| 00:16.51 | *** join/#asterisk n0cturnal (~n0c@2403:dc00:ffff:fffc:226:18ff:fe3a:3bad) |
| 00:20.07 | neurosys | Intersting. 1.8 has phone provisioning... |
| 00:22.45 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 00:37.09 | *** join/#asterisk jblack (~jblack@71.181.209.104) |
| 00:57.13 | ectospasm | neurosys: earlier ones did too, but you usually had to pay for it |
| 00:58.25 | neurosys | well I would just set option66 in my dhcp server and roll out poly configs from a scripts i made. |
| 01:08.27 | *** join/#asterisk aidinb (~Aidin@unaffiliated/aidinb) |
| 01:16.43 | *** join/#asterisk smeet2002 (~smeet2002@dsl-69-172-120-166.acanac.net) |
| 01:16.58 | smeet2002 | hi there |
| 01:17.24 | *** join/#asterisk greezmunkey (~greezmunk@173.241.161.62) |
| 01:20.12 | smeet2002 | does anybody know how to fix "invalid conversion from 'const ssl_method ' to 'ssl_method '" while installing pwlib? |
| 01:23.21 | smeet2002 | I have to go out for a while...if anybody unfreeze, pls answer |
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| 01:40.03 | *** join/#asterisk eject_ck (~eject_ck@haitek.cosmonova.net.ua) |
| 01:47.41 | greezmunkey | p3nguin: An interesting turn in testing further. A softphone on a lappy connected to the same AP, using the PhoneA config you (generously) helped me with registers, and works! The only thing I can think of the the fact that the lappy uses dhcp to get its ip address info. I am in the process of setting up a dedicated dhcp server to interact with the spectralink phones to see if they will come up. P.S. PhoneB (same config from last night) can call th |
| 01:47.42 | pabelanger | smeet2002: should be fixed in the latest branch |
| 01:58.25 | *** join/#asterisk Teevo (~Dr_Steve@134.36.233.220.static.exetel.com.au) |
| 01:58.34 | Teevo | Hello people. |
| 02:00.17 | Teevo | So has anyone played with the vtiger PBX module that intergrates with Asterisk? |
| 02:18.27 | Teevo | Quiet in here today... |
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| 02:35.25 | *** join/#asterisk tyman (~tyler@99-28-157-10.lightspeed.frsnca.sbcglobal.net) |
| 02:47.28 | smeet2002 | @pabelanger Thanks! Do you mean there is no way to install H323 on Asterisk 1.8 right now? |
| 02:49.53 | Niklas- | I have an issue getting IAX trunking to work with Asterisk 1.6.2.14. I have a TE410P card and both the kernel and asterisk modules are loaded. 'timing test' works and shows it is using the DAHDI module. Any suggestions on what i can do wrong? |
| 02:50.17 | *** join/#asterisk rrb3942 (~rbullock@cpe-67-252-100-238.stny.res.rr.com) |
| 02:52.09 | tzafrir | Niklas-, how can you tell something is wrong with IAX? |
| 02:52.19 | tzafrir | (that said, I'm off now) |
| 02:53.54 | Niklas- | iax2 show peers, doesn't show (T). And when starting up, i get theese errors: Unable to support trunking on user 'peer1' without a timing interface |
| 02:55.11 | pabelanger | smeet2002: svn checkout branch/1.8 |
| 03:19.03 | Niklas- | Seems like an 1.6.2.14 bug, i upgraded to 1.6.2.15-rc1 and then it worked |
| 03:40.13 | *** join/#asterisk aidinb (~Aidin@unaffiliated/aidinb) |
| 03:52.53 | smeet2002 | @pabelanger...I am not familiar with svn...from where I need to do this command? asterisk CLI? or linux cli? |
| 03:56.41 | pabelanger | smeet2002: You can google 'subversion' about how to use it. Here is the repo: http://svn.digium.com/svn/asterisk/branches/1.8/ |
| 03:59.49 | smeet2002 | @pabelanger Thank you...I will.... |
| 04:05.34 | smeet2002 | the only thing I can't understand..pwlib is not Asterisk's part...how it can be fixed there? |
| 04:09.55 | Teevo | If anyone can help me troubleshoot a vtiger asterisk intergration problem, I've got it to the point where it will call the extension and then say goodbye without connecting the call, and it doesn't recognise incomming calls at all. |
| 04:11.14 | *** part/#asterisk smeet2002 (~smeet2002@dsl-69-172-120-166.acanac.net) |
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| 05:00.22 | greezmunkey | holy manhole covers Batman, it worked! WooHoo! |
| 05:06.42 | *** join/#asterisk Kate6 (~six@184-100-222-210.ptld.qwest.net) |
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| 05:07.43 | Kate6 | Can anyone tell me why Asterisk seems to behave oddly every time I include the star character in an extension? |
| 05:08.37 | shamelessn00b | the same reason as if you type google inside google you'd crash the internet |
| 05:08.42 | shamelessn00b | :) |
| 05:09.24 | *** join/#asterisk atan (~atan@unaffiliated/atan) |
| 05:09.46 | Kate6 | I wanna do stuff like "exten => *77.,1,Set(something=something)" "exten => *77.,2,Goto(${EXTEN:3},1)" |
| 05:10.29 | Kate6 | If I put the star there, as soon as I've dialed the "*77" the phone makes an error sound and the Asterisk debug says the extension "*77" can't be found. |
| 05:10.37 | Kate6 | If I make it "777" instead of "*77" it works as I want it to. |
| 05:10.58 | Kate6 | Is there some weird behavior associated with the star that I'm not finding in the documentation? |
| 05:16.45 | shamelessn00b | Kate6: try _*777 |
| 05:16.56 | shamelessn00b | _*77 |
| 05:17.08 | greezmunkey | Kate6: It shounds like you made a wild card, try 677 and see what it does. |
| 05:17.25 | greezmunkey | s/shounds/sounds |
| 05:17.32 | shamelessn00b | _*77 works |
| 05:17.55 | Kate6 | The dot is there to match the rest of the number. |
| 05:18.23 | Kate6 | i.e. I want to be able to dial *77 area code number, have the *77 tell Asterisk to set some variables to particular values, then use a goto to actually dial the number. |
| 05:18.54 | Kate6 | _777. works correctly for that purpose. |
| 05:19.19 | Kate6 | _*77. makes it so if I try to dial "*77 area code number" I get a weird tone immediately after the "*77" on the phone. |
| 05:19.30 | Kate6 | And the Asterisk debug output says the extension "*77" was not found. |
| 05:20.00 | Kate6 | ... Do I need to do _[*]77? Is the * the wildcard you're talking about? |
| 05:20.47 | Kate6 | ... |
| 05:21.18 | shamelessn00b | What is it you want to do |
| 05:21.32 | shamelessn00b | first the user dials *77 |
| 05:21.47 | shamelessn00b | and then they should get a tone, and then afterwards they have to dial another extension? |
| 05:22.20 | Kate6 | Um, no, essentially I just want to be able to prefix a number with a star code in order to set some variables. |
| 05:22.29 | Kate6 | Specifically I'd like to be able to turn on the pitch shift effect with a prefix code. |
| 05:22.41 | Kate6 | And I'd like to be able to use a prefix to dial out using a different SIP account. |
| 05:23.40 | Kate6 | exten => _333.,1,Set(outbound_gmail=gmail-nocid) |
| 05:23.41 | Kate6 | exten => _333.,2,Goto(${EXTEN:3},1) |
| 05:23.59 | Kate6 | This works. If I change it to "_*33" instead of "_333", it no longer works. |
| 05:27.12 | shamelessn00b | _333! |
| 05:27.30 | Kate6 | ?? |
| 05:28.05 | shamelessn00b | http://www.voip-info.org/wiki/view/Asterisk+Dialplan+Patterns |
| 05:28.41 | shamelessn00b | _333! will match 333 too, _333. will match any 4 digit extension starting with 333 |
| 05:29.31 | Kate6 | Actually, "_333." will match any extension of 4 or more digits starting with 333. |
| 05:29.48 | Kate6 | But for some reason "_*33." won't match any extension starting with *33. |
| 05:30.34 | Kate6 | The page you linked me to doesn't mention any special significance for the * character. |
| 05:30.45 | shamelessn00b | _*333! matches *333 |
| 05:31.17 | Kate6 | I'm aware. |
| 05:31.44 | shamelessn00b | and for me _*333. is working |
| 05:31.55 | shamelessn00b | I dialed *3334 |
| 05:32.05 | shamelessn00b | and it took me to the defined extension |
| 05:32.30 | Kate6 | So you're saying you aren't really sure of a specific special meaning to the star character, but that through experimentation you've noticed that "_*33." does not work but "_*333." does? |
| 05:32.35 | *** join/#asterisk jblack (~jblack@71.181.209.104) |
| 05:32.54 | Kate6 | Ok, let me try it like that, one second. |
| 05:33.06 | shamelessn00b | exten => _*456.,1,Playback(/usr/src/jack_capture-0.9.56/Prompt_2) |
| 05:33.07 | shamelessn00b | <PROTECTED> |
| 05:33.09 | shamelessn00b | <PROTECTED> |
| 05:33.10 | shamelessn00b | <PROTECTED> |
| 05:33.12 | shamelessn00b | <PROTECTED> |
| 05:33.13 | shamelessn00b | <PROTECTED> |
| 05:33.30 | shamelessn00b | *4567 |
| 05:33.45 | shamelessn00b | it just pauses for a while |
| 05:33.58 | shamelessn00b | you can change the time using the timeout function IIRC |
| 05:34.00 | Kate6 | [Nov 25 21:33:33] NOTICE[10090]: chan_sip.c:21289 handle_request_invite: Call from 'sipurasip' to extension '*33' rejected because extension not found in context 'outgoing'. |
| 05:34.20 | Kate6 | exten => _*333.,1,Set(outbound_gmail=gmail-nocid) |
| 05:34.20 | Kate6 | exten => _*333.,2,Goto(${EXTEN:3},1) |
| 05:34.21 | shamelessn00b | can you pastebin your extensions.comf |
| 05:34.23 | shamelessn00b | ? |
| 05:34.40 | kaldemar | Kate6: what kind of a phone are you using? |
| 05:35.13 | Kate6 | kaldemar: A SIP ATA with a Uniden 5.8 GHz cordless phone attached to it. |
| 05:35.14 | shamelessn00b | hi kaldemar |
| 05:36.07 | Kate6 | shamelessn00b: http://pastebin.com/9ryemXA5 |
| 05:36.15 | kaldemar | Kate6: is the * character coming through to asterisk? check with sip debug. |
| 05:36.53 | Kate6 | kaldemar: Well, considering that it's saying "Call ... to extension '*33" rejected", isn't it safe to say that it is? |
| 05:37.03 | kaldemar | * is no wildcard, it's just a char |
| 05:37.29 | kaldemar | Kate6: yes |
| 05:37.37 | Kate6 | kaldemar: Mind you it's saying that as soon as I've entered "*33" into the phone, even though I have a "_*333." extension defined. It doesn't let me get any further. |
| 05:38.07 | Kate6 | It's like the star causes it to match immediately after three tones have been sent. |
| 05:38.33 | kaldemar | maybe the phone has a dialplan in it that makes it dial immediately. |
| 05:38.39 | shamelessn00b | no even if you have dialed *333 you'd have to wait a while |
| 05:38.59 | shamelessn00b | till the timeout expires cuz asterisk is expecting more than 4 digits |
| 05:39.15 | shamelessn00b | and if you don't enter the fourth digit, this extension wont match |
| 05:39.23 | Kate6 | kaldemar: That's an interesting idea. |
| 05:39.38 | kaldemar | or the ata more likely. |
| 05:39.44 | Kate6 | kaldemar: Lemme look through the ATA's web panel. |
| 05:40.34 | Kate6 | (*xx|[3469]11|0|00|[2-9]xxxxxx|1xxx[2-9]xxxxxxS0|xxxxxxxxxxxx.) |
| 05:40.41 | Kate6 | Is that the *xx at the beginning? |
| 05:40.55 | kaldemar | looks like it |
| 05:41.01 | Kate6 | Hah. |
| 05:41.08 | Kate6 | Now I feel silly. |
| 05:41.23 | Kate6 | Thank you kaldemar. :) |
| 05:43.29 | kaldemar | don't thank before it works. :) |
| 05:43.33 | shamelessn00b | lol |
| 05:43.48 | Kate6 | *nods* |
| 05:43.54 | shamelessn00b | So I'm just tweaking my asterisk-sphinx 4 connector a bit now |
| 05:44.08 | shamelessn00b | testing in noisy environments etc |
| 05:44.24 | shamelessn00b | see how my VAD performs |
| 05:44.52 | shamelessn00b | cellular phones have very sensitive microphones |
| 05:45.36 | shamelessn00b | I mean, I was using a boom mic attached to my PC, and I had the threshold value set to 9, and it was very good at compensating for background noise |
| 05:45.49 | shamelessn00b | when I tested on cell phone I had to set the threshold to 20 |
| 05:46.09 | shamelessn00b | in order for it to be effective against background noise |
| 06:03.25 | Kate6 | kaldemar: Took me a few moments to tweak the dialplan on the ATA correctly, but it's working now. |
| 06:03.38 | Kate6 | kaldemar: So yes, thank you! You hit the nail right on the head. *hugs* |
| 06:03.40 | *** join/#asterisk Z_God (~julius@2001:610:1908:8000:21e:ecff:fe5d:679e) |
| 06:03.55 | ChannelZ | kiss and run |
| 06:08.44 | greezmunkey | Hi, I'm trying to chase down an issue registering a Spectralink 8020 to *. I have a working dialplan, and I can make and take calls to the 8020's, but with a hack in extensions.conf where I hard code the phones ip address: exten => 1001,1,Dial(SIP/1001@192.168.11.21,30) The sip register requests show up in the cli, but asterisk does not respond to the request at all. sip.conf has host=dynamic defined. |
| 06:16.10 | *** join/#asterisk tyman (~tyler@99.28.157.10) |
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| 06:39.00 | diemos | hi all |
| 06:55.04 | *** join/#asterisk Alagar (~avaithiya@193.220.61.145) |
| 07:20.13 | ChannelZ | ahoy |
| 07:21.09 | ChannelZ | greezmunkey: Dial(SIP/192.168.11.21/1001) |
| 07:21.38 | greezmunkey | ChannelZ: ok |
| 07:21.47 | ChannelZ | And you didn't show your register line |
| 07:22.08 | ChannelZ | (or the peer in sip.conf) |
| 07:22.54 | greezmunkey | ChannelZ: The thing is that these Spectralink phones send register messages to asterisk, but asterisk does not send any replies. |
| 07:23.18 | greezmunkey | ChannelZ: yet I have a softphone using the ap, and it works fine. |
| 07:23.56 | ChannelZ | You see the regisration attemps and no error, or how do you know they're making it to * ? |
| 07:23.58 | greezmunkey | ChannelZ: The sip.conf, extension.conf settings for the soft phone and the wifi phone are functionally identical. |
| 07:24.12 | greezmunkey | ChannelZ: I am at the *cli |
| 07:24.30 | greezmunkey | ChannelZ: I have also run wireshark traces of this behavior |
| 07:24.55 | ChannelZ | and? you're not really saying anything here |
| 07:25.57 | greezmunkey | ChannelZ: THere is just a lot of information to try to relate in a few sentenses. The bottom line is that I am not sure how to troubleshoot this. |
| 07:26.59 | ChannelZ | Well I asked 3 specific questions that you didn't answer |
| 07:27.28 | greezmunkey | ChannelZ: I'll pastebin, sorry. |
| 07:28.02 | ChannelZ | The second two.. do you see *NOTHING* on the console when the device tries to register? (Are you running with some verbose on?) |
| 07:28.30 | ChannelZ | do "sip set debug on" - do you see registration attempts from the device? If you don't, then the traffic isn't even getting there. |
| 07:29.33 | greezmunkey | ChannelZ: I did that, the registration attempts from the spectralink arrives, one after another with no responce. I'm gathering some info to send you... |
| 07:29.45 | *** join/#asterisk a1fa (~a1fa@unaffiliated/a1fa) |
| 07:30.45 | ChannelZ | * should respond in some way |
| 07:31.55 | greezmunkey | ChannelZ: Here you go...http://pastebin.com/vymAuJNP |
| 07:32.05 | greezmunkey | ChannelZ: I agree that it should. |
| 07:34.30 | *** join/#asterisk angryuser_laptop (~angryuser@90-156-167-83.reverse.alphalink.fr) |
| 07:35.31 | greezmunkey | ChannelZ: Extensions.conf incorporates the hack I mentioned earlier. Before that I could make a call from the spectralink to the softphone, but no vice versa. |
| 07:35.47 | greezmunkey | s/but no/but not |
| 07:38.36 | ChannelZ | what IP is 11.200 |
| 07:38.47 | greezmunkey | ChannelZ: the AP |
| 07:39.29 | ChannelZ | AP? as in wireless? |
| 07:39.53 | *** join/#asterisk bjornts (~BTS@it010226.klientdrift.uib.no) |
| 07:39.58 | greezmunkey | ChannelZ: Yes, access point. It's a ddwrt linksys in ap mode. |
| 07:40.16 | greezmunkey | ChannelZ: the spectralink phones are wifi |
| 07:40.39 | greezmunkey | The softphone is also, the laptop is accociated to the same Ap |
| 07:40.41 | ChannelZ | well the one curious thing I see is the spectralink provides no auth info, but I'm not sure why * is just not responding |
| 07:41.45 | greezmunkey | ChannelZ: I am new to deciphering sip packets. Where should it be...I'll stare and compare the softphone to the other... |
| 07:42.30 | ChannelZ | (and by the way if the access point was really acting as an access point, both devices would have their own IP. It must be doing NAT) |
| 07:43.07 | ChannelZ | Well generally in the phone you give it the 'proxy' name (your * box) and then the peer name and password it's supposed to be using |
| 07:43.29 | *** join/#asterisk qjb (~qjb@a83-163-158-168.adsl.xs4all.nl) |
| 07:44.18 | greezmunkey | ChannelZ: I see that as well, but the wifi devices are on the same subnet as asterisk! |
| 07:44.38 | *** join/#asterisk jsjc (~jsjc@13.Red-213-96-102.staticIP.rima-tde.net) |
| 07:46.02 | ChannelZ | that's fine I'm just saying it appears you're not using a WAP but just a wireless router and your wireless devices are doing NAT through it.. or else your softphone and the wireless phone would have different IPs (as far as Asterisk is concerned) |
| 07:46.35 | greezmunkey | ChannelZ: I honestly agree with that. Maybe I'll default the ap and try again. |
| 07:46.42 | ChannelZ | or it IS really running as an AP and you've got duplicate IPs. |
| 07:50.20 | ChannelZ | although your spectralink seems to know it's IP of 192.168.11.21 |
| 07:50.34 | ChannelZ | and your softphone of 11.102 |
| 07:50.35 | greezmunkey | ChannelZ: no, duplicates. I setup dhcpd on the * box, mac address reserved for 192.168.11.21, and 22 |
| 07:51.02 | greezmunkey | ChannelZ: There is something wierd going on. I'm going to default the ap, brb |
| 07:51.20 | ChannelZ | yeah I don't know how any of this is occuring |
| 07:52.51 | greezmunkey | ChannelZ: The phones require access to a tftp server. I set up dhcpd here, and the tftp server. When I start one of the Slink phones, set to dhcp, it gets an address (seen in /var/log/messages), then accesses the tftp server just fine! |
| 07:55.06 | *** join/#asterisk hrhrhr (~c1@unaffiliated/hrhrhr) |
| 07:55.11 | greezmunkey | ChannelZ: even wierder, check this out: NOTICE[4061]: chan_sip.c:13434 check_auth: Correct auth, but based on stale nonce received from '"1000"<sip:stevec@192.168.11.10:5060> |
| 07:55.40 | ChannelZ | when did that happen |
| 07:55.44 | greezmunkey | ChannelZ: It is definately a network problem |
| 07:55.54 | greezmunkey | ChannelZ: been seeing that for hours. |
| 07:57.13 | ChannelZ | did the messages in the sip debug you pasted earlier occur absolutely sequentially? IE line 77 on, you didn't copy/paste portions, nothing happened in between that you removed? |
| 07:58.34 | greezmunkey | yes, more of the same (sorry for the delay, I had my hands full defaulting the ap) |
| 07:59.00 | greezmunkey | ChannelZ: I can have the ap back up in a few minutes. |
| 07:59.50 | *** join/#asterisk reber (~reber@212-198-99-56.rev.numericable.fr) |
| 08:00.37 | ChannelZ | hmm. I'm not sure what reason * would not show a response being transmitted as a rejection or throwing an error, rather than just ignoring it outright. I don't necessarily see anything malformed about the registration attempts per se |
| 08:00.42 | ChannelZ | shrugs |
| 08:01.34 | greezmunkey | ChannelZ: I don't get it either. p3nguin halped me a lot yesterday - the dialplan and sip.conf are his handywork |
| 08:01.55 | greezmunkey | ChannelZ: I think the * is cool, it's just network wierdness. |
| 08:02.15 | ChannelZ | and what you pasted is with 'sip set debug on' yes? (not limited to an IP or specific peer) |
| 08:02.51 | greezmunkey | Yes |
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| 08:08.47 | greezmunkey | ChannelZ: ok, ap is back up after a 30/30/30 reset. Configured just enough to bring it on line. I need to reset wireshark, ect. I'll let you know what I find. |
| 08:09.23 | greezmunkey | softphone is up - ok |
| 08:09.38 | greezmunkey | starting Slink1001 |
| 08:12.24 | greezmunkey | ChannelZ: same thing...this is the trace, starting with the softphone: http://pastebin.com/AJbatAEX |
| 08:12.36 | greezmunkey | bizarre |
| 08:13.17 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 08:15.33 | jsjc | I been looking for a freelancer for a very simple simple setup >1hour job. Anybody does freelancer jobs and it has 1h slot today? |
| 08:15.38 | *** join/#asterisk Tim_Toady (~moi@178.128.21.207.dsl.dyn.forthnet.gr) |
| 08:16.01 | ChannelZ | greezmunkey: this might be a bug |
| 08:16.10 | greezmunkey | ChannelZ: the wierd thing is that I can call from/to the softphone and the Slink phone! |
| 08:16.24 | greezmunkey | ChannelZ: I am on 1.8.??? |
| 08:16.43 | greezmunkey | ChannelZ: maybe I should install 1.6 ??? |
| 08:16.43 | kaldemar | the stale nonce is about the subscribe message, not the register. |
| 08:18.23 | greezmunkey | ChannelZ: But if I remove the "101@X.X.X.X" in extensions.conf, I can only call from the Slink to the softphone, not vice-versa... |
| 08:18.41 | greezmunkey | er 1001@(ip address) |
| 08:18.45 | *** join/#asterisk dimm (~miniadmin@unaffiliated/dimm) |
| 08:18.45 | kaldemar | the phone uses the nonce from the unathorized response to the INVITE to generate a challenge response in the SUBSCRIBE so asterisk sends a new nonce. |
| 08:18.46 | ChannelZ | There's just something odd about those register packets from the spectralink and * is ignoring them. I'm manually transmitting the same packet to my * and it's basically sitting there doing nothing as well. |
| 08:19.12 | greezmunkey | ChannelZ: whoa, you can do that? |
| 08:19.35 | dimm | how can i view log only from one peer ? (is 'sip set debug peer <peer number>' is good for me?) |
| 08:20.07 | greezmunkey | dimm |
| 08:20.14 | greezmunkey | dimm: that does work. |
| 08:20.36 | greezmunkey | dimm: I saw my peers after hitting tab |
| 08:21.03 | ChannelZ | it's a bug/?? in 1.8.0. I'm sending it to my 1.6.2.x box and it's getting angry |
| 08:21.21 | greezmunkey | ChannelZ: angry? |
| 08:21.45 | ChannelZ | I get a response, it complains there's no peer matching what it's trying to register as |
| 08:22.05 | ChannelZ | under 1.8.0 it just ignores me like nothing has happened |
| 08:22.24 | greezmunkey | Here's my version info: Asterisk 1.8.0 built by root @ localhost.localdomain on a i686 running Linux on 2010-10-22 17:12:10 |
| 08:22.55 | greezmunkey | Running on: Linux asterisk2.local 2.6.18-194.26.1.el5 #1 SMP Tue Nov 9 12:54:40 EST 2010 i686 i686 i386 GNU/Linux |
| 08:23.05 | greezmunkey | Centos 5.5 |
| 08:24.23 | greezmunkey | ChannelZ: would you like me to gather more information for you to "send up the line"? |
| 08:25.56 | greezmunkey | ChannelZ: Or, would it make more sence for me to load 1.6, and see if it does the same thing first? |
| 08:27.26 | greezmunkey | sense.. |
| 08:27.38 | ChannelZ | well I can't do anything about it |
| 08:27.55 | greezmunkey | ChannelZ: That's ok, I was just offering. |
| 08:28.01 | ChannelZ | I'm trying to scrutinize the registration packet and figure out what that is in it could be causing it to be ignored |
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| 08:32.27 | ChannelZ | It's the lack of ;tag=xxxx on the From: header |
| 08:32.55 | greezmunkey | ChannelZ: I dont know if matters, but here are the setting passed to the phones via tftp: http://pastebin.com/ynCumCjL |
| 08:33.48 | greezmunkey | ChannelZ: Yet the 1.6 box parses the request, where 1.8 does not??? |
| 08:33.59 | ChannelZ | doesn't mean anything to me without those phones |
| 08:34.06 | greezmunkey | ChannelZ: agreed. |
| 08:35.03 | greezmunkey | ChannelZ: The config files for these phones are well documented, but I don't know if that was against 1.8 or not. I can find no mention of which version they support. |
| 08:35.42 | kaldemar | from SIP RFC 8.1.1.3: "The From field MUST contain a new "tag" parameter, chosen by the UAC." |
| 08:36.21 | ChannelZ | and there you go |
| 08:36.30 | greezmunkey | kaldemar: so I have a malformed packet issue. |
| 08:36.44 | ChannelZ | Look for new firmware or otherwise complain at Spectralink |
| 08:36.49 | ChannelZ | (or whatever) |
| 08:37.48 | greezmunkey | ChannelZ: you and kaldemar may have something there. I have what they ssay is the "latest" sip software, but I bet they have a patched load laying around somewhere. |
| 08:39.33 | ChannelZ | Barring that send them an email and point them at the RFC. |
| 08:39.47 | greezmunkey | The topic makes no mention of where this channel is logged, anyone know? |
| 08:39.53 | kaldemar | the old RFC didn't require the From tag. |
| 08:39.54 | ChannelZ | I guess you could build 1.6.2.something and see if it will still function anyway |
| 08:41.13 | greezmunkey | kaldemar: I will find out more about this, but for right now, we're sip2.0 from what i can tell. What was the old rfc? |
| 08:41.15 | kaldemar | "A UAS MUST be prepared to receive a request without a tag in the From field, in which case the tag is considered to have a value of null." |
| 08:41.28 | kaldemar | "This is to maintain backwards compatibility with RFC 2543, which did not mandate From tags." |
| 08:41.36 | kaldemar | 12.1.1 |
| 08:42.04 | greezmunkey | kaldemar: so we are back to ...bug? |
| 08:43.12 | kaldemar | well... on both sides, it seems. |
| 08:43.56 | greezmunkey | kaldemar: agreed, obviously the phone isn't sending it. |
| 08:45.44 | greezmunkey | kaldemar: rfc 3261 was the old one? |
| 08:45.45 | kaldemar | if Spectralink claims to support RFC 3261, they have a bug. from the asterisk side, i'd guess that a bug report would be classified as a feature request. |
| 08:46.10 | greezmunkey | kaldemar: I see that error, I have it backwards. |
| 08:46.15 | kaldemar | 3261 is the new one, 2543 the old one. they use running numbers. |
| 08:47.28 | greezmunkey | Well, I'll find out (maybe) tomarrow. I doubt I'll reach anyone, but I'll try. I have access to an engineer or two at Polycom. |
| 08:47.54 | ChannelZ | hmm interesting |
| 08:49.02 | greezmunkey | kaldemar: I downloaded both rfc's, and have marked what you sited. |
| 08:49.22 | ChannelZ | DEBUG[6281]: chan_sip.c:7398 find_call: REGISTER request has no from tag, dropping callid: ..... |
| 08:49.39 | ChannelZ | DEBUG[6281]: chan_sip.c:23871 handle_request_do: Invalid SIP message - rejected , no callid, len 666 |
| 08:50.54 | ChannelZ | I see the same actual code in 1.6.2.x though which is interesting |
| 08:51.27 | ChannelZ | Yet it seemed to be behaving differently so there must be a difference elsewhere in the code |
| 08:56.14 | ChannelZ | huh. 1.6.2 doesn't output that same debug, it just acts like there's no matching peer for the registration (which there isn't in my case) yet most of the code looks similar. |
| 08:56.35 | ChannelZ | Oh well. Either way it seems to be the phone's fault |
| 08:56.40 | greezmunkey | heh |
| 08:57.49 | ChannelZ | you *might* be able to get up and working under an older * build but hard to say |
| 08:58.39 | greezmunkey | ChannelZ: well, I'll try it for the excercise, but I still plan on bringing this up to Polycom. |
| 08:59.01 | greezmunkey | ChannelZ: do you know where this channel is logged? |
| 09:00.05 | ChannelZ | ?? possibly nowhere |
| 09:00.28 | ChannelZ | Save out from your IRC client |
| 09:00.59 | shamelessn00b | hmm |
| 09:01.19 | greezmunkey | ChannelZ: yeah, I can't find it anywhere. I'll have to check out how to grab this from irssi... |
| 09:02.06 | greezmunkey | Eh, copy paste works. |
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| 09:03.00 | greezmunkey | ChannelZ: and kaldemar : Thank you for helping me with this, you guys rock. |
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| 09:03.25 | schmidts | good morning |
| 09:03.30 | kaldemar | greezmunkey: channel logs: http://purl.rikers.org/%23asterisk/ |
| 09:03.45 | greezmunkey | You rock even more! |
| 09:03.47 | ChannelZ | greezmunkey: http://burner.com/spectralink.txt - those are the relevent lines of tonight |
| 09:04.49 | ChannelZ | (late) |
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| 09:10.22 | greezmunkey | One quick question, are the sip and extensions.conf files I'm using backward compatible to 1.6.x ? |
| 09:13.00 | ChannelZ | mostly |
| 09:13.38 | greezmunkey | ChannelZ: I'll deal with it, thanks again! |
| 09:14.49 | ChannelZ | good lucky |
| 09:15.21 | greezmunkey | ChannelZ: This is the fun stuff. I like the pain, I guess :) |
| 09:16.19 | ChannelZ | heh well you certainly found an interesting one |
| 09:17.18 | greezmunkey | I try not to bore ppl if I can avoid it. |
| 09:19.41 | ChannelZ | :) |
| 09:20.36 | ChannelZ | On the bright side tonight I learned netcat can do UDP |
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| 09:40.56 | greezmunkey | ChannelZ: you're not going to believe this, (maybe you will...) in 1.6, they register just fine... |
| 09:42.30 | greezmunkey | omg, well at least I learned some valuable troubleshooting skills. |
| 09:46.58 | ChannelZ | I believe it I"m just not sure why, as at least at a glance chan_sip in both versions seem to have the same code that would trap that missing tag. But something is obviously different somewhere |
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| 09:49.00 | greezmunkey | ChannelZ: Now * transmits back to the Slink phones: SIP/2.0 200 OK |
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| 09:50.42 | greezmunkey | ChannelZ: well, now (at least) I can move forward with some dialplan, and gateway programming. Which, after all is a lot more fun than chasing ghosts. |
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| 09:56.15 | greezmunkey | ChannelZ: well, on that note... I'm going to get some sleep, and play with this more in the morning. Thanks again :) |
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| 09:57.07 | ChannelZ | sure |
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| 10:18.41 | EmleyMoor | What's the best way to save the astdb and transfer it (or at least most of it) to another asterisk system? |
| 10:19.10 | ChannelZ | it's generally in /var/lib/asterisk |
| 10:19.34 | EmleyMoor | Ah, so just transferring that as is is generally adequate? |
| 10:19.59 | ChannelZ | It'll work with asterisk stopped when you do it. Whether or not it's wise to is another matter |
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| 10:21.00 | EmleyMoor | The system I will be transferring it to is intended to replace the existing one - so I can't see a problem |
| 10:21.29 | ChannelZ | The only thing of question in there is cached SIP registrations but in your case it's probably not a problem |
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| 10:58.28 | Hatrix76 | hi guy's and pro's ... how can i pickup a phone with AMI or with other scripting? We want our callcenter users to be able to pickup their phones without the need to touch them, from their web-based admin interface ... |
| 10:58.59 | Hatrix76 | I read that FOP is able to pickup phones, so I am asking myself how is the best way to do it ... |
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| 11:04.32 | petern_ | dunno but that would be nice indeed |
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| 11:31.46 | robl^laptop | Hatrix76: that's not an asterisk feature, per se.. it depends on the phone model |
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| 11:35.11 | robl^laptop | Hatrix76: most SIP phones support "auto answer" or some phones support pushing events to them. for example, Polycom phones support auto answer just by using SIPAddHeader() command. |
| 11:37.55 | robl^laptop | some other phones may have a built in web server that allows commands be sent to control the phone.. you can use asterisk's curl() function to send an http: request to some of those phones to tell it to answer a call |
| 11:47.24 | Hatrix76 | robl^laptop yeah, I thought so, i was confused on how Callcenter applications (openACD, others) do it from their web interface, but it could be that they open a channel on login, the agent is always off-hook and only the audio get's bridged to the agent when they accept a call via the web-interface. |
| 11:48.07 | robl^laptop | Hatrix76: that's another option.. |
| 11:50.48 | robl^laptop | but asterisk can't directly/arbitrarily force a phone to go off hook. either the agent is already connected to asterisk (off hook) and asterisk just bridges random calls.. or... it relies on using standard asterisk features to invoke a phone brand/model specific feature |
| 11:51.25 | creativx | x-lite doesnt have auto answer unless you pay for it afaik |
| 11:51.31 | creativx | softphones ftw |
| 11:52.14 | Hatrix76 | robl^laptop well, i will just install one of these call-center solutions and check how they are doing it ... |
| 11:52.27 | Hatrix76 | creativx i what das ftw mean? |
| 11:52.32 | creativx | for ze win |
| 11:52.56 | Hatrix76 | ok, seems that I am to old for this ... |
| 11:53.24 | creativx | that depends, did you discover the internet this year? |
| 11:53.25 | creativx | ;) |
| 11:53.53 | creativx | i suppose somewhere on the internet you will find all the memes.. urbandictionary.com might help |
| 11:53.54 | creativx | :) |
| 11:58.59 | Hatrix76 | well, i know nearly all of the old-schole mnemonics (like, from the beginning of the internet) ... like FWIW, classics like RTFM, all this stuff, but the youngens today ... for ze win ... neither the abbreviation nor the spelled out version does actually mean something to me ... well, you know you get old when .... |
| 12:09.08 | schmidts | hatrix76 http://www.urbandictionary.com/define.php?term=ftw ;) |
| 12:11.43 | E-bola | hehe¨ |
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| 12:41.18 | Hatrix76 | Hehehe, I am reading the snom documentation regarding call pick-up, and it seems that the way BLF works is interesting, it sends some SIP messages around and at last a SIP call-pickup INVITE message ... does this mean that if my phone receives a call-pickup INVITE message, it goes actually offhook and answers the call? If that's true, I can implement a Web Interface which just sends a |
| 12:41.19 | Hatrix76 | call-pickup INVITE message to the phone, and if it's allready ringing, it should just pick-up, am I right? I'm not that deep into SIP (looks at schmidts) ... |
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| 12:46.42 | file | Hatrix76, you are trying to force the phone to answer the call? |
| 12:47.42 | robl^laptop | Hatrix76: no.. not exactly.. its talking about how a secretary/boss scenerio may work. for example, a secretary has a BLF key for her boss.. if she sees it bringing, she can intercept the ringing call by pressing the BLF key... |
| 12:47.54 | file | quite. |
| 12:48.25 | robl^laptop | tar and gzips file |
| 12:48.30 | file | eep |
| 12:48.38 | robl^laptop | morning, file! have plenty of turkey and pumpkin pie? |
| 12:48.43 | file | nope |
| 12:48.45 | file | it's not thanksgiving here |
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| 12:49.38 | schmidts | hatrix76 you can use chanisavail or devicestate to check the state of a phone before dialing it, it has nothing todo with only sip. |
| 12:49.48 | robl^laptop | ohh! I was thinking you were in the states. |
| 12:49.56 | file | nope |
| 12:50.26 | schmidts | hatrix76 but you can also tell the phone to send back a busy if a call is active (CALL-WAITING off or something like this) |
| 12:50.29 | Hatrix76 | robl^laptop, yes, but the way how SIP seems to work (at least in the snom documentation) the last part of the BLF communication is a message with SIP call-pickup INVITE ... I was thinkging along the lines of using this message to force the automatic pickup |
| 12:50.55 | file | a message with SIP call-pickup INVITE from the phone to Asterisk |
| 12:52.53 | Hatrix76 | schmidts we have a standard call-center scenario, phones ring defined by the strategies, app_queues, all that stuff. What we want now is that the agent does not have to touch the phone and that Login/Logout/Pickup can be done via the web-browser, this would allow us at the same time to pickup the phone, check the phone-number and display the caller-data from the crm in the agent-browser view. |
| 12:53.39 | schmidts | hatrix76 with pickup you mean answer a call or pickup a call from another phone? |
| 12:53.51 | Hatrix76 | answer the call |
| 12:53.54 | file | you want integration between the SIP phone and your application, and you want to control the phone at a deeper level (like tell it to answer a call from your application) |
| 12:54.20 | Hatrix76 | @file, yep, that's basically what I want ... just checking around the best way of doing it ... |
| 12:54.27 | file | generally you can't |
| 12:54.40 | file | SIP phones are smart and unless they provide a mechanism for deeper integration you can't 'force' them to do anything |
| 12:54.41 | schmidts | you can use the sip autoanswer header for this |
| 12:54.54 | schmidts | like paging |
| 12:54.58 | file | yes that is the closest you can do but still not exactly what Hatrix76 wants |
| 12:55.17 | file | that would mean as soon as the call reaches the phone it's answered, that does not mean that you click a button in an application and it is answered |
| 12:55.42 | Hatrix76 | yeah, thats right, we would like to have the click ... |
| 12:55.44 | schmidts | hatrix76 why do you want to use a hardphone? why not use a softphone |
| 12:56.40 | Hatrix76 | it's another requirement in our call-center, they got burned with softphones reacting badly when the load on the local computer (for whatever reasons) kept going up ... they want a physical phone |
| 12:57.00 | schmidts | and they want the phones ringing? |
| 12:57.48 | schmidts | a solution could be something like a WebIf where all agents see the queue, but only online and if someone takes this call, its dialed from asterisk like paging to the agents phone |
| 12:58.04 | Hatrix76 | If I can sent a URL to the snom phones which tells them to pick up it would be ok as well, ... yes, the phones should ring, then the agent should be able to accept the call via the web-interface, the call should get answered, the agent can talk right away with the headset, the browser is fetching the customer data |
| 12:59.10 | schmidts | hatrix76 you maybe can do this a little tricky ;) call all phones in a normal way and if someone press the button to get the call, stop calling all phones and just call this phone with auto answer set |
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| 12:59.57 | schmidts | hatrix76 another way would be to start a call from snom to for example a meetme room and bridge the call in this room |
| 13:01.14 | Hatrix76 | schmidts, hmm, well, the first option sounds fun(ier), I should be able to do the same thing without calling everyone? like, calling a phone, it rings, if the agent clicks on accept, call the same extension again with the sip autoheader ... how|is this implementable via the app_queue? |
| 13:01.53 | Hatrix76 | this would have the benefit that you just can answer the phone as well, without the web-interface ... |
| 13:01.55 | robl^laptop | speaking of autoanswer - whats the best / cleanest way to do auto-paging? i.e., I want to be able to dial an extension which causes asterisk to use playback a recorded audio file to multiple phones and hangs up. I'm thinking of the scenerio where it would use Page(), SIPAddHeader to auto answer, and play a recording. I don't want the caller to be able to say anything. should I just cause a call file to be created? |
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| 13:02.50 | schmidts | hatrix76 it has nothing to do with app_queue, you have to controll the call via an AMI connection and let it stop calling the queue just call this single agent |
| 13:04.54 | Hatrix76 | schmidts hmmm ... ok, let's pick this up later, ... brb |
| 13:05.07 | schmidts | ok |
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| 13:20.10 | Dovid | whats the correct way to build the asterisk init script ? |
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| 13:20.51 | russellb | make config |
| 13:20.52 | russellb | i think |
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| 13:52.59 | *** join/#asterisk madduck (~madduck@debian/developer/madduck) |
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| 13:52.59 | *** mode/#asterisk [+oo pabelanger Qwell] by zelazny.freenode.net |
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| 13:54.50 | EmleyMoor | Is there an example for configuring DAHDI for an AEX4 card, 3 FXS 1FXO + EC? |
| 13:54.55 | phix | russellb: make menuconfig |
| 13:55.23 | russellb | phix: that doesn't install an init script |
| 13:56.18 | *** join/#asterisk merlin8282 (~merlin828@AStrasbourg-554-1-213-45.w86-204.abo.wanadoo.fr) |
| 13:56.53 | merlin8282 | Hi. Is it possible with asterisk to trigger a call recording while you're in a call ? |
| 13:57.20 | merlin8282 | I mean, with 1.4 |
| 13:57.36 | russellb | yes. |
| 13:57.54 | merlin8282 | I already saw the function monitor(), but that seems to work only on the entire call. |
| 13:58.06 | russellb | see features.conf, set a key sequence for "automon" |
| 13:58.12 | russellb | then enable the feature with the appropriate option to Dial() |
| 13:58.32 | merlin8282 | Ok, i'll try it, thanks russellb. |
| 13:58.51 | russellb | you're welcome. |
| 14:05.10 | krion | hum, wonder something about a difference between voicemail in asterisk 1.4 and 1.6 |
| 14:05.30 | krion | in 1.4, we don't get the envelope heared before the message is played |
| 14:05.40 | krion | in 1.6 it's the default |
| 14:05.48 | russellb | might be a new option in voicemail.conf |
| 14:06.06 | russellb | see CHANGES / UPGRADE.txt to see new options or intentional documented changes in behavior |
| 14:06.18 | krion | don't see nothing in the conf in order to get the same behaviour with 1.4 |
| 14:06.30 | krion | ok russellb thanks i will |
| 14:11.10 | krion | in fact it's asterisk 1.2... |
| 14:12.34 | *** join/#asterisk rgsteele (~rgsteele@apo.aweber.com) |
| 14:15.33 | krion | asterisk 1.2 voicemail.conf |
| 14:15.42 | krion | oops sorry |
| 14:19.40 | russellb | krion: the "saycid" option in voicemail.conf looks like it might be relevant ... |
| 14:19.47 | russellb | oh wait, no |
| 14:19.53 | *** join/#asterisk rrb3942 (~rbullock@cpe-67-252-100-238.stny.res.rr.com) |
| 14:20.02 | russellb | ; envelope=no ; Turn on/off envelope playback before message playback. [ON by default] |
| 14:20.09 | russellb | there you go :-) |
| 14:20.15 | russellb | I got that from voicemail.conf.sample in 1.6.2 |
| 14:28.42 | Hatrix76 | schmidts hi, sorry that I had to run off ... do you have a few more minutes to spare? we where talking about AMI controlling a call, how would I go about that? |
| 14:30.06 | Hatrix76 | schmidts most importantly, how do I stop a call from ringing, and start the ring again with the autoanswer header set ... that's what I have trouble to understand with |
| 14:38.41 | *** join/#asterisk jetlag (~jetlag@pool-173-61-245-217.cmdnnj.east.verizon.net) |
| 14:41.46 | schmidts | hatrix76 i dont know ami very well but i think there is something like a goto command to put a call from one extension (the queue) to another one. at this exten you can set the sip header for auto answer and just dial the agent you want |
| 14:43.07 | *** join/#asterisk angryuser (~angryuser@LPuteaux-151-42-27-99.w193-251.abo.wanadoo.fr) |
| 14:45.27 | Hatrix76 | schmidts, thank you, I will play around with this a bit. |
| 14:46.37 | krion | russellb: yes i find that too, FYI my voicemail.conf is in mysql |
| 14:46.55 | krion | i try to set it up to yes and then do a reload app_voicemail.so but it didn't fix my problem |
| 14:51.27 | merlin8282 | hmmm. Is there a CTI-like application for using with asterisk ? Or does it depend on the phone / phone manufacturer ? |
| 14:51.56 | merlin8282 | I heard of FOP, but i'm not sure if it's that what our client needs. |
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| 15:12.42 | EmleyMoor | I can see my AEX410P in lspci and so on, but cannot work out how to see the attached FXS/FXO modules, nor an easy way to set them up. Can anyone advise? |
| 15:14.53 | *** join/#asterisk cesar_CR (~cesar@201.191.254.20) |
| 15:16.38 | *** join/#asterisk moy_ (~moy@UNVLON55-1176057127.sdsl.bell.ca) |
| 15:20.28 | EmleyMoor | This seems much harder than zaptel ever was |
| 15:41.36 | *** join/#asterisk ketas (ketas@ketas6-sixxs.si.pri.ee) |
| 15:44.20 | *** join/#asterisk v1s (~v1s@202.84.107.67) |
| 15:44.23 | EmleyMoor | I have tried to configure chan_dahdi.conf and dahdi seems to have vanished from asterisk |
| 15:44.44 | v1s | what is the best sounding minimum bandwidth codec? |
| 15:45.52 | E-bola | lol |
| 15:46.22 | *** join/#asterisk rrb3942 (~rbullock@cpe-67-252-100-238.stny.res.rr.com) |
| 15:46.35 | EmleyMoor | dahdi_cfg -v gives this: http://paste.debian.net/100815/# |
| 15:47.11 | russellb | v1s: Most people choose G.729 for that. |
| 15:47.13 | EmleyMoor | How can I verify what is actually seen on my card? |
| 15:47.34 | russellb | Have you loaded the appropriate kernel module? |
| 15:47.35 | *** join/#asterisk aidinb (~Aidin@unaffiliated/aidinb) |
| 15:48.23 | EmleyMoor | dahdi is loaded |
| 15:48.36 | russellb | ok, you will also need the driver that is for that specific card. |
| 15:49.24 | russellb | Specifically, for the AEX410P, you need to load wctdm24xxp |
| 15:52.05 | *** join/#asterisk cesar_CR (~cesar@201.191.254.20) |
| 15:52.17 | EmleyMoor | russellb: Done that - what next? |
| 15:52.38 | russellb | now try your dahdi_cfg again |
| 15:52.47 | russellb | assuming you set up /etc/dahdi/system.conf already |
| 15:53.46 | EmleyMoor | Right - that looks a bit more promisinc |
| 15:53.51 | EmleyMoor | promising |
| 15:54.06 | russellb | also, check the output of 'dmesg' |
| 15:54.15 | russellb | when the driver loaded, it probably said something about what modules it saw |
| 15:56.11 | EmleyMoor | http://paste.debian.net/100816/ - looks fairly good but what is VPM100? |
| 15:56.29 | russellb | hardware based echo canceller |
| 15:56.36 | EmleyMoor | (also, what are the "Failed"... messages all about?) |
| 15:56.51 | EmleyMoor | I do have that - but that claims "not present" |
| 15:57.18 | russellb | the not present part is probably related to the failures you see |
| 15:57.22 | russellb | it couldnt' find the firmware for it |
| 15:57.32 | russellb | "make install" of dahdi is supposed to download and install the firmware for you |
| 15:58.29 | EmleyMoor | But, failing that? |
| 15:58.38 | russellb | no idea |
| 15:58.51 | E-bola | Hmm if i use the init script for debian with the default file supplied, asterisk starts in the foreground, and not as a daemon? |
| 16:00.17 | EmleyMoor | Hmmm... non-free |
| 16:00.40 | russellb | what, the firmware? |
| 16:04.10 | EmleyMoor | Well, in this case, just a package with a script to get it... |
| 16:04.15 | EmleyMoor | Still no go |
| 16:04.52 | russellb | why don't you try just installing DAHDI from source? |
| 16:06.54 | EmleyMoor | Because it's a "long" "try just" |
| 16:06.56 | E-bola | just found a bug in the contributed init scripts |
| 16:07.08 | russellb | EmleyMoor: long? it will take like 2 minutes. |
| 16:07.21 | E-bola | tzafrir_laptop: you here? |
| 16:07.30 | russellb | $ svn co http://svn.digium.com/svn/dahdi/linux-complete/trunk linux-complete ; cd linux-complete ; make ; sudo make install |
| 16:08.02 | *** join/#asterisk lost_soul (~noymfb@cpe-67-249-141-227.twcny.res.rr.com) |
| 16:08.18 | E-bola | Cant you set default verbosity without starting asterisk in console mode? |
| 16:08.49 | russellb | yes, by setting it in asterisk.conf |
| 16:09.27 | E-bola | yup just figured, but if u enable the option in the contributed default file, it means using the init script starts asterisk in console mode |
| 16:09.34 | E-bola | because it passes -v at startup |
| 16:09.48 | russellb | yeah, that's not a good thing to do |
| 16:10.18 | *** part/#asterisk v1s (~v1s@202.84.107.67) |
| 16:10.29 | E-bola | Indeed, im not sure why anybody woudl find that a standard thing to do, so i think it should atleast have a warning explaining that the consequence is that its started in foreground |
| 16:10.54 | russellb | submit a patch :-) |
| 16:11.21 | E-bola | Cant remember how to make one :) |
| 16:11.27 | E-bola | just a diff file or? |
| 16:11.30 | russellb | yeah. |
| 16:11.39 | E-bola | Alright, ile fix one up |
| 16:11.42 | russellb | check out from svn, make changes, then ... $ svn diff > my_patch.txt |
| 16:11.49 | EmleyMoor | No tzafrir_laptop? |
| 16:11.54 | tzafrir_laptop | E-bola, yes |
| 16:12.16 | tzafrir_laptop | be right here |
| 16:12.25 | E-bola | tzafrir_laptop: nevermind i guess, i thought there was a debian specific issue, but its more like something easily missunderstandable about the initscript/default file |
| 16:12.30 | E-bola | making a patch as we speak |
| 16:13.22 | EmleyMoor | Ah... |
| 16:14.18 | EmleyMoor | tzafrir_laptop: Do you know how to get the firmware to get the echo canceller on my AEX410P working, on a Debian system, short of ditching packages and building from scratch? |
| 16:14.31 | EmleyMoor | (squeeze if it helps) |
| 16:15.19 | E-bola | russellb: where to send patch? |
| 16:15.29 | russellb | upload to https://issues.asterisk.org |
| 16:15.54 | E-bola | Do i need to create an issue first? |
| 16:16.01 | russellb | yup |
| 16:16.04 | E-bola | Alrighty |
| 16:16.09 | russellb | create an issue and attach it |
| 16:16.28 | russellb | You'll also need to fill out the contributor license agreement form if you haven't already |
| 16:16.36 | russellb | you should see a link up top after you log in |
| 16:16.44 | russellb | "Sign License" or something like that |
| 16:16.49 | tzafrir_laptop | EmleyMoor, did you see the script in the dahdi-linux packages to download the firmwares? |
| 16:16.58 | tzafrir_laptop | Something under /usr/share/dahdi/ |
| 16:17.06 | EmleyMoor | tzafrir_laptop: Saw it, ran it, no go |
| 16:18.09 | EmleyMoor | It did some dowloading but either not what was needed or not to the right place |
| 16:18.27 | tzafrir_laptop | What do you need: |
| 16:19.37 | EmleyMoor | Whatever will fix http://paste.debian.net/100816/ |
| 16:19.39 | tzafrir_laptop | EmleyMoor, Maybe the issue is http://svn.debian.org/viewsvn/pkg-voip/dahdi-linux/trunk/debian/patches/no_firmware_download?view=markup ? |
| 16:20.11 | EmleyMoor | Hmmm... could be... |
| 16:20.44 | E-bola | heh |
| 16:20.53 | E-bola | you have to enter your street address to submit a patch.... my god |
| 16:21.39 | russellb | yup ... just have to go through that once |
| 16:21.49 | E-bola | russellb: https://issues.asterisk.org/view.php?id=18383 |
| 16:21.53 | russellb | we can only accept patches if you grant us a license to use them for whatever |
| 16:21.56 | E-bola | I hope windows didnt fuckup the txtfile formatting |
| 16:22.21 | russellb | looks fine |
| 16:22.22 | russellb | thanks |
| 16:22.37 | russellb | will probably get merged in the next week or so |
| 16:22.45 | russellb | can't do it right now |
| 16:23.05 | E-bola | no worries |
| 16:23.40 | EmleyMoor | t |
| 16:23.44 | seanbright | q |
| 16:23.50 | EmleyMoor | tzafrir_laptop: I think I can see what to do |
| 16:28.30 | EmleyMoor | :-) |
| 16:29.27 | EmleyMoor | How do I confirm from within asterisk that it can see all my FX[SO] ports? |
| 16:29.38 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 16:34.59 | seanbright | dahdi show status |
| 16:35.02 | seanbright | dahdi show channels |
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| 16:36.10 | E-bola | Whats wrong if i cant select timerfd as timing source in make menuconfig? Im running debian lenny with a 2.6.32 kernel so it should work? |
| 16:36.35 | russellb | Glibc might not be new enough |
| 16:36.48 | E-bola | ahh ya just saw that in the notes |
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| 16:38.25 | EmleyMoor | No dahdi there it seems |
| 16:38.45 | seanbright | chan_dahdi loaded? |
| 16:38.49 | seanbright | chan_dahdi.so? |
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| 16:41.19 | EmleyMoor | seanbright: Seemingly not... how do I sort it? |
| 16:44.13 | EmleyMoor | (it obviously has been loaded in the past) |
| 16:48.14 | *** join/#asterisk Ad-Hoc (~nimbus@194.219.217.162.dsl.dyn.forthnet.gr) |
| 16:48.52 | EmleyMoor | is really stuck |
| 16:50.20 | EmleyMoor | Why would chan_dahdi not load? |
| 16:51.26 | *** join/#asterisk leafartn (~chatzilla@190.145.253.51) |
| 16:51.45 | leafartn | hi everyone |
| 16:51.50 | *** join/#asterisk thansen (~thansen@173.84.218.244) |
| 16:52.18 | leafartn | how can i log in a agent from asterisk maganer (AMI) in asterisk 1.8 ??? |
| 16:52.34 | seanbright | EmleyMoor: if you start asterisk directly you can see error messages |
| 16:52.42 | seanbright | asterisk -cvvvvvdddddg |
| 16:52.53 | seanbright | (after stopping it, obviously) |
| 16:53.20 | EmleyMoor | I found out why... needed dahdi_cfg run again |
| 16:53.59 | EmleyMoor | It can see my FXS ports but doesn't show my FXO - why would that be? |
| 16:56.56 | *** join/#asterisk garymc (~chatzilla@host81-139-141-137.in-addr.btopenworld.com) |
| 16:56.59 | EmleyMoor | It seems to be completely ignoring it |
| 16:57.36 | *** join/#asterisk guilhermebr (~Guilherme@200.103.96.98) |
| 16:57.56 | EmleyMoor | (might it somehow need a line in to detect it?) |
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| 17:03.45 | EmleyMoor | No sign of channel 4 |
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| 17:04.24 | EmleyMoor | I can paste anything if it will help |
| 17:05.15 | EmleyMoor | I know channel 1 is working - plugged my phone in |
| 17:06.01 | EmleyMoor | This non-appearing fxo is getting to em |
| 17:06.03 | EmleyMoor | me |
| 17:06.42 | leafartn | anyone? |
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| 17:11.18 | EmleyMoor | My FXS ports are there but my FXO isn't - I think I have something wrong in chan_dahdi.conf but cannot spot it - would pasting it somewhere help or is there something I shourd check first? |
| 17:12.16 | seanbright | there is a script |
| 17:12.28 | seanbright | called dahdi_genconf (which you shouldn't run yet) |
| 17:13.07 | EmleyMoor | What should I doL |
| 17:13.08 | EmleyMoor | do? |
| 17:13.17 | seanbright | i'm getting there |
| 17:13.21 | seanbright | ~pb |
| 17:13.21 | 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. |
| 17:13.33 | seanbright | paste all of the output from your dmesg file that relates to the card |
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| 17:15.12 | EmleyMoor | http://paste.debian.net/100825/ |
| 17:15.46 | seanbright | ok, cool |
| 17:15.55 | seanbright | so backup your dahdi configuration |
| 17:16.11 | seanbright | /etc/dahdi and /etc/asterisk/chan_dahdi.conf |
| 17:16.14 | seanbright | and whatever else you've done |
| 17:16.31 | seanbright | and then run dahdi_genconf which should give you a boilerplate configuration based on your installed hardware |
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| 17:21.32 | EmleyMoor | How exactly do I include one conf in another? |
| 17:24.56 | EmleyMoor | Great - it's there |
| 17:26.14 | EmleyMoor | Any significant changes to sip.conf, iax.conf or jabber.conf, between 1.4 and 1.6? |
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| 17:40.47 | EmleyMoor | Where are AGI scripts normally kept? |
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| 17:53.25 | EmleyMoor | OK - if asterisk is correctly set up on a Debian system, it should load at boot, including all needed dahdi config - right? |
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| 17:58.57 | WindBack | I have two asterisk trunked with a IAX trunk. Sometimes ,I'm reciving a FRAME_CONTROL (-1) message and the call goes down. Any idea? |
| 18:00.28 | EmleyMoor | It appears asterisk is using a default dialplan rather than the one in extensions.conf. How do I change this? |
| 18:02.42 | EmleyMoor | Hmmm... sip isn't loaded either |
| 18:03.40 | diemos | I'm pretty sure most of the asterisk geeks are out today. |
| 18:03.52 | diemos | Possibly buying gifts for their loved ones. |
| 18:04.28 | EmleyMoor | Is there something I need to do to switch asterisk out of "default" mode? It seems to be ignoring some confs |
| 18:05.34 | merlin8282 | EmleyMoor: maybe this has something to do with extensions.ael ? |
| 18:08.12 | EmleyMoor | Where therefore is any part of my dialplan? |
| 18:09.55 | EmleyMoor | is on the verge of backing out |
| 18:10.24 | merlin8282 | EmleyMoor: forget it, i've to sleep :/ |
| 18:11.38 | EmleyMoor | Hmmm... |
| 18:11.52 | EmleyMoor | Obviously I copied some stuff to the wrong place |
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| 18:14.44 | EmleyMoor | No sip still |
| 18:16.26 | EmleyMoor | ... and iax, though present, seems to do nething |
| 18:16.29 | EmleyMoor | nothing |
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| 18:22.56 | EmleyMoor | I have no peers and no users - despite them being defined in iax.conf |
| 18:29.11 | EmleyMoor | All dahdi stuff is now OK - but without sip and iax, I am really stuck |
| 18:32.25 | EmleyMoor | It makes a complete mockery of asterisk for sip to be missing and iax2 to be "no good" |
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| 18:35.15 | EmleyMoor | Ah - permissions! |
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| 18:37.54 | *** mode/#asterisk [+o Corydon76-dig] by ChanServ |
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| 19:37.59 | EmleyMoor | Is there a good way to get Asterisk to work with Festival now? |
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| 19:42.37 | fireman_biff | Are there any issues with using AsteriskNOW with non-Digium hardware? (Rhino, to be specific) |
| 19:42.39 | EmleyMoor | I used to use it before, but now it produces no output |
| 19:42.51 | EmleyMoor | (Festival) |
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| 20:01.24 | EmleyMoor | Damn - will have to make some recordings |
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| 20:07.38 | Jamuel | hi--was wondering if any one could tell me what I'm doing wrong when compiling 1.6.2.15 RC 1 and addons 1.6.2.1 |
| 20:08.45 | Jamuel | the modules created from addons (notably cdr_addon_mysql.so are not loadable |
| 20:09.59 | Jamuel | I keep seeing this in the * log: loader.c: Module 'cdr_addons_mysql so' was not |
| 20:09.59 | Jamuel | compiled with the same compile-time options as this version of Asterisk. |
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| 20:11.12 | p3nguin | Seems pretty clear to me. |
| 20:12.26 | Jamuel | I compiled them at the same time using the same configure options and nothing different in the make |
| 20:12.28 | Jamuel | just make all |
| 20:16.01 | p3nguin | I can see how that poses a problem, then. |
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| 20:22.41 | greezmunkey | p3nguin: It turns out the the Slink phones do not include a ;tag=xxxxxx in their REGISTER packet(s). ChannelZ and kaldemar found it. In the meantime I downgraded to 1.6.2, and they register just fine. Wierd. |
| 20:23.16 | p3nguin | What branch were you using? |
| 20:23.59 | greezmunkey | uh, does 1.8.0 answer that? |
| 20:24.23 | greezmunkey | p3nguin: sorry, I am still new to this. |
| 20:25.04 | p3nguin | sorta |
| 20:25.52 | greezmunkey | p3nguin: here is a log from last night: http://pastebin.com/2fhUDDxV |
| 20:26.10 | greezmunkey | p3nguin: it's pretty short. |
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| 20:31.04 | p3nguin | So you're able to use the phones now? |
| 20:32.22 | greezmunkey | p3nguin: yes, I also have a 3com 3108, it works as well. |
| 20:33.28 | greezmunkey | p3nguin: I have the 3com, and softphone on one ssid, and the two Slinks on another. Pretty cool. |
| 20:33.49 | p3nguin | What's the benefit of that? |
| 20:34.36 | greezmunkey | p3nguin: none, the 3com only accept 63 chars for wpa2, the Slinks only accept 64, so I created two ssids |
| 20:34.53 | p3nguin | *shrug* |
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| 20:51.32 | greezmunkey | p3nguin: I installed a sounds package, and now have /var/lib/asterisk/sounds/en/blahblahblah yet when I load my dialplan including this: extn => 222,1,Playback(/var/lib/asterisk/sounds/en/housekeeping) * throws this: |
| 20:51.40 | greezmunkey | ugh |
| 20:51.53 | greezmunkey | cont. pbx_load_config: ==!!== Unknown directive: extn at line 30 |
| 20:52.38 | greezmunkey | Do I need to include the path to the sounds directory in asterisk.conf? |
| 20:53.12 | p3nguin | There's no reason to screw with the settings in the provided asterisk.conf. |
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| 20:53.37 | greezmunkey | p3nguin: nvm, I typoed it... Trying again... |
| 20:54.06 | p3nguin | And you need not provide the full path, either. |
| 20:54.11 | greezmunkey | yeah, extn should have been exten |
| 20:54.26 | p3nguin | Playback(housekeeping) should be fine. |
| 20:54.26 | greezmunkey | p3nguin: I'll try it w/o the path |
| 20:54.59 | greezmunkey | grins |
| 20:58.34 | greezmunkey | thank you [TK]D-Fender for pointing out my typo :) |
| 20:58.54 | p3nguin | hmm? |
| 20:59.29 | drmessano | lol |
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| 21:23.04 | greezmunkey | omg, this rocks. |
| 21:25.08 | ChannelZ | with its cock out |
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| 21:47.11 | greezmunkey | nice, first default extension after dialing an invalid extension, and first hunt group - grins |
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| 21:54.02 | E-bola | How do u include srtp in asterisk 1.8? |
| 21:58.18 | ChannelZ | It's probably just there, you just need to config it all up |
| 21:59.33 | ChannelZ | Assuming it built |
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| 22:03.09 | ChannelZ | http://srtp.sourceforge.net/srtp.html |
| 22:08.58 | Noxchi | I am New to Asterisk, recently I downloaded AsteriskNow under VmWare workstation, Dose anyone know how can I start Asterisk GUI ? |
| 22:09.29 | *** join/#asterisk qjb (~qjb@a83-163-158-168.adsl.xs4all.nl) |
| 22:10.29 | ChannelZ | it should be already AFAIK, just go to http://localhost:8080 I think |
| 22:23.58 | Noxchi | ChannelZ, from where should I use the browser to try this command http://localhost:8080? is there Linux command to do this ? |
| 22:43.51 | drmessano | You dod it from another workstation |
| 22:43.54 | drmessano | Not the local one |
| 22:55.23 | ChannelZ | Lynx! haha |
| 22:55.52 | ChannelZ | but yes, bad choice of words, I mean localhost to mean "whatever IP your *now is" which under VM could be anything |
| 22:56.00 | ChannelZ | so you fill in the blank |
| 22:57.28 | EmleyMoor | In asterisk.conf, is it possible to specify two directories to be searched in sequence for, say, AGIs? |
| 22:58.50 | greezmunkey | I'm reading the bestpractices.txt, looking at $(EXTEN). Does $(EXTEN) only take the value of defined extensions within their context? It looks to me that that is the case. |
| 22:59.29 | greezmunkey | s/within their context/within its context |
| 23:19.12 | paulc | If I use Dial with U(...) to Gosub somewhere on answer, it says I can return GOTO in GOSUB_RESULT but it's not clear how to specify where.. can someone point me in the right direction? |
| 23:24.04 | EmleyMoor | How do I resolve the requirement for calltoken support? |
| 23:25.38 | paulc | And the answer is "GOTO:context^exten^priority" - use the source Luke |
| 23:30.22 | EmleyMoor | What is the purpose of calltokens anyway? |
| 23:36.37 | EmleyMoor | Hmmm - no go even circmventing them... |
| 23:37.32 | EmleyMoor | [Nov 26 23:34:33] NOTICE[6605]: chan_iax2.c:10101 socket_process: Rejected connect attempt from 77.240.48.214, request '<number>@from-iax' does not exist |
| 23:39.42 | EmleyMoor | What would cause that? |
| 23:40.38 | paulc | does <number> exist in your [from-iax] context? |
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| 23:42.12 | EmleyMoor | paulc: Yes |
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| 23:44.10 | E-bola | pbx.c:4936 increase_call_count: Available system memory (~0MB) is below the configured low watermark (5MB) |
| 23:44.19 | E-bola | Why do i get that on a new install? |
| 23:44.31 | E-bola | tons of memory free, and not knowingly configured any call limits |
| 23:46.12 | E-bola | google isnt finding anything at all :/ |
| 23:46.31 | EmleyMoor | I'm having various troubles - not been able to dufinitively call out over IAX, but worse, not receiving either |
| 23:46.52 | EmleyMoor | is almost at the point of assuming the context name is now disliked |
| 23:50.14 | E-bola | Hmmm, for soem reason asterisk wasnt finding me free mem correctly |
| 23:50.23 | EmleyMoor | Hmmm... seems to be hanging up unexpectedly now it's hitting the dialplan |
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| 23:54.23 | EmleyMoor | Can't tell whether my end is hanging it up or theirs |
| 23:55.43 | EmleyMoor | Does 1.6 not like NoOp? |
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