| 00:04.02 | *** join/#asterisk jstapleton (~jstapleto@c-24-125-171-223.hsd1.va.comcast.net) |
| 00:06.11 | *** join/#asterisk micols (~mio@rlogin.dk) |
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| 00:13.52 | jstapleton | anyone know where I can get SIP firmware for Cisco 7975? |
| 00:18.23 | *** join/#asterisk guilhermebr (~Guilherme@189.63.65.56) |
| 00:19.28 | *** join/#asterisk Qwell (~north@pdpc/sponsor/digium/Qwell) |
| 00:19.29 | *** mode/#asterisk [+o Qwell] by ChanServ |
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| 00:33.22 | xheliox | From Cisco? :P |
| 00:33.45 | *** join/#asterisk mathslinux (~mathslinu@120.42.46.126) |
| 00:39.22 | *** join/#asterisk [intra]lanman (~lanman@freeswitch/developer/intralanman) |
| 00:43.37 | ldiamond | Im looking for a VOIP (SIP) provider with support for SMS. Anybody knows any? |
| 00:43.58 | ldiamond | With DIDs in Canada (if only google voice was available here!) |
| 00:52.45 | *** join/#asterisk boodu (~antoine@host-115-126-167-240.adsl.nautile.nc) |
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| 01:08.27 | ACK-NAK | ~infobot |
| 01:08.28 | infobot | ack-nak, i love abuse, feed me!, or whack, yo |
| 01:08.54 | ACK-NAK | ~infobot |
| 01:08.55 | infobot | ack-nak, i love abuse, feed me!, or whack, yo |
| 01:09.06 | ACK-NAK | ~options |
| 01:09.27 | ACK-NAK | ;-) |
| 01:12.25 | *** join/#asterisk Beirdo (~gjhurlbu@unaffiliated/beirdo) |
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| 01:50.55 | carrar | Y*A*W*N |
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| 02:09.29 | xheliox | no yawning. |
| 02:11.35 | russellb | yawns in the general direction of xheliox |
| 02:12.28 | xheliox | well that's just very contrarian. |
| 02:15.10 | ChannelZ | what about knawing? |
| 02:15.29 | russellb | no! |
| 02:16.48 | xheliox | No bodily functions, whatsoever. |
| 02:16.50 | xheliox | Breathing only! |
| 02:17.07 | russellb | ~roulette |
| 02:17.08 | infobot | ACTION watches russellb pull the trigger: BANG! |
| 02:17.08 | xheliox | No blinking, coughing, sneezing, yawning, farting, queefing (sp?), or any variation there of. |
| 02:17.11 | russellb | :-( |
| 02:17.23 | xheliox | points and laughs at russellb's grey matter |
| 02:17.56 | xheliox | ~roulette |
| 02:17.57 | infobot | ACTION watches xheliox pull the trigger: Click! |
| 02:18.02 | xheliox | winks |
| 02:26.01 | ldiamond | Anyone know of a provider that supports SMS? |
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| 03:18.37 | brettnem | hey, has anyone seen in 1.6 where calls to a queue with a strat of linear or rrmemory doesn't ring anything, but ringall works just fine? :/ |
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| 03:57.50 | dmz | anyone here use blind transfer? every time i try to use it, i hit "#" then any other # immediatly puts the caller into voicemail, any suggestions? |
| 03:59.27 | ChannelZ | hmm I do but as a SIP feature, not dtmf. |
| 03:59.53 | ChannelZ | What does the console say is happening? |
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| 04:12.04 | xheliox | Blue. |
| 04:12.37 | dmz | nothing really |
| 04:12.38 | dmz | i'll debug a bit more later |
| 04:12.38 | dmz | gotta get a few thigns done before i crash, thought i'd see if anyone ran into issues w/it |
| 04:14.32 | p3nguin | Any reason you want to do DTMF transfers? |
| 04:20.04 | xheliox | p3nguin: one would presume a lack of a xfer button :P |
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| 05:12.22 | chuckp | ?SendFax |
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| 05:44.44 | daltio | guys i want to integrate asteriks with seimens high path 3000 ip enabled pbx |
| 05:44.46 | daltio | any idea |
| 05:44.46 | *** join/#asterisk Faustov (user@gentoo/user/faustov) |
| 05:44.52 | daltio | it supports sip protocol |
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| 05:54.01 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
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| 06:19.47 | daltio | guys |
| 06:19.51 | daltio | i am stuck any idea |
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| 06:35.36 | ChannelZ | well if it does SIP, hook them up and go.. |
| 06:35.48 | ChannelZ | you haven't asked a specific question |
| 06:48.01 | pepselap | i can't tell why my cid insists on showing my extension as the number :/ |
| 06:50.48 | pepselap | damn, i thought it might've been my ATA. |
| 06:50.59 | pepselap | but blanking out the phone number field did nothing :/ |
| 06:55.14 | ChannelZ | You can implicitly set the CID in sip.conf |
| 06:55.51 | ChannelZ | The ATA (assuming it's an ATA->SIP box like a SPA3102 et al) should be able to provide its own |
| 06:55.56 | pepselap | wouldn't i set the CID of an incoming call in the extensions.conf? |
| 06:56.22 | pepselap | incoming calls show up as the extension of the phone for the number.. the name shows up as expecteed |
| 06:56.27 | pepselap | expected |
| 06:56.34 | pepselap | oh.. wait, maybe it doesn't |
| 06:57.07 | pepselap | can you still Set(CALLERID(all)=Etc <etc>) in 1.6? |
| 06:57.55 | pepselap | either way, looks like i'm doing (num)= seeing that all failed.. exten => dj_pepse,n(allow),Set(CALLERID(num)=<${CALLERID(num):2}>) |
| 07:01.24 | *** join/#asterisk stix (~stix@exchange2003.corporate.billetkontoret.dk) |
| 07:03.31 | pepselap | it was showing the <>'s in the name. |
| 07:03.49 | pepselap | apparently set(callerid()) might be different in this version. |
| 07:04.29 | *** join/#asterisk sawgood (~sawgood@173-13-158-25-sfba.hfc.comcastbusiness.net) |
| 07:05.48 | pepselap | oh i have to put a comma now. |
| 07:05.56 | ChannelZ | You generally set the CID of the device, which sets what it *sends* to others... you receive what people send. |
| 07:06.06 | *** join/#asterisk Faithful (~Faithful@202.6.145.116) |
| 07:07.03 | pepselap | channelz: right, but in my case the extension is getting inserted in there at some point |
| 07:07.25 | pepselap | also i wanted to knock out the +1 that i get |
| 07:07.35 | pepselap | because I can't dial +1 back. |
| 07:08.13 | ChannelZ | well I can't read minds, I have no idea what your configs are and you haven't shown them |
| 07:10.01 | pepselap | feh. that comma example is bullpucky. |
| 07:10.52 | ChannelZ | you don't put <> around the number |
| 07:11.04 | ChannelZ | Set(CALLERID(num)=1234567890) |
| 07:11.12 | ChannelZ | Set(CALLERID(name)=Some Fool) |
| 07:11.15 | pepselap | what if you want to set all? |
| 07:11.36 | pepselap | Set(CALLERID(all)=${CALLERID(num):2} <${CALLERID(num):2}>) |
| 07:11.40 | pepselap | no bueno? |
| 07:11.57 | ChannelZ | that's fine but you were setting only num before |
| 07:12.10 | pepselap | indeed. and somehow it was ending up as the name : |
| 07:12.12 | pepselap | err :) |
| 07:13.11 | pepselap | damn. it gets me what i want for the name, but the number is still the extension number |
| 07:13.45 | pepselap | now, i think someone said something about 'fromuser=' in sip.conf.. |
| 07:14.12 | pepselap | but without fromuser, authuser, and username, i can't get it to ring |
| 07:14.29 | pepselap | for the extension |
| 07:21.49 | *** join/#asterisk Netgeeks (~chris@gw1.netgeeks.net) |
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| 07:23.52 | *** mode/#asterisk [+o Qwell] by ChanServ |
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| 07:25.11 | *** join/#asterisk UQlev (~yuriy@212.50.100.76) |
| 07:26.28 | ChannelZ | it's "callerid=xxxx <###>" |
| 07:26.34 | *** join/#asterisk sgimeno (~chatzilla@226.Red-80-33-64.staticIP.rima-tde.net) |
| 07:27.01 | pepselap | oh without (all)? |
| 07:27.12 | ChannelZ | in the config yes.. read the sample config |
| 07:27.23 | ChannelZ | Who are you calling and from where? |
| 07:28.12 | sawgood | what is the 'difference' between dialing with /Local vs. /SIP |
| 07:28.19 | sawgood | what does /Local 'mean'? |
| 07:28.20 | pepselap | without (all) it doesn't work at all |
| 07:28.42 | ChannelZ | Local is a virtual channel that lets you feed back into the dialplan |
| 07:28.46 | ChannelZ | SIP is a SIP channel |
| 07:28.52 | pepselap | I'm calling from my cellphone to my *. My cellphone number shows up as name, the extension of the phone I'm looking at shows up as the number |
| 07:29.23 | sawgood | When I try to call out to my cell phones (as virtual extensions) ... setting them up requires I use the command, " Local/95551212 |
| 07:29.29 | sawgood | I had not seen that before |
| 07:29.46 | ChannelZ | your number should show up as the number and your name should be nothing (from a cell phone in the US anyway). RE: we have no idea what your config and/or dialplan are doing |
| 07:29.57 | *** join/#asterisk SiNGLer (~singler@81-7-123-162.static.zebra.lt) |
| 07:30.18 | *** join/#asterisk gr0mit (~tim@extrt.txrx.org.uk) |
| 07:30.29 | pepselap | yes, it most definitely should |
| 07:30.42 | ChannelZ | If you dial Local/95551212 that is basically acting as if you picked up a phone and dialed 95551212 into the dialplan.. so it then jumps to whatever extension matches that in the same context, and does whatever that is supposed to do. |
| 07:30.47 | pepselap | my SIP provider doesn't pass name CID anyway |
| 07:30.53 | pepselap | hence why i'm setting the name to the number too |
| 07:31.06 | ChannelZ | So ultimately you are (probably) dialing as SIP/something but you're taking the long way around |
| 07:31.07 | sawgood | ChannelZ: Thank you ... nice explaination! |
| 07:32.37 | pepselap | does fromuser= in sip.conf do something like this? |
| 07:32.53 | pepselap | under the extension .."section" |
| 07:32.55 | ChannelZ | actually nix the 'no name from a cell phone', my brain is not working. Cell phones don't do CNAM lookups. |
| 07:33.05 | ChannelZ | fromuser has nothing to do with caller ID |
| 07:33.13 | ChannelZ | you're fucking up your authentication/peer matching |
| 07:33.23 | pepselap | how? |
| 07:33.29 | pepselap | err.. wat? |
| 07:34.03 | ChannelZ | I'm done trying to help you on this since it's all shots in the dark since you haven't shown any configs, console output as to what is happening on an incoming call, etc. |
| 07:34.49 | pepselap | .. what would you like to know? |
| 07:35.18 | ChannelZ | nothing, I'm going to bed. |
| 07:37.13 | [TK]D-Fender | [03:33]<ChannelZ>fromuser has nothing to do with caller ID <- it does |
| 07:37.32 | pepselap | d-fender: essplain por favor |
| 07:37.47 | *** join/#asterisk Benwa (~Benwa@cable-78.29.254.179.coditel.net) |
| 07:38.14 | [TK]D-Fender | pepselap: What is on the other side of this peer? |
| 07:38.39 | pepselap | hm, not sure what you mean by other side. SIP provider -> * -> ATA -> pos phone |
| 07:38.58 | [TK]D-Fender | pepselap: Too big a pickture isolate the side that is screwing up |
| 07:39.41 | pepselap | it's definitely at *, and what's funny is when the call comes in (i'm at verbosity 3), it shows the line setting the callerid with the expected numbers |
| 07:39.58 | pepselap | but then when the extension rings, it's the extension's number in there instead |
| 07:40.00 | [TK]D-Fender | pepselap: Wrong answer, try again |
| 07:40.12 | sawgood | pepselap: in a nutshell ... are you trying to have Caller ID show on an incoming or an outgoing call? |
| 07:40.22 | pepselap | sawgood: incoming call |
| 07:40.27 | [TK]D-Fender | pepselap: Your PHONE's peer should NOT have "fromuser=" at all |
| 07:40.40 | pepselap | if i comment it out, the phone doesn't ring |
| 07:40.47 | sawgood | pepselap: so you want caller ID name (CNAM) to arrive on incoming calls over a SIP trunk? |
| 07:40.52 | [TK]D-Fender | pepselap: pastebin the peer |
| 07:41.10 | pepselap | sawgood: no |
| 07:41.57 | pepselap | http://pastebin.ca/1893285 |
| 07:42.08 | sawgood | Well, if you want CNAM (name not number) to show up on an incoming call ... what type of line is it that the call arrives to the Asterisk box with? |
| 07:42.11 | pepselap | sawgood: i want the number to come up as the number :) |
| 07:42.34 | pepselap | sawgood: the number is instead showing up as the extension number, not the caller's number |
| 07:42.42 | sawgood | So, caller ID number (not name) on incoming calls to an Asterisk box via a SIP trunk? |
| 07:43.23 | pepselap | yes. the number is coming in correctly from the sip provider, that part's fine |
| 07:43.54 | pepselap | i have no problems sending or receiving CID outside of * |
| 07:44.15 | [TK]D-Fender | pepselap: username=101 |
| 07:44.17 | [TK]D-Fender | fromuser=101 |
| 07:44.18 | sawgood | So, your Asterisk box is 'stripping' the Caller ID number from the incoming call and putting the extension number of the other caller's extension in its place? |
| 07:44.19 | [TK]D-Fender | authuser=101 |
| 07:44.21 | [TK]D-Fender | Trash these |
| 07:44.27 | pepselap | if i comment them out, the phone doesn't ring |
| 07:44.40 | [TK]D-Fender | pepselap: and do "type=peer" |
| 07:44.43 | SiNGLer | pepselap: remove fromuser, this replaces your callerid |
| 07:44.52 | pepselap | sawgood: putting the extension number of the -ringing- extension in its place |
| 07:44.54 | [TK]D-Fender | pepselap: Do these and pastebin a call with SIP DEBUG <- |
| 07:45.04 | pepselap | d-fender: i'll try peer. |
| 07:46.03 | pepselap | heh, so peer made the set(callerid() not do anything. |
| 07:46.06 | pepselap | lemme turn on sip debug |
| 07:46.19 | *** join/#asterisk spenguin[work] (~penguin@122.182.0.38) |
| 07:46.21 | pepselap | (even tho in the console it showed the set callerid as expected) |
| 07:46.25 | spenguin[work] | hey |
| 07:46.26 | spenguin[work] | test |
| 07:47.01 | spenguin[work] | if I run asterisk as root, how do I make sure it can access the dahdi device nodes? |
| 07:47.05 | spenguin[work] | err |
| 07:47.08 | pepselap | what level debug? |
| 07:47.10 | spenguin[work] | asterisk as a normal user |
| 07:47.20 | SiNGLer | pepselap: Set works as expected, your callerid gets "replaced" later |
| 07:47.41 | pepselap | ah |
| 07:47.53 | pepselap | how do I stop it from getting replaced? hehe |
| 07:48.02 | [TK]D-Fender | .... |
| 07:48.05 | [TK]D-Fender | PASTEBIN |
| 07:48.09 | pepselap | what level debug |
| 07:48.24 | SiNGLer | (10:44:39) SiNGLer: pepselap: remove fromuser, this replaces your callerid |
| 07:48.52 | *** join/#asterisk Tim_Toady (~moi@178.128.16.115.dsl.dyn.forthnet.gr) |
| 07:49.08 | pepselap | singler: i did that.. with type=friend it wouldn't ring, with type=peer it now completely ignores what set(callerid did |
| 07:49.32 | pepselap | the name shows up as +1number and the number shows UNAVAILABLE |
| 07:50.00 | SiNGLer | pepselap: then go with [TK]D-Fender's sugestion |
| 07:50.13 | SiNGLer | and pastebin dialplan |
| 07:50.19 | pepselap | which is to paste a call with debug, but i'm trying to figure out what debug level.. 3? :) |
| 07:50.30 | SiNGLer | "sip set debug on" |
| 07:50.36 | [TK]D-Fender | pepselap: verbose 10, SIP DEBUG <------------- |
| 07:50.39 | SiNGLer | or something like that |
| 07:50.53 | pepselap | o i thought you meant core set debug <#> |
| 07:51.04 | [TK]D-Fender | NOT CORE |
| 07:51.43 | *** join/#asterisk NiugeS (~NiugeS@mail.cjs-solutions.com) |
| 07:53.33 | *** join/#asterisk Godfather_ (~Godfather@193.153.129.150) |
| 07:54.25 | spenguin[work] | I seem to have SUBSYSTEM="dahdi", OWNER="asterisk", GROUP="asterisk", MODE="0660" in udev rules |
| 07:54.33 | spenguin[work] | still asterisk couldnt access the device |
| 07:54.44 | spenguin[work] | had to chwon it manually |
| 07:58.13 | pepselap | d-fender: http://pastebin.ca/1893294 |
| 07:58.14 | *** join/#asterisk sulex (~sulex@office.blindata.ch) |
| 07:58.59 | *** join/#asterisk Elwell (~elwell@freenode/staff/elwell) |
| 08:00.26 | Elwell | hi folks - how can I get my dialplan to accept + and substitue 00 (ie so I can dial saved contacts on wiifone that are in the format +33... |
| 08:00.54 | [TK]D-Fender | not evena complete call... ok, not wasting time fighting for decent debug. I'm off |
| 08:01.01 | Elwell | (it's hard to google for '+' |
| 08:02.17 | SiNGLer | Elwell: check voip-info.org for string replacement |
| 08:02.22 | pepselap | elwell: exten => _+whatever,X,Dial(SIP/peer/00${EXTEN:1}) |
| 08:05.44 | *** join/#asterisk sekil (~sekil@80.93.247.26) |
| 08:06.02 | Elwell | ta |
| 08:06.04 | pepselap | singler: no idea what might be causing callerid to get effed up? or at least not sent the way it shows up in the console.. |
| 08:07.11 | SiNGLer | pepselap: show your dialplan, and show console output (with and without "fromuser") |
| 08:07.56 | pepselap | singler: checkit, [Jul 2 01:10:38] -- Executing [dj_pepse@frombeer:2] Set("SIP/15202701872-08dec798", "CALLERID=6026784450 <6026784450>") in new stack |
| 08:08.09 | pepselap | singler: so you'd think that the phone would show what it says right there |
| 08:08.25 | pepselap | instead it says +16026784450 for name and UNAVAILABLE for number |
| 08:09.25 | pepselap | when you say dialplan, do you mean extensions.conf? |
| 08:09.26 | SiNGLer | try Set(CALLERID(num)=6026784450) |
| 08:09.32 | pepselap | yeh, i tried that too |
| 08:09.47 | SiNGLer | pepselap: yes, extensions.conf |
| 08:09.51 | SiNGLer | and use pastebin :) |
| 08:10.13 | *** join/#asterisk darksk1ez (~mhb@darkskiez-1-pt.tunnel.tserv5.lon1.ipv6.he.net) |
| 08:10.14 | pepselap | Set(CALLERID(num)=${CALLERID(num):2} |
| 08:10.21 | SiNGLer | did you tried my suggestion with removed fromuser? |
| 08:10.34 | *** join/#asterisk sulex (~sulex@office.blindata.ch) |
| 08:10.37 | pepselap | yeh, all of three are gone now and i have to use type=peer |
| 08:10.52 | pepselap | with type=friend, the phone doesn't ring without fromuser, authuser, and username |
| 08:11.12 | *** join/#asterisk Da-Geek (~Da-Geek@nat/redhat/x-nscnqlwujpnrqicz) |
| 08:11.24 | SiNGLer | pastebin your dialplan |
| 08:11.28 | pepselap | so, my extensions.conf is spread out to like 30 flles. gimme a sec to gather everything relevant |
| 08:11.40 | pepselap | i use #include :D |
| 08:11.55 | SiNGLer | paste just from Set to dial |
| 08:14.02 | pepselap | singler: there's nothing between them :) that's two lines. |
| 08:14.10 | SiNGLer | paste them then |
| 08:14.24 | pepselap | that line i pasted earlier then Dial(SIP/101) |
| 08:14.25 | pepselap | that's it |
| 08:18.02 | SiNGLer | with fromuser removed try http://pastebin.ca/1893299 |
| 08:18.16 | SiNGLer | and show console output |
| 08:18.39 | pepselap | k, sec. btw, http://pastebin.ca/1893300 |
| 08:20.51 | pepselap | singler: word, that has worked, sir |
| 08:21.05 | pepselap | i wonder if switching back to friend will yield the same result. |
| 08:21.22 | SiNGLer | if you will keep removed fromuser, it should work |
| 08:22.36 | pepselap | yeh, without that it won't ring at all. o well. type=peer it is |
| 08:22.49 | pepselap | weird that setting callerid(all) doesn't work with that though |
| 08:23.00 | pepselap | FREAKIN OSS! :D |
| 08:23.28 | SiNGLer | :) |
| 08:23.39 | *** join/#asterisk wonderworld (~ww@ip-62-143-22-226.unitymediagroup.de) |
| 08:23.57 | pepselap | you don't know either? :) |
| 08:24.06 | pepselap | why, that is |
| 08:24.10 | SiNGLer | never had such problem |
| 08:24.33 | pepselap | mebbe it's a bug |
| 08:26.48 | *** join/#asterisk Polysics (~luca@ip-192-59.sn3.eutelia.it) |
| 08:27.07 | *** join/#asterisk Faithful (~Faithful@202.6.145.116) |
| 08:27.20 | Polysics | hello |
| 08:27.35 | Polysics | how can i handle calling out to cellphones, please? |
| 08:27.53 | Polysics | i mean, i have a list of phones to call |
| 08:28.04 | Polysics | if first one does not answer, i need to call the second after X time |
| 08:28.05 | pepselap | singler: thanks much for the excellent suggestion. |
| 08:28.14 | SiNGLer | pepselap: np |
| 08:28.24 | Polysics | am i right that if the called cellphone has voicemail, i am basically out of luck? |
| 08:28.34 | Polysics | that is, it will count as a "connected" call |
| 08:28.56 | Polysics | is there any way to detect the call trasnfer that a carrier does when it send you to the voicemail? |
| 08:29.33 | mboehn | Polysics: Looked at http://www.voip-info.org/wiki/view/Asterisk+cmd+FollowMe ? |
| 08:31.03 | pepselap | comeandFollowMe..towhereverthelightbreaksthrough.. |
| 08:32.22 | Polysics | mboehn, looks like it's what i need :-) |
| 08:32.36 | Polysics | now i just have to find out how to change the message |
| 08:33.05 | Polysics | problem is, i need a followme.conf entry for each user |
| 08:33.46 | Polysics | this should be used for emergency calls to licensed translators, and the whole system is db-based and in a constant state of change |
| 08:34.03 | Polysics | i suppose there is no way to make it dynamic |
| 08:34.20 | Polysics | i wonder if you can script the "call, then ask for a number press" by hand |
| 08:34.21 | SiNGLer | Polysics: check, maybe it supports realtime |
| 08:35.06 | Polysics | the wiki does have a follow-meexample without follow-me |
| 08:35.23 | Polysics | and realtime is not mentioned in the wiki page |
| 08:36.25 | *** join/#asterisk Lecherke (~lecherke@62.112.212.211) |
| 08:36.36 | *** join/#asterisk stix (~stix@exchange2003.corporate.billetkontoret.dk) |
| 08:36.54 | Lecherke | hi all |
| 08:44.58 | Polysics | i can call an AGI from a macro, correct? |
| 08:45.20 | *** join/#asterisk CoderForLife (~Miranda@cpe-174-101-155-51.cinci.res.rr.com) |
| 08:45.39 | Polysics | a macro is just a part of the dialplan |
| 08:45.51 | SiNGLer | yes, you can |
| 08:46.27 | *** join/#asterisk mrchrisadams (~Adium@CPE-121-217-148-241.lnse2.cht.bigpond.net.au) |
| 08:48.08 | daltio | ChannelZ you are asking with me? |
| 08:48.08 | Polysics | ok, then i probably should run this without followme, but by using the M option of Dial - still, great pointer, thanks a lot |
| 08:48.39 | daltio | how i integrate seimens highpath 3000 with asterisk |
| 08:49.28 | NiugeS | guys... i'm looking for someone to come in and do an asterisk / asterisknow installation.. any suggestions on where I can find someone to do it? |
| 08:50.43 | Polysics | NiugeS, probably here, as long as you are in a not-so-remote area of the world .-) |
| 08:50.57 | *** join/#asterisk vrban (~chatzilla@p50997436.dip0.t-ipconnect.de) |
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| 09:13.23 | NiugeS | Polysics i'm in London |
| 09:13.27 | NiugeS | Or southeast London |
| 09:19.00 | dohd | one of those places in the world that is not completely breaking down :-) |
| 09:19.04 | dohd | unlike this country |
| 09:20.28 | NiugeS | generally speaking what are the costs for an asterisknow installation.. rough price? |
| 09:24.45 | Polysics | here in italy, totally unrelated market, i have seen anything between 500 and 2000 euros |
| 09:25.02 | Polysics | on top of hardware, that is |
| 09:31.06 | *** join/#asterisk UQlev (~yuriy@212.50.100.76) |
| 09:40.07 | NiugeS | i have the hardware already... acutally i have asterisknow installed just need someone to configure it.. |
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| 10:19.28 | cjk | i have a pri line split into several channel groups. if i do not execute a regular outbound call on each group, incoming calls stop working |
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| 11:03.26 | *** part/#asterisk c0rnoTa (~c0rnoTa@109.188.52.223) |
| 11:11.52 | *** join/#asterisk ParanoyaM (~ParanoyaM@83-145-135-95.pool.ukrtel.net) |
| 11:12.40 | ParanoyaM | hi guys, can anybody help me, all calls from h323 are rejected with reason Anonymous call from but peer is described in h323.conf |
| 11:13.41 | ParanoyaM | it looks like: setup_incoming_call: Anonymous call from '@@XX.XX.XXX.XXX' rejected |
| 11:13.55 | ParanoyaM | but XX.XX.XXX.XXX is correct IP |
| 11:16.11 | ParanoyaM | anybody? |
| 11:16.54 | *** join/#asterisk Qwell (~north@pdpc/sponsor/digium/Qwell) |
| 11:16.54 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 11:17.21 | ParanoyaM | i tried all types, peer, user, h323 |
| 11:23.53 | *** join/#asterisk Martinblr (~Miranda@61.12.17.170) |
| 11:24.33 | Martinblr | my asterisk is behind a router, through stun i am able to register softphone... some time there is two way audio.. some times there is no audio |
| 11:27.26 | *** join/#asterisk skyion (~bradc@siza.thusa.net) |
| 11:30.12 | ParanoyaM | nobody knows why calls rejected? |
| 11:30.55 | *** join/#asterisk coppice (~chatzilla@m121-203-215-144.smartone-vodafone.com) |
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| 11:37.41 | ParanoyaM | hey guys, please |
| 11:37.43 | ParanoyaM | help me |
| 11:44.14 | *** join/#asterisk KermitTheFragger (~KermitThe@118-197.bbned.dsl.internl.net) |
| 11:49.02 | Martinblr | ParanoyaM: what is your issue..? |
| 11:49.14 | ParanoyaM | all my calls are rejected |
| 11:49.22 | ParanoyaM | but peer is descrived in h323.conf |
| 11:51.57 | Martinblr | your extensions are registered rite? |
| 11:52.39 | ParanoyaM | sure |
| 11:53.01 | ParanoyaM | setup_incoming_call: Anonymous call from '@@IP.IP.IP.IP' rejected |
| 11:53.11 | ParanoyaM | it even don't try to go to extensions |
| 11:54.09 | Martinblr | can you able to view your peer extensions in asterisk console |
| 11:54.23 | Martinblr | if it unreachanble then there would be problem |
| 11:55.39 | ParanoyaM | how to check this? |
| 11:57.25 | Martinblr | normally for sip we used to check as sip show peers |
| 11:57.47 | *** join/#asterisk guilhermebr (~Guilherme@ns2.aser.com.br) |
| 11:58.45 | ParanoyaM | Martinblr: yes it is ok, if i turn on AllowAnonimous then calls are ok |
| 11:59.53 | *** join/#asterisk bn-7bc (bjarne@cl-199.osl-01.no.sixxs.net) |
| 12:00.06 | Martinblr | are you registering your extension behind any router or firewall |
| 12:00.31 | ParanoyaM | no |
| 12:00.48 | Martinblr | it means the extensions are not registered properly thats why if you enable anonymous it is working |
| 12:01.28 | ParanoyaM | if i make call from sip to sip everything perfect |
| 12:04.47 | bn-7bc | ParanoyaM: sorry for missing the start, can you tell me what the problem is? |
| 12:05.27 | ParanoyaM | i am unable to make a call from h323 to sip |
| 12:05.36 | ParanoyaM | setup_incoming_call: Anonymous call from '@@IP.IP.IP.IP' rejected |
| 12:05.46 | ParanoyaM | but when i turn on allowanonimous |
| 12:05.51 | ParanoyaM | everything ok |
| 12:07.33 | bn-7bc | ParanoyaM: hmm did yhe h232 device register? |
| 12:07.47 | ParanoyaM | no |
| 12:07.55 | ParanoyaM | just sending a call with fs and tunneling |
| 12:07.58 | bn-7bc | upp correv´ct thet to h323 |
| 12:08.42 | ParanoyaM | ? |
| 12:09.55 | *** join/#asterisk Ziaeon (~Alchemist@adsl-074-166-171-132.sip.mia.bellsouth.net) |
| 12:10.08 | bn-7bc | ParanoyaM: well that exsplains why enebeling anynymous fixed it, configure the h323 device to register abd everyting will work even without allowanonimous |
| 12:10.29 | ParanoyaM | can you explain me, how to make it to register/ |
| 12:10.37 | ParanoyaM | how to configure asterisk |
| 12:13.24 | bn-7bc | ParanoyaM: hold on need to check docs (hav not delt with h323 yet) |
| 12:14.56 | *** join/#asterisk binbash_ (~peter@ip4da5c213.direct-adsl.nl) |
| 12:20.53 | *** join/#asterisk ariel_ (~chatzilla@63.214.236.169) |
| 12:22.20 | ariel_ | Morning Folks |
| 12:25.49 | SiNGLer | evening ariel_ :) |
| 12:28.54 | *** join/#asterisk Failrar (~Failrar@5ED66E6D.cable.ziggo.nl) |
| 12:29.32 | ParanoyaM | bn-7bc: did find anything? |
| 12:30.19 | bn-7bc | ParanoyaM: not a complete examle hold on |
| 12:32.24 | *** join/#asterisk jozza (~chatzilla@BSN-77-186-224.dsl.siol.net) |
| 12:32.41 | bn-7bc | ParanoyaM: Try adding a user in h323.conf just like you do wheb adding a sip user in sip.conf and set the client up with the relevant username and password |
| 12:33.34 | bn-7bc | anyone here have any exsperence wit h232 i don't want to leaf ParanoyaM down the wrong path |
| 12:34.15 | ParanoyaM | yeap, i would be very thankful |
| 12:35.09 | *** join/#asterisk Godfather_ (~Godfather@193.153.129.150) |
| 12:36.45 | *** join/#asterisk jayvee (~jayvee@azaroth.sunriseroad.net) |
| 12:38.28 | jozza | hi, how can i create a call file on ast machine from a windows machine with php? the problem i'm having now is that the owner of the call file is different than ast so ast wont dial |
| 12:38.38 | jayvee | How can I add a user to Asterisk? Before you tell me that's a stupid question and I should RTFM, be aware that I *have* RTFM, and I've added a [my_username] section in users.conf, and I'm still getting a "Forbidden" error when I try to register Linphone with Asterisk. |
| 12:39.09 | jayvee | Or is users.conf not the right place to put users? |
| 12:39.24 | skyion | which sip client do most of you use for linux? |
| 12:41.35 | *** join/#asterisk fish-bulb (~qcstewart@nat/digium/x-enivusebzyygipzu) |
| 12:42.31 | Godfather_ | jayvee, did you read it carefully? |
| 12:43.05 | jayvee | Godfather_: I think so. |
| 12:43.30 | jayvee | Funny thing is, one of the more detailed ones I'm reading http://www.automated.it/guidetoasterisk.htm doesn't even *mention* the word 'user'. |
| 12:43.51 | jayvee | and isn't that, like, the most basic function that there would be? |
| 12:44.36 | *** part/#asterisk Lecherke (~lecherke@62.112.212.211) |
| 12:44.38 | Godfather_ | jayvee, register it in sip.conf |
| 12:44.40 | SiNGLer | jayvee: if you use SIP phone, go for sip.conf ;) |
| 12:45.15 | bn-7bc | ParanoyaM: did you try what i sayd |
| 12:45.19 | pabelanger | jayvee: drop users.conf and add it directly using sip.conf |
| 12:45.31 | ParanoyaM | not yet, one moment |
| 12:45.35 | jayvee | <jayvee> Or is users.conf not the right place to put users? |
| 12:45.40 | jayvee | and I thought that was an outlandish thought :) |
| 12:45.44 | jayvee | I'll try that. |
| 12:46.14 | pabelanger | jayvee: users.conf is a general way to add phones to asterisk. You are limited on what you can use. |
| 12:46.27 | Godfather_ | jayvee, its a right place, but usually people do it in sip.conf or iax.conf |
| 12:46.50 | pabelanger | jayvee: IE: AsteriskGUI uses it |
| 12:48.13 | jayvee | okay well I added the same lines to sip.conf, and I'm still getting forbidden. I guess I should try running it with -vvvv and see what I get. |
| 12:48.40 | jayvee | Peer 'joe' is trying to register, but not configured as host=dynamic |
| 12:49.01 | jayvee | so is that like a security thing, where I have to whitelist clients? |
| 12:50.24 | jayvee | cool, adding host=dynamic to the user works...even in users.conf! |
| 12:52.08 | *** join/#asterisk rossand (~aross@CPE485b390978ce-CM00159a025ad4.cpe.net.cable.rogers.com) |
| 12:53.45 | *** part/#asterisk rossand (~aross@CPE485b390978ce-CM00159a025ad4.cpe.net.cable.rogers.com) |
| 12:54.27 | *** join/#asterisk ariel_ (~chatzilla@63.214.236.169) |
| 13:01.53 | jayvee | So what's the status of IPv6 in Asterisk? |
| 13:02.36 | pabelanger | jayvee: on reviewboard, needs testers |
| 13:03.02 | jayvee | so there is code? :) |
| 13:04.46 | jayvee | ah yes, I see the reviewboard item |
| 13:04.52 | *** join/#asterisk viq (~viq@unaffiliated/viq) |
| 13:05.47 | *** join/#asterisk Bryanstein (bryan@shellium/admin/bryanstein) |
| 13:05.57 | pabelanger | jayvee: Yup, I have not tested it myself, plan too. I need to do some IPv6 work on my LAN first |
| 13:06.36 | jayvee | it's a mammoth amount of work to change all the code |
| 13:07.12 | *** join/#asterisk Z_God (~julius@2001:888:141f:0:221:5dff:fe2a:6806) |
| 13:07.30 | jayvee | I'm a fairly amateur coder, but I do know that it's hard — I got DikuMUD working with IPv6. |
| 13:07.43 | jayvee | come to think of it, I should publish my patch |
| 13:09.29 | pabelanger | jayvee: Just a lot of testing :) Making sure not to break IPv4 |
| 13:10.21 | jayvee | the amount of ipv6 sip support out there is surprisingly little |
| 13:10.41 | jayvee | gnome meeting used to support it, and then when it became ekiga, they removed the support :-S |
| 13:11.11 | pabelanger | o.0 |
| 13:11.13 | cjk | i have a pri line split into several channel groups. if i do not execute a regular outbound call on each group every hour , incoming calls stop working |
| 13:11.21 | pabelanger | removed IPv6 support? |
| 13:11.24 | jayvee | yeah |
| 13:11.32 | jayvee | when they moved to using the libopal library |
| 13:11.38 | *** join/#asterisk slidesinger (~slidesing@c-68-44-99-50.hsd1.nj.comcast.net) |
| 13:11.51 | jayvee | libopal theoretically supports ipv6 now, but in practice the ekiga frontend still doesn't support it |
| 13:11.53 | pabelanger | cjk: any errors from your PRI? |
| 13:12.14 | *** join/#asterisk Mhaddog (~Mhaddog@z65-50-118-232.ips.direcpath.com) |
| 13:12.16 | cjk | pabelanger, not really, it works great |
| 13:12.34 | evilbit | is it considered best practices to have a noload statement in modules.conf for anything not used... i.e. res_config_mysql.so ? |
| 13:12.44 | cjk | pabelanger, i have this on every pri that is split |
| 13:12.56 | cjk | evilbit, i would say so, but only my opinion |
| 13:13.22 | pabelanger | evilbit: ya, it helps to lower the footprint of Asterisk |
| 13:13.36 | evilbit | yeah, that's what I was thinking too. thx :-) |
| 13:14.03 | *** join/#asterisk vrban (~chatzilla@p50997436.dip0.t-ipconnect.de) |
| 13:14.30 | evilbit | skinny is for cisco phones, right? |
| 13:15.18 | evilbit | and app_minivm isn't needed if just using the standard voicemail? |
| 13:16.59 | pabelanger | cjk: Does your D-channel go out of service? How to you fix the problem? |
| 13:17.22 | cjk | pabelanger, i do originate/g1/MYTESTNUMER application echo |
| 13:17.33 | cjk | so i launch a call to a number that does not answer |
| 13:17.36 | cjk | so i dont have costs |
| 13:17.40 | cjk | and then everything works fine |
| 13:17.47 | cjk | looks a little like NAT |
| 13:17.49 | cjk | :) |
| 13:19.38 | pabelanger | cjk: I would check to see if you d-chan goes out of service |
| 13:20.41 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
| 13:20.41 | cjk | pabelanger, on 3 different pbxes on 3 different locations. I bet its either a bug or a misconfiguration on my side |
| 13:21.12 | vrban | hello, T38MaxBitRate is allways 14400 (in the re-INVTE to t38), but befoer ReciveFAX, i set FAXOPT(maxrate)=9600 . also in res_fax.conf i set maxrate=9600. but it has no effect. i this issue known? |
| 13:22.37 | *** join/#asterisk chuckf (~chuckf@ubuntu/member/chuckf) |
| 13:22.41 | vrban | iam using res_fax_digium-1.6.2.0_1.2.0-opteron_sse3_64 on Asterisk 1.6.2.6 |
| 13:29.45 | *** join/#asterisk g_r_eek (~g_r_eek@ppp-94-67-87-226.home.otenet.gr) |
| 13:30.41 | pabelanger | vrban: Contact digium for support, since you can :) |
| 13:32.53 | vrban | i allready done, I thought that it might be faster here |
| 13:34.17 | pabelanger | vrban: Maybe, I haven't used it before. |
| 13:37.13 | vrban | i going to test what happen when i hard code it to 9600 |
| 13:37.17 | vrban | - ast_str_append(&a_modem, 0, "a=T38MaxBitRate:%d\r\n", t38_get_rate(p->t38.our_parms.rate)); |
| 13:37.18 | vrban | + ast_str_append(&a_modem, 0, "a=T38MaxBitRate:9600\r\n"); /* hard coded to 9600 */ |
| 13:38.11 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 13:38.11 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 13:39.17 | *** join/#asterisk echinos (~echinos@67.196.136.211) |
| 13:42.02 | evilbit | anyone have recommendations for a GUI (web based ok) management front end to asterisk? |
| 13:42.24 | *** part/#asterisk skyion (~bradc@siza.thusa.net) |
| 13:42.31 | leifmadsen | I don't recommend GUI frontends to Asterisk |
| 13:42.49 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 13:42.49 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 13:43.05 | leifmadsen | putnopvut: morning! |
| 13:43.53 | evilbit | ok, thanks but that kinda falls outside of the question ;-) |
| 13:44.07 | leifmadsen | it depends what you're trying to do too |
| 13:44.09 | evilbit | I'm more looking for stats I guess then management |
| 13:44.23 | evilbit | somewhere a person can log in and see what's going on |
| 13:44.29 | leifmadsen | you can use FreePBX, but it takes over your configuration. You can use Asterisk GUI, but it's not being supported at the moment. |
| 13:44.48 | evilbit | is AMP still around? |
| 13:44.49 | leifmadsen | You can use something like QueueMetrics to see stats of queue agents |
| 13:45.00 | evilbit | ok, I'll look up that one |
| 13:45.35 | evilbit | thx |
| 13:47.48 | chuckp | evilbit: we have used QueueMetrics, its a decent program but lacks on some very basic reporting we get from customers. Seems made for more very large scale call centers |
| 13:48.36 | evilbit | chuckp: ah, ok. thanks. This is a small operation so maybe that's not so good. Total of 7ppl |
| 13:50.07 | putnopvut | leifmadsen: it's you! |
| 13:50.14 | *** join/#asterisk jasonjjohnsonjr (~jasonjjoh@asa1.jsmc.org) |
| 13:50.29 | chuckp | evilbit: we ended up creating an XML parse of the CDR for some of our clients then pump that into an XLS |
| 13:50.30 | leifmadsen | putnopvut: it sure is! |
| 13:50.35 | leifmadsen | putnopvut: working at the cottage :) |
| 13:50.59 | chuckp | the qloaderd app from QueueMetrics (free) monitors the queue log and inputs it into a table in the DB, which provides a lot of info |
| 13:51.01 | putnopvut | leifmadsen: is this cottage on a lake. |
| 13:51.02 | putnopvut | ? |
| 13:51.13 | chuckp | just make sure you add the IVR and DID references so you can track where calls come from |
| 13:52.04 | leifmadsen | putnopvut: yes it is! I'll send you a picture |
| 13:52.29 | evilbit | hmm, anyone used OrderlyQ? |
| 13:53.16 | leifmadsen | putnopvut: can you receive MMS? |
| 13:54.11 | leifmadsen | actually nevermind, I'll email |
| 13:56.34 | *** join/#asterisk jpeeler (~jpeeler@asterisk/digium-software-dev/jpeeler) |
| 13:58.42 | *** join/#asterisk t_dot_zilla (~chatzilla@firewall-a.buf.ny.i-evolve.net) |
| 14:00.48 | t_dot_zilla | anybody know where i can get some royalty free s3m for MOH ? |
| 14:02.14 | jasonjjohnsonjr | If I install an echo cancel board on a digium 4 port T1 card why would that cause my ATT T1s to go to yellow alarm? Without the hardware echo cancel everything works fine. |
| 14:02.29 | *** join/#asterisk sgimeno (~chatzilla@95.122.8.54) |
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| 14:14.52 | leifmadsen | t_dot_zilla: s3m? you mean like... a MOD ? |
| 14:15.19 | leifmadsen | jasonjjohnsonjr: not sure -- call Digium for support since you paid for the hardware |
| 14:18.52 | jayvee | The config file is quite well self-documented, and there is plenty of info for Asterisk if you know the lingo. But what about for those of us that are both new to Asterisk, *and* the terms involved. Are there any guides on "How to configure Asterisk, and oh by the way, this is what an extension is"? |
| 14:21.01 | jasonjjohnsonjr | leifmadsen: Ok. That was my next step, I was just curious if anyone had run into this before. Thanks. |
| 14:22.52 | *** join/#asterisk moy (~moy@UNVLON55-1176057127.sdsl.bell.ca) |
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| 14:24.53 | *** part/#asterisk rocksfrow (~kyle@74-92-146-97-WashingtonDC.hfc.comcastbusiness.net) |
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| 14:27.25 | t_dot_zilla | what does this error mean? res_musiconhold.c:1093 init_app_class: Unable to open pseudo channel for timing... Sound may be choppy. |
| 14:27.39 | t_dot_zilla | leifmadsen: yeah |
| 14:28.08 | leifmadsen | t_dot_zilla: well first of all, S3M files have not been supported by much software since about 1989 |
| 14:28.17 | *** join/#asterisk Deeewayne (~dwayne@75-150-14-49-Atlanta.hfc.comcastbusiness.net) |
| 14:28.17 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 14:28.33 | leifmadsen | t_dot_zilla: that error means you don't have a timing interface installed (such as dahdi_dummy, etc...) |
| 14:30.41 | stix | leifmadsen, are you from Denmark? |
| 14:31.11 | leifmadsen | Canada. My late grandfather was from Copenhagen |
| 14:31.13 | t_dot_zilla | i'm using Asterisk 1.6.1.12 |
| 14:31.31 | stix | leifmadsen, oh that explains it :) |
| 14:31.32 | leifmadsen | t_dot_zilla: you sure are :) |
| 14:31.53 | *** join/#asterisk Tim_Toady (~moi@178.128.16.115.dsl.dyn.forthnet.gr) |
| 14:31.56 | t_dot_zilla | i thought the timing was built into asterisk 1.6.1 |
| 14:31.59 | leifmadsen | stix: heh ya, I have a very Danish name but don't speak a word of Danish other than "god morgen", "ja", and "nej" :) |
| 14:32.42 | leifmadsen | t_dot_zilla: you still have to have the appropriate kernel modules installed in order to compile res_timing_timerfd |
| 14:32.46 | stix | leifmadsen, okay, if you can order a couple of beers and say thank you, you have come a long way :) |
| 14:33.16 | leifmadsen | stix: heh, I'm good at ordering beers in English :) |
| 14:33.17 | *** join/#asterisk emora (~emora@213.236.9.114) |
| 14:33.38 | leifmadsen | stix: however when I visit Copenhagen I'll make sure I look up how to order beers and say thank you :) |
| 14:33.46 | leifmadsen | stix: oh, and I know skoal :) |
| 14:33.58 | leifmadsen | burn! |
| 14:34.13 | *** join/#asterisk stix (~stix@exchange2003.corporate.billetkontoret.dk) |
| 14:35.04 | WIMPy | leifmadsen: You want to order øl and say tak. Now you can be on your way :-)) |
| 14:35.23 | leifmadsen | how do you pronounce that? "ole" ? |
| 14:35.37 | leifmadsen | or more like "ule" ? |
| 14:35.44 | leifmadsen | "ewle" |
| 14:35.59 | stix | øl :) |
| 14:36.21 | WIMPy | No, it mor like the beginning soun of "errr...", but *very* short. |
| 14:36.25 | leifmadsen | my Swedish friend always Danes sound like they have porridge in their mouth :) |
| 14:36.50 | *** join/#asterisk bmg505 (~leon@196-209-7-27.dynamic.isadsl.co.za) |
| 14:36.55 | stix | probably not pretty to listen to |
| 14:37.16 | WIMPy | In germany we say you have to take a hot potato in your mouth to speak danish. |
| 14:37.41 | leifmadsen | :) |
| 14:39.15 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 14:40.22 | leifmadsen | I should try out that Rosetta Stone software to learn some Danish |
| 14:40.50 | *** join/#asterisk elliot98 (~elliot@unaffiliated/elliot98) |
| 14:40.55 | elliot98 | hiya all! |
| 14:41.32 | stix | never heard of that, but it looks interesting |
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| 14:58.06 | Polysics | is upgrading from an 1.6.2 version to another simply a case of make; make install? |
| 14:58.15 | Polysics | and would the above work from 1.6.1 to 1.6.2? |
| 14:59.42 | ChannelZ | mostly but there might be a couple of config changes that affect you, see the 'UPGRADE' documents |
| 15:00.20 | ChannelZ | If you use SFA or FFA or any other separate modules, you'll need to fetch 1.6.2 compat versions of those too |
| 15:00.43 | Polysics | i think i have never heard of those, so i can assume i am not .-) |
| 15:00.44 | Gnarfy | Is there a way to disable "Packet2Packet bridging" (as in forcing asterisk to always make the call go through itself), which doesn't involve using Dial() with t,T,h,H,w,W or L ? |
| 15:01.19 | WIMPy | Gnarfy: canreinvite=no |
| 15:01.49 | Gnarfy | If that isn't working for me, i should probably update my asterisk? (using version 1.4.22.2) |
| 15:02.26 | *** join/#asterisk g_r_eek (~g_r_eek@ppp-94-67-65-187.home.otenet.gr) |
| 15:05.37 | grinder13 | hello! i have the following problem since yesterday: Two Asterisk servers. I am trying to connect them via IAX. A registered IAX user in the first server tries to call registered IAX extension in the second server, but keeps getting "No authority found" by his own server (the first one). When the same IAX registered user calls an IAX extension in the same server, there is no problem. You can see my confs of the two servers here: http://pastebin.com/ |
| 15:05.38 | grinder13 | bXSx1Myh , http://pastebin.com/Ay9eWaCR |
| 15:06.10 | grinder13 | oops, here are the links: http://pastebin.com/bXSx1Myh , http://pastebin.com/Ay9eWaCR |
| 15:08.48 | *** join/#asterisk Godfather_ (~Godfather@193.153.129.150) |
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| 15:45.17 | xheliox | has a strong desire to drink CocaCola and Fly Emirates. |
| 15:45.24 | *** part/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net) |
| 15:47.26 | *** join/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net) |
| 15:48.29 | p3nguin | http://partiallyclips.com/2010/06/30/men-at-computer/ |
| 15:49.32 | *** join/#asterisk wcselby (~wcselby@216.110.88.194) |
| 15:49.33 | wcselby | o/ |
| 15:50.18 | *** join/#asterisk [intra]lanman (~lanman@freeswitch/developer/intralanman) |
| 15:52.46 | *** join/#asterisk markitoxs (~miranda83@spitfire-gw.nsswl.ftuk.net) |
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| 15:53.06 | markitoxs | is there any way to show the sip peers that only match a string? |
| 15:54.37 | markitoxs | as in as in 2000-STRING, 2001-STRING, etc// |
| 15:56.48 | *** join/#asterisk jtodd (azrgngshcg@asterisk/community-director-and-tie-dye-shirt-lover/jtodd) |
| 15:56.48 | *** mode/#asterisk [+o jtodd] by ChanServ |
| 15:57.02 | wcselby | markitoxs - asterisk -rx "sip show peers" | grep STRING |
| 15:57.07 | wcselby | from the command line |
| 15:58.12 | markitoxs | wcselby, yeah, thats the only way i could figure out, but we have already a asterisk -r running so it would change the verbosity |
| 15:58.26 | markitoxs | is it not possible within the asterisk console i guess |
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| 16:02.13 | dohd | and there was much rejoicing in a small yet very orange country somewhere near a random northsea |
| 16:02.47 | *** join/#asterisk dkirker-openmobl (~dkirker@openmobl/ceo/dkirker) |
| 16:03.15 | wcselby | lol dohd |
| 16:03.15 | *** join/#asterisk moy (~moy@UNVLON55-1176057127.sdsl.bell.ca) |
| 16:03.30 | wcselby | they were just discussing that here (lots of brazillian fans at this client) |
| 16:04.43 | p3nguin | It changes the verbosity level? Weird. |
| 16:05.18 | p3nguin | Only changing the verbosity level should change the verbosity level. |
| 16:05.56 | *** join/#asterisk nightwalk (~nightwalk@daimon.vixel.org) |
| 16:06.00 | *** join/#asterisk micols (~mio@rlogin.dk) |
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| 16:23.50 | grinder13 | can anybody check this log, please? http://pastebin.org/375560 is related to my problem with the two asterisk servers and the "No authority found" issue. the confs are here: http://pastebin.com/bXSx1Myh , http://pastebin.com/Ay9eWaCR |
| 16:29.30 | *** join/#asterisk Z_God (~julius@schwartzenberg.xs4all.nl) |
| 16:32.29 | *** join/#asterisk ruben23 (~ITadmin@125.212.40.2) |
| 16:33.22 | ruben23 | hi |
| 16:54.48 | Naikrovek | far out i hate trixbox |
| 16:54.54 | Naikrovek | i gotta lose this machine |
| 16:55.06 | Naikrovek | and by lose i mean annihilate |
| 17:01.18 | wcselby | hahaha |
| 17:01.22 | wcselby | Naikrovek - what's going on? |
| 17:01.35 | wcselby | other than you finding a high place to "bounce" your server from? |
| 17:01.39 | Naikrovek | heh |
| 17:01.47 | Naikrovek | well two things, one matters, one doesn't |
| 17:02.02 | Naikrovek | thing that matters: phone audio is garbled for INCOMING audio only |
| 17:02.04 | Naikrovek | not sure why |
| 17:02.09 | Naikrovek | setting up netflow and monitoring on that link |
| 17:02.52 | Naikrovek | thing that doesn't matter: i turned on recording of calls so i could prove that we're getting garbled audio, but all i could prove is that my recordings are useless because they go way way way out of sync |
| 17:03.02 | wcselby | haha |
| 17:03.04 | Naikrovek | that's probably a symptom of something |
| 17:03.08 | Naikrovek | but i dont' know what, yet |
| 17:03.09 | wcselby | yeah probably |
| 17:03.14 | wcselby | powerful enough box? |
| 17:03.22 | wcselby | please tell me this isn't the ec2 box...... :P |
| 17:03.43 | Naikrovek | if i am indeed losing incomping packets, that could explain why the caller's side of the conversation progresses faster than real-time on the recording |
| 17:04.04 | Naikrovek | not the EC2 box. this is a dual quad xeon of recent vintage |
| 17:04.14 | Naikrovek | it's maybe 2 years old |
| 17:04.47 | wcselby | true |
| 17:04.52 | Naikrovek | i /should/ see more or less synchronous traffic incoming and outgoing on the link that phone traffic goes out on |
| 17:04.55 | wcselby | is it only over a certain incoming link? |
| 17:05.01 | Naikrovek | but i see FAR fewer incoming packets than outgoing |
| 17:05.04 | *** join/#asterisk fcolista (~quassel@93-63-253-26.ip30.fastwebnet.it) |
| 17:05.10 | Naikrovek | yes |
| 17:05.26 | Naikrovek | i have a T1 dedicated for phones (temporarily, we're upgrading to fiber but i needed more BW for phones in the meantime) |
| 17:05.40 | wcselby | is this to an ITSP? |
| 17:05.43 | Naikrovek | it is |
| 17:05.48 | wcselby | hmmmm |
| 17:05.50 | Naikrovek | i'm working with them |
| 17:05.53 | wcselby | yeah |
| 17:05.56 | Naikrovek | on determining the cause |
| 17:06.08 | Naikrovek | everything points to packet loss on the audio they send to me |
| 17:06.17 | Naikrovek | and there are no errors on the circuit |
| 17:06.24 | wcselby | happens even with only one call in progress? |
| 17:06.32 | Naikrovek | yes |
| 17:06.59 | Naikrovek | router shows 938 incoming packets, 328761 outgoing |
| 17:07.05 | wcselby | sounds like an upstream issue. you've checked with the circuit provider that they're not seeing any issues or restricting any traffic? |
| 17:07.07 | Naikrovek | the only thing happening on this link is phone traffic |
| 17:07.30 | Naikrovek | i'm talking to the ITSP because I did extensive testing on this before I put the phones on it |
| 17:07.43 | Naikrovek | everything about the link itself seems to be fine |
| 17:09.59 | *** join/#asterisk aidinb (~Aidin@wsip-98-190-45-225.oc.oc.cox.net) |
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| 17:13.07 | *** join/#asterisk Slugs_ (~Slugs@unaffiliated/slugs) |
| 17:13.55 | Slugs_ | hi, is there a command to restart only 1 T1 as opposed to restating all with 'dahdi restart'? |
| 17:14.04 | Slugs_ | restarting* |
| 17:15.27 | *** join/#asterisk txwikinger (~quassel@sblug/member/txwikinger) |
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| 17:22.26 | *** join/#asterisk kfife (~Miranda@kfife.com) |
| 17:22.49 | kfife | Bumber sticker: Reduce global warming: Don't use flash! |
| 17:23.17 | kfife | 5 pound of coal are burned for every 10 minutes of 1080p flash |
| 17:23.21 | kfife | :-) |
| 17:24.09 | kfife | WTF!! my PPTP Server does not support mixed case??! |
| 17:24.20 | kfife | ...passwords that is |
| 17:24.47 | Naikrovek | aren't you the talkative one |
| 17:25.12 | kfife | Damn. THat means it's going to only take someone 5 billion years (instead of 100 Trillion) to brute force my password. |
| 17:25.23 | kfife | Shit Shit Shit. I feel so vulnerable |
| 17:25.44 | xheliox | Clients.. can't live with them, can't find any more storage space to hide the bodies. |
| 17:27.16 | *** join/#asterisk moy (~moy@UNVLON55-1176057127.sdsl.bell.ca) |
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| 17:31.54 | *** join/#asterisk markfeatherston_ (~mark@71.4.41.98.ptr.us.xo.net) |
| 17:35.40 | xheliox | howdy Mark |
| 17:36.32 | *** join/#asterisk aruntomar (~aruntomar@115.118.232.105) |
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| 17:41.32 | aruntomar | we've a remote asterisk server. while making calls we get to hear the sound after 30 seconds, how can i debug what's wrong |
| 17:42.09 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 17:42.16 | Naikrovek | you mean you can't hear the audio for 30 seconds, or you can't hear it AFTER 30 seconds |
| 17:43.01 | aruntomar | Naikrovek, i can't hear the sound for the first 30 sec, after that i'm able to hear the sound, weird ;) |
| 17:43.10 | markfeatherston_ | xheliox: Howdy! |
| 17:43.13 | Naikrovek | that is weird |
| 17:43.55 | markfeatherston_ | aruntomar: sip debug on the asterisk console should let you see what might be happening around the disconnect |
| 17:44.06 | *** join/#asterisk daubers (~daubers@daubers.co.uk) |
| 17:44.28 | markfeatherston_ | aruntomar: though some routers will time out udp connections before asterisk has a chance to renew it, so see if you can change your router's timeout to >60 seconds. |
| 17:44.43 | daubers | Hey all having some issues with an spa3000 not dialing out |
| 17:46.26 | markfeatherston_ | aruntomar: wait, i read that as you couldn't hear it after 30 seconds... now I'm confused. Either way, sip debug and look at what might be happening around that time |
| 17:46.41 | Naikrovek | yeah it's a weird issue |
| 17:46.46 | Naikrovek | he can hear it for 30 seconds, then can't |
| 17:46.51 | Naikrovek | argh |
| 17:47.06 | Naikrovek | he can't hear it for 30 seconds THEN he can <----- THIS ONE |
| 17:47.10 | markfeatherston_ | lol |
| 17:47.25 | Naikrovek | either way weird issue |
| 17:47.34 | Naikrovek | firewall or maybe some dns thing. dns timeouts can sometimes take 30s |
| 17:47.38 | Naikrovek | usually 1m but sometimes 30s |
| 17:48.04 | markfeatherston_ | dns should be cached by the asterisk server though right? |
| 17:48.17 | Naikrovek | DNS should be cached by the DNS server at least |
| 17:48.21 | markfeatherston_ | true |
| 17:48.22 | Naikrovek | i don't know if linux caches dns |
| 17:48.48 | Naikrovek | but the sip signalling should take 30s too |
| 17:48.55 | Naikrovek | well... maybe not |
| 17:49.18 | Naikrovek | depends if the audio comes from the same server he registers to |
| 17:49.20 | Naikrovek | i dunno |
| 17:49.24 | Naikrovek | i'm confusing myself now |
| 17:49.26 | markfeatherston_ | heh |
| 17:49.40 | markfeatherston_ | aruntomar: do you use sip reinvite? |
| 17:50.23 | aruntomar | markfeatherston_, no |
| 17:50.46 | *** join/#asterisk Benwa (~Benwa@cable-212.76.247.170.coditel.net) |
| 17:51.24 | aruntomar | i enabled debugging for sip channel |
| 17:51.33 | aruntomar | got a lot of stuff & info |
| 17:51.42 | aruntomar | but i don't know, what to look for in it |
| 17:52.28 | markfeatherston_ | aruntomar: look for what packets are going through when the audio starts coming through |
| 17:53.21 | markfeatherston_ | http://en.wikipedia.org/wiki/Session_Initiation_Protocol you should probably read through this or at very least, learn what the delayed packet is supposed to do |
| 17:54.05 | *** join/#asterisk Sorcier_FXK (~sorcierfx@unaffiliated/sorcierfxk) |
| 17:56.25 | pabelanger | aruntomar: Are you behind a NAT? |
| 17:57.32 | aruntomar | asterisk server is behind nat, and i'm not |
| 18:01.08 | *** join/#asterisk gloin (me@unaffiliated/gloin) |
| 18:03.13 | gloin | weird - I installed 1.6.2 with the samples, everything came up with all the sample dialplan stuff, then I shut down asterisk, moved the samples out of /etc/asterisk, created a really simple sip and extensions conf, start asterisk back up, and "dialplan show" has like nothing in it |
| 18:03.51 | gloin | is clearly missing something obvious, but can't see it |
| 18:03.53 | gloin | -= 0 extensions (0 priorities) in 1 context. =- |
| 18:03.55 | Gnarfy | aruntomar: have you got the right ports forwarded? port 5060 and the udp ports defined in rtp.conf (10k to 20k for example)? maybe a keepalive packet after 30 secs opens up the rtp stream or something.... |
| 18:04.45 | gloin | basically, used this as the model for my dialplan and sip conf: |
| 18:04.46 | gloin | http://www.beardy.se/2010/05/30/an-introduction-to-asterisk-the-open-source-telephony-project |
| 18:06.56 | markfeatherston_ | gloin: is it loading all of your modules correctly? |
| 18:07.15 | markfeatherston_ | gloin: try sip show peers, does that work as a valid command? I've messed up the modules before like that |
| 18:07.27 | gloin | ah |
| 18:07.33 | gloin | yeah, no such command |
| 18:07.42 | gloin | I suppose I have to create a modules.conf or similar |
| 18:09.24 | *** join/#asterisk rocksfrow (~kyle@74-92-146-97-WashingtonDC.hfc.comcastbusiness.net) |
| 18:09.37 | rocksfrow | so, i have really weird calling issues going on |
| 18:09.48 | rocksfrow | when trying to call a fax machine |
| 18:10.03 | rocksfrow | could i paste my intense pri debug for somebody to help me review? |
| 18:10.15 | rocksfrow | (in pastbin) |
| 18:10.46 | aruntomar | Gnarfy, i get all the relevant ports forwarded correctly |
| 18:11.28 | rocksfrow | http://pastebin.com/ZmS8rJ5V |
| 18:11.42 | rocksfrow | Could somebody please help me debug this? I'm thinking maybe I should call the telco? |
| 18:11.43 | gloin | markfeatherston_: thanks, it was the modules indeed. [modules] context with autload=yes was all it needed |
| 18:13.27 | [TK]D-Fender | gloin: you should have modeules.cofn rtp.conf, indications, and bunch of other essential ones from the samples. |
| 18:14.40 | [TK]D-Fender | gloin: I recommend copying the samples back and starting by only replacing the channel driver modules like SIP, IAX, DAHDI, and your dialplan ones to start |
| 18:14.44 | gloin | [TK]D-Fender: probably not a bad idea, but figuring out where the various elements of the dialplan live in the samples has proven to be a challenge for me, so I'm trying to take the opposite approach |
| 18:15.03 | *** join/#asterisk netmax (~netmax@is.linux-administrator.com) |
| 18:15.05 | gloin | start minimal, build only what I need |
| 18:15.26 | gloin | the samples creates a fairly complex dialplan. |
| 18:15.47 | [TK]D-Fender | gloin: dialplan = extensions*.* |
| 18:16.02 | rocksfrow | # |
| 18:16.02 | rocksfrow | q931_hangup: other hangup |
| 18:16.05 | [TK]D-Fender | gloin: Feel free to OCMPLETELY trash those and start from scratch |
| 18:16.27 | rocksfrow | what is 'other hangup' ? |
| 18:17.02 | Slugs_ | Is there a command to restart only 1 T1 as opposed to restarting all with 'dahdi restart'? |
| 18:17.22 | gloin | [TK]D-Fender: will try, thanks. |
| 18:17.25 | WIMPy | rocksfrow: I have no idea, but nothing you receive before that message is a reason for a hangup. |
| 18:17.52 | rocksfrow | WIMPy, really? |
| 18:17.56 | rocksfrow | call the telco? |
| 18:18.18 | n3hxs | rocksfrow If your fax worked before, then your carrier might have changed the routing through a system that does not adaquately support fax. Been there done that. |
| 18:18.20 | WIMPy | rocksfrow: No. It's Asterisk hanging up, but I cannot see any reason. |
| 18:18.53 | rocksfrow | WIMPy, is there anywhre else i can look to debug? |
| 18:18.58 | rocksfrow | I have a theory though |
| 18:19.16 | gloin | [TK]D-Fender: ahh, much cleaner. Thanks! |
| 18:19.25 | WIMPy | So what's that theory? |
| 18:19.54 | rocksfrow | well, it's a freepbx system, and i recently installed udpates which came with a new 'fax options' module |
| 18:20.00 | rocksfrow | let me disable that |
| 18:20.02 | rocksfrow | and try again |
| 18:20.21 | rocksfrow | i wonder if asterisk has some weird fax configuration? |
| 18:20.30 | rocksfrow | is that even possible? that it's sending something the receiving end doesn't like? |
| 18:21.08 | Naikrovek | yeah that's possible |
| 18:21.27 | Naikrovek | newer fax machines support higher resolution, for example |
| 18:21.37 | Naikrovek | old machines barf on thigns like that |
| 18:21.37 | WIMPy | rocksfrow: I theory I could imagine it might be possible it doesn't like the inband information. But that's absolutely normal to occour. |
| 18:22.11 | WIMPy | Naikrovek: * hangs up befor the call is answered. |
| 18:22.24 | Naikrovek | i gotta stop jumping in the middle of conversations |
| 18:22.26 | Naikrovek | sorry |
| 18:22.34 | rocksfrow | well |
| 18:22.39 | rocksfrow | disabling it didn't make a difference |
| 18:22.45 | rocksfrow | but i would imagine freepbx already messed up the config |
| 18:22.51 | rocksfrow | and that's just stopping me from modifying it |
| 18:22.56 | rocksfrow | let me tweak with these settings |
| 18:24.25 | WIMPy | rocksfrow: There is however an error that might stop the called fax machine from answering the call: The BC is set to speech. |
| 18:24.28 | rocksfrow | minimum transfer rate? |
| 18:24.46 | rocksfrow | WIMPy, ah...that could make sense |
| 18:24.51 | rocksfrow | but.. |
| 18:24.51 | rocksfrow | hrm |
| 18:25.00 | rocksfrow | but i didnt change anything |
| 18:25.01 | rocksfrow | you know? |
| 18:25.09 | rocksfrow | could i run through these possible configs with you? |
| 18:25.21 | rocksfrow | could a blank fax header break it? |
| 18:25.35 | WIMPy | 100% of all errors came from the keyboard. - But not all from the own one :-) |
| 18:25.37 | rocksfrow | i had nothing for 'default fax header', and 'default local station' |
| 18:25.58 | rocksfrow | max and min tranfer rate are set at 14400 |
| 18:26.07 | WIMPy | There is no connection so whatever the fax ist or however it's configured won't matter, yet. |
| 18:26.16 | rocksfrow | ...oh really? |
| 18:26.17 | rocksfrow | wtf.. |
| 18:26.52 | WIMPy | Maybe it's a bad version of libpri? What did you update? |
| 18:27.10 | Slugs_ | Is there a command to restart only 1 T1 as opposed to restarting all with 'dahdi restart'? |
| 18:27.19 | rocksfrow | only freepbx updates |
| 18:27.23 | rocksfrow | let me check for os updates |
| 18:27.35 | rocksfrow | there is an update available |
| 18:27.37 | rocksfrow | for libpri |
| 18:27.41 | rocksfrow | maybe that'll fix? lol |
| 18:27.57 | WIMPy | Might be woth a try. |
| 18:30.43 | rocksfrow | damn, same congestion error |
| 18:30.47 | rocksfrow | busy sig |
| 18:31.11 | WIMPy | busy? |
| 18:31.19 | rocksfrow | i get a busy signal from the phone |
| 18:31.23 | rocksfrow | as i always have |
| 18:32.11 | WIMPy | Sme as when trying to fax? |
| 18:32.52 | rocksfrow | oh well shit man |
| 18:33.00 | rocksfrow | i might be a retard, i've been calling from my phone |
| 18:33.03 | rocksfrow | to see if it'll even ring |
| 18:33.06 | rocksfrow | i can call it from my cell |
| 18:33.08 | aruntomar | guys, i even tried with iax protocol, still i hear the sound after 30 sec, tried to debug iax also |
| 18:33.10 | rocksfrow | and it rings a few times b4 the fax picks up |
| 18:33.19 | rocksfrow | i can't do the same from a phone here |
| 18:34.23 | aruntomar | i think b'cas my asterisk server is behind nat, although all the relevant ports are forwarded, do u guys still think it's a nat issue |
| 18:34.50 | WIMPy | rocksfrow: Unless you've got the same going on as shown in your paste, that's likely due to the BC. |
| 18:35.42 | rocksfrow | okay let me actually fax it and then give you the log of that |
| 18:36.18 | WIMPy | (even tho it should say incompatible destination, not congestion, but that might have gone missing somewhere) |
| 18:36.29 | rocksfrow | well |
| 18:36.47 | rocksfrow | damnit, lol |
| 18:36.51 | rocksfrow | let me try faxing it and see what the log says |
| 18:36.57 | rocksfrow | is intense necessary? |
| 18:38.21 | WIMPy | is only interested in the lines containing the hex strings. |
| 18:38.38 | WIMPy | I think that makes it a yes :-) |
| 18:41.13 | aruntomar | [Jul 3 00:08:42] NOTICE[17183]: chan_sip.c:13388 handle_response_peerpoke: Peer 'xxxx' is now Reachable. (133ms / 2000ms) |
| 18:41.54 | aruntomar | 2000ms is like 33.3 s |
| 18:42.09 | aruntomar | can this be the cause of sound delay |
| 18:45.17 | rocksfrow | WIMPy, same exact thing |
| 18:45.47 | rocksfrow | <PROTECTED> |
| 18:46.36 | rocksfrow | do i need to install the digium ffa module? |
| 18:46.40 | rocksfrow | how can i check if i have this installed or not |
| 18:47.30 | WIMPy | I never bothered about fax. |
| 18:47.58 | rocksfrow | WIMPy, huh? |
| 18:48.18 | WIMPy | But it does not explain that "other hangup". |
| 18:49.02 | WIMPy | I never tried to get a fax in to, out of or even through Asterisk. |
| 18:49.10 | rocksfrow | oh really? |
| 18:49.26 | rocksfrow | man i've been faxing out of the box with an asterisknow box for months |
| 18:49.29 | rocksfrow | now all the sudden |
| 18:49.47 | WIMPy | I don't care about the two faxes per year, offering overpriced radar warners :-) |
| 18:50.03 | rocksfrow | lol |
| 18:51.17 | Slugs_ | hardwire, ping |
| 18:51.34 | *** join/#asterisk lost_soul (shackett@devio.us) |
| 18:56.51 | *** join/#asterisk Godfather_ (~Godfather@193.153.129.150) |
| 18:57.39 | Godfather_ | o/ |
| 19:01.38 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 19:02.33 | *** join/#asterisk [sr] (~Unknowned@pal-213-228-140-150.netvisao.pt) |
| 19:02.35 | [sr] | hi :) |
| 19:03.33 | WIMPy | [sr]: You talking about the temperature? |
| 19:03.54 | [sr] | lol |
| 19:03.55 | [sr] | no |
| 19:04.03 | [sr] | just gretting people here |
| 19:10.19 | [sr] | hum i want to enable asterisk verbose logger |
| 19:10.46 | pabelanger | [sr]: *CLI> core set verbose 15 |
| 19:10.55 | [sr] | merci |
| 19:11.08 | [sr] | i was looking for the *core* options |
| 19:11.53 | [sr] | ok already saw what i want |
| 19:12.10 | [sr] | when i enter the passwd to access the voicemail using 3CX softphone client, it doubles what i type |
| 19:12.10 | [sr] | grrrr |
| 19:12.19 | [sr] | it's their bug (3cx) |
| 19:14.39 | [sr] | they don't have any place on their page to report this... damn! |
| 19:24.02 | Slugs_ | I have 83 asterisk boxes in the feild, what would be the easiest way to get all the version information from them? |
| 19:24.28 | [TK]D-Fender | Slugs_: asterisk -rx -"core show version" |
| 19:24.41 | rocksfrow | how can i get yum to force install ffa? |
| 19:24.42 | rocksfrow | Error: asterisk14-res_fax conflicts with asterisk14-addons-core |
| 19:24.57 | Slugs_ | yes.. but that would mean logging into to all manually |
| 19:25.07 | rocksfrow | research tells me it is a GPL thing, and forcing is OK, but you can't force with YUM |
| 19:25.13 | rocksfrow | is it possible to install an rpm from a yum repo? |
| 19:25.21 | rocksfrow | rpm -i --force style |
| 19:25.33 | Slugs_ | [TK]D-Fender, i have a L3 L1 support contract |
| 19:26.02 | wcselby | it's a little wet outside |
| 19:26.26 | wcselby | http://houston.justweather.com/radar/ |
| 19:26.30 | chuckf | Slugs_: clusterssh might be what you want |
| 19:26.41 | Slugs_ | [TK]D-Fender, i want to automate the logging in process and extract the information |
| 19:26.43 | [TK]D-Fender | Slugs_: Well where do you think * is going to advertise this in detail? |
| 19:27.14 | Slugs_ | no clue |
| 19:27.17 | [TK]D-Fender | Slugs_: there is some level in SIP debug. then again.. they'd all have to be registered or contactable and running aversion that will detail what version in the UA |
| 19:27.44 | *** join/#asterisk bcrosby (~bcrosby@shell.infiniteloop.ca) |
| 19:27.48 | [TK]D-Fender | Slugs_: * is a single engine, it isn't a "built for imemdiate central polling in a standardized way". |
| 19:28.00 | [TK]D-Fender | Slugs_: You've got the tools that you've got. CLI, and AMI |
| 19:28.04 | bcrosby | can anyone offer me suggestions on a way to debug why asterisk is not registering with a SIP provider as a client? |
| 19:28.15 | bcrosby | I seem to have everything in order, but having a hard time figuring out why it's not registering |
| 19:28.24 | [TK]D-Fender | bcrosby: Yes... look at the SIP DEBUG in * CLI for the attempt |
| 19:28.33 | [TK]D-Fender | bcrosby: And PASTEBIN is your friend. |
| 19:28.33 | bcrosby | it just does OPTIONS and that's it |
| 19:28.35 | [TK]D-Fender | ~pb |
| 19:28.36 | 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.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. |
| 19:29.07 | Slugs_ | [TK]D-Fender, core show version is perfect, i was just wondering if there was an easy script i could write to automate to running of that command on 83 hosts |
| 19:29.40 | Slugs_ | to=the |
| 19:30.45 | [TK]D-Fender | Slugs_: Depends how you can issue it, either on OS CLI, or via AMI. |
| 19:30.55 | [TK]D-Fender | Slugs_: And as to how you get the list of hosts. |
| 19:31.18 | Slugs_ | ok |
| 19:31.49 | bcrosby | [TK]D-Fender: Any suggestions on why it's not doing anthing more than OPTIONS? |
| 19:32.20 | [TK]D-Fender | bcrosby: what does "sip show registry" say? |
| 19:32.30 | bcrosby | no registrations |
| 19:32.43 | Slugs_ | [TK]D-Fender, socond question, can you restart just 1 T1 if you have 4? I don't want to restart all of them |
| 19:33.19 | [TK]D-Fender | bcrosby: NO entries in there? |
| 19:33.22 | bcrosby | none |
| 19:33.39 | [TK]D-Fender | bcrosby: Then you don't have a valid register lin in the right place in your sip.conf |
| 19:33.46 | bcrosby | hmm ok |
| 19:33.56 | [TK]D-Fender | Slugs_: No |
| 19:34.02 | bcrosby | it has to appear under the [general] section? |
| 19:34.26 | Slugs_ | ty |
| 19:35.13 | [TK]D-Fender | bcrosby: Yes |
| 19:35.30 | bcrosby | hmm ok |
| 19:35.37 | bcrosby | i'll revist the register line, thanks |
| 19:39.01 | bcrosby | [TK]D-Fender: should I be setting up a type=peer config in sip.conf for asterisk to actually do any registrations? |
| 19:39.02 | *** part/#asterisk warmbowski (~warmbowsk@97.65.109.26) |
| 19:39.28 | [TK]D-Fender | bcrosby: that has nothing to do with registering |
| 19:39.58 | [TK]D-Fender | bcrosby: that only provides something to auth calls they might happen to send you AFTER you've registered and you actually have a call. |
| 19:40.27 | *** join/#asterisk lost_soul (shackett@devio.us) |
| 19:40.43 | bcrosby | ok cool |
| 19:44.06 | bcrosby | http://pastebin.com/3xjAWxFS |
| 19:44.12 | bcrosby | incase you wanted to see what it was doing |
| 19:44.40 | [TK]D-Fender | [15:32]<[TK]D-Fender>bcrosby: what does "sip show registry" say? <--- |
| 19:44.51 | [TK]D-Fender | bcrosby: OPTIONS = nothing to do with registration. |
| 19:44.58 | bcrosby | right |
| 19:45.03 | bcrosby | 0 SIP registrations |
| 19:45.13 | bcrosby | im 100% sure the register line is correct |
| 19:45.17 | *** join/#asterisk Arsenick (~rp@modemcable022.82-21-96.mc.videotron.ca) |
| 19:46.59 | [TK]D-Fender | bcrosby: And I'm 10% sure you've done something wrong. And * agrees |
| 19:47.05 | [TK]D-Fender | %100* |
| 19:47.11 | bcrosby | lol |
| 19:47.23 | bcrosby | yeah im trying to figure out what |
| 19:47.39 | [TK]D-Fender | bcrosby: How's that working out for you? </drphil> |
| 19:53.51 | *** join/#asterisk lesouvage (~lesouvage@cc741036-b.groni1.gr.home.nl) |
| 19:57.10 | *** join/#asterisk lost_soul (shackett@devio.us) |
| 19:59.28 | *** join/#asterisk jmacz (~jmacz@190.144.75.22) |
| 20:00.32 | Joe_CoT | bcrosby, are you behind a NAT? did you set externip? |
| 20:00.44 | bcrosby | not behind nat |
| 20:03.06 | *** join/#asterisk Docteh (~mage@S0106001b24ce7281.vc.shawcable.net) |
| 20:05.07 | *** join/#asterisk Alric (~nbowyer@64.6.54.218) |
| 20:05.33 | *** join/#asterisk netmax (~netmax@is.linux-administrator.com) |
| 20:05.37 | bcrosby | heh |
| 20:05.41 | bcrosby | got it working |
| 20:07.10 | Joe_CoT | bcrosby, what was the problem? |
| 20:07.26 | bcrosby | originally I just tacked on my config to the bottom of the default sip.conf |
| 20:07.32 | bcrosby | I created a new file with just my config |
| 20:07.58 | Joe_CoT | if you tacked it onto the end, then the register wouldn't have been in the [general] section |
| 20:08.21 | bcrosby | ahh. I thought the [general] section would have continued throughout the entire config |
| 20:08.24 | bcrosby | my mistake |
| 20:09.27 | Joe_CoT | it'd stop as soon as there's another section, and there are a couple in the sample as examples |
| 20:10.20 | bcrosby | ahh ok |
| 20:12.28 | gloin | heh |
| 20:12.39 | gloin | Somebody had a lot of fun recording tt-monty-knights |
| 20:13.16 | p3nguin | I fart in your general direction. |
| 20:13.22 | gloin | faarrrt |
| 20:13.24 | *** join/#asterisk Cain (~Geek@unaffiliated/cain) |
| 20:24.48 | Docteh | would prompting people calling to press a button work for stopping robocalls? |
| 20:25.46 | pabelanger | Docteh: playback an out-of-order tone |
| 20:26.41 | ChannelZ | ARGH |
| 20:27.06 | *** join/#asterisk billum (~bthies@122.167.71.238) |
| 20:27.19 | ChannelZ | OT: How in the hell do you make Mac OS stop changing the permissions on directories/files it makes on a network server (SMB)? The GD thing keeps making everything group read-only |
| 20:29.14 | Docteh | pabelanger: I just want to set up a light barrier, i don't have any regular robocalls coming in, seems to be a different number every time. |
| 20:30.55 | billum | Hi, I wondered if anyone would be so kind as to entertain a dahdi question: is it possible to distinguish between different boards in /etc/dahdi/system.conf? I have two boards installed and recognized (with dahdi_hardware) but dahdi_cfg seems to interpret all lines in system.conf with respect to only one board, leading to an error. More details available... (thanks!) |
| 20:31.00 | ChannelZ | Yeah jus do a 'press 3 to continue' or something |
| 20:31.19 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 20:31.21 | ChannelZ | robocalls don't understand english |
| 20:31.22 | [TK]D-Fender | Docteh: Do robo-calls send random DTMF to you? |
| 20:32.01 | *** join/#asterisk QubeZ (~nkasu@64.128.254.34) |
| 20:32.03 | QubeZ | hello all |
| 20:32.23 | *** part/#asterisk ryanlin (~ryanlin@you.better.not.nulroute.us) |
| 20:32.28 | QubeZ | im having an issue where when I move my asterisk server to gigabit, i lose dtmf... ie: cannot send tones so no password's for my vm work or 888 IVR's etc.. |
| 20:32.40 | QubeZ | when I put server back on my FastEthernet port, it works |
| 20:32.42 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 20:33.12 | [TK]D-Fender | QubeZ: NETWORKING issue. Same IP? What about firewall rules? |
| 20:33.27 | QubeZ | [TK]D-Fender: no access-list or firewall, same IP |
| 20:33.34 | WIMPy | Same interface name? |
| 20:33.37 | [TK]D-Fender | QubeZ: Somehting else is wrong |
| 20:33.46 | QubeZ | same vlan etc.. I literally just move it from fa 3/4 to gi 2/3 and dtmf stops working |
| 20:33.54 | QubeZ | WIMPy: same eth0 is plugged in |
| 20:33.56 | [TK]D-Fender | VLAN... oh wonderful |
| 20:34.07 | QubeZ | access port |
| 20:34.35 | ChannelZ | did your routing tables not catch up yet, media is going off into space? |
| 20:35.21 | QubeZ | ChannelZ: i can make calls and receive calls |
| 20:35.31 | QubeZ | just when i dial my vm and it asks for password, dtmf is not sent |
| 20:35.49 | ChannelZ | cool |
| 20:35.54 | QubeZ | right now my phone is registered to the server and able to make outbound and internal calls just no dtmf. |
| 20:36.29 | WIMPy | QubeZ: This is a Cisco switch? |
| 20:40.02 | QubeZ | WIMPy: yup |
| 20:40.31 | WIMPy | Are you sure the port configurations are identical? |
| 20:41.06 | QubeZ | WIMPy: yup, just checked |
| 20:41.36 | WIMPy | Have you checked the documented bugs for the switch and the line card? |
| 20:41.38 | QubeZ | have spanning-tree portfast on both, switchport mode access, switchport access vlan 11 |
| 20:41.44 | QubeZ | WIMPy: not yet |
| 20:48.50 | QubeZ | i dont get it... this cannot be a bug... refuse to believe with Cisco that this is possibly a bug |
| 20:48.51 | *** join/#asterisk g_r_eek (~g_r_eek@ppp-94-67-69-235.home.otenet.gr) |
| 21:02.34 | gloin | Is there a dialplan function for "do nothing for one second?" |
| 21:02.52 | [TK]D-Fender | gloin: Wait(1) |
| 21:03.46 | gloin | [TK]D-Fender: thanks again |
| 21:11.24 | *** join/#asterisk Godfather_ (~Godfather@193.153.129.150) |
| 21:12.10 | *** join/#asterisk pabelanger-lap (~pabelange@CPE001fe2a8fd1d-CM0012254094b2.cpe.net.cable.rogers.com) |
| 21:14.41 | *** join/#asterisk Buttons840 (~buttons@c-76-27-4-93.hsd1.ut.comcast.net) |
| 21:15.08 | Buttons840 | what functions or keywords should i be looking for if i want to set my own caller id? |
| 21:16.14 | Buttons840 | i remember the AMI originate function has a field for callerid, but i don't remember what arguements it accepts |
| 21:17.06 | pabelanger-lap | Buttons840: core show function CALLERID |
| 21:18.34 | *** join/#asterisk pyite (~dschreibe@unaffiliated/pyite) |
| 21:23.55 | *** join/#asterisk pyite (~dschreibe@unaffiliated/pyite) |
| 21:24.39 | ruben23 | hi how do i check my sip debug for my one way audio problem.. |
| 21:25.01 | Joe_CoT | ruben23, that's not sip, that's rtp |
| 21:25.13 | Joe_CoT | sip does the connection, rtp handles the audio |
| 21:25.49 | Joe_CoT | rtp set debug on (you probably want to do rtp set debug ip <ip> instead) |
| 21:25.52 | fenrus | sounds like nat is involved |
| 21:25.58 | fenrus | if the audio is only one way :) |
| 21:26.03 | ruben23 | Joe_CoT:i have set all the required for asterisk NAT, open all ports but still one way audio the client im calling cannot hear me, but i can hear him |
| 21:26.06 | fenrus | or perhaps firewall-rules blocking the rtp |
| 21:26.30 | ruben23 | all is open already port-forward rtp media 10000-20000 |
| 21:26.39 | Joe_CoT | ok, well try rtp debug then |
| 21:26.50 | fenrus | forwarded?, to where ? |
| 21:27.00 | fenrus | the phone should get the rtp, not the asterisk |
| 21:27.02 | *** join/#asterisk Netgeeks (~chris@gw1.netgeeks.net) |
| 21:27.19 | ruben23 | but he case is here, sometime they can hear but frequent they cant hear .. |
| 21:27.19 | [TK]D-Fender | ~sipnat |
| 21:27.20 | infobot | [~sipnat] Quick guide on configuring * + 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 |
| 21:27.21 | [TK]D-Fender | ^^^^6^ |
| 21:27.21 | *** join/#asterisk voxter (~voxter@macpro.daytonhome.voxter.net) |
| 21:27.57 | ruben23 | Joe_CoT: rtp media is forwarded to my local asterisk server. |
| 21:29.00 | Joe_CoT | ruben23, I think you keep pinging the wrong person. I'm just telling you to use rtp debug, which was your original question. If you can hear them, you're receiving rtp packets, so I don't think the port forwarding is an issue. |
| 21:30.42 | ruben23 | <PROTECTED> |
| 21:30.59 | Joe_CoT | you should. It should be back and forth between you and another ip |
| 21:31.29 | *** join/#asterisk uqlev (~yuriy@91.184.221.31) |
| 21:32.20 | ruben23 | http://pastebin.com/0DNyyYP7 |
| 21:32.51 | ruben23 | how do i interpret that back and forth IP.. |
| 21:32.56 | *** join/#asterisk Buttons840 (~buttons@c-76-27-4-93.hsd1.ut.comcast.net) |
| 21:33.19 | Joe_CoT | ruben23, I'm guessing 192.168.254.5 is your phone's ip right? |
| 21:34.33 | leifmadsen | infobot: no, sipnat is 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 |
| 21:34.34 | infobot | okay, leifmadsen |
| 21:34.50 | leifmadsen | the use of '*' instead of saying 'Asterisk' shoudl be outlawed |
| 21:35.42 | Joe_CoT | hi leifmadsen, I've got your book sitting in front of my keyboard |
| 21:35.50 | leifmadsen | w00t! :) |
| 21:35.56 | leifmadsen | I can't wait until we finish the 3rd edition |
| 21:36.02 | leifmadsen | won't be for several months yet |
| 21:36.10 | Joe_CoT | cool, i'll look forward to it |
| 21:36.14 | leifmadsen | rewriting it from scratch |
| 21:36.36 | ruben23 | Joe_CoT:yes thats the phone IP |
| 21:36.40 | xheliox | leifmadsen: I second the banning of the use of '*'.. |
| 21:36.48 | xheliox | All in favor? |
| 21:36.52 | leifmadsen | aye |
| 21:37.03 | xheliox | It is so ordered. |
| 21:37.10 | xheliox | Now.. what shall the punishment be? |
| 21:37.11 | leifmadsen | Motion passed |
| 21:37.14 | leifmadsen | DEATH! |
| 21:37.18 | xheliox | Isn't that too easy? |
| 21:37.21 | leifmadsen | (by icecream) |
| 21:37.37 | xheliox | Shouldn't we make them use an inferior phone system? |
| 21:37.37 | *** join/#asterisk neurosys (~neurosys@c-71-196-20-208.hsd1.fl.comcast.net) |
| 21:37.46 | Joe_CoT | compiling and running asterisk on windows? |
| 21:38.23 | xheliox | Joe_CoT: Are you saying I'm not supposed to be running Asterisk on my Windows 98 box? |
| 21:39.08 | Joe_CoT | xheliox, if you really do, then I think you've earned the right to use "*" anyway :P |
| 21:39.43 | xheliox | Hey now, the only Linuxeseseses anyone needs is cygwin and a nice spybot filled installation of Windows. |
| 21:40.05 | xheliox | But none of that new fancy Windows Me stuff. ;) |
| 21:41.10 | ruben23 | Joe_CoT:..? |
| 21:43.38 | Joe_CoT | ruben23, I'm trying it on my server to see what the rtp output looks like |
| 21:44.26 | ruben23 | ok |
| 21:45.44 | Joe_CoT | ruben23, here's mine: http://pastebin.com/mT9QLMSn |
| 21:46.12 | Joe_CoT | notice how there's 2 ips involved, and a from and to for both. Did you do it just for a specific ip or just on? |
| 21:46.58 | Docteh | huh asterisk-sounds-core gets version bumped occasionally but the CHANGES file doesn't mention anything |
| 21:47.22 | *** join/#asterisk troy42 (troy@fitzroy.yort.com) |
| 21:47.37 | ruben23 | Joe_CoT: just on |
| 21:47.56 | ruben23 | not on a specific ip, how to do it on a specific IP..? |
| 21:52.02 | Joe_CoT | ruben23, if you didn't do a specific IP, then you're fine on the debugging. But there should be another leg there -- to and from your sip provider (or whatever provider you're using) |
| 21:53.29 | Joe_CoT | ruben23, have you tried doing an echo test? |
| 21:55.00 | [TK]D-Fender | Joe_CoT: pastebin your SIP DEBUG for your failed call. |
| 21:55.32 | Joe_CoT | ruben23, ^ |
| 21:57.06 | ruben23 | ok |
| 21:59.35 | ruben23 | http://pastebin.com/6vNes1a2 |
| 22:00.16 | *** join/#asterisk ruben23 (~ITadmin@125.212.40.2) |
| 22:00.36 | ruben23 | http://pastebin.com/6vNes1a2 |
| 22:00.42 | ruben23 | thats the sip debug |
| 22:01.55 | Joe_CoT | ruben23, g2g. Remaining suggestions are to check that the rtpstart and end in your rtp.conf match the ports your forwarded, and try doing an echo test independent of your sip provider: http://www.voip-info.org/wiki/view/Phone+Numbers |
| 22:08.05 | [TK]D-Fender | ruben23: Issues between * and a device on your local LAN only? |
| 22:09.14 | ruben23 | when liek 4 phone try to dial to my local asterisk behind NAT, sometime they are heard but most frequently they are not heard by the person they call |
| 22:09.51 | [TK]D-Fender | ruben23: You didn't show the COMPLETE call. |
| 22:10.01 | *** join/#asterisk Buttons840 (~buttons@c-76-27-4-93.hsd1.ut.comcast.net) |
| 22:10.37 | ruben23 | how do i capture, its all the screen i can copy. |
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| 22:22.15 | [TK]D-Fender | ruben23: 2 magic words for you : SCROLLBACK BUFFER |
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| 22:32.35 | kc8pxy | i'm trying to punch a hole in my firewall for my asterisk server, and I'm forgetting which rtp ports i need to dnat. which ports are standard? |
| 22:34.47 | *** join/#asterisk riddlebox (d8d7cdda@gateway/web/freenode/ip.216.215.205.218) |
| 22:35.23 | ruben23 | [TK]D-Fender: how to used that |
| 22:35.40 | riddlebox | hello all, come someone help me out here, I am trying to connect an IP Office to Asterisk with their 3rd party sip license and I get asterisk to register but a call doesnt complete? |
| 22:35.43 | riddlebox | http://pastebin.ca/1893624 |
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| 22:46.17 | riddlebox | [TK]D-Fender: can you take a look at http://pastebin.ca/1893624 and let me know if it gives any clues to why my calls come in to asterisk but nothing happens....? |
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| 23:09.16 | seanjohn | would the time being off on sip devices conflict if the time is correct on asterisk? |
| 23:10.16 | seanjohn | i have had to delete 7 accounts to make them register, even though everything for credentials was correct. I just noticed the timezone was incorrect on all 7 |
| 23:15.21 | jdoe | in asterisk 1.6, call-limit <-- does it count inbound? outbound? inbound+outbound? |
| 23:22.42 | seanjohn | jdoe, it has been the same since 1.4; it limits both IF you have set your trunks outbounds to be counted with your inbounds, on the same context. Otherwise, its only outbound |
| 23:24.28 | jdoe | seanjohn: Ah. I'm just looking at the voip-info documentation and it doesn't actually explain what busy-limit and call-limit do. Or rather, it says they limit calls but makes no mention of how that number is reached. |
| 23:25.28 | Buttons840 | from my novice perspective it seems silly to have to control asterisk through both the AMI and AGI; am i missing something? what were the reasons for designing it this way? |
| 23:25.55 | [TK]D-Fender | Buttons840: Perhaps you could be less vague about "controlling" |
| 23:26.00 | *** join/#asterisk pyite (~dschreibe@unaffiliated/pyite) |
| 23:26.05 | [TK]D-Fender | Buttons840: Because AMI and AGI are completely different. |
| 23:27.32 | jdoe | Buttons840: AGI lets you extend asterisk to do things it doesn't by default. AMI lets you... well, run asterisk commands and see what asterisk is doing. |
| 23:27.46 | *** join/#asterisk Buttons840 (~buttons@c-76-27-4-93.hsd1.ut.comcast.net) |
| 23:27.52 | jdoe | as to why they're separate, well, he was right. |
| 23:29.05 | [TK]D-Fender | Buttons840: AGI lets you do a lot more in between interacting with the caller. AMI allows you to monitor general server activity, manipulate ANY call, etc |
| 23:29.29 | [TK]D-Fender | Buttons840: AMI can't substitute actual call processing. Hence apples & oranges |
| 23:31.32 | Buttons840 | i see; my current script (which has fulfilled my needs for some time) uses the dialplan for call processing and the script watches the AMI and monitors the progress of the calls then reports the progress -- controlled by the dialplan, monitored by the AMI script |
| 23:31.40 | Buttons840 | is this a poor approach? |
| 23:32.12 | Buttons840 | i'm guessing the dialplan lacks the flexibility of a proper agi script |
| 23:32.59 | [TK]D-Fender | Buttons840: Dialplan is a CALL. AMI just monitors activiy. Why should a CALL be monitoring things? |
| 23:33.26 | jdoe | Buttons840: whether it's a poor approach or not depends on what exactly you're doing, I guess. |
| 23:33.40 | jdoe | Buttons840: ... why can't you do what you want with the CDR logs? |
| 23:34.52 | Buttons840 | jdoe: well, i really wanted to avoid the agi, so i came up with this and it works |
| 23:35.20 | jdoe | jdoe: okay, but I'm asking why you can't get the info you want from looking at the CDR logs directly, and leaving asterisk alone. |
| 23:35.27 | jdoe | ... wow, brain fail. |
| 23:35.33 | jdoe | Buttons840: okay, but I'm asking why you can't get the info you want from looking at the CDR logs directly, and leaving asterisk alone. |
| 23:35.57 | jdoe | Buttons840: if you just want to know how far a call got, quality details etc. iirc you can get that from the CDR directly. |
| 23:37.48 | Buttons840 | jdoe: i see what your saying; i'm using asterisk as an automated call bot (i'm not sure what the proper term is) and i can currently call on 14 people at a time over the pstn; as soon as a call ends the script sees it on the ami and originates a new call |
| 23:38.16 | Buttons840 | jdoe: if the cdr records updated fast enough i guess i could use them for the same purpose |
| 23:39.03 | Buttons840 | it works, i'm just not sure whether it's a horrible hack or not; guess it doesn't matter |
| 23:40.52 | Buttons840 | about "calls" and the ami; i see a single call can produce several newchannel events (ami events) and also several hangup events; do these corrispond to each other? for example if i have 2 newchannels events will i eventually see 2 corrisponding hangups? |
| 23:40.54 | jdoe | Buttons840: uhh... you can tweak how often the call logs get flushed to disk, take a look at the cdr configs. |
| 23:41.11 | jdoe | Buttons840: they should be joined by the unique id. |
| 23:42.41 | Buttons840 | jdoe: are scripts which simply monitor the cdr common? |
| 23:49.19 | Buttons840 | "The uniqueid field is not guaranteed to be unique across the different CDR entries, even though the name suggests exactly that. In fact Uniqueid has never been guaranteed to be unique across all CDRs: It is only unique across a channel, and a channel may have multiple CDRs." from the viop-info.org wiki; which admitedly isn't the most complete, or reliable |
| 23:49.55 | jdoe | Buttons840: depends on how big you are, I guess. I used to work for a provider, we'd do batched reports daily. Any major problems with call quality we'd hear about through the CSRs. |
| 23:50.12 | jdoe | Buttons840: if you need to know *right now* you could hack together something with perl and File::Tail pretty quick. |
| 23:50.19 | jdoe | ... or whatever you want, really. |
| 23:51.48 | Buttons840 | python :) i think i'm the only asterisk user who chooses python |
| 23:51.51 | *** join/#asterisk lost_soul (shackett@devio.us) |
| 23:52.31 | jdoe | well, whatever works. |
| 23:54.22 | Buttons840 | when i first began i figured that a single "call" would always have the same uniqueid, but this turned out to be not true (mabye because a single call can create multiple channels?), so every time i found a new uniqueid, i'd querry a special variable through the AMI and then use that variable to map the uniqueid and events to the correct call |
| 23:54.25 | jdoe | Buttons840: ... I'm not a developer, but it sure *looks* like uniqueid is actually unique. |
| 23:55.50 | jdoe | the format is ctime.number, where number is just a counter. |
| 23:56.36 | *** join/#asterisk Mhaddog (~Mhaddog@z65-50-118-232.ips.direcpath.com) |
| 23:57.02 | jdoe | ... so unless you restart asterisk a couple of times and process calls within a second, uniqueid should be unique. |