| 00:01.19 | *** join/#asterisk pdtpatr1ck (~pdtpatric@ip184-179-74-42.oc.oc.cox.net) |
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| 00:29.48 | devdvd | jaddison |
| 00:30.06 | devdvd | nm lol |
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| 00:47.45 | Ice_Strike | @file you there? |
| 00:48.24 | Ice_Strike | I could not get damn DTMF Send working :( |
| 00:48.49 | Ice_Strike | I keep getting a few second pause/silent and then I can hear the operater talking |
| 00:49.37 | Ice_Strike | See debug log http://pastebin.com/FwvFFHvA |
| 00:49.42 | Ice_Strike | does that mean anything wrong? |
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| 01:40.54 | Tagor | I just upgraded lenny to squeeze but I seem to be unable to receive calls now. The only thing I see in the CLI is 'Using SIP RTP CoS mark 5'. Anyone know what this means and what's wrong? |
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| 02:15.51 | *** join/#asterisk needascreenname_ (~fee@unaffiliated/shockshock) |
| 02:16.17 | Tagor | Never mind, found it. Damn a lot of things changed in Asterisk :P |
| 02:17.03 | p3nguin | ice_strike: What channel tech are you using to dial out? |
| 02:20.06 | needascreenname_ | hey |
| 02:20.14 | needascreenname_ | which voip providers support cid spoofing? |
| 02:21.32 | *** join/#asterisk ks3 (ks3@tatooine.digitallotus.com) |
| 02:22.06 | Ice_Strike | p3nguin what do you mean? |
| 02:29.13 | Tagor | Is there an easy way to disable the playback of vm-intro.gsm? |
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| 03:01.26 | *** join/#asterisk Assid (~kvirc@unaffiliated/assid) |
| 03:01.44 | Assid | has anyone been successful in connecting an NEC legacy ip pbx to asterisk |
| 03:07.52 | Kobaz | mmm |
| 03:12.56 | ChannelZ | You're probably the one person here trying :) |
| 03:13.13 | ardweebno | Oh snap! |
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| 03:33.09 | Assid | hmm |
| 03:33.16 | Assid | highly annoying system |
| 03:34.06 | ChannelZ | What is the issue? How are you trying to interconnect them? |
| 03:34.46 | Assid | well it required me to procure a voip card.. which i did |
| 03:34.53 | Assid | *on trial* |
| 03:35.15 | Assid | im trying to have it connect to my asterisk server.. which is on a non standard tcp port |
| 03:35.15 | ChannelZ | a voip card? For the NEC you mean? |
| 03:35.19 | Assid | which doesnt work |
| 03:35.25 | Assid | yes.. for the NEC.. |
| 03:35.35 | ChannelZ | Is it SIP? |
| 03:35.50 | Assid | yep |
| 03:36.18 | Assid | the interface asks for me for the sip port etc |
| 03:36.56 | ChannelZ | but what, it ignores the port number you use? |
| 03:37.34 | Assid | i dont see it connect |
| 03:38.08 | Assid | and the stupid thing is crazy.. it asks for the sip registration information in multiple places |
| 03:38.09 | ChannelZ | Are they on the same physical network? |
| 03:38.41 | Assid | nah.. sip server is on the internet |
| 03:39.13 | ChannelZ | So the * box is remote.. firewalls involved? |
| 03:39.28 | ChannelZ | (seeing as you said it's on a non-standard port number..) |
| 03:40.36 | Assid | outgoing connection.. and my outgoing firewall is whitelisting both the * ip AND the NEC system ; so 2 rules to make sure this doesnt happen |
| 03:41.42 | ChannelZ | wait.. so you're trying to get the * to register TO this NEC thing? |
| 03:41.53 | Assid | no.. the NEC to the * |
| 03:42.07 | Assid | * on the internet remember |
| 03:42.14 | Assid | nec ->>outgoing to * |
| 03:43.13 | ChannelZ | sorry so the firewall is on the NEC side.. and nothing on the * side? |
| 03:44.30 | Assid | NEC side |
| 03:44.45 | Assid | but its allowing all outgoing ports for the NEC |
| 03:46.27 | ChannelZ | So either traffic is getting stopped before it gets out, or there is no traffic being generated in the first place :) |
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| 03:47.37 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 03:47.40 | Assid | nah.. sometie while i was playing with it.. i managed to get it to register.. but it kept unregistering after 20-30 seconds |
| 03:48.54 | Assid | how do i know if the sip client is using tcp or udp to connect |
| 03:49.24 | ChannelZ | sip is 99% of the time UDP and you have to go out of your way to use TCP as far as * is concerned |
| 03:50.05 | ChannelZ | But I suppose to see what's going for sure if you don't know what this NEC thing is doing, is to use wireshark or something |
| 03:51.49 | Assid | hmmm okay.. sip debug says udp? |
| 03:52.16 | Assid | REGISTER sip:216.185.x.x;transport=udp SIP/2.0 |
| 03:52.33 | ectospasm | SIP is generally UDP, but it doesn't have to be. Doesn't make sense for RDP to be TCP... |
| 03:52.44 | ectospasm | s/RDP/RTP |
| 03:52.59 | ectospasm | s/$/\// |
| 03:54.07 | Assid | okay im doing 1 thing at a time |
| 03:54.11 | Assid | first udp->tcp |
| 03:54.21 | Assid | 2;transport=tcp SIP/2.0 |
| 03:54.24 | Assid | that changed |
| 03:54.30 | Assid | now.. port change |
| 03:55.03 | ectospasm | bindport is pretty straightforward to change. |
| 03:55.56 | Assid | ectospasm: not for the nec |
| 03:59.37 | ectospasm | Ah, I only know Asterisk |
| 04:01.49 | Assid | okay now that i got the port changed.. now i need to know.. why im getting 401 unauthorized |
| 04:02.21 | ectospasm | 401 Unauthorized could be normal, it may be expecting the other side to send the INVITE again with the authorization headers... |
| 04:02.38 | ectospasm | s/s...$/.../ |
| 04:02.45 | Assid | yeah but its not registering |
| 04:03.25 | ectospasm | right, some registrars send 401 Unauthorized in response to the first INVITE. Some may send 407 Proxy Authentication Required... |
| 04:03.43 | ectospasm | ...or REGISTER, not just INVITE. |
| 04:05.10 | Assid | well its not effectively registering |
| 04:05.24 | Assid | i managed to get the stupid thing to atleast try and talk |
| 04:05.53 | ectospasm | have you examined the traffic in wireshark? |
| 04:06.32 | Assid | nope... :( |
| 04:06.43 | Assid | i was hoping not to have to end up doing that |
| 04:06.48 | ectospasm | you should, it may be enlightening. |
| 04:06.51 | Assid | i get lost in wireshark |
| 04:07.06 | Assid | as far as sip traffic is concerned |
| 04:07.24 | ectospasm | just realized he doesn't have wireshark installed... |
| 04:07.27 | ectospasm | fixing that now. |
| 04:09.21 | ectospasm | Assid: give me a moment, I'll walk you through looking at a SIP capture in wireshark. |
| 04:09.31 | ectospasm | Assid: do you have Wireshark installed? |
| 04:10.13 | Assid | no.. lemme get it |
| 04:11.09 | ectospasm | be sure to tell me what version you have, since it may differ from mine quite a bit |
| 04:28.24 | *** join/#asterisk Assid (~kvirc@unaffiliated/assid) |
| 04:28.28 | Assid | stupid internet |
| 04:28.32 | Assid | ectospasm: you still around |
| 04:28.37 | ectospasm | I am. |
| 04:29.35 | Assid | okay loaded1.6.7 on my winbox |
| 04:29.55 | Assid | first ineed a tcpdump on my * box right |
| 04:31.13 | *** part/#asterisk Bullmoose (~Bullmoose@70-58-133-94.bois.qwest.net) |
| 04:33.48 | ectospasm | Assid: it'd help if I was looking at your pcap. Can you put it on dropbox or something so I can grab it? DCC isn't a good option because I'm not local to this VPS |
| 04:34.20 | ectospasm | ...not to mention I don't know where weechat would store it. |
| 04:41.17 | ectospasm | dammit, why do they keep dropping? |
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| 04:45.03 | Assid | argh |
| 04:45.05 | Assid | ectospasm: |
| 04:45.11 | Assid | okay now wjat |
| 04:45.27 | Assid | had to go reboot the entire network,, was going crappy |
| 04:45.27 | ectospasm | send me the pcap |
| 04:45.36 | ectospasm | yeah, I get that at times. |
| 04:45.46 | ectospasm | rebooting the router is the way to go sometimes. |
| 04:46.27 | Assid | yeah.. just did that.. but i got a multi homed network |
| 04:46.31 | Assid | 3 internet connections :D |
| 04:47.18 | ectospasm | I live alone. I'm the network admin, the cook, the housekeeping, and the renter. |
| 04:47.30 | Assid | so gotta reboot alot of crap |
| 04:47.45 | Assid | well.. atleast you dontg have people yelling at you if you forget something on the table |
| 04:48.02 | Assid | okay so what package do i install on debian |
| 04:48.11 | Assid | tcpdump? |
| 04:48.46 | ectospasm | tcpdump is good |
| 04:49.10 | ectospasm | although wireshark can capture |
| 04:49.32 | Assid | remote * box.. |
| 04:49.36 | ectospasm | Assid: and I've got wireshark 1.6.7 |
| 04:51.18 | Assid | remote * box.. :( |
| 04:51.29 | ectospasm | so use tcpdump |
| 04:51.42 | ectospasm | ...and transfer it to your workstation with wireshark. |
| 04:52.09 | ectospasm | ...don't do the transfer with wireshark, just transfer the pcap to the workstation that has wireshark. |
| 04:52.39 | Assid | you remember the command ? |
| 04:52.58 | Assid | tcpdump -vvvv > sample.pcap doesnt log the actual sip transaction |
| 04:54.31 | ectospasm | no |
| 04:55.10 | ectospasm | tcpdump -D # to tell you what interfaces are available. The first listed here will tell you which interface it will listen to if you don't explicitly tell it. |
| 04:55.35 | ectospasm | then, tcpdump -s0 -w tcpdump.pcap |
| 04:55.46 | ectospasm | that will capture everything from the first interface listed with -D |
| 04:56.02 | ectospasm | ...if you have to select an interface, use -i <iface> |
| 04:59.14 | *** join/#asterisk Wiretap (~wiretap@unaffiliated/wiretap) |
| 04:59.32 | Assid | yeah got that.. hold up |
| 04:59.47 | Assid | check pm |
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| 09:39.48 | weinerk | Hi. Please help. DeadAGI + pcntl_signal(SIGHUP,"handler") = doesn't work. |
| 09:39.48 | weinerk | The handler does get executed eventually, but not immediatelly after SIGHUP. I am on Asterisk 1.4.33 . |
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| 09:55.29 | usam | Hi, Im having problem with D-channel which is going down all the time (5-6 times pr day) .. Im using sangoma A102D against DMT-1 channelbank. I got a free E1 cross cable from manufactor, can it be that the cable is bad? |
| 09:58.53 | WIMPy | Are you getting other errors as well? |
| 10:01.51 | usam | WIMPy: I have had Sangoma techteam to remote config the e1 card , they said there is not problem with the card. Normally, when i get the 'sig_pri.c: Span 1: D-channel is down!' Im unable to place calls over the PRI |
| 10:02.17 | WIMPy | Sure |
| 10:02.33 | usam | btw, Im getting this error in the last couple of days 'chan_dahdi.c: PRI Span: 1 !! Don't know how to pre-handle message type Unknown Message Type (0x0)' |
| 10:02.43 | WIMPy | I'd suspect a timing issue. |
| 10:03.10 | WIMPy | Ok, so there are other errors. |
| 10:03.53 | usam | Asterisk is MASTER, DMT-1 is slave |
| 10:04.28 | WIMPy | Ok |
| 10:04.38 | WIMPy | Or you could have IRQ processing issues. |
| 10:05.53 | usam | im using swatch to monitor the log file if d-channel is down, restart asterisk, sometimes it works, sometimes it doesnt... |
| 10:06.14 | usam | the restart of asterisk workis, but the d-channel wont come up |
| 10:07.54 | usam | IRQ, could be ... I have 3 NIC on the system and i cant relocate IRQ on them ... its a dell server, asterisk running as domU |
| 10:27.05 | usam | WIMPy: A102D is the only one that is using IRQ 16 |
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| 11:32.23 | Ice_Strike | Hi |
| 11:33.07 | WIMPy | lo |
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| 12:57.28 | needascreenname_ | is there any way of figuring out what the number i have to dial is to get an outside line? |
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| 14:50.21 | cVsup | can i disable incoming calls to specified dahdi channel? |
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| 16:58.29 | picard276 | hey guys i had a question about hangup() |
| 16:58.37 | picard276 | whenever i try to do hangup() with a mobile phone.. it does not work.. |
| 16:58.57 | picard276 | it just says "the number you have dialed is not in service." (on my mobile phone calling into my asterisk system)... it does not actually hangup the call |
| 16:59.00 | picard276 | is there anyway to fix this? |
| 17:08.07 | [TK]D-Fender | Answer then hangup |
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| 17:22.38 | picard276 | TK D-Fender i do not want to answer |
| 17:29.49 | [TK]D-Fender | picTelco sees * not refuse the call and not accept the call so it considers it an "out of order" |
| 17:35.56 | *** join/#asterisk DarKnesS_WolF (~sherif@unaffiliated/sherif) |
| 17:36.54 | DarKnesS_WolF | so no there is no more asterisk-addon ? |
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| 17:42.26 | Ice_Strike | Does anyone know good VOIP provider that allow you to change any callerid without validation |
| 17:42.53 | Shazzamy | Quick question here, is there a bot in this channel that supports the !seen command? |
| 17:43.04 | Shazzamy | !seen Alfarin |
| 17:43.27 | Shazzamy | I guess not. Just looking for the person, adios! |
| 17:43.31 | *** part/#asterisk Shazzamy (~Shazzamy@unaffiliated/shazzamy) |
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| 17:55.43 | ChannelZ | stalker |
| 17:56.56 | autofsckk | somebody tried asterisk in pfsense? experiences? |
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| 18:23.41 | *** mode/#asterisk [+o mjordan] by ChanServ |
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| 18:45.01 | edge | If i set the timeout option on a queue and that value in seconds is reached. Does the person then leave the queue and continue to the next priority? |
| 18:50.59 | ectospasm | edge: yes, that's how it's supposed to work... |
| 18:51.30 | edge | ectospasm: is "supposed to" meaning that it sometimes doesn't? |
| 18:51.51 | ectospasm | no, just that I haven't tested it in a while, so can't be 100% certain. |
| 18:52.07 | ectospasm | edge: see "core show application Queue" |
| 18:53.08 | edge | ectospasm: i'm thinking about making 2 call queues and only having 1/4 of my users in the first queue. The main job of those 1/4 is answering the phones. But should they not answer it fast enough the second queue contains 100% of my users |
| 18:53.59 | ectospasm | so use penalties |
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| 18:54.30 | ectospasm | see annotated queues.conf for details. |
| 18:54.47 | edge | thanks ectospasm |
| 18:55.51 | ectospasm | you can use relatively complex logic on whether to send a call to the deeper penalty group |
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| 19:05.49 | edge | ectospasm: so assigning penalties to members will allow me to set who i want to be tired first, but how long will it try my penalty 1 users for before switching to 1 and 2? |
| 19:06.50 | ectospasm | I think you can look in queueupenalties.conf, but I may be wrong, lemme look it up. |
| 19:08.36 | ectospasm | edge: queuerules.conf |
| 19:10.49 | edge | ectospasm: i assume that , that is new in 1.8 as most sites i was using don't seem to have documentation on it. which means its time for me to actually download the sample configs and read them. |
| 19:11.41 | p3nguin | If the members with lower penalties are available, members with higher penalties may never be tried. |
| 19:12.00 | weinerk | Hi. Please help. DeadAGI + php pcntl_signal(SIGHUP,"handler") = doesn't work. |
| 19:12.00 | weinerk | The handler does get executed in the end, but not immediatelly after SIGHUP. Asterisk 1.4.33 . |
| 19:13.07 | edge | p3nguin: thats fine, as they are busy doing other work, and should only be asked (rung) to tak a call if my other employees don't take the call within a special amount of time |
| 19:13.09 | p3nguin | There's a small section on how penalties don't do what you think they will do here: http://www.voip-info.org/wiki/view/Asterisk+call+queues |
| 19:13.10 | ectospasm | weinerk: Asterisk 1.4 is end of life (EOL) |
| 19:13.46 | p3nguin | Using penalties like that will not work like you think they will work. |
| 19:13.53 | p3nguin | See the section on that wiki. |
| 19:14.43 | edge | p3nguin: so i really want to be using call cascading? |
| 19:14.46 | *** join/#asterisk Tagor (Tagor@s55978a13.adsl.wanadoo.nl) |
| 19:14.56 | Tagor | Does ReceiveFAX only work with T38? |
| 19:15.04 | p3nguin | probably |
| 19:15.18 | p3nguin | err... |
| 19:15.24 | p3nguin | tagor: no, of course not |
| 19:15.29 | p3nguin | edge: probably |
| 19:15.56 | Tagor | I used rxfax before, which worked fine. My sip provider doesn't support T38 |
| 19:17.13 | Tagor | Am I right rxfax doesn't work with 1.6 anymore? |
| 19:17.14 | ectospasm | At least in FAX/FFA, non-T.38 faxing is unreliable over SIP. |
| 19:17.24 | Tagor | I know, but it worked fine |
| 19:17.44 | ectospasm | YMMV, but it's more likely to fail sooner than later. |
| 19:17.52 | Tagor | So I was wondering if ReceiveFAX works without T38? |
| 19:18.05 | ectospasm | Tagor: yes, but it's unreliable. |
| 19:18.24 | Tagor | Thanks, as unreliable as when I was using rxfax? |
| 19:19.09 | ectospasm | possibly, I don't know much about rxfax... |
| 19:23.13 | edge | p3nguin: the queuerules.conf.sample talks about how after 30 seconds the QUEUE_MAX_PENALTY can be increased. So if i had users in penalty 1, 2, and 3. After 30 seconds i could make the QUEUE_MAX_PENALTY 2 (while leaving min to 1) and then both people in 1 and 2 would be ringing. Assumeing my ring stragety is ringall (and it is) |
| 19:26.06 | ectospasm | Tagor: T.38 is designed to work around network issues (such as latency, dropped packets, etc.), whereas straight G.711 faxing over SIP does not have the same provisioning. Thus, G.711 faxing over SIP is unreliable. |
| 19:27.19 | Tagor | ectospasm: I know that, but when my sip provider doesn't use T38, I can't use it ;) I was just wondering whether ReceiveFAX will work without T38 |
| 19:28.16 | ectospasm | ...as I said, it will be unreliable. Again, YMMV. |
| 19:28.39 | Tagor | ectospasm: I know, thanks :) |
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| 19:42.36 | zamba | after reboot of my server, i'm now longer able to get dahdi working |
| 19:42.43 | zamba | Unable to open DAHDI pseudo device |
| 19:42.45 | zamba | that's what it says |
| 19:43.21 | WIMPy | Looks like you didn't load it then. |
| 19:43.29 | zamba | how do i do that? :p |
| 19:43.38 | zamba | i believe this has to be some kernel issue |
| 19:43.51 | WIMPy | modprobe dahdi |
| 19:44.03 | zamba | FATAL: Module dahdi not found. |
| 19:44.10 | WIMPy | and the module(s) for whatever hardware you've got. |
| 19:44.17 | zamba | got no hardware |
| 19:44.19 | zamba | this is dummy |
| 19:44.48 | WIMPy | So you upgraded your kernel? You have to remake dahdi then. |
| 19:45.11 | zamba | ah, got it |
| 19:45.23 | zamba | this is a gotcha everytime when you're running ubuntu :p |
| 19:54.40 | zamba | how do i load the dahdi module in asterisk? |
| 19:54.55 | *** part/#asterisk edge (~edge@97-64-216-2.client.mchsi.com) |
| 19:55.38 | WIMPy | module load res_timing_dahdi |
| 20:04.57 | chrisq | how do i escape a "." in extension matching? |
| 20:05.51 | chrisq | like: exten => _domain.com_username |
| 20:06.23 | WIMPy | [.] |
| 20:06.47 | WIMPy | But I don't see a pattern in yur example. |
| 20:09.23 | *** join/#asterisk screenn (~screenn@37.46.237.217) |
| 20:22.09 | chrisq | WIMPy: no thats more like what i want to match |
| 20:23.26 | chrisq | domain name followed by an underscore and the username |
| 20:23.48 | chrisq | as asterisk doesnt seem to be able to change to "To" sip header |
| 20:23.58 | chrisq | i have to do dialplan tricks |
| 20:28.00 | jaytee | if an Asterisk 1.8.11.1 server and all phones (Polycom) are behind a NAT firewall and I use Flowroute as the ITSP, what should I have for nat in my general section of sip.conf and what should I have for nat= in my settings for the flowroute account? Also, should I use the externaddr='myoutsideip" in the general section? With Asterisk 1.6 I used externip= and canreinvite=no. I'm thinking for |
| 20:28.00 | jaytee | my internal devices I need directmedia=nonat but I'm not sure what to set for the flowroute account for directmedia. |
| 20:36.30 | ectospasm | zamba: if you want DAHDI timing, follow WIMPy's suggestion. If you want to use a DAHDI telephony device, load chan_dahdi.so |
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| 21:50.13 | Ice_Strike | hi |
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| 23:26.05 | nfsnobody | hey guys. Can anyone point me in the direction of a guide that helps convert a dialplan to LUA? |
| 23:36.40 | [TK]D-Fender | PBX_LUA does what it does ... there is no "convert" |
| 23:45.35 | nfsnobody | yes I understand that [TK]D-Fender - I want to change how my dialplan is written through from the standard dialplan context to LUA though, and I'm looking for some good information about converting it |
| 23:47.49 | [TK]D-Fender | <PROTECTED> |
| 23:53.09 | nfsnobody | .... So I have a dialplan written in the standard context and I want to rewrite it in LUA. Can anyone point me in the direction of a guide? |
| 23:53.57 | [TK]D-Fender | extensions.lua.sample |
| 23:54.00 | [TK]D-Fender | ~book |
| 23:54.00 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 23:54.13 | nfsnobody | thanks :) |
| 23:54.18 | nfsnobody | ~buybook |
| 23:54.18 | infobot | You can buy "Asterisk: The Definitive Guide" at http://oreilly.com/catalog/9780596517342 so go buy it SERIOUSLY |
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| 23:58.50 | nfsnobody | so aside from extensions.lua.sample (which is handy, but not complete) and the book (which simply mentions that LUA is an option and doesn't go into it) can anyone point me to a good asterisk LUA related resource? :) |
| 23:59.52 | [TK]D-Fender | www.google.com . |