| 00:02.51 | *** join/#asterisk philippel_mac (~p_lindhei@50-46-123-25.evrt.wa.frontiernet.net) |
| 00:15.25 | dlublink | I think I have something |
| 00:19.53 | dlublink | it says "Echo Canceller(s): MG2" when initiallizing the PRI. Can I disable this mg2 echo cancelling ? |
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| 02:58.36 | egeste | I'm having trouble with asterisk and flite on my centos 5.5 box, running asterisj 1.4.4 - the Flite application does not show when I "show application [tab]" |
| 02:58.41 | egeste | asterisk* |
| 02:59.57 | egeste | I have installed flite, flite-devel, app_flite (rpm), asterisk-flite (rpm) and recompiled flite from source |
| 03:00.00 | egeste | still no go |
| 03:00.09 | egeste | er, recompiled asterisk |
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| 03:58.57 | titter | Can someone help me figure out why the music class isn't setting correctly? Is it because I am sending it to a local chan first for JB purposes? http://pastebin.com/HvAivZM4 - 1.8.3.3. |
| 04:01.22 | *** join/#asterisk Defraz (~Defraz@184-155-137-27.cpe.cableone.net) |
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| 04:04.58 | titter | I also tried Dial(Local/conf@internal/nj,,m(none)) as a fyi |
| 04:06.17 | jql | umm |
| 04:07.02 | jql | I'm not quite sure what you expect out of that |
| 04:07.18 | jql | you're trying to disable hold music upon joining a conference room? |
| 04:08.27 | titter | Yep |
| 04:09.12 | titter | It works if I just do exten => 1000,1,Set(CHANNEL(musicclass)=none) exten => 1000,n,MeetMe(${EXTEN},dicq) |
| 04:09.29 | titter | So I am assuming it has something to with dialing to the local channel first |
| 04:10.07 | jql | yeah, you're changing the direction in which it takes effect. I think |
| 04:10.11 | jql | confusing a bit, yeah |
| 04:13.17 | jql | let me think this out |
| 04:14.36 | jql | exten 1000's CHANNEL(musicclass) specifies what the caller would here. exten 1000 Dial(x,,M(setholdmusic)) would set what the conf context would here |
| 04:14.44 | jql | I think I have that right |
| 04:16.06 | titter | Actually you know what, I think it may be working ... I just remembered I am home, and not at my office and I have my Polycom configured differently. It is going iax to my main pbx which has the conf ... need to add the class before it dials the iax I bet. |
| 04:16.51 | jql | so, theoretically, you want exten => 1000,1,Dial(Local/conf@internal/nj,,M(cancelmusic)) [macro-cancelmusic] exten => s,1,Set(CHANNEL(musicclass)=none) |
| 04:16.52 | *** part/#asterisk benngard (~mabe@h30n2-g-ml-a11.ias.bredband.telia.com) |
| 04:17.06 | jql | well, good luck. :) |
| 04:17.25 | titter | That was it |
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| 06:18.04 | BelTechs | hello Im trying to debug my trunk incoming calls Im using asterisk 1.8 and varphonex trunk. Any is appreciated. http://pastebin.com/mdLLxZbY |
| 06:19.33 | BelTechs | Outgoing works fine. Port are forwarded |
| 06:22.12 | kaldemar | BelTechs: your asterisk is not set up to function behind a NAT. forwarding ports is not enough. ask in #freepbx how to do it with freepbx. |
| 06:24.58 | *** join/#asterisk puzzled (~patrick@5ED2AE82.cm-7-3c.dynamic.ziggo.nl) |
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| 06:25.36 | BelTechs | ok i see where in fpbx and I now set nat to yes in sip settings. Same problem |
| 06:25.40 | BelTechs | http://pastebin.com/dKgufpbE |
| 06:26.02 | kaldemar | asterisk needs to no an external ip to use in the messages. |
| 06:26.32 | BelTechs | I have my external ip setup |
| 06:27.47 | *** join/#asterisk jizzzum6 (~jizzzum6@2001:1938:232:0:99e6:e825:8b92:3cbf) |
| 06:28.29 | kaldemar | the problem is not the same in your second paste. your asterisk seems to require authentication for the incoming call. the registration still fails though. |
| 06:28.57 | BelTechs | In the 2nd post I set nat to yes. |
| 06:31.36 | kaldemar | setting nat to yes is not enough. |
| 06:31.53 | BelTechs | hmm |
| 06:32.56 | BelTechs | Kaldemar: I have set up this exact same pbx using *1.6 same settings all worked. |
| 06:33.05 | kaldemar | like i said, asterisk needs to know the external ip address that is used. it is configured with externalip or externaladdr in asterisk. no idea on how to do it in freepbx. |
| 06:34.02 | p3nguin | externip/externaddr or extenhost |
| 06:34.13 | p3nguin | s/exten/extern |
| 06:34.18 | p3nguin | dammit |
| 06:34.24 | p3nguin | externip/externaddr or externhost |
| 06:34.36 | *** join/#asterisk Exten (~Exten@mail.gdc.co.il) |
| 06:35.28 | Exten | Hello, can anybody help me with a new asterisk installation ? asterisk is behind an ADSL router, when i call an extension it does the playback - but i hear nothing ... |
| 06:35.53 | Exten | Retransmitting #6 (NAT) to 80.250.146.53:33840: |
| 06:35.56 | Exten | SIP/2.0 200 OK |
| 06:36.03 | Exten | etc... |
| 06:37.26 | kaldemar | Exten: describe your setup networkwise. are there NAT's involved? is asterisk behind a NAT? what do you use for calling? what does the extension do? |
| 06:37.52 | Exten | yes |
| 06:38.06 | sxpert | (NAT shows, once again, its evil head) |
| 06:38.39 | Exten | asterisk is on an adsl router, it has an 10.0 ip so its nat - i use X-Lite the extension does playback(demo-congrats) |
| 06:39.06 | p3nguin | Configure Asterisk correctly to work behind NAT. |
| 06:39.14 | sxpert | and you're attempting from outside ? |
| 06:39.21 | Exten | yes |
| 06:39.25 | p3nguin | If necessary, configure the peer (phone) for NAT as well. |
| 06:39.36 | p3nguin | ~sipnat |
| 06:39.36 | infobot | [~sipnat] Quick guide on configuring Asterisk + SIP behind NAT: http://www.aocomputing.net/?p=3 otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions . Typically, you MUST configure these settings with appropriate values in the [general] section of sip.conf: nat, canreinvite, externhost or externip, and localnet. |
| 06:39.37 | Exten | nat=yes |
| 06:39.41 | sxpert | or switch to ipv6 ;) |
| 06:40.15 | p3nguin | If you are using 1.8, canreinvite is now directmedia, and externip is now externaddr. |
| 06:40.23 | BelTechs | ah... |
| 06:40.59 | p3nguin | The deprecated terms should still work, but you should expect to see a complaint in the CLI about using the old terms. |
| 06:41.32 | BelTechs | ok adding externaddr killed trunk registration |
| 06:41.33 | Exten | Ok lets try the iptables thingie .. |
| 06:41.39 | *** join/#asterisk jg1234 (~jan@dslc-082-082-037-188.pools.arcor-ip.net) |
| 06:41.43 | p3nguin | exten: Did you forward the ports? |
| 06:41.56 | Exten | on the adsl ? yes |
| 06:42.13 | p3nguin | exten: UDP 5060 and the UDP range that is found in rtp.conf need to be forwarded to Asterisk's IP address. |
| 06:42.22 | p3nguin | The range is usually 10000-20000. |
| 06:42.48 | Exten | yes also did the rtp |
| 06:44.13 | p3nguin | Having no audio is most commonly a result of misconfiguration or complete lack of NAT settings. |
| 06:44.32 | *** join/#asterisk eject_ck (~eject_ck@62.205.134.210) |
| 06:44.48 | sxpert | p3nguin: you don't get the messages when running asterisk as a daemon such as the system install in debian |
| 06:45.03 | *** join/#asterisk kwk (~kleine@carbon.gonicus.de) |
| 06:45.16 | p3nguin | *shrug* I don't see why not. |
| 06:47.09 | Exten | ok, did all the iptables and general sip.conf . damnit. |
| 06:50.12 | Exten | Retransmitting #6 (NAT) to 62.90.210.232:3204: |
| 06:50.12 | Exten | SIP/2.0 200 OK |
| 06:50.26 | kaldemar | ~pb |
| 06:50.26 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com , http://pastebin.us , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ ; or install pastebinit with yum or aptitude. |
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| 06:59.11 | Exten | http://pastebin.com/6MT0rK7U :) |
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| 07:02.35 | kaldemar | Exten: what does your sip.conf look like? |
| 07:03.20 | *** join/#asterisk zkn (~zkn@195.222.14.202) |
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| 07:08.14 | Exten | http://pastebin.com/FHa56kat |
| 07:13.16 | kaldemar | Exten: externip=10.0.0.1 <--- 10.0.0.1 is the internal address of your asterisk box, not the external one. |
| 07:13.33 | kaldemar | external ip is the public ip of your router. |
| 07:13.49 | Exten | gotcha |
| 07:13.51 | Exten | sec.. |
| 07:13.58 | kaldemar | localnet is also defined to be 192.168.0.0/255.255.0.0 which is clearly not your network. |
| 07:15.25 | *** join/#asterisk eject_ck (~eject_ck@62.205.134.210) |
| 07:15.26 | Exten | it works |
| 07:16.05 | Exten | by the way, it works what the current localnet 192.168.0.0/255.255.0.0, and when i disable it, it dosent |
| 07:16.11 | Exten | what=with |
| 07:18.04 | Exten | Thanks !@ :) |
| 07:18.41 | kaldemar | Exten: don't disable it, configure it to be what your network is. if your network is 10.0.0.0/24, set it to 10.0.0.0/255.255.255.0. |
| 07:19.04 | *** join/#asterisk eject_ck (~eject_ck@62.205.134.210) |
| 07:20.50 | kwk | Hello |
| 07:21.31 | *** part/#asterisk eject_ck (~eject_ck@62.205.134.210) |
| 07:21.44 | kwk | has anybody ever setup a speech quality monitoring solution (e.g. based on PESQ) with asterisk? |
| 07:23.22 | jg1234 | i am still having problems with asterisk not hanging up isdn call correctly |
| 07:23.23 | Exten | my understading of network is a bit lacking ... :) when you say 10.0.0.0/24 - what do you mean by that range ? |
| 07:23.24 | jg1234 | http://pastebin.com/m5AgcLnG |
| 07:23.37 | p3nguin | 10.0.0.0/255.255.255.0, like he said. |
| 07:24.15 | p3nguin | A subnet mask of 255.255.255.0 is a 24-bit mask, a.k.a. /24. |
| 07:24.57 | jg1234 | with this patch its hanging up things correctly, if the call was answered or if the isdn channel was called |
| 07:26.45 | jql | jg1234: interesting patch. does it cause any problems for you? |
| 07:27.23 | jg1234 | s/was called/is calling/ |
| 07:27.28 | Exten | Ok , did it - you guys are very helpful - thank you |
| 07:27.50 | jg1234 | jql: i have more problems without it |
| 07:28.39 | *** join/#asterisk \DSAFEW\ (~DSAFEW_@ip72-208-176-219.ph.ph.cox.net) |
| 07:29.08 | zkn | <PROTECTED> |
| 07:29.10 | jg1234 | without it asterisk is messing up the whole s0-line because its constantly sending 1010... |
| 07:31.17 | kaldemar | zkn: then asterisk is more likely to handle requests wrong. but this was only about the localnet setting. |
| 07:31.30 | p3nguin | I would guess that an IP address without a mask would assume a 32-bit mask. |
| 07:31.48 | kaldemar | zkn: you don't need to define a mask for every address in the configs. |
| 07:31.55 | jql | I'd guess that as well |
| 07:32.32 | p3nguin | s/an/a 32-bit/ |
| 07:32.59 | p3nguin | But unless you try it, the world may never know. |
| 07:33.34 | zkn | so subnet mask has to defined for the localnet IP |
| 07:33.45 | *** join/#asterisk tamiel (~tamiel@213.30.183.226) |
| 07:34.06 | p3nguin | A network requires a subnet mask. |
| 07:34.31 | p3nguin | Without a subnet mask or with a 32-bit subnet mask, it's just an IP address and not a network. |
| 07:35.09 | p3nguin | And since it's not called localip, let's go ahead and define the subnet mask. |
| 07:40.09 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 07:41.45 | *** join/#asterisk shark (d5cf2336@gateway/web/freenode/ip.213.207.35.54) |
| 07:42.14 | Guest33474 | morning or evening to all |
| 07:43.03 | Guest33474 | i want to forward unknown nr to another nr from asterisk! Does some one nkow's how to do that ??? |
| 07:43.45 | *** join/#asterisk Defraz (~Defraz@184-155-137-27.cpe.cableone.net) |
| 07:44.42 | pethkaqeni | plz |
| 07:44.44 | jql | pethkaqeni: what did you try? |
| 07:45.25 | kaldemar | pethkaqeni: be more accurate. explain "forward unknown nr to another nr". |
| 07:45.36 | pethkaqeni | i have googled that |
| 07:45.39 | pethkaqeni | but nothing |
| 07:45.40 | pethkaqeni | ok |
| 07:45.51 | pethkaqeni | im trying to eplain |
| 07:45.52 | pethkaqeni | nothing since now |
| 07:46.58 | pethkaqeni | i want that when an unknown number call my numbers this call to be automaticly forwardet at another nr |
| 07:47.22 | pethkaqeni | sorry for my grammar |
| 07:47.56 | jql | either you want exten => i,1,Dial(SIP/another-nr) or exten => _X.,1,Dial(SIP/another-nr) ? |
| 07:48.22 | jql | might want to try those and let us know what you want instead. :) |
| 07:48.46 | pethkaqeni | thnx im trying right now and let u know ;) |
| 07:49.27 | ChannelZ | I think he probably means he wants to send calls with particular CallerID numbers to a specific extension? |
| 07:49.41 | ChannelZ | (or no CallerID) |
| 07:49.43 | jql | yet a third option |
| 07:49.49 | pethkaqeni | yes in fact |
| 07:50.14 | pethkaqeni | my prob is with no id numbers |
| 07:50.19 | kaldemar | pethkaqeni: you need to be precise so that people don't have to guess what you're trying to do. |
| 07:50.56 | jql | that's more of a GotoIf thing |
| 07:51.13 | pethkaqeni | ok from top |
| 07:51.18 | pethkaqeni | i have no ID callers that are borring me |
| 07:51.34 | pethkaqeni | i want to forward automaticly to the main office nr |
| 07:51.36 | pethkaqeni | so |
| 07:51.37 | pethkaqeni | !!! |
| 07:51.59 | jg1234 | maybe someone could point out to me the difference between hanging up a channel that is in AST_STATE_RING and one that is in AST_STATE_UP |
| 07:52.20 | kaldemar | pethkaqeni: GotoIf($["${CALLERID(num)}" = ""]?context,exten,priority) |
| 07:52.51 | *** join/#asterisk freckle (~jon@95.172.10.10) |
| 07:53.38 | jql | jg1234: what do you mean by difference? |
| 07:54.04 | jql | jg1234: they're different protocol states. different billing states, too, to your carrier |
| 07:54.43 | *** join/#asterisk WiretapMac (~wiretap@unaffiliated/wiretap) |
| 07:55.38 | jg1234 | i am probably to focused on that state because its the only difference i could find |
| 07:56.33 | jg1234 | AST_STATE_UP -> correct hangup (with my patch) AST_STATE_RING -> no hangup |
| 07:57.15 | jg1234 | its very likely that this is not the reason |
| 07:57.19 | jql | jg1234: in q.931, either one is going to cause a -> RELEASE <- DISCONNECT -> DISCONNECT ACK |
| 07:57.38 | jql | *shrug* |
| 07:58.33 | jql | or is it the other way around? probably doesn't matter |
| 07:58.51 | *** join/#asterisk sekil (~sekil@80.93.247.26) |
| 08:01.06 | kaldemar | jg1234: where is the call coming from? |
| 08:02.25 | jg1234 | jql well i think thats all done by libpri |
| 08:02.54 | jg1234 | kaldemar i am a little confused right now |
| 08:03.18 | jg1234 | i will check my setup again |
| 08:06.57 | jg1234 | oh and let my point out that in asterisk-1.6.2.17.2 everything it working just fine |
| 08:07.07 | *** join/#asterisk Tim_Toady (~moi@62.1.175.227) |
| 08:08.09 | jg1234 | kaldemar its an incomming call from another isdn "phone" |
| 08:09.36 | kaldemar | jg1234: what do you see in the protocol debug? |
| 08:16.25 | jg1234 | kaldemar http://pastebin.com/qixvm3VN |
| 08:20.26 | jg1234 | kaldemar or did you mean something else ? |
| 08:20.28 | *** join/#asterisk timahvo1 (~rogue@41.223.57.76) |
| 08:22.06 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
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| 08:27.13 | verywiseman | how can i know current call sessions , and how can i end any one of them? |
| 08:27.27 | jg1234 | kaldemar http://pastebin.com/cBWvMBCt |
| 08:29.09 | jacc0 | @verywiseman: core show channels |
| 08:29.39 | *** join/#asterisk puzzled (~patrick@puzzled.xs4all.nl) |
| 08:30.45 | BelTechs | Hi I am still attempting to debug a trunk. The trunk outbound is ok and the inbound is not working. please see http://pastebin.com/08Z3DCvH for latest sip debug |
| 08:30.53 | BelTechs | thanks in advance |
| 08:32.19 | k3asd` | hi |
| 08:36.44 | kwk | hallo |
| 08:36.47 | *** part/#asterisk kwk (~kleine@carbon.gonicus.de) |
| 08:37.38 | *** join/#asterisk timahvo1 (~rogue@41.223.57.75) |
| 08:37.40 | *** join/#asterisk kwk (~kleine@carbon.gonicus.de) |
| 08:38.50 | *** join/#asterisk ferdna (~yup@cpe-67-10-220-35.elp.res.rr.com) |
| 08:40.09 | kaldemar | jg1234: the other end should answer a RELEASE with a RELEASE COMPLETE. now it sends a RELEASE, maybe that confuses asterisk. |
| 08:40.26 | kaldemar | jg1234: does that happen regardless of where you make the call from? |
| 08:44.21 | Chainsaw | jacc0: The crasher is in 1.8.4-rc3 as well. I just tried that one in production (briefly obviously). |
| 08:45.23 | *** join/#asterisk timahvo1 (~rogue@41.223.57.76) |
| 08:46.44 | Chainsaw | jacc0: Should be reproducable by having a peer with transport=TCP,UDP and having a firewall block TCP access. It takes a few minutes to time out, and that timeout seems to be what brings it down with a null pointer dereference. |
| 08:46.57 | Chainsaw | jacc0: I really could do with a core dump to prove it that I could take to teknoviking though :/ |
| 08:52.38 | jacc0 | @Chainsaw:thnx |
| 08:58.23 | jacc0 | @BelTechs: there is this text in your log: 87.SIP/2.0 401 Unauthorized |
| 08:58.35 | jacc0 | @BelTech: check username and password |
| 09:00.03 | jacc0 | @Chainsaw: some of the crashes I'm was experiancing seem like they are fixed in 1.8.4-rc3 |
| 09:00.36 | Chainsaw | jacc0: Particularly transfer-related ones I take it? That "unable to break RTP bridge" message looks very familiar. |
| 09:01.23 | jacc0 | [Apr 12 19:06:48] WARNING[4007]: channel.c:6493 ast_do_masquerade: Channel type 'NULL' does not have a fixup routine (for Bridge/SIP/172.20.143.211-0000001a<ZOMBIE>)! Bad things may happen. |
| 09:01.27 | Chainsaw | jacc0: 1.8.4 should be good for me, upstream has also applied one of the 10 patches from the patchset. Down to 9 at long last. |
| 09:01.41 | Chainsaw | jacc0: Oh, ouch. That's worse than what I'm used to. |
| 09:03.25 | jacc0 | and I had someting strange last sunday/monday; asterisk wasn't able to start |
| 09:03.49 | jacc0 | I removed astdb and is started again |
| 09:04.02 | jacc0 | *it started |
| 09:05.40 | Chainsaw | jacc0: That would be a disaster for me. My astdb has a *lot* of essential data like country/area codes. |
| 09:06.03 | Chainsaw | jacc0: (You get location information for any inbound call, if the data isn't in the CRM) |
| 09:06.49 | *** join/#asterisk Denial (Denial@drgi.co.uk) |
| 09:07.15 | jacc0 | in astdb? |
| 09:07.32 | jacc0 | it is regenerated when you start asterisk |
| 09:08.32 | jg1234 | kaldemar sry that i have to ask this, but to be sure ">" is receiving and "<" is sending right ? |
| 09:08.42 | jacc0 | I've been putting astdb to the test :) |
| 09:08.54 | jacc0 | 26.000.000 entries without problems |
| 09:09.45 | jacc0 | only when you have a value of 4096 bytes long asterisk takes up 85% ram on a 4gb ram system |
| 09:09.57 | jacc0 | :S |
| 09:10.20 | Chainsaw | jacc0: I have... 1793 entries. |
| 09:10.27 | Chainsaw | jacc0: 26 million seems a lot :) |
| 09:10.41 | jacc0 | just make sure the values don't get to long |
| 09:10.43 | jacc0 | ;) |
| 09:11.28 | jacc0 | you better use the key name to store values; asterisk has no problem with keynames of 4096 bytes |
| 09:11.29 | jacc0 | :P |
| 09:12.30 | jacc0 | value length is limited to 4096; I guess it is because a dialplan row can be max 4096 bytes |
| 09:13.06 | jacc0 | maybe you could make it longer bij using push en shift |
| 09:13.06 | Chainsaw | jacc0: Your middle name appears to be "torture test". I like that. |
| 09:13.27 | jacc0 | i've also tested the total ascii table : noprblems there |
| 09:14.00 | kaldemar | jg1234: the opposite |
| 09:14.08 | jacc0 | I dont seem to be able to reproduce what caused asterisk not to start |
| 09:14.36 | Chainsaw | [2011-04-27 10:12:27] NOTICE[21840]: res_musiconhold.c:661 monmp3thread: Request to schedule in the past?!?! <- This is also a firm favourite. I don't even use MP3 for MOH. |
| 09:17.31 | jg1234 | kaldemar ok , http://pastebin.com/QqB7Pdqy this is with asterisk 1.6.. and it hanging up correctly, or at least its not messing up my s0-bus after that |
| 09:19.30 | *** join/#asterisk carloimperia (~carloimpe@93-62-218-170.ip24.fastwebnet.it) |
| 09:21.37 | *** join/#asterisk timahvo1 (~rogue@41.223.57.77) |
| 09:29.09 | *** join/#asterisk devil_evoxxx (~d3v1l@157.27.183.122) |
| 09:29.13 | devil_evoxxx | hi all guys! :) |
| 09:30.16 | jacc0 | hi devil |
| 09:38.22 | *** join/#asterisk SiNGLer (~singler@81-7-123-162.static.zebra.lt) |
| 09:52.30 | devil_evoxxx | jacc0, have you got any idea about old isdn pbx that for dialing out firs keep the line and next compose the number |
| 09:53.15 | devil_evoxxx | because, i need to transform an old isnd pbx in voip tecnology.. |
| 09:55.00 | devil_evoxxx | and when the old pbx dial out, an incoming call come to asterisk but not with the corrext exten |
| 09:55.06 | devil_evoxxx | it comes in s extesnsion :( |
| 10:05.22 | zkn | what is that : chan_iax2.c:5117 iax2_read: I should never be called! |
| 10:08.49 | jg1234 | kaldemar without my patch asterisk never hangs up |
| 10:09.22 | jg1234 | and in the case of "release collision" it should be able to hang up |
| 10:09.31 | jg1234 | like the 1.6.. version does |
| 10:17.32 | jacc0 | @devil: nope |
| 10:23.09 | *** join/#asterisk nibel (~nibel@i59F44705.versanet.de) |
| 10:23.13 | nibel | hi |
| 10:23.46 | nibel | i have a urgent problem and I hope someone could help me with that |
| 10:24.23 | nibel | can you tell me something about this error message? app_dial.c:1076 dial_exec_full: Unable to create channel of type 'Zap' (cause 34 - Circuit/channel congestion) |
| 10:24.26 | nibel | <PROTECTED> |
| 10:24.48 | nibel | i get it everytime I try to dial |
| 10:25.08 | nibel | but the setup was working some time ago... |
| 10:26.01 | *** join/#asterisk cyborg-one (1000@188-115-188-178.broadband.tenet.odessa.ua) |
| 10:26.14 | *** join/#asterisk WiretapMac (~wiretap@unaffiliated/wiretap) |
| 10:29.43 | nibel | ok found out that my spans are down -.- |
| 10:29.59 | nibel | dont know how to bring them up |
| 10:34.32 | *** join/#asterisk ruyo (~psantos@83.132.152.91) |
| 10:40.24 | *** join/#asterisk k3asd` (~k3asd`@static-217-133-87-244.clienti.tiscali.it) |
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| 11:33.56 | z4nD4R | hi all :) |
| 11:34.17 | z4nD4R | somebody to help with call redirection and call ide forwarding? |
| 11:35.48 | jacc0 | @z4nD4R>: don't ask if anyone can help, just ask your question |
| 11:35.59 | z4nD4R | ok :) |
| 11:36.35 | z4nD4R | i have this in my dialplan |
| 11:37.07 | z4nD4R | exten => 123,n,Dial(SIP/Office,10) |
| 11:37.07 | z4nD4R | exten => 123,n,Dial(SIP/trunk-03/some_number) |
| 11:37.54 | z4nD4R | redirection works verry well, but .. on phone i see number: "some_number" and not.. the real caller number |
| 11:40.59 | z4nD4R | Question is.. if i can direct this caller id to trunk. i am not sure, if this is possible... |
| 11:43.32 | jg1234 | kaldemar are you still there ? |
| 11:46.04 | *** join/#asterisk psilvao (~psilvao@190.20.25.210) |
| 11:46.19 | psilvao | hi |
| 11:46.19 | *** join/#asterisk carloimperia (~carloimpe@93-62-218-170.ip24.fastwebnet.it) |
| 11:46.32 | psilvao | somebody use tdm400P + freetdm? |
| 11:49.36 | \DSAFEW\ | nibel, what version asterisk are you using? Zap is depreciated |
| 12:06.27 | leifmadsen | freetdm is some alternative to zaptel or dahdi |
| 12:06.34 | leifmadsen | I've never heard of anyone using it |
| 12:06.44 | leifmadsen | ask in #freeswitch |
| 12:09.15 | jg1234 | http://pastebin.com/HApPsiQq |
| 12:11.01 | jg1234 | that patch is working best for me |
| 12:12.22 | jg1234 | i think someone should really check dahdi_hangup(.. because the "goto hangup_out" really skips a lot of stuff that was done in older versions |
| 12:12.51 | leifmadsen | jg1234: have you provided your patch to the issue tracker and described the issue? |
| 12:13.07 | jg1234 | ;) no |
| 12:13.20 | leifmadsen | then no one who can commit the issue can look at your patch :) |
| 12:13.49 | jg1234 | really why not |
| 12:15.08 | *** join/#asterisk slidesinger (~slidesing@c-68-44-99-163.hsd1.nj.comcast.net) |
| 12:17.02 | leifmadsen | jg1234: because of the license policy that requires patches be put through the systems we have in place |
| 12:17.17 | jg1234 | is my email address going to be posted on any mailing lists or something like that |
| 12:17.20 | leifmadsen | all patches must be submitted by someone who has a license on file |
| 12:17.26 | leifmadsen | jg1234: no |
| 12:17.40 | jg1234 | ok |
| 12:17.43 | leifmadsen | jg1234: you just fill out the online license agreement which simply goes to the Digium legal department |
| 12:17.49 | leifmadsen | it's a link at the top of https://issues.asterisk.org |
| 12:18.09 | jg1234 | yeah i am currently doing that |
| 12:18.23 | *** join/#asterisk ariel_ (~chatzilla@pdpc/supporter/active/abatista) |
| 12:19.26 | jg1234 | i was just wondering because of the kernel-mailing list i have lost some email addresses due to spam (i know that you are not responsable for the kernel mailing list) |
| 12:19.36 | leifmadsen | certainly not :) |
| 12:19.43 | leifmadsen | you're only subscribed to mailing lists you sign up for |
| 12:19.55 | leifmadsen | at that point it is your problem to protect your identity |
| 12:25.52 | *** join/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
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| 12:31.49 | jg1234 | well i guess the report is enough |
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| 13:11.00 | k3asd` | hi |
| 13:12.37 | jacc0 | hi |
| 13:12.43 | Chainsaw | Yes, hello. |
| 13:12.44 | k3asd` | hi jacc0 |
| 13:12.57 | leifmadsen | howdy |
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| 13:22.30 | Faustov | will _X. catch 1-digit extensions? |
| 13:23.52 | kaldemar | Faustov: no. "." is one or more. |
| 13:24.01 | Faustov | thx |
| 13:26.46 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 13:27.27 | nibel | hi |
| 13:27.33 | nibel | i have a urgent problem and I hope someone could help me with that |
| 13:27.58 | nibel | can you tell me something about this error message? app_dial.c:1076 dial_exec_full: Unable to create channel of type 'Zap' (cause 34 - Circuit/channel congestion) |
| 13:28.03 | nibel | <PROTECTED> |
| 13:28.11 | nibel | i get it everytime I try to dial |
| 13:28.51 | kaldemar | nibel: what brought you to the conclusion that the span is down? |
| 13:29.25 | nibel | pri show span 1 says : Status: Provisioned, Down, Active |
| 13:29.45 | nibel | kaldemar: i dont know how to bring them up... |
| 13:29.46 | kaldemar | nibel: and you haven't changed any settings since it worked? |
| 13:30.09 | nibel | kaldemar: y but i also phoned my carrier who says everything is alright... |
| 13:30.38 | nibel | kaldemar: i also have backup of my config and use it right now... |
| 13:30.49 | nibel | a* |
| 13:31.46 | *** join/#asterisk golam (4531720a@gateway/web/freenode/ip.69.49.114.10) |
| 13:33.26 | nibel | kaldemar: its really urgent. I know how to configure a dialplan or sip but have no idea about wanpipe oder zap |
| 13:35.34 | *** join/#asterisk ruben23 (~RLACUMBA@121.97.111.142) |
| 13:37.15 | \DSAFEW\ | nibel, do you have any modules which need loading? |
| 13:37.54 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 13:43.09 | nibel | \DSAFEW\: what do you exactly mean? |
| 13:43.36 | \DSAFEW\ | nibel, has the machine rebooted since it last worked? perhaps modules need to be loaded |
| 13:43.58 | nibel | \DSAFEW\: rebooted several times :/ |
| 13:44.01 | \DSAFEW\ | nibel, like, zaptel module and hardware |
| 13:44.21 | nibel | \DSAFEW\: zaptel is loaded |
| 13:44.28 | \DSAFEW\ | nibel, look over the modules.conf and see if you can modprobe them manually |
| 13:44.55 | *** join/#asterisk Juggie (~Juggie@unaffiliated/juggie) |
| 13:44.55 | \DSAFEW\ | nibel, you said something about a missing device? |
| 13:46.04 | nibel | \DSAFEW\: no but pri show span says status: down |
| 13:47.16 | *** join/#asterisk golam (4531720a@gateway/web/freenode/ip.69.49.114.10) |
| 13:47.30 | \DSAFEW\ | what does zttool say? |
| 13:48.20 | \DSAFEW\ | same thing I gather from the docs |
| 13:48.39 | nibel | \DSAFEW\: Ok wanpipe1 card0 |
| 13:48.49 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 13:48.58 | nibel | \DSAFEW\: Red wanpipe2 card1 |
| 13:49.23 | nibel | \DSAFEW\: Ok wanpipe3 card2 |
| 13:49.39 | nibel | \DSAFEW\: Red wanpipe4 card3 |
| 13:49.48 | \DSAFEW\ | perhaps there's a connection problem, is card1/4 plugged in? |
| 13:49.50 | nibel | \DSAFEW\: that's what zttol says |
| 13:50.22 | *** join/#asterisk DaveCanoe (~Dave@strike.eicat.ca) |
| 13:50.33 | nibel | \DSAFEW\: yes it is, but when i remember right all wanpipe go to status red when i plug it out |
| 13:50.51 | *** join/#asterisk Arsenick (~y@fedora/Arsenick) |
| 13:50.57 | \DSAFEW\ | plug it out? |
| 13:51.00 | \DSAFEW\ | is out? |
| 13:51.22 | nibel | \DSAFEW\: no it's plugged in at the moment |
| 13:51.46 | \DSAFEW\ | nibel, what did you think the problem is again? |
| 13:52.45 | nibel | \DSAFEW\: probably the zaptel.conf is wrong |
| 13:53.09 | \DSAFEW\ | do you have wgetpaste? |
| 13:53.19 | \DSAFEW\ | nibel, that'd be very handy right now |
| 13:54.12 | nibel | \DSAFEW\: no sorry but i could load the conf up to pastebin, btw im very very glad u are helping me |
| 13:54.35 | nibel | \DSAFEW\: im a little bit despaired at the moment |
| 13:55.07 | *** part/#asterisk benngard (~mabe@213.88.138.230) |
| 13:55.31 | \DSAFEW\ | so how did it stop working? one day randomly it lost all connectivity to the trunk? |
| 13:56.04 | \DSAFEW\ | if you're using the backups, how come you think it's a zaptel.conf problem? |
| 13:58.12 | nibel | \DSAFEW\: actually i have no idea. We have a pmx here on the one slot a hipath system is connected on the other one this asterisk server which is only used occassinally but now we need it |
| 13:59.16 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 13:59.29 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 13:59.59 | nibel | \DSAFEW\: but sometime ago we had a network failure, the system got restarted but with the wrong default kernel, i assumed there was a problem with zaptel.conf and edited it but then discovered the wrong kernel was loaed so i started the right one and used the config backup but i have still the same problem |
| 14:02.27 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 14:05.06 | \DSAFEW\ | nibel, were you still pastebinning? |
| 14:05.53 | nibel | gathering the infos giht now mom |
| 14:06.01 | *** join/#asterisk JonRob (~jon@89.167.143.232) |
| 14:06.02 | nibel | moment* |
| 14:06.55 | nibel | \DSAFEW\: this is zapata.conf http://pastebin.com/s1QyGgND |
| 14:07.55 | nibel | \DSAFEW\: this is zaptel.conf http://pastebin.com/gU7gcZwF |
| 14:08.13 | *** join/#asterisk cyford (~cyford@96-25-169-243.gar.clearwire-wmx.net) |
| 14:09.36 | kaldemar | nibel: is the signalling pri_net on purpose on spans 1 and 2? |
| 14:09.50 | \DSAFEW\ | oh one thing I learned recently, the ael files are also included in the configuration if they are present, are those recent or from backups? |
| 14:10.32 | nibel | kaldemar: i dont know i did not set up the machine myself unfortunately |
| 14:10.54 | kaldemar | nibel: what are the spans connected to? |
| 14:11.18 | nibel | \DSAFEW\: they were also present in the original configuration and are also in the backups |
| 14:11.28 | kaldemar | and show what causes the "Unable to create channel of type 'Zap'" message. |
| 14:11.38 | nibel | kaldemar: e1 pmx |
| 14:13.18 | kaldemar | the telco equipment is usually the network side and customer is cpe. your zaptel.conf is even configured to take clock from the line. |
| 14:14.08 | nibel | kaldemar: is that a good or a bad thin? |
| 14:14.23 | nibel | kaldemar: sry for my dumbness... |
| 14:14.40 | nibel | kaldemar: never worked on a isdn line before... |
| 14:15.16 | kaldemar | it's normal for the cpe side since telco's provide timing. |
| 14:15.26 | kaldemar | what causes the cause 34 message? |
| 14:15.44 | *** join/#asterisk skten (~skten@118.11.233.220.static.exetel.com.au) |
| 14:16.27 | nibel | kaldemar: mom ihave here the error message from my php/ami script first |
| 14:16.36 | nibel | http://pastebin.com/D2bXDS5q |
| 14:16.52 | nibel | the script just does a originate from ami interface |
| 14:16.52 | skten | Evening all, has anyone here played with two B410Ps back to back and constantly see 'Changing state from awaiting establishment to tei assiged) Yet no reports of down D chan? |
| 14:17.22 | *** join/#asterisk nicoAMG (~nicoamg@201.237.49.131) |
| 14:17.39 | kaldemar | nibel: so group 1 is used for outgoing calls. from zapata.conf we see that channel =>63-77,79-93, and |
| 14:18.01 | kaldemar | channel =>94-108,110-124 belong to group 1. |
| 14:18.02 | \DSAFEW\ | brb food |
| 14:18.33 | nibel | kaldemar: at the moment i only need outgoing calls so we can concentrante on that |
| 14:18.39 | kaldemar | those spans should be connected to the telco device. from zaptel.conf we see that those are spans 3 and 4. |
| 14:18.43 | nibel | kaldemar: everything else is not needed |
| 14:19.31 | kaldemar | pri show span 3 and pri show span 4 will tell you about those spans. |
| 14:19.50 | *** part/#asterisk TobSnyder (~schneider@dslb-088-073-180-175.pools.arcor-ip.net) |
| 14:20.44 | nibel | kaldemar: this is pri show span 3 http://pastebin.com/hM9aGRye |
| 14:21.39 | nibel | kaldemar: this is pri show span 4 http://pastebin.com/azF0mnTL |
| 14:22.27 | kaldemar | interesting. your zapata.conf doesn't have a switchtype defined. are you in NL? |
| 14:22.56 | *** join/#asterisk Ean (~Ean@unaffiliated/ean) |
| 14:23.01 | nibel | kaldemar: no im in de, i was wondering about that too |
| 14:23.27 | \DSAFEW\ | huh... what's a switchtype? |
| 14:23.30 | *** join/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 14:23.50 | kaldemar | nibel: put switchtype=euroisdn in zapata.conf somewhere *above* all channel => lines. |
| 14:24.41 | nibel | kaldemar: do i have to issue a asterisk or just a zaptel restart then? |
| 14:25.08 | kaldemar | nibel: asterisk restart |
| 14:27.02 | nibel | kaldemar: in which context do i have to put it? above channel but under "[channels]" |
| 14:27.37 | kaldemar | under [channels] |
| 14:28.15 | nibel | kaldemar: done |
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| 14:30.31 | nibel | kaldemar: did not help and i get with "dial xxx@from-internal" the 34 error again |
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| 14:31.57 | kaldemar | nibel: what kind of settings has the telco advised you to use? |
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| 14:35.48 | nibel | kaldemar: it's a edss1 with 30 channels plus d-channel |
| 14:36.21 | nibel | kaldemar: i dont know much more i just have the backups of the config that worked in the past |
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| 14:41.02 | nibel | kaldemar: this is the log from issueing a dial on the cmdline http://pastebin.com/26yC74P2 |
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| 14:51.49 | jaytee | I haven't changed anything on my asterisk server or in my configs but this morning I started having problems with my inbound calls. I keep getting this message. |
| 14:52.01 | jaytee | [Apr 27 10:44:07] WARNING[3303]: chan_sip.c:3825 retrans_pkt: Maximum retries exceeded on transmission 1611573244_55241559@4.55.17.35 for seqno 23423 (Critical Response) -- See doc/sip-retransmit.txt. |
| 14:52.21 | jaytee | my provider is Flowroute. Outbound call work fine. |
| 14:52.43 | *** join/#asterisk espent (~espent@46.80-203-219.nextgentel.com) |
| 14:52.44 | jaytee | I restarted my server but still have the same issue. |
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| 14:56.46 | WiretapSeven | jaytee, looks like NAT config |
| 14:56.48 | WiretapSeven | well |
| 14:56.53 | WiretapSeven | as much as I can tell from that one line |
| 14:58.42 | ChannelZ | is a firewall between your server and the net? (or did one get placed there unknowingly?) |
| 14:59.26 | sled-dog | ~book |
| 14:59.27 | infobot | For more information about the Asterisk book, see ~thebook |
| 14:59.33 | sled-dog | ~thebook |
| 14:59.34 | 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://ofps.oreilly.com/titles/9780596517342/, or ~buybook |
| 14:59.45 | mocker | ~buybook |
| 14:59.45 | infobot | You can buy "Asterisk: The Definitive Guide" at http://oreilly.com/catalog/9780596517342 so go buy it SERIOUSLY |
| 15:00.01 | sled-dog | I've got an older version of it, don't worry |
| 15:00.14 | Freeaqingme_ | why cant ~book just show the same message as ~thebook ? |
| 15:00.24 | leifmadsen | because then I have to update to separate things |
| 15:00.30 | leifmadsen | s/to/two/ |
| 15:00.34 | Freeaqingme_ | hehe |
| 15:00.35 | Freeaqingme_ | fair enough |
| 15:00.36 | leifmadsen | it's stupid having the same information in multiple locations |
| 15:00.52 | Freeaqingme_ | agreed |
| 15:00.52 | leifmadsen | (that are not the same resource) |
| 15:01.13 | Freeaqingme_ | leifmadsen, do you know what is suggested these days, the use of ajam or ami? |
| 15:02.06 | *** join/#asterisk bchia (~chatzilla@nat/digium/x-xditvsmtgryhbxih) |
| 15:03.47 | *** join/#asterisk msetim (~setim@187.112.150.148) |
| 15:04.35 | msetim | I'm trying to use voicemail + postgresql 8.3 + lo to store recording messages but I'm getting this error: res_odbc.c: SQL Execute returned an error -1: HY000: Could not commit (in-line) a transaction (40) |
| 15:04.45 | msetim | It was working fine with postgresql 8.1 |
| 15:05.53 | *** join/#asterisk dimm (~appleworm@unaffiliated/dimm) |
| 15:06.00 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 15:08.30 | \DSAFEW\ | nibel, can you show us the dialplan please? |
| 15:09.17 | nibel | \DSAFEW\: i will but i need 5 minutes becuase the machine is restarting right now |
| 15:09.34 | \DSAFEW\ | nibel, sure |
| 15:11.38 | *** join/#asterisk knorkeknie (~uli@84.150.188.107) |
| 15:13.02 | *** join/#asterisk Drumm96 (~drumm@host217-44-206-252.range217-44.btcentralplus.com) |
| 15:17.19 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 15:18.08 | *** join/#asterisk knorkeknie (~uli@p5496BC6B.dip.t-dialin.net) |
| 15:19.35 | *** join/#asterisk leroybuckingham (637c543e@gateway/web/freenode/ip.99.124.84.62) |
| 15:20.29 | nibel | \DSAFEW\: that's the extensions.conf http://pastebin.com/YPC8U0sY |
| 15:20.38 | jaytee | WiretapSeven, bingo! you were right. My cable modem which has it's own firewall and uses DHCP had rebooted during the night. I've since set the NIC for the WAN port on my * box to static and changed the config on the cable modem. |
| 15:20.45 | nibel | \DSAFEW\: it's a mess |
| 15:21.25 | nibel | \DSAFEW\: i plan to clean it up after i got it running |
| 15:21.50 | jaytee | <PROTECTED> |
| 15:22.20 | *** join/#asterisk knorkeknie (~uli@84.150.188.107) |
| 15:22.28 | jaytee | and Russell's recipe for Goat Cheese and Bison meat Lasagna is to die for! |
| 15:24.17 | knorkeknie | hi there, my first time to join here, so ... hi there ;) |
| 15:25.39 | leifmadsen | o/ |
| 15:25.39 | nibel | \DSAFEW\: at the moment i just need from-internal and from-internal2 from that dialplan on this machine |
| 15:26.11 | knorkeknie | started with asterisk some days ago and need some help |
| 15:26.40 | \DSAFEW\ | nibel, yeah, what dials with from-internal? |
| 15:26.47 | \DSAFEW\ | nibel, can you use that to dial? |
| 15:28.55 | \DSAFEW\ | nibel, exten => i,1,Congestion |
| 15:28.57 | \DSAFEW\ | huh? |
| 15:29.23 | \DSAFEW\ | I'm sorry I'm very tired, so this dialplan doesn't make sense to me now |
| 15:29.31 | \DSAFEW\ | my brain is on light duty only |
| 15:29.41 | nibel | \DSAFEW\: moment pls |
| 15:29.57 | nibel | \DSAFEW\: just look at this ok http://pastebin.com/26yC74P2 |
| 15:30.43 | *** join/#asterisk Jaburto (~MrTelepho@S01060018e7e1c1b7.ls.shawcable.net) |
| 15:31.06 | Jaburto | Can I register my purchased fax for asterisk licenses on a backup machine? |
| 15:31.56 | Freeaqingme_ | Jaburto, I think you would have to ask digium (or wherever you ordered it) |
| 15:31.58 | leifmadsen | Jaburto: I think they can only be registered to a single machine |
| 15:32.06 | leifmadsen | and yes - that's a commercial support question |
| 15:32.42 | Jaburto | ok |
| 15:32.54 | Jaburto | because I was able to register the free fax for asterisk on both |
| 15:33.08 | Freeaqingme_ | Jaburto, if you cannot register it twice, you could perhaps also install the free license on your backup system |
| 15:33.20 | nibel | \DSAFEW\: can u look on my last pastebin to give me a last idea |
| 15:33.44 | Jaburto | I'll have to buy some more then |
| 15:34.01 | *** join/#asterisk bn-7bc (bjarne@macbook-pro.lan-sx.noare-1.holmedal.net) |
| 15:35.53 | Jaburto | If you have 2 asterisk boxes with the exact same configuration and you clients were setup to connect based on SRV records how could you proxy the invites between machines to see which one the client is connected to? |
| 15:37.08 | Jaburto | I made up a working dialplan but what was happening is the remote asterisk box was trying to authenticate the client instead of the source asterisk machine where the sip message was piggy backed |
| 15:38.51 | Jaburto | I think it would work great if the authentication logic was hacked up a little bit |
| 15:39.30 | Jaburto | Or should I go pop another xanax? |
| 15:39.53 | \DSAFEW\ | nibel, so does the asterisk log say anything? |
| 15:40.17 | \DSAFEW\ | nibel, about the CPE? |
| 15:46.31 | *** join/#asterisk dfamorato (~dfamorato@173-9-190-190-miami.txt.hfc.comcastbusiness.net) |
| 15:48.40 | *** join/#asterisk sourcode (~code@ppp-115-87-212-151.revip4.asianet.co.th) |
| 15:49.24 | *** join/#asterisk JonathanRose (~jonathan@nat/digium/x-fwkodizszpvtlhbo) |
| 15:50.25 | *** join/#asterisk fragileX (~bill@204.134.45.21) |
| 15:51.37 | nibel | \DSAFEW\: where shall i look for cpe? |
| 15:52.19 | \DSAFEW\ | nibel, I was asking for a log file about asterisk something like /var/log/asterisk/messages |
| 15:53.01 | \DSAFEW\ | nibel, if you want to rotate that and restart asterisk, it would be smaller to paste |
| 15:53.35 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 15:53.49 | *** join/#asterisk knorkeknie (~uli@p5496F37A.dip.t-dialin.net) |
| 15:54.41 | *** part/#asterisk Jaburto (~MrTelepho@S01060018e7e1c1b7.ls.shawcable.net) |
| 15:57.28 | nibel | \DSAFEW\: i paste the last startup until now |
| 15:58.57 | *** join/#asterisk Sertys (~sertys@89.252.247.42) |
| 16:04.41 | nibel | \DSAFEW\: http://pastebin.com/NbNnTKz3 /var/log/asterisk/messages |
| 16:06.24 | *** join/#asterisk gruvfunk (~chatzilla@user-160uac8.cable.mindspring.com) |
| 16:09.43 | *** join/#asterisk Eitan (~Eitan@adsl-99-22-192-148.dsl.lsan03.sbcglobal.net) |
| 16:10.15 | nibel | damn it |
| 16:10.25 | nibel | i'm so tired |
| 16:10.35 | *** join/#asterisk puzzled (~patrick@puzzled.xs4all.nl) |
| 16:12.32 | beek | hands nibel some No-Doze |
| 16:16.15 | *** join/#asterisk zamba (marius@flage.org) |
| 16:16.18 | zamba | you guys heard about lync? |
| 16:18.03 | Qwell | Another Microsoft voice thingie? I can't wait! The last several they've done did so well. |
| 16:20.41 | zamba | Qwell: well.. this has some potential, i believe |
| 16:20.55 | zamba | but i hate microsoft just as much as the next guy in here :) |
| 16:21.06 | zamba | just wondered if anyone had any experience with it |
| 16:22.25 | beek | I'd rather just use two tin cans and some string than use anything from MS. |
| 16:22.42 | \DSAFEW\ | nibel, try commenting out this in the zapata |
| 16:23.01 | \DSAFEW\ | echocancel=yes |
| 16:23.01 | \DSAFEW\ | echocancelwhenbridged=yes |
| 16:23.14 | \DSAFEW\ | or set them to no |
| 16:24.30 | \DSAFEW\ | restart the asterisk service and see if that helps |
| 16:24.31 | nibel | \DSAFEW\: i'll try that, btw you such awesome person helping me here for such a long time |
| 16:24.44 | nibel | + are |
| 16:24.49 | *** join/#asterisk fhmiv (~fhmiv@c-67-173-205-151.hsd1.ga.comcast.net) |
| 16:26.48 | nibel | \DSAFEW\: no it did not help and "pri show span 3" still says: Status: Provisioned, Down, Active |
| 16:26.59 | \DSAFEW\ | nibel, you'll be doing this next week probably, lol |
| 16:27.33 | nibel | \DSAFEW\: when i sortedd this problem out until then XD |
| 16:28.01 | *** join/#asterisk devil_evoxxx (~d3v1l@host193-21-dynamic.180-80-r.retail.telecomitalia.it) |
| 16:28.23 | devil_evoxxx | hi guys..there someone of digium support center? |
| 16:29.03 | Qwell | devil_evoxxx: There are, but this is not an official support channel. |
| 16:29.53 | devil_evoxxx | Qwell, i know :) thankyou for your reply. I open a support case for a b410pf card |
| 16:30.15 | devil_evoxxx | 4 or 5 day ago |
| 16:30.55 | devil_evoxxx | i have resend an email because we had some problem with our mailserver |
| 16:32.12 | devil_evoxxx | and i want to say if the support is already open and upgraded with my last email ( sent today) |
| 16:35.29 | leifmadsen | ruben23: |
| 16:35.31 | leifmadsen | oops |
| 16:35.45 | leifmadsen | russellb: sip:polycom@shifteight.org I think works too |
| 16:35.59 | leifmadsen | russellb: or ISN: 7659*460 |
| 16:36.48 | russellb | oic |
| 16:36.53 | \DSAFEW\ | nibel, does a dahdi restart do anything? |
| 16:36.53 | *** join/#asterisk fholmes (~fholmes@adsl-99-97-195-110.dsl.hstntx.sbcglobal.net) |
| 16:39.03 | citywok | Any digium guy's around? What's the latest firmware on the AA50 supposed to be? I've got a beta firmware still from an RMA unit. |
| 16:39.19 | russellb | http://www.digium.com/en/supportcenter/ |
| 16:39.25 | russellb | because i have no idea |
| 16:42.22 | \DSAFEW\ | nibel, look into maybe resetinterval= , Im' fallin asleep now |
| 16:42.55 | nibel | \DSAFEW\: where do i have to look for resetinterval? |
| 16:43.00 | \DSAFEW\ | nibel, might be upstream, call the provider again and make them suffer your insane old astrrisks |
| 16:43.15 | \DSAFEW\ | nibel, I would google it, forget what it's for |
| 16:44.59 | nibel | \DSAFEW\: i have no dahdi in the asterisk ctl |
| 16:45.52 | nibel | \DSAFEW\: but dundi... |
| 16:46.20 | \DSAFEW\ | zaptel is old name for dahdi |
| 16:46.35 | nibel | \DSAFEW\: it is a sangoma card |
| 16:46.45 | leifmadsen | wanpipe can still use dahdi |
| 16:46.52 | nibel | ah right |
| 16:46.56 | leifmadsen | zaptel is like 2 years old now |
| 16:47.09 | leifmadsen | I'd not be doing any new deployments on it |
| 16:47.14 | nibel | leifmadsen: this one is even older |
| 16:47.23 | leifmadsen | define: "this one" |
| 16:47.34 | nibel | leifmadsen: it's a doomed 1.2 machine |
| 16:48.19 | nibel | leifmadsen: and with doomed i mean doomed |
| 16:49.02 | leifmadsen | and with that, I wipe my hands of it ;) |
| 16:49.04 | \DSAFEW\ | http://pastebin.com/gU7gcZwF http://pastebin.com/s1QyGgND http://pastebin.com/hM9aGRye http://pastebin.com/YPC8U0sY http://pastebin.com/NbNnTKz3 |
| 16:49.21 | nibel | thx for the summary |
| 16:49.31 | \DSAFEW\ | oh yes one of the bugreports I found said this problem was patched |
| 16:49.39 | \DSAFEW\ | but this worked before |
| 16:49.50 | \DSAFEW\ | and you aren't reinstalling asterisk I guess |
| 16:49.51 | *** join/#asterisk digilink (~digilink@vps.stephennet.net) |
| 16:50.00 | nibel | \DSAFEW\: im considering it |
| 16:50.09 | nibel | \DSAFEW\: can u give me alinkt to the bug report |
| 16:50.17 | \DSAFEW\ | nibel, well, that dialplan baffles my sleepy head |
| 16:50.20 | \DSAFEW\ | good luck |
| 16:50.39 | *** join/#asterisk bchia (~chatzilla@nat/digium/x-ivuxdogmdconpcgk) |
| 16:50.43 | \DSAFEW\ | nibel, no I just read it was patched, nothing new here |
| 16:51.00 | \DSAFEW\ | searched for various parts of your 34 error |
| 16:51.22 | \DSAFEW\ | you are not up, your link isn't up |
| 16:51.25 | \DSAFEW\ | that's the problem |
| 16:51.26 | nibel | \DSAFEW\ hm |
| 16:51.43 | *** join/#asterisk paulc (~paulc@unaffiliated/paulc) |
| 16:51.57 | nibel | \DSAFEW\: i know but i find no reason or error log for that... |
| 16:52.05 | \DSAFEW\ | see if that timeout thing works for your zapata or w/e |
| 16:52.23 | \DSAFEW\ | nibel, perhaps it's in the zaptel log |
| 16:52.52 | nibel | \DSAFEW\: there is no /var/log/zaptel |
| 16:53.08 | nibel | \DSAFEW\: do u mean setinterval or timeout? |
| 16:55.27 | *** join/#asterisk el3slave (~email@ip68-4-133-145.oc.oc.cox.net) |
| 16:58.40 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
| 17:02.10 | Kobaz | anyone have an example of a chan_dahdi with a simple pri for 1.8 |
| 17:02.35 | *** join/#asterisk NightMonkey (debian-tor@pdpc/supporter/professional/nightmonkey) |
| 17:02.55 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v013-117.mobile.uci.edu) |
| 17:04.07 | *** join/#asterisk rogersja (~RogersJa@S0106000f6695039f.gv.shawcable.net) |
| 17:06.25 | *** join/#asterisk nickfennell (~nick@i-195-137-23-30.freedom2surf.net) |
| 17:06.26 | nickfennell | hihi |
| 17:06.29 | nickfennell | Quick question |
| 17:06.57 | nickfennell | Would we all assume that the most risky part of using VoIP is between the EP and the provider in terms of security |
| 17:07.23 | Kobaz | depends |
| 17:07.27 | nickfennell | ie, when it involves the intraweb |
| 17:07.44 | carrar | not if the SIP signaling is in a VPN's |
| 17:07.44 | Kobaz | where does the provider send the call... is the providers' provider link secure |
| 17:07.46 | carrar | VPN |
| 17:07.49 | *** join/#asterisk sakajawebe (~chazz@nat/digium/x-cbbienjewtxiiyxt) |
| 17:08.06 | nickfennell | carrar, without VPN or any other security in place |
| 17:08.29 | nickfennell | and Kobaz, I assume that the provider has E1/T1 within their network and all media gateways exist within provider network |
| 17:08.53 | nickfennell | so once EP traffic reaches provider, it should be secure |
| 17:08.58 | Freeaqingme_ | nickfennell, I think most 'hacks' occur because people have too easy passwords,/and/ their pbx allows spoofed callerids to dial outbound |
| 17:09.00 | nickfennell | but, between the provider and the EP |
| 17:09.05 | nickfennell | Freeaqingme, indeed. |
| 17:09.34 | carrar | Secure voip involves encrypting the voice |
| 17:09.39 | carrar | end to end |
| 17:09.46 | nickfennell | Correct. |
| 17:09.46 | Kobaz | nickfennell: you shouldn't assume anything |
| 17:09.48 | carrar | encryptig the audio |
| 17:10.09 | carrar | which doesn't work over T1 |
| 17:10.11 | devil_evoxxx | i'm try to configure b410pf card, for trasforming a old pbx into voip tech. I'm using dahdi and i want to say if the options immediate=yes (present on zapte), is present still in dahdi and where i can configure that. |
| 17:10.13 | carrar | from sip |
| 17:10.26 | nickfennell | I'm trying to build a generic case example where the most common scenario is 'x' and the security should be implemented at 'y' |
| 17:10.41 | Freeaqingme_ | if you want your T1 to be safe you'd have to start using (and designing) ss8 |
| 17:10.41 | nickfennell | well E1/T1 would be PSTN break out |
| 17:10.57 | carrar | PSTN is not secure |
| 17:10.58 | nickfennell | ISDN30 equiv. |
| 17:11.04 | Freeaqingme_ | it is, but T1/ss7 by definition is insecure |
| 17:11.05 | carrar | why would you thik it is? |
| 17:11.11 | nickfennell | Yeah but that's a totally different issue |
| 17:11.15 | nickfennell | Which goes well past VoIP |
| 17:11.23 | nickfennell | I can't do anything about securing PSTN |
| 17:11.30 | Freeaqingme_ | you can |
| 17:11.31 | Freeaqingme_ | by not using it |
| 17:11.39 | Freeaqingme_ | it's all about priorities |
| 17:11.41 | nickfennell | but I can do something about securing up to the provider. If the providers core network isn't secure then that's their issue not mine |
| 17:12.02 | *** join/#asterisk deadpigeon (~jonathan@office.xpressamerica.net) |
| 17:12.11 | nickfennell | Do you see where I'm coming from |
| 17:12.45 | carrar | London? |
| 17:13.17 | nickfennell | where $humour >=0; die(); |
| 17:13.38 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 17:13.40 | jaytee | how can you tell if he's from London? bad teeth? |
| 17:13.46 | carrar | yeah |
| 17:13.51 | carrar | and |
| 17:14.02 | carrar | he's on the side of the street with a laptop waiting for the marriage people |
| 17:14.11 | nickfennell | OOo I like this |
| 17:14.18 | carrar | heh |
| 17:14.28 | jaytee | 60 mil for a wedding. what a joke! |
| 17:14.40 | Kobaz | nickfennell: vpn |
| 17:14.40 | nickfennell | What can we say, we have style |
| 17:14.46 | Kobaz | what specific question do you have? |
| 17:14.49 | nickfennell | Kobaz, I have better than VPN :) |
| 17:14.56 | Kobaz | carrier pidgeon? |
| 17:14.57 | paulc | I can't believe you went there with the bad teeth reference |
| 17:15.04 | nickfennell | Kobaz, it wasn't any specific, I just wanted some opinions |
| 17:15.24 | Kobaz | messenger on horseback with the official crown seal? |
| 17:15.27 | nickfennell | but typically, as with most conversations, it becomes convoluted and pointless |
| 17:15.28 | carrar | What do you want to protect? the conversation or the server? |
| 17:16.13 | nickfennell | nvm people, this is clearly beyond your heuristic capabilities |
| 17:16.17 | carrar | haha |
| 17:16.18 | carrar | yes |
| 17:17.00 | nickfennell | I'll leave you to your pigeon holing and stereotypical ideologies |
| 17:17.07 | carrar | back to register... you want fries with that |
| 17:17.08 | Kobaz | we're here to help |
| 17:17.13 | nickfennell | and that you have. |
| 17:17.17 | Kobaz | k |
| 17:17.22 | nickfennell | cheers Kobaz |
| 17:18.17 | Kobaz | the teeth thing was a little off |
| 17:18.28 | Kobaz | americans probably have much worse teeth on average than the rest of the world |
| 17:18.37 | Kobaz | all this sugar in everything |
| 17:18.39 | nickfennell | Americans have bigger problems than teeth |
| 17:18.40 | nickfennell | lol |
| 17:18.53 | nickfennell | It's OK though. Wet get it ;) |
| 17:19.03 | msetim | I'm trying to use voicemail + postgresql 8.3 + lo to store recording messages but I'm getting this error: res_odbc.c: SQL Execute returned an error -1: HY000: Could not commit (in-line) a transaction (40) |
| 17:19.03 | msetim | <PROTECTED> |
| 17:19.06 | nickfennell | Just try harder next time :P |
| 17:19.57 | drmessano | I missed another ideology class? |
| 17:19.58 | drmessano | Shit, I am gonna fail the final |
| 17:21.30 | Kobaz | no make up sessions either |
| 17:23.41 | Kobaz | damn, i dont have my knoppix usb stick |
| 17:24.49 | *** join/#asterisk fhmiv (~fhmiv@c-67-173-205-151.hsd1.ga.comcast.net) |
| 17:24.57 | Qwell | Knoppix still exists? And people still use it? |
| 17:26.12 | Kobaz | yeah |
| 17:26.14 | Kobaz | recovery |
| 17:26.21 | Kobaz | and drive copying |
| 17:26.24 | Kobaz | etc |
| 17:27.58 | tzafrir | hi |
| 17:28.47 | tzafrir | Some off-topic questions: I need some information about Avaya systems |
| 17:29.35 | tzafrir | They use some proprietary voip protocol of their own, right? |
| 17:29.38 | Freeaqingme_ | now that is indeed offtopic :P |
| 17:29.56 | tzafrir | Does wireshark understand it? |
| 17:30.08 | tzafrir | Does it use RTP for the audio? |
| 17:30.26 | *** join/#asterisk Tozz_ (Tozz@hardwire.duocast.net) |
| 17:30.52 | Kobaz | the avaya ip office stuff is sip |
| 17:31.47 | tzafrir | I'll be looking at an installation with some Avaya hard phones tommorow (so I figure the rest of the system is Avaya) |
| 17:31.57 | tzafrir | Is it SIP? |
| 17:34.15 | Kobaz | is it an ip office system? |
| 17:34.38 | Kobaz | the definity phones are non-ip but digital |
| 17:34.45 | Kobaz | they run on single pair cat3 |
| 17:34.52 | tzafrir | Not really sure. It's VoIP |
| 17:34.55 | Kobaz | the ip office phones are sip |
| 17:35.37 | Kobaz | without more information i can't tell you anything else |
| 17:35.47 | RypPn | h323 |
| 17:36.04 | RypPn | If its a 4600 |
| 17:36.45 | Kobaz | anyway |
| 17:36.47 | Kobaz | back to asterisk |
| 17:36.52 | Kobaz | did something change with dahdi in 1.8.3 |
| 17:36.53 | Kobaz | [2011-04-27 13:32:56] ERROR[3549]: chan_dahdi.c:16838 process_dahdi: Unknown signalling method 'pri_cpe' at line 36. |
| 17:37.12 | *** join/#asterisk not_schmaltz (~Athens@74-92-102-230-Philadelphia.hfc.comcastbusiness.net) |
| 17:37.58 | *** join/#asterisk eduzimrs (~eduzimrs@201.22.86.124.static.gvt.net.br) |
| 17:38.22 | Qwell | Kobaz: Do you have libpri installed? |
| 17:38.31 | Qwell | You may need to update it. |
| 17:38.41 | not_schmaltz | Hi, I'm working on creating an IVR application (web app in php), that will need different logic for different clients. What should I use to create the dialplans -- I need them to be dynamic.. |
| 17:38.41 | Kobaz | that's what i was thinking |
| 17:38.48 | Kobaz | i'm trying to go from 1.6.0 to 1.8.3 |
| 17:38.54 | Kobaz | i have a really old libpri probably |
| 17:39.28 | Kobaz | 1..10 |
| 17:39.31 | Kobaz | 1.4.10 |
| 17:39.49 | Kobaz | oh [ ]ChangeLog-1.4.1018-Apr-2009 18:15 16K |
| 17:39.53 | Kobaz | yeah that is old |
| 17:40.31 | *** join/#asterisk Shazaum (~1@unaffiliated/shazaum) |
| 17:41.03 | tzafrir | Qwell, asteriks will completely fail to use an older libpri, or just not use newer features? |
| 17:42.02 | *** join/#asterisk benngard (~mabe@h30n2-g-ml-a11.ias.bredband.telia.com) |
| 17:43.05 | Qwell | tzafrir: don't know, but something that old, I wouldn't be surprised if it just bombed completely. |
| 17:43.44 | Kobaz | teh bombs |
| 17:43.58 | *** join/#asterisk ks3 (~ksandy@74.203.195.1) |
| 17:44.17 | tzafrir | Kobaz, what happens if you explicitly configure --with-libpri ? |
| 17:44.18 | Kobaz | oh crap, what can i substitute for a torx driver |
| 17:44.28 | tzafrir | Though upgrading libpri wouldn't hurt |
| 17:44.39 | Kobaz | yeah i;m upgrading |
| 17:44.50 | Kobaz | this is a box in a failover setup, not using it right now |
| 17:45.17 | Kobaz | figured might as well upgrade asterisk too see if that resolves the problems we've been having on this system |
| 17:45.29 | Kobaz | since i'm already replacing the hard drives with bigger ones and etc |
| 17:47.25 | Kobaz | stupid torx |
| 17:47.28 | asilva | Has anyone made upgrades from 1.6.x to 1.8.x without problems ? |
| 17:47.40 | *** join/#asterisk mclaro (~mclaro@190.183.222.194) |
| 17:50.14 | Kobaz | i haven't |
| 17:50.28 | Kobaz | i used 1.8.1 and crashed it within a few minutes |
| 17:50.30 | not_schmaltz | cakephp |
| 17:51.15 | Kobaz | trying 1.8.3.3 now |
| 17:51.41 | Kobaz | i'm hitting the end of a bunch of my development projects, so any crashes I do hit, i can start working on reporting and fixing |
| 17:52.35 | asilva | Kobaz, crashes without any reason ? |
| 17:52.42 | Kobaz | there's always a reason |
| 17:52.42 | asilva | just start & crash ? |
| 17:52.47 | Kobaz | no it would start |
| 17:52.52 | asilva | ahahah sure... but's not what i meant |
| 17:52.59 | Kobaz | i was trying out attended transfers after having Bridge()'d a call |
| 17:53.21 | Kobaz | i think that was fixed in 1.8.2 |
| 17:53.31 | Kobaz | i remember seeing a fix for an attended transfer crash |
| 17:54.51 | Kobaz | okay dahdi 2.4.1.2 libpri 1.4.11.5 |
| 17:55.02 | *** join/#asterisk wonderworld (~ww@port-92-201-150-156.dynamic.qsc.de) |
| 17:55.03 | Kobaz | chan_dahdi.c:16838 process_dahdi: Unknown signalling method 'pri_cpe' at line 36. |
| 17:56.03 | Qwell | did you re-run configure? |
| 17:57.21 | Kobaz | doing that now, i just upgraded dahdi and libpri in place |
| 17:57.25 | Kobaz | upgrading dahdi on my build server |
| 18:00.25 | *** join/#asterisk mickecarlsson (~Micke@h131n2c1o1101.bredband.skanova.com) |
| 18:00.40 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 18:01.34 | mickecarlsson | has a question about realtime voicemal, any takers? |
| 18:03.50 | Sertys | realtime voicemail |
| 18:03.59 | Sertys | i like the way it sounds |
| 18:04.09 | Sertys | realtime voicemail is called conversation |
| 18:04.27 | mickecarlsson | lol |
| 18:05.07 | mickecarlsson | Actually, it section [general] read from realtime database or from voicemail.conf when using realtime voicemail |
| 18:05.07 | *** join/#asterisk cj (~cjac@adsl-207-32-169-17.rockisland.net) |
| 18:05.25 | cj | so... diameter... |
| 18:06.01 | Kobaz | oh |
| 18:06.02 | Kobaz | mama |
| 18:06.03 | Kobaz | er |
| 18:06.04 | Kobaz | haha |
| 18:06.11 | Kobaz | i dont need a torx, this hard drive just pops right out |
| 18:06.12 | Kobaz | yay |
| 18:06.19 | Sertys | :) |
| 18:08.45 | cj | https://reviewboard.asterisk.org/r/268/ |
| 18:08.59 | cj | <3 ChipX86 |
| 18:09.13 | *** join/#asterisk ferdna (~yup@cpe-67-10-220-35.elp.res.rr.com) |
| 18:09.35 | tzafrir | Kobaz, one more off-topic Avaya question: |
| 18:10.11 | tzafrir | If I sniff traffic there, can I expect to figure out a range of IP addresses from broadcasts? |
| 18:10.35 | tzafrir | (apart from ARPs, which I may or may not see) |
| 18:10.51 | Kobaz | perhaps |
| 18:10.57 | Kobaz | i don't know really what sort of data they send |
| 18:11.12 | Kobaz | arps may be your only bet |
| 18:11.22 | Kobaz | but i don't know if they use broadcast traffic |
| 18:12.35 | Kobaz | Qwell: okay, rebuilt dahdi, recompiled asterisk. still doesn't like pri_cpe |
| 18:13.17 | Kobaz | time to visit chan_dahdi line 16838 |
| 18:13.45 | Kobaz | #ifdef HAVE_PRI |
| 18:13.46 | Kobaz | hmm |
| 18:14.04 | Qwell | recompiled, or re-ran configure and recompiled? |
| 18:14.29 | Kobaz | i already had it configured |
| 18:14.36 | Kobaz | or does it need reconfiguring after a new dahdi? |
| 18:14.45 | Kobaz | i dont have HAVE_PRI defined |
| 18:16.09 | Kobaz | okay reconfigured, HAVE_PRI is defined now |
| 18:20.25 | *** join/#asterisk wonderworld (~ww@port-92-201-150-156.dynamic.qsc.de) |
| 18:22.23 | Kobaz | yay, okay that worked |
| 18:29.04 | knorkeknie | hey ther, im trying to find out how to use DEVICE_STATE.... just wanna ensure that if one phone is busy another sip-phone is called... but even if im phoning the device-state is not_inuse... |
| 18:31.19 | knorkeknie | just doing a simple NoOp(${DEVICESTATE(SIP/223)}) in an extension... whenn im calling another phone with my device 223 and i call with my mobile the state is still NOT_INUSE |
| 18:32.00 | knorkeknie | did i misunderstand DEVICE_STATE or where is the problem ? |
| 18:34.49 | cj | tzafrir: use lldpd |
| 18:35.07 | cj | tzafrir: I've got a bunch of 1120e phones |
| 18:37.24 | tzafrir | cj, This is what Cisco uses. Avaya also uses it? |
| 18:37.44 | tzafrir | Though switches there are cisco, so it won't hurt |
| 18:39.49 | Kobaz | lldp is a standard |
| 18:40.49 | *** join/#asterisk psilvao (~psilvao@190.20.25.210) |
| 18:41.34 | cj | tzafrir: cisco might have switched to it. it used to use CDP. LLDP is the vendor-neutral version |
| 18:43.02 | *** join/#asterisk p3nguin (~xwQ5kwYl6@cobalt.esxi.hosts.a2infotech.com) |
| 18:43.06 | cj | tzafrir: http://paste2.org/p/1387722 |
| 18:44.29 | cj | look like the info you're seeking? |
| 18:48.30 | tzafrir | cj, yes, indeed :-) |
| 18:48.56 | cj | good deal |
| 18:49.01 | tzafrir | now, if I can only convince them to connect my laptop to that network for such a demonstration... |
| 18:49.04 | cj | sudo apt-get install lldpd :) |
| 18:51.21 | cj | okay... so... why are these phones responding 305? |
| 18:51.22 | cj | :( |
| 18:54.56 | Qwell | Because Nortel doesn't know SIP. |
| 19:06.40 | Kobaz | why can't people stop calling in when you're doing maintenance |
| 19:07.19 | leifmadsen | why don't you remember to turn off your phones when doing maintenance? :) |
| 19:07.57 | *** part/#asterisk bchia (~chatzilla@nat/digium/x-ivuxdogmdconpcgk) |
| 19:08.10 | *** join/#asterisk bchia (~chatzilla@nat/digium/x-ivuxdogmdconpcgk) |
| 19:09.23 | cj | leifmadsen: feel like helping me troubleshoot this 305 issue with the nortel 1120e? |
| 19:09.54 | cj | McNamara says he got it working: |
| 19:09.55 | cj | http://blog.michaelfmcnamara.com/2011/01/avaya-ip-1100-series-ip-phone-upgrade-to-sip/ |
| 19:10.01 | cj | so I expect I'm doing something wrong. |
| 19:10.44 | Kobaz | hmm |
| 19:10.54 | Kobaz | looks like dialplan quoting works differently in 1.8 |
| 19:11.08 | *** join/#asterisk haroldp (~haroldp@99-46-24-87.lightspeed.renonv.sbcglobal.net) |
| 19:11.26 | haroldp | hullo |
| 19:13.32 | Kobaz | is there a dialplan variable for asterisk version |
| 19:16.15 | haroldp | I have a voip phone on a private NAT-ed IP logged into an asterisk server running on my router box. The asterisk server sends calls out through a trunk to a voip service. I can place calls, but I can't receive them. Any ideas? |
| 19:17.42 | leifmadsen | cj: not really |
| 19:17.43 | Kobaz | i guess quotes are literal now |
| 19:17.48 | Kobaz | in 1.8? |
| 19:17.51 | Kobaz | Set(foo="") |
| 19:18.04 | Kobaz | actually sets foo to the string "" not the empty string |
| 19:22.03 | *** join/#asterisk sled-dog (~luser@adsl-074-165-241-009.sip.msy.bellsouth.net) |
| 19:22.25 | sled-dog | yealink = garbage? # my first thought, anyway... |
| 19:24.49 | cj | leifmadsen: yeah, me neither. *sigh* |
| 19:24.58 | Kobaz | a bunch of people at astricon liked yealink |
| 19:25.05 | sled-dog | hrm |
| 19:25.13 | Kobaz | whatever quality they are, they have to be better than grandstream... right? |
| 19:25.15 | Qwell | Kobaz: How many of them were shunned? |
| 19:25.21 | sled-dog | Kobaz: you got that right |
| 19:25.29 | Kobaz | either people liked yealink, or they never used them |
| 19:27.24 | Kobaz | Qwell: so for 1.8, is that correct? " for Set() is now literal? |
| 19:31.42 | *** join/#asterisk mac-mini (~mac-mini@unaffiliated/macmini/x-648924) |
| 19:32.03 | Kobaz | i like the new dahdi channel format DAHDI/i1/8147351234-5 |
| 19:36.49 | *** join/#asterisk p3nguin (~xwQ5kwYl6@cobalt.esxi.hosts.a2infotech.com) |
| 19:42.43 | haroldp | I'm seeing this in the asterisk messages log when a call comes in: [Apr 27 12:41:36] WARNING[89899] chan_sip.c: username mismatch, have <teliax>, digest has <haroldp> |
| 19:44.10 | haroldp | "teliax" is my voip providor. don't think I have that set as a username anywhere in my config |
| 19:44.33 | Kobaz | the digest is what's inside the sip message |
| 19:44.42 | Kobaz | the have <teliax> is what you have defined in your asterisk |
| 19:44.53 | Kobaz | it's telling you the provider is sending a username that doesn't match your local config |
| 19:45.12 | cj | leifmadsen: is there any way to run asterisk without disabling selinux? I've got some clients that would disapprove. |
| 19:45.30 | leifmadsen | cj: sure, configure SElinux |
| 19:45.36 | Kobaz | heh |
| 19:45.45 | *** join/#asterisk wedhorn (~wedhorn@123-243-209-34.static.tpgi.com.au) |
| 19:45.56 | sxpert | cj: good luck jim ;) |
| 19:46.03 | wedhorn | exit |
| 19:46.04 | haroldp | in sip.conf I have a [teliax] block defined, but is have username=haroldp |
| 19:46.08 | cj | sxpert: inorite? |
| 19:46.15 | *** part/#asterisk wedhorn (~wedhorn@123-243-209-34.static.tpgi.com.au) |
| 19:46.16 | haroldp | "it has" rather |
| 19:46.38 | leifmadsen | cj: I didn't document it because I have zero SElinux experience |
| 19:46.48 | leifmadsen | and it seemed like more work than I was comfortable doing |
| 19:46.49 | sxpert | cj: aka "good luck in configuring se-linux to work with asterisk" |
| 19:47.00 | Kobaz | haroldp: your provider is sending you haroldp |
| 19:47.06 | Kobaz | haroldp: you need to have [haroldp] |
| 19:47.09 | cj | ok. maybe I'll blog all about it and let you publish it in the next version for a spot in the thanks blurb :) |
| 19:47.14 | leifmadsen | so the answer is of course, "sure you can", I just can't be the one to help you |
| 19:47.24 | Kobaz | you need to match the username the provider is sending you |
| 19:47.25 | leifmadsen | cj: that's certainly doable :) |
| 19:47.28 | haroldp | ok. |
| 19:47.31 | Kobaz | anyway. busting out |
| 19:47.37 | cj | leifmadsen: I've got regulatory obligations that make it worth my time :) |
| 19:47.41 | Kobaz | either change it on your end, or have the provider change it on theirs |
| 19:47.47 | Kobaz | goes away |
| 19:48.17 | cj | the index isn't helping me find the "how to configure your tdm400 card" bit. could you point me in the right direction? |
| 19:48.41 | leifmadsen | cj: integration chapter |
| 19:48.54 | cj | I looked under dahdi but only found a build ref. cool, I'll look there. |
| 19:49.10 | leifmadsen | cj: sorry, it's called Outside Connectivity now |
| 19:49.27 | leifmadsen | I think User Device Configuration may also have something |
| 19:49.38 | leifmadsen | (I didn't write those chapters really, or if I did, it was many moons ago) |
| 19:50.37 | cj | ok. I'll come back when I've read those chapters. I'm at the point that I get these messages in the console: |
| 19:50.40 | cj | [Apr 27 12:33:45] WARNING[20315]: chan_dahdi.c:7608 handle_alarms: Detected alarm on channel 4: Red Alarm |
| 19:50.43 | cj | [Apr 27 12:33:56] NOTICE[20315]: sig_analog.c:3680 analog_handle_init_event: Alarm cleared on channel 4 |
| 19:51.32 | leifmadsen | that looks normal |
| 19:51.38 | leifmadsen | if it's clearing the alarm |
| 19:51.49 | leifmadsen | means "I don't have sync", "Now I have sync" |
| 19:52.05 | cj | I would expect the phone to be picked up and handed on to some screaming monkeys at this time |
| 19:52.07 | leifmadsen | or on analog, "I don't have signal", "Now I have signal" |
| 19:52.12 | *** join/#asterisk anonymouz666 (~anonymouz@187-28-37-118.poolip.RJO.embratel.net.br) |
| 19:52.49 | cj | http://paste2.org/p/1387798 |
| 19:54.08 | haroldp | Kobaz: Progress (aka, found other breakage)! Thanks for your help. |
| 19:54.15 | *** join/#asterisk jplank (~G_Bove@208-104-67-26.dyn.fttp.comporium.net) |
| 19:54.24 | cj | http://paste2.org/p/1387799 |
| 19:54.32 | cj | ok. now for the reading. |
| 19:55.48 | haroldp | And fixed! w00h00! Now I can constantly ruin my concentration with lots of incoming calls! |
| 20:08.20 | *** join/#asterisk Aut0ExeC (~Jack@24.244.156.75) |
| 20:09.12 | *** join/#asterisk Aut0ExeC (~Jack@24.244.156.75) |
| 20:09.26 | Aut0ExeC | hi guys... how can I check to see if my asterisk box has been haxed? |
| 20:09.56 | *** part/#asterisk haroldp (~haroldp@99-46-24-87.lightspeed.renonv.sbcglobal.net) |
| 20:10.22 | Freeaqingme_ | ~thebook |
| 20:10.22 | 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://ofps.oreilly.com/titles/9780596517342/, or ~buybook |
| 20:10.29 | Freeaqingme_ | I should add that to my favorites... |
| 20:11.06 | Aut0ExeC | i just wanna know if my box is haxed or not |
| 20:11.19 | Aut0ExeC | ive been getting random calls |
| 20:14.26 | Freeaqingme_ | leifmadsen, in ~thebook you write "All AMI messages, including manager events, manager actions, and manager action responses, are encoded in the same way. The messages are text-based, with lines terminated by a carriage return and a line-feed character", but it seems it's just \n ? |
| 20:16.25 | leifmadsen | looks at russellb |
| 20:16.28 | leifmadsen | I didn't write that chapter ;) |
| 20:18.44 | Freeaqingme_ | oh, sorry |
| 20:18.48 | Freeaqingme_ | I guessed |
| 20:18.50 | Freeaqingme_ | russellb, ^^ |
| 20:21.15 | leifmadsen | I forgot but russell is out this afternoon |
| 20:21.32 | Freeaqingme_ | np |
| 20:21.35 | leifmadsen | Freeaqingme: isn't \n a linefeed character? |
| 20:21.36 | Freeaqingme_ | will bug him later :twisted: |
| 20:23.29 | Freeaqingme_ | leifmadsen, I believe so |
| 20:24.58 | sled-dog | I'm having a problem getting one calls from one * box to another: http://joeykelly.net/hacks/dual-asterisk-problem.txt # I've been through the various google hits, but can't make it work |
| 20:25.01 | *** join/#asterisk LedZeplin (jbearer@74.46.1343.static.theplanet.com) |
| 20:26.10 | Aut0ExeC | to make sure my box isnt hacked shoudl I just cehck my cdr logs? |
| 20:26.30 | LedZeplin | I would like to have an object like a queue but maybe not actually a queue. I would like it to ring a set of phones for x seconds. then I'd like it to escelate to more phones, and then escelate to even more phones before timing out and going to voicemail. any tips on how to attack this? |
| 20:27.15 | *** join/#asterisk WiretapSeven (~Wiretap@unaffiliated/wiretap) |
| 20:30.28 | rogersja | :leifmadsen just curious why your sip.conf examples in the book show qualify=no ? |
| 20:32.10 | leifmadsen | rogersja: I just turn it off typically to reduce traffic and such |
| 20:32.22 | leifmadsen | session-timers should really be used for what qualify used to be used for |
| 20:33.26 | rogersja | i see |
| 20:34.19 | rogersja | leifmadsen: do you go into session timers anywhere? |
| 20:34.50 | leifmadsen | I don't think we discussed that really |
| 20:34.53 | leifmadsen | maybe briefly... |
| 20:34.57 | leifmadsen | something to add to the next edition |
| 20:36.27 | gruvfunk | greetz, anyone in here experience problem with Call Monitor files in ARI not showing the icon to press and play, not showing check box to flag for deletion, select all greyed out, etc? (I've come a long way in 2 days to get the Call Monitor files to even show up, now I'd just like to play them) |
| 20:36.44 | gruvfunk | and yes, i'm over on #freepbx too |
| 20:37.23 | Kobaz | so far so good on 1.8.3.3 |
| 20:37.38 | rogersja | leifmadsen: next edition!! published by Oct/11 :P |
| 20:37.51 | Kobaz | leifmadsen: no crashes yet :) |
| 20:38.31 | *** join/#asterisk ibercom (d9d85207@gateway/web/freenode/ip.217.216.82.7) |
| 20:40.32 | leifmadsen | Kobaz: :) |
| 20:40.38 | leifmadsen | rogersja: heh, I doubt it ;) |
| 20:41.16 | gruvfunk | My personal PBX is also flawlessly running (only glitch is mp3 playback) : Asterisk 1.8.3.3-1digium1~lucid |
| 20:41.34 | *** join/#asterisk zkn (~zkn@82.131.28.206.cable.starman.ee) |
| 20:41.42 | Freeaqingme_ | russellb, nvm on theline ending issue, seems php is hating me |
| 20:41.49 | gruvfunk | (PBX I'm having issues with is a customer's custom-built CentOS, * 1.8.1.1, FPBX 2.8.1.4 |
| 20:42.53 | rogersja | leifmadsen: new title: The even more definitive guide then the previous one. |
| 20:42.58 | golam | Trying to find some information about fax for asterisk usingT.38. Any pointers?? |
| 20:43.04 | rogersja | ps. i want title credits for that one |
| 20:44.02 | jaytee | I know it's possible to have a SIP phone setup at another location behind a NAT'd firewall register to my * server at this location but is it possible to have more than one at a given location? Could I have 3 phones in a satellite office register to my * server? |
| 20:44.07 | rogersja | golam: take a look at ~the book |
| 20:44.31 | golam | which book |
| 20:44.40 | jaytee | ~book |
| 20:44.40 | infobot | For more information about the Asterisk book, see ~thebook |
| 20:44.47 | jaytee | ~thebook |
| 20:44.47 | 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://ofps.oreilly.com/titles/9780596517342/, or ~buybook |
| 20:45.03 | WiretapSeven | gruvfunk, in my personal experience that only happens when theyre too short to play |
| 20:45.14 | WiretapSeven | i.e. don't exist because they fall under the threshold |
| 20:45.51 | gruvfunk | WiretapSeven: 35 and 49 seconds? I'll try it longer... |
| 20:46.05 | WiretapSeven | the threshold is normally set to '3' |
| 20:46.11 | WiretapSeven | but it may have been messed with |
| 20:46.23 | gruvfunk | pointer to where the setting is ? |
| 20:46.31 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 20:47.39 | Kobaz | leifmadsen: who came up with the DAHDI/i1/<callerid>-<id> format? |
| 20:47.40 | jaytee | quittin time be back later |
| 20:47.47 | *** join/#asterisk r0d3nt (r0d3nt@foster.stonedcoder.org) |
| 20:48.29 | Kobaz | leifmadsen: i would like to give them a high five, and maybe some home made cookies |
| 20:48.31 | WiretapSeven | gruvfunk, I can't remember |
| 20:55.17 | leifmadsen | Kobaz: sorry, not sure :) |
| 20:55.23 | leifmadsen | looks at rmudgett maybe |
| 20:55.35 | leifmadsen | I think someone was complaining about that format the other day though |
| 21:02.27 | paulc | GotoIf, testing for equality... = or == ? (I seem to have both in a dialplan fragment, but it seems to work?) |
| 21:05.45 | *** part/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
| 21:13.55 | *** join/#asterisk p3nguin (~xwQ5kwYl6@cobalt.esxi.hosts.a2infotech.com) |
| 21:18.01 | leifmadsen | paulc: always just = |
| 21:18.04 | leifmadsen | == is not valid |
| 21:18.21 | leifmadsen | =, !=, >=, <=, >, < are the valid checks |
| 21:19.01 | paulc | Thanks.. figured as much having RTFM.. (and am now getting rid of my "=="'s, which may explain why some of my stats are wonky :) |
| 21:23.45 | *** join/#asterisk mr_ian (~mr_ian@S0106001b63f49383.du.shawcable.net) |
| 21:25.45 | *** join/#asterisk d_preston215 (~chatzilla@static-76-161-250-54.t1.cavtel.net) |
| 21:25.50 | d_preston215 | Is there anyway to combine a recorded voicemail message, with something that says the extension they are connected to? |
| 21:27.19 | *** join/#asterisk fauxalliance (~fauxallia@142.163.132.151) |
| 21:28.35 | WiretapSeven | d_preston215, you mean "the person at extensions one, two, three, is not available" |
| 21:29.01 | d_preston215 | Yeah. |
| 21:29.13 | p3nguin | core show application VoiceMail |
| 21:29.18 | p3nguin | See option u. |
| 21:29.48 | d_preston215 | Or like "You have reached extension XXX". After that, the recorded voicemail message plays. |
| 21:30.06 | p3nguin | It's your extension, so you can make it do anything you want it to do. |
| 21:30.55 | p3nguin | If you don't know already, it's done in extensions.conf. |
| 21:31.10 | d_preston215 | Via dialplans. |
| 21:31.35 | p3nguin | That's what extensions.conf contains. |
| 21:33.02 | d_preston215 | Is there a specific dialplan I should be looking at? |
| 21:33.41 | WiretapSeven | d_preston215, have you considered something like freepbx so you don't have to worry about config files? |
| 21:33.43 | p3nguin | Why do I get the feeling that you didn't write your own dial plan? |
| 21:35.01 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 21:36.12 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 21:37.13 | *** join/#asterisk jpsharp (jsharp@ohno.mrbill.net) |
| 21:38.09 | jpsharp | Question: I have a couple of queues set up, but I have agents who are leaving their phones off hook to skip getting calls. How can I catch them? |
| 21:39.04 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v013-117.mobile.uci.edu) |
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| 21:40.04 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
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| 21:40.52 | WiretapSeven | jpsharp, most callcentres fix this problem I believe by taking away the handset completely and configuring the phone to autoanswer and use headset |
| 21:41.31 | leifmadsen | or make them login to chan_agent so when they are logged in, they are always online |
| 21:41.37 | leifmadsen | then the system just connects caller to the agent |
| 21:41.47 | leifmadsen | if the agent "hangs up" then they won't appear as available |
| 21:41.57 | p3nguin | When the login that way, the agent listens to moh while waiting for calls to come in. |
| 21:42.03 | p3nguin | s/the/they/ |
| 21:42.23 | leifmadsen | p3nguin: unless you assign them to a MoH that simply plays back silence |
| 21:42.35 | leifmadsen | that's not really a problem that can't be worked around if your agents don't want that |
| 21:42.35 | p3nguin | well, it's still moh... |
| 21:42.39 | leifmadsen | so? |
| 21:42.39 | p3nguin | but yes, that's correct. |
| 21:42.45 | *** join/#asterisk seraphie (~erin@207.98.195.107) |
| 21:42.45 | leifmadsen | I don't see the problem |
| 21:42.57 | p3nguin | Who said there's a problem? |
| 21:43.06 | leifmadsen | you seemed to imply it |
| 21:43.11 | p3nguin | If they are not listening to the headset or speakerphone, they'll miss the calls. You'll surely be able to figure out who's missing calls. |
| 21:43.31 | jpsharp | My client is using ATA186's + POTS phones for the Agents and he's using fixed SIP ids in queue.conf, rather than Agent login-based. |
| 21:43.38 | _Corey_ | I usually turn that off when I use AgentLogin |
| 21:43.38 | *** join/#asterisk hairyraven (~nobody@95.73.200.3) |
| 21:44.30 | _Corey_ | Auto-answer is best in high-volume call centers, but in low-volume situations it's probably better to actually ring the agent |
| 21:45.04 | _Corey_ | if they're getting two calls an hour, you're much more likely to get them talking to their neighbor than greeting the caller |
| 21:45.24 | jpsharp | There's 9 agents, so it is by far not a high volume operation. |
| 21:45.45 | _Corey_ | It's all about call volume |
| 21:45.54 | _Corey_ | 9 agents can be fully utilized |
| 21:46.00 | p3nguin | Nine agents could still handle high volume. |
| 21:46.37 | jpsharp | True. Let me rephrase, then. At any given time, there's 2-3 active calls in the queue with 9 active agents. |
| 21:47.06 | jpsharp | And at night, there's 10-15 minute gaps between calls. |
| 21:47.07 | p3nguin | So you've got like 12 calls at any given moment? |
| 21:47.29 | jpsharp | No, 2-3. |
| 21:47.40 | _Corey_ | jpsharp: To answer your original question... I'd probably nail them with logs |
| 21:48.10 | p3nguin | With that low of volume, it wouldn't be hard to check CDR to see what is going on. |
| 21:48.13 | jpsharp | I'll have to load one of the Queue log parsers and see what it spits out. |
| 21:48.39 | _Corey_ | Yeah, the queue log stuff may be overkill but Queuemetrics is pretty nice |
| 21:48.41 | cj | have any of you used the RADIUS CDR bits yet? |
| 21:48.49 | p3nguin | Why would people be trying to skip out on calls when they don't get that many calls in the first place? |
| 21:48.56 | cj | er, Diameter, not RADIUS |
| 21:49.07 | cj | p3nguin: because they're hung over? |
| 21:49.15 | p3nguin | Don't ask me. |
| 21:49.22 | p3nguin | I'm the one asking why. |
| 21:49.33 | jpsharp | they're slackers. |
| 21:49.44 | _Corey_ | Trying to figure out what motivates call center agents to be lazy is like trying to figure out why we have four seasons |
| 21:49.52 | p3nguin | Are you going to reprimand them when you catch them? |
| 21:50.15 | sxpert | jpsharp: underpaid slackers ? |
| 21:50.17 | jpsharp | I can't directly, I'm just the Asterisk contractor guy. |
| 21:50.29 | Kobaz | awwww |
| 21:50.31 | p3nguin | What I would do in that situation is send more calls to them. |
| 21:50.43 | Kobaz | the moh corruption bug is in 1.8.3.3 also |
| 21:50.45 | Kobaz | damnit |
| 21:50.48 | p3nguin | Give the slackers something to do. |
| 21:50.54 | _Corey_ | lol |
| 21:51.15 | drmessano | Set lower penalties for them |
| 21:51.24 | drmessano | MARK THEM ZERO |
| 21:51.42 | p3nguin | That'll just send the calls elsewhere when they've left their phone off-hook. |
| 21:51.55 | drmessano | Electro-shock ? |
| 21:52.11 | p3nguin | That could be effective, when administered properly. |
| 21:52.14 | _Corey_ | generally a nasty supervisor is the best approach |
| 21:52.36 | drmessano | Metrics system that administers known biotoxins in their immediate area in the event of poor performance? |
| 21:52.46 | _Corey_ | although every customer i've proposed an integrated cattle-prod has been receptive to the idea |
| 21:53.14 | _Corey_ | i'd say there's a market opportunity if we take some initiative on the agent electro-motivation |
| 21:53.15 | drmessano | Yeah, maybe biotoxins are taking it too far. I should put those back in the shed |
| 21:54.18 | jpsharp | bluetooth-enabled shock collars. |
| 21:54.32 | drmessano | chan_immobile? |
| 21:54.34 | drmessano | I love it |
| 21:54.38 | jpsharp | yes. |
| 21:54.45 | _Corey_ | i was thinking chair restraints |
| 21:55.07 | drmessano | Asterisk already has bluetooth support.. Just need to integrate the shock collars |
| 21:55.11 | p3nguin | Could one really take punitive measures TOO FAR? I'm not so sure. |
| 21:55.21 | _Corey_ | p3nguin: NEVER |
| 21:55.28 | _Corey_ | there is no 'too far' |
| 21:55.37 | drmessano | p3nguin, in this economy, you can get away with a lot.. People want their jobs and I love to hear screaming. |
| 21:55.44 | _Corey_ | always more agents waitingf |
| 21:55.54 | p3nguin | If the result is death, it's his or her own fault! |
| 21:55.54 | _Corey_ | haha, seriously |
| 21:56.22 | p3nguin | Just make sure they sign the waiver during the intro. |
| 21:56.28 | drmessano | "Bamboo under the fingernails? Really?" "You want to keep your job, right?" "Yeah yeah, fine" |
| 21:56.39 | _Corey_ | i can imaging a soothing allison recording warning the agent that the shock is coming |
| 21:56.44 | _Corey_ | s/imaging/imagine/ |
| 21:56.52 | drmessano | Hell yes |
| 21:57.01 | _Corey_ | very HAL9000 |
| 21:57.20 | drmessano | "Put down your drink, here comes Mr. Prickly" |
| 21:57.26 | drmessano | BZZZZAP |
| 21:58.46 | drmessano | Of course, there's something to be said for just screwing with them |
| 21:59.04 | _Corey_ | that gets old fast, they're usually not very bright |
| 21:59.12 | drmessano | "Why is there a fire extinguisher next to my desk.. and all the desks?" "In case your pants catch on fire when we shock your chair" |
| 21:59.34 | _Corey_ | well, that may work :) |
| 21:59.45 | *** join/#asterisk mr_ian (~mr_ian@S0106001b63f49383.du.shawcable.net) |
| 22:00.14 | drmessano | "Why is there a wallet on my desk?" "Seizure risk when you get above 10,000 volts. Stay busy and it wont be a problem" |
| 22:02.27 | drmessano | No different than putting a folded up cardboard box in all their offices... "Saves time in case we fire you" |
| 22:03.13 | drmessano | This is why my boss won't let me interact with the staff directly anymore... |
| 22:03.14 | drmessano | :( |
| 22:03.20 | _Corey_ | haha |
| 22:05.29 | _Corey_ | All conversations about agent productivity usually go this direction eventually |
| 22:05.39 | *** join/#asterisk Defraz (~Defraz@184-155-137-27.cpe.cableone.net) |
| 22:05.52 | drmessano | When we got our automation system at the radio stations, we had joked for weeks about each PC replacing one of the jocks.. so the boxes come in and one night someone wrote one of the jocks names on each of the boxes.. I got called into the GM's office. I told him "That's funny as hell, and I wish I had done it, but it wasnt me" |
| 22:08.16 | Nugget | heh |
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| 22:23.18 | *** join/#asterisk asilva (~andre@2801:88:1000:2::18) |
| 22:23.33 | asilva | downloads.digium.com - DOWN ??? |
| 22:24.39 | asilva | asterisk.org also down |
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| 22:30.29 | pabelanger | yes |
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| 22:36.56 | *** join/#asterisk nighty^ (~nighty@x122091.ppp.asahi-net.or.jp) |
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| 22:47.49 | Katty | hai |
| 22:53.59 | p3nguin | What's for supper? |
| 22:57.40 | *** join/#asterisk lost_soul (~noymfb@cpe-74-78-191-114.twcny.res.rr.com) |
| 22:58.29 | rogersja | asilva: downloads.digium.com should be back up |
| 22:58.33 | rogersja | i can access it |
| 22:59.36 | p3nguin | It's too bad that asilva will never know about this. |
| 23:00.10 | rogersja | oh well, his fault, he gave up to quick |
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| 23:38.00 | cj | http://paste2.org/p/1388043 |
| 23:38.03 | cj | anyone familiar with this? |
| 23:38.11 | cj | avaya/nortel phones sending 305 to asterisk |
| 23:38.22 | *** join/#asterisk logicwrath (~no@68.62.24.205) |
| 23:38.44 | logicwrath | does #asterisk really need +r its kind of annoying when net splits and my script doesnt identify |
| 23:48.22 | p3nguin | What if you identify using a different method? |
| 23:50.21 | logicwrath | example? |
| 23:51.07 | p3nguin | Are you reconnecting to the server or something which is removing your identification? |
| 23:52.03 | p3nguin | I haven't seen anyone split from here recently, so I don't know where your problem actually lies. |
| 23:52.06 | logicwrath | yes, either my connection fails briefly or freenode splits briefly |
| 23:52.27 | logicwrath | every day i come back to freenode unidentified |
| 23:52.38 | p3nguin | If you are reconnecting to the server, send your password as the server password. |
| 23:53.08 | p3nguin | If you send your password as the server password, you should be identified immediately. |
| 23:53.31 | p3nguin | And since your client obviously knows how to reconnect, this won't be a scripting issue. |
| 23:54.03 | logicwrath | im not sure what you mean "server password" |
| 23:54.15 | logicwrath | how does that differ from /ns identify |
| 23:55.13 | p3nguin | When you input the network and/or server information into your IRC client, there's a field to supply a password. Put your nick password there. When the client connects to the server, you'll automatically be identified. |
| 23:55.25 | logicwrath | hmm ill check for that |
| 23:56.07 | logicwrath | i usually connect by hand /server irc.freenode.net, then my client remembers my last server and will reconnect automatically |
| 23:56.36 | p3nguin | Maybe /help server or /server help would reveal the syntax required to send a password. |
| 23:58.04 | p3nguin | In irssi, I would use something like /connect -ssl -network freenode irc.freenode.net 7000 mysecretpassword |
| 23:58.16 | logicwrath | i found a spot to send a command after joining a network, and i input the identify command there. i am running a mirc script that auto identify's for me when i connect, the problem is when im already connected something happens (ping timeout) or something, and when i get reconnected the mirc script doesnt identify for some reason... i just checked the log, and my nick is already in use, so |
| 23:58.16 | logicwrath | i must be timing out |
| 23:59.02 | logicwrath | i like reviewing the chat log for the day and i always lose that when i dont get rejoined |
| 23:59.27 | pabelanger | logicwrath: run an IRC proxy; bip |
| 23:59.53 | pabelanger | then you connect and disconnect directly to your proxy |