelanger | deployment of a VoIP network across co-los |
| 22:23.35 | snadge | cool |
| 22:23.46 | snadge | kamailio is something i should look at then |
| 22:23.59 | snadge | the technology chief wants me to check out netscaler |
| 22:24.15 | snadge | because its a brand name.. and somewhere in its marketing spiel goobledigook it meantions SIP |
| 22:24.30 | snadge | so it must be good ;) |
| 22:24.40 | snadge | citrix wants you to pay money for it |
| 22:26.05 | pabelanger | ya, you'll want to us something like kamailio or opensips |
| 22:26.12 | pabelanger | not sure what you can all do with netscaler |
| 22:26.23 | pabelanger | but you'll likely need to be re-writing SIP headers |
| 22:26.26 | snadge | likely more.. but the question is do we need that and how much does it cost |
| 22:26.36 | snadge | yes, i read some documentation that mentions reverse nat |
| 22:27.08 | snadge | so.. the netscaler probably only does what kamailio or opensips does.. with sip in particular |
| 22:27.14 | pabelanger | there is also specific integration stuff you need to do with asterisk |
| 22:27.16 | snadge | obviously netscaler is a massive product that does lots of things |
| 22:27.21 | pabelanger | right |
| 22:27.44 | snadge | if we needed to load balance things other than SIP, it might be worth it |
| 22:27.56 | snadge | i dont know.. ive been asked to look at it so i will, including how much it costs |
| 22:28.08 | snadge | and whether it does anything that the free ones dont do.. which i doubt |
| 22:28.20 | pabelanger | well, pay for the product or pay to hire the consultant |
| 22:28.23 | pabelanger | will be about the same |
| 22:28.29 | snadge | there is a free 30 day trial |
| 22:28.39 | snadge | so that presumably will be enough to see whether it does anything good or not |
| 22:28.41 | snadge | hopefully |
| 22:30.15 | ppc | snadge: that sounds like a giant project |
| 22:30.44 | pabelanger | ppc, it is |
| 22:31.51 | snadge | yeah i know.. i basically signed up because im a linux sysadmin, and the job was local |
| 22:32.01 | snadge | doh.. oh well ;) |
| 22:33.43 | snadge | apparently dialers are a bit of an issue.. some customers like to send a hundred call connection attempts per second |
| 22:33.59 | snadge | of course, a small percentage of those calls are actually successful.. but that doesnt matter |
| 22:34.03 | snadge | the server still has to process them :p |
| 22:34.17 | *** join/#asterisk imcdona (~Thunderbi@c-50-159-37-24.hsd1.wa.comcast.net) |
| 22:34.59 | snadge | also its not so bad when one person decides to do that.. its when 3 or 4 people etc |
| 22:35.07 | snadge | all decide to run their dialers at the same time |
| 22:35.18 | snadge | facepalm worthy ;) |
| 22:36.22 | *** part/#asterisk vittorio88 (~vittorio@net-2-34-114-99.cust.dsl.vodafone.it) |
| 22:36.49 | snadge | this is the kind of stuff you don't have to deal with.. when you arn't dealing with the general public |
| 22:37.05 | snadge | if it was an office situation.. you'd be like.. what are you doing you douchebag.. stop that |
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| 22:53.02 | ledoktre | greetings everyone. I'm having a weird problem that I'm hoping is just something simple on my part. Downloaded Certified LTS 1.8.15-cert3 this morning and went to compile and load it. Problem is, sip wasn't working. I figured out that the module was no where to be found. I thought it must be something I turned off in `make menuselect` - a dependency or something, so I did `make clean` and re-installed from a fresh directory. No luck. I did not |
| 22:53.02 | ledoktre | install libpri, but I am not and have no intentions of using any PRI channels. Ideas? |
| 23:10.09 | ppc | weird, everytime i reboot my phone it needs to have all the settings put back in it |
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